AF’s Weblog

December 30, 2010

Compressors: How They Really Work

It’s one of the most used, and most misunderstood, signal processors. While people use it to make a recording “punchier,” it often ends up dulling the sound instead because the controls aren’t set optimally. And it was supposed to go away when the digital age, with its wide dynamic range, appeared.

Yet the compressor is more popular than ever, with more variations on the basic concept than ever before. Let’s look at what’s available, pros and cons of the different types, and applications.

Introduction

Compression was originally invented to shoehorn the dynamics of live music (which can exceed 100 dB) into the restricted dynamic range of radio and TV broadcasts (around 40-50 dB), vinyl (50-60 dB), and tape (40dB to 105 dB, depending on type, speed, and noise reduction used). As shown in Fig. 1, this process lowers only the peaks of signals while leaving lower levels unchanged, then boosts the overall level to bring the signal peaks back up to maximum. (Bringing up the level also brings up any noise as well, but you can’t have everything.)

Fig. 1: The first, black section shows the original audio. The middle, green section shows the same audio after compression; the third, blue section shows the same audio after compression and turning up the output control. Note how softer parts ot the first section have much higher levels in the third section, yet the peak values are the same.

Even though digital media such as the CD have a decent dynamic range, people are accustomed to compressed sound. Compression has been standard practice to help soft signals overcome the ambient noise in typical listening environments; furthermore, analog tape has an inherent, natural compression that engineers have used (consciously or not) for over half a century.

There are other reasons for compression. With digital encoding, higher levels have less distortion than lower levels—the opposite of analog technology. So, when recording into digital systems (tape or hard disk), compression can shift most of the signal to a higher overall average level to maximize resolution.

Compression can create greater apparent loudness (commercials on TV sound so much louder than the programs because they are compressed without mercy). Furthermore, given a choice between two roughly equivalent signal sources, people will often prefer the louder one. And of course, compression can smooth out a sound—from increasing piano sustain to compensating for a singer’s poor mic technique.

Now let’s look at some compressor basics…

Compressor Types

Compressors are available in hardware (usually a rack mount design or for guitarists, a “stomp box”) and as software plug-ins for existing digital audio-based programs. Following is a description of various compressor types.

  • “Old faithful.” Whether rack-mount or software-based, typical features include two channels with gain reduction amount meters that show how much your signal is being compressed, and most of the controls mentioned above.
  • Multiband compressors. These divide the audio spectrum into multiple bands, with each one compressed individually. This allows for a less “effected” sound (for example, low frequencies don’t end up compressing high frequencies), and some models let you compress only the frequency ranges that need to be compressed.
  • Vintage and specialty compressors. Some swear that only the compressor in an SSL console will do the job. Others find the ultimate squeeze to be a big bucks tube compressor. And some guitarists can’t live without their vintage Dan Armstrong Orange Squeezer, considered by many to be the finest guitar sustainer ever made. Fact is, all compressors have a distinctive sound, and what might work for one sound source might not work for another. If you don’t have that cool, tube-based compressor from the 50s of which engineers are enamored, don’t lose too much sleep over it: Many software plug-ins emulate vintage gear with an astonishing degree of accuracy.

Whatever kind of audio work you do, there’s a compressor somewhere in your future. Just don’t overcompress—in fact, avoid using compression as a cop out for bad mic technique or dead strings on a guitar. I wouldn’t go as far as those who diss all kinds of compression, but it is an effect that needs to be used subtly to do its best.

To read the full article see:  Compressors Demystified

December 27, 2010

DJ-Tech U2 Station MK2 Review

Filed under: DJ — Tags: , , , , , , , , , , — audiofanzine @ 9:25 am

How about mixing MP3 files without a computer? That’s the goal of the DJ-Tech U2 Station MK2…

DJ-Tech’s commitment is to offer affordable products for the masses, as their slogan clearly states: “not only for DJ.” That’s why the manufacturer offers fun products that meet the needs of most people, from beginners to a bit more experienced DJs.

The U2 Station MK2 is the second version of the DJ-Tech U2 Station. The concept of this small mixer is to allow the user to mix digital audio files in MP3 format. Offered at a reasonable price, the device works as a full standalone mixer without any computer support.

The 14.1″ x 9.4″ mixer has a nice, black glossy finish — you can almost see yourself in it, but watch out for finger marks! The control elements have different quality levels: faders and EQ controls are not very smooth, but the backlit switches react very well and feel pleasant under your fingers.

Inside the product’s box you’ll also find a CD with Magix Audio Cleaning (SE). This software only works on a PC (no Mac version) running Windows 2000, XP or Vista (no Seven support). In fact, the software provided with the product is Audio Cleaning 9.02 while the current version is 16.00. I find it a bit awkward to provide such an old version. Nonetheless, this version is enough to convert your CDs into MP3, which is the program’s main purpose. However, Magix’s support reacted very fast and gave me version 16 for free, which I could install on Seven without a hitch. This software will allow you to import all your analog audio sources (vinyls, tapes, etc.), clean them (of noise, hum, etc.) and convert them into MP3 files so you can use them directly with the U2 Station. The unit hosts an internal USB sound card that allows you to connect all sorts of old analog gear to your computer.

Sound Sources

Each channel has four different inputs:

DJ-Tech U2 Station mk2

- Line/phono input (switchable via a small selector on the rear panel of the U2 Station). It’s an excellent idea to provide a facility that allows you to connect analog sound sources to a digital mixer. However, notice that not all functions are available for this input, most notably the BPM counter, which works only with digital audio.

- Two USB ports (A and B ) on the top panel. Note that it is not necessary to connect two USB devices to be able to mix: only one USB device is enough. You can use all songs included in the USB device with both players, fully independent from each other.

Your USB hard drives and keys must be formated in FAT 16 or 32, the only formats supported by the U2 Station. One might have expected an Apple iPhone/iPod connection, like on other DJ-Tech products, but that’s not the case this time around! You got it right, the mixer provides you with two independent MP3 players. They constitute the core of the U2 Station.

Bloody Sunday for My MP4!

The mixer’s main advantage is that it’s a fully standalone concept made possible thanks to the USB port for hard and pen drives with up to 250 GB in capacity. But watch out, there is an important limitation: this mixer can play back only MP3 files! It does support all possible MP3 files from 32 to 320 kb/s bit rates (CBR and VBR) but no other formats like AAC (.m4a), OGG, WAV, etc. All non-supported files will be ignored when you browse the files so they won’t disturb you during your search.

Now let’s take a closer look…

Conclusion

The small U2 Station MKII is a controller you can take with you anywhere. It’s an excellent product. It lacks almost nothing, all essential features are present, except for a button to sync the BPM and beat of the two MP3 players automatically. This feature, as well as AAC compatibility, is a real drawback considering that the product is designed for beginners who mix for their grandma and cousins. Otherwise, this mixer has everything a professional product does and it offers many features you’ll only find on much more expensive mixers or tools that use sophisticated software and require a computer.

Advantages:

  • Mixing from a USB device
  • No need for a computer
  • Possibility to mix real line/phono audio sources
  • Portability
  • Three good quality effects
  • Scratch function with the jog wheel
  • Crossfader (auto start, curve control)
  • Overall finish
  • Sound quality

Disadvantages:

  • Somme controls don’t work smoothly
  • No auto sync for the two MP3 players
  • No AAC file support
  • No iPhone/iPod connection

To read the full detailed article see:  DJ-Tech U2 Station MK2 Review

December 22, 2010

A Guitarist’s Guide to Multiband Distortion

If you’re a guitarist and you’re not into multiband distortion…well, you should be. Just as multiband compression delivers a smoother, more transparent form of dynamics control, multiband distortion delivers a “dirty” sound like no other.

Not only does it give a smoother effect with guitar, it’s a useful tool for drums, bass, and believe it or not, program material – some people (you know who you are!) have even used it with mastering to add a distinctive, unique “edge.”

As far as I know, the first example of multiband distortion was a do-it-yourself project, the Quadrafuzz, that I wrote up in the mid-’80s for Guitar Player magazine. It remains available from PAiA Electronics (www.paia.com), and is described in the book “Do It Yourself Projects for Guitarists” (BackBeat Books, ISBN #0-87930-359-X).

I came up with the idea because I had heard hex fuzz effects with MIDI guitar, where each string was distorted individually, and liked the sound. But it was almost too clean, yet I wasn’t a fan of all the intermodulation problems with conventional distortion. Multiband distortion was the answer. However, we’ve come a long way since the mid-’80s, and now there are a number of ways to achieve this effect with software.

How it Works

Like multiband compression, the first step is to split the incoming signal into multiple frequency bands (typically three or four). These usually have variable crossover points, so each band can cover a variable frequency range. This is particularly important with drums, as it’s common to have the low band zero in on the kick and distort it a bit, while leaving higher frequencies (cymbals etc.) untouched.

Then, each band is distorted individually (incidentally, this is where major differences show up among units). Then, each band will usually have a volume control so you can adjust the relative levels among bands. For example, it’s common to pull back on the highs a bit to avoid “screech,” or boost the upper midrange so the guitar “speaks” a little better.

With guitar, you can hit a power chord and the low strings will have minimal intermodulation with the high strings, or bend a chord’s higher strings without causing beating with the lower ones.

Now let’s take a closer look at some plugins…

Rolling Your Own

You’re not constrained to dedicated plug-ins. For example, Native Instruments’ Guitar Rig has enough options to let you create your own multiband distortion. A Crossover module allows splitting a signal into two bands; placing a Split module before two Crossover modules gives the required four bands. Of course, you can go nuts with more splits and create more bands. You can then apply a variety of amp and/or distortion modules to each frequency split.

Yet another option is to copy a track in your DAW for as many times as you want bands of distortion. For each track, insert the filter and distortion plug-ins of your choice. On advantage to this approach is each band can have its own aux send controls, as well as panning. Spreading the various bands from left to right (or all around you, for surround fans!) adds yet another level of satisfying mayhem.

Here a guitar track has been “cloned” three extra times in Sonar, with each instance feeding an EQ and distortion plug-in. These have been adjusted, along with panning, to create multi-band distortion.

And Best of All….

Thanks to today’s fast computers, sound cards, and drivers, you can play guitar through plug-ins in near-real time, so you can tweak away while playing crunchy power chords that rattle the walls. Happy distorting!

To read the full detailed article see:  A Guitarist’s Guide to Multiband Distortion

December 20, 2010

Propellerhead Reason 5 Review

What features does the fifth version of the historical Propellerhead software have to offer? Overview.

We won’t retell the full Reason story, but we must acknowledge that Propellerhead shows an impressive consistency in the sense that they never derailed from their original philosophy: provide a standalone application that doesn’t allow the integration of third-party software (however open to the outside world via ReWire) and provides almost anything you need to produce electronic music.

The launch of Record (review to come) reinforces Reason’s position: instead of importing audio recordings into Reason, they can be embedded into Record, which is meant to remedy Reason’s “deficiencies.”

The fifth version of the virtual studio includes virtual synths, samplers, effects, etc., as well as some improvements and new features. Let’s have a look.

Introducing Reason 5

Propellerhead Reason 5

Reason 5 comes in a box including a DVD, a quick start guide (no printed user’s manual but an HTML help instead…) and a sheet of paper with the license and registration numbers required to activate the software and have access to updates. Note: it’s a good thing that the manufacturer tries to save paper not providing too many printed documents. But if that’s case, why do they deliver the product in such a big cardboard box? Considering the number of products sold, isn’t it a big waste of paper? I don’t really get it…

There’s no need to comment on the installation: everything is clear enough so anyone can open their first project after just fifteen clicks or so.

Powerful Sampler

Propellerhead Reason 5

Each new version brings with itself some graphic and useful improvements. On the top of the rack, you’ll find four buttons to open/close advanced audio and Midi parameters, as well as a Big Meter that can be set as a VU, PPM, Peak, VU+Peak, or PPM+Peak meter with in/out channel selection. Yes, indeed: Reason 5 finally supports audio, not for track recording like a sequencer but for making its samplers “real” samplers. Actually, many manufacturers misuse language when they state that their sample players/editors are real samplers — regardless of the incredible possibilities they provide.

It’s different at Propellerhead: with this new version, the NN-XT, NN19, Redrum and Kong (new module, see below) can record audio from any input or directly from one of the rack modules, with independent monitoring of the incoming signal. You can even route audio data directly within the computer, using Soundflower, for example (you can also create a loop with the audio card but it’s not that practical). The ability to sample the modules could inspire many manufacturers to use Reason’s possibilities to create sample banks from its very versatile instruments. Not that this wasn’t possible before, but you needed ReWire, external editors, etc.

Propellerhead Reason 5

Now it all happens inside. Select the input or the module by routing it to the sampling input on the rear side of the rack, click the waveform button (or use the tools window) and it will start recording immediately. By the way, a control to start recording manually would be much appreciated. We can imagine some extreme setups, considering that Sampling supports any stereo signal: you can rig modules and get the signal out of the main out (or the sends) of a mixer connected to several mixers, etc. So the internal possibilities are actually… endless. Once the signal has been recorded, click the Edit button to open the integrated sample editor.

Propellerhead Reason 5

Waveform display, selection, loop options, crop, normalize, invert, fade in/out and three play modes (normal, loop, forward/backward loop): only basic features (you feel like using a good old hardware sampler) but it’s enough to prepare a sample. Afterwards, you can make all resynthesis editing in any module, in which case the samples become available for all compatible instruments — including the outside world (AU, VST, etc.). Although the samples are saved by default with the actual song, you can export them to any WAV compatible tool.

Now let’s take a closer look…

Conclusion

The main advantage this version has to offer is the introduction of real sampling within the program and in all modules that deal with audio data. Hats off Propellerhead! This will hopefully have some impact among competitors. However, we also hope for an update (or a future version?) with a more sophisticated sample editor providing more features.

When it comes to new modules, Dr Octorex is very disappointing, considering that it cannot play several loops simultaneously; but, on the other hand, Kong does a very good job if you keep in mind that Reason is a software tool dedicated primarily to electronic music production. We still miss the possibility to have real pads of four layers each.

Regarding all other new features, the development team has been proving its mastery for years, and Reason 5 takes full advantage of this fact — just like all its predecessors. In short, if you want to record samples directly into a module to use them immediately, or if you want a powerful instrument dedicated to drum sound design, Reason 5 is the tool for you. If you are still hesitant, the manufacturer offers a free demo version so you can try it out.

Advantages:

  • Sampling embedded directly into the modules
  • Integrated sample editing
  • Kong module
  • Many drum samples
  • Three different sound synthesis engines in Kong
  • Dr OctoRex module
  • Improved editing
  • Possibility to export the samples recorded
  • Multicore support
  • Blocks
  • Multitrack Midi recording
  • HTML help

Drawbacks:

  • No simultaneous Rex loop playback in Dr OctoRex
  • No real four-layer pads
  • Snare drum PM and bass drum PM modules not very convincing
  • Sample recording cannot be triggered manually
  • No printed manual
  • Still no 64-bit Rewire

To read the full detailed article with sound samples see:  Reason 5 Review

December 17, 2010

Gifts for Geeks

Clock is ticking, and there is still time to please and be pleased. Here are some ideas for Christmas gifts for musicians and gear heads to fit all tastes and wallet sizes.

Computer Music

Line 6 MIDI Mobilizer

Line 6 MIDI Mobilizer : and your iThing speaks MIDI

Together with an Apple iPhone, iPad, or iPod touch, and the free MIDI Memo Recorder app, MIDI Mobilizer can play, record, and backup MIDI information any time, any place. Whether you want to capture a quick musical idea or back up the settings of all your MIDI gear, MIDI Mobilizer is a simple and compact solution for everything MIDI.  Price: $70

Peavey AmpKit Link

Peavey AmpKit Link :

Turn your iPhone into a virtual amp for $30. The sound quality is fair considering the price. The marketing strategy of offering a free amp and then have us pay for additional amps is not so bad, considering that guitar players usually have their favorite amps and do not play with 15 different models.

Plugin Lexicon

Plugin Lexicon :

The new software package makes all the effects processing of Lexicon’s PCM96 available as a plug-in designed to add “inspirational new sounds to a user’s DAW that are not available anywhere else.”  The PC- and Macintosh-compatible PCM Native Effects Plug-In Bundle is designed to work with DAWs like Pro Tools and Logic, as well as with any other VST, Audio Unit or RTAS-compatible host.  Price: $1200.

Apogee One

Apogee One : All in one in your pocket

ONE is described as a single input, stereo output USB music interface designed to work seamlessly with Apples iTunes, GarageBand, Logic, Final Cut or any Core Audio compliant application on a Mac. Unlike any product in its category, ONE features an internal reference condenser microphone, ideal for capturing inspired musical moments, according to Apogee. ONE also includes a microphone preamp, an instrument input for guitar, bass, and keyboards, and a studio-quality stereo output for headphones or powered monitors.  Price:  $249

 

Native Komplete 7

Native Komplete 7 : The Bundle of the Decade?

The latest version of the Komplete bundle combines a range of NI products, while the Komplete 7 Elements collection is designed to set a new price point for music production enthusiasts on a budget.  The seventh generation of Komplete now comprises 24 individual products, including the latest Reaktor 5.5 version as well as the new Reaktor Prism, Rammfire, Reflektor, Traktor’s 12 and Vintage Organs. Other products now contained in Komplete include the Abbey Road 60s Drums vintage drum library, the performance effect The Finger, the electric pianos and an electric bass by sampler Thomas Scarbee, the four acoustic pianos from the Classic Piano Collection, the cinematic Acoustic Refractions instrument and the Reaktor Spark synthesizer, amounting to about 10,000 sounds and 90 GB of studio-grade sample material overall.  Price: $559.

Guitar Pro 6

Guitar Pro 6 :

Version 6 is definitely a major update for Guitar Pro. What used to be a small software tool has become the ultimate reference in its category thanks to its intuitive user interface, well thought-out features and an absurdly low price. Should you upgrade your previous Guitar Pro version for $29.95? Yes, a thousand times yes! You’ll benefit from a better design and a much better sounding and efficient audio engine than in previous versions. Should you buy the full version for $59.95 if you don’t own a guitar tab editor? Yes, a thousand times yes!

Pro Tools 9

Pro Tools 9 : Compatible Soundblaster (among others) !

Pro Tools 9 is an open platform that doesn’t require an Avid/M-Audio interface anymore, but can work with or without any Core Audio or ASIO compatible interface – on Mac AND PC.  The new version enables bigger mixes with more tracks, and pro features including Automatic Delay Compensation, multitrack Beat Detective, full Import Session Data dialog, DigiBase Pro, and other separately priced add-ons—now standard.  Price: $599 for the full version.

Pianoteq Play

Pianoteq Play :

Pianoteq Play is a virtual piano based on the physically modeled Pianoteq software instrument, appraised by many musicians for its close intimacy and responsiveness.

Modarrt says there is no need to tweak settings and parameters, as Pianoteq Play is delivered with “perfectly designed instruments.”  Pianoteq Play supports all Pianoteq instruments, and the grand pianos K1, C3, and M3 are embedded.  Price:  $99

RME Babyface

RME Babyface :

RME succeeded in launching a compact and rugged interface with remarkable sound quality. At about $750, this baby provides two quality mic preamps and converters, ADAT in/out, a jog wheel, a transport bag, and a pair of nice-looking VU-meters. Add TotalMix FX —the virtual mixer that allows you to manage all 22 channels and process the signals (EQ, filter, reverb, and echo)— to the package and you get the best mobile audio interface on the market.

Akai APC 20

Akai APC 20 : Enter the Matrix

Yes, the APC40 is much more comprehensive than the APC20! But if you have only $200 for a Live controller, the APC20 has only one competitor in the form of the Novation Launchpad. The latter is less expensive but doesn’t have any faders, which makes it less interesting…

DJing and Live Sound

Traktor Kontrol S4

Traktor Kontrol S4 :

Combining an extended version of the existing Traktor Pro software with a dedicated hardware controller, the Traktor Kontrol S4 is aiming to provide an all-in-one solution for digital DJs. The controller comprises a four-channel mixer, an integrated 24-bit/96kHz audio interface based on NI’s Audio 4 DJ, and interface sections for looping, cueing, track browsing and effects control.  Price: $1000.

Hercules DJ Console 4-MX

Hercules DJ Console 4-MX :

Hercules launched this year the newest version of their DJ Console line for Pro DJs, the DJ Console 4-Mx, a controller featuring large jog wheels (each equipped with touch sensor) a built-in audio interface tailored for DJing, and control over 2 and 4 virtual decks.  The DJ Console 4-Mx has steel and aluminium crafted body with a variety of controls including 89 controls in 2-deck mode and 150 controls in 4-deck mode.  Price: $450.

Pioneer DJM-2000

Pioneer DJM-2000 :

Let’s be clear: this is a great piece of gear! Well thought-out, nicely finished and with a great sound, it offers countless possibilities to allow the most demanding DJ’s to have endless fun. With this product, Pioneer targets night clubs with big budgets who want to offer the best to their DJ’s. The latter will have the possibility to prepare their sets before performing, and to come to the club with only a CD or a USB key — no need for a computer.  Price: $2500.

Denon DN-X1700

Denon DN-X1700 :

The DN-X1700 is a four-channel tabletop mixer with rubberised knobs, 60mm Alps K Series channel faders, 45mm FLEX cross fader, a color LCD display, extended 24-point LED channel and output metering, and LED ring metering around the control knobs.  In operation, the principal features related to the power and flexibility of the DN-X1700 are its Matrix Input Assignment with digital input and MIDI/USB audio, independent and parametric three-band EQ with Kill on each channel, and dual independent EFX processors.  Price: $1800.

Fender Passport 500 Pro

 

Fender Passport 500 Pro :

The eight-channel Passport 500 PRO is the new top-of-the-line Passport system:

  • A port that lets you record your performance with CD quality (.wav) straight to a USB flash drive.
  • CD-quality .wav and mp3 file playback.
  • Sub-out jack for an external powered sub-woofer.
  • Redesigned speaker system with 10″ woofer and improved clarity.
  • Price: $1000.

 

Presonus StudioLive 24.4.2

Presonus StudioLive 24.4.2 :

StudioLive 24.4.2 sports the same user interface, feature set, and I/O configuration as the StudioLive 16.4.2 but with several additions and enhancements. The main difference is that the new mixer provides 24 input channels and 10 aux buses, whereas the StudioLive 16.4.2 has 16 channels and 6 auxes. In addition, the new mixer’s Fat Channel has fully parametric EQ, rather than semi-parametric, and the gate and limiter have been enhanced. Instead of one stereo 31-band graphic EQ on the main bus, you get four dual 31-band graphic EQs that can be assigned to the mains, subgroups, and aux buses.  Price: $3,300.

To see many more gift ideas see:  Gift for Geeks- Xmas Shopping 2010

Multitrack Drum Libraries

Filed under: Drums/Percussion, Samplers, Software — Tags: , , — audiofanzine @ 3:51 pm

More drum libraries are showing up in multitrack format from companies like Discrete Drums, Wizoo, East-West, Reel Drums, etc. Although these are sold on the basis of being useable out of the box for drum parts (with the additional advantage of being mixable), I see them more as a gold mine of raw materials for creating custom drum loops. Being able to process individual tracks separately is certainly a major advantage when deriving loops from multitracked parts, and of course, proper looping allows using these parts at different tempos.

For example, I just did a “remix” of the Discrete Drums sample library for the company, who had received numerous requests for “dirtier,” lower-resolution versions aimed for more hardcore hip-hop and dance musicians. Hard disk recording programs are ideal for doing this type of remixing; this article will concentrate on using Sonar, but most techniques apply to other programs, and specific examples are given for Acid as well.

Dealing with Human Error

Drum libraries played by real drummers are great, because of the additional “human feel” compared to using machines. But due to timing inaccuracies, it sometimes takes a little tempo tweaking to line up measure markers with downbeats.

 

This illustration shows a loop whose stated tempo was 79 BPM, but in the upper view, note how the downbeat at the beginning of measure 9 (the loop end point in this particular case) hits a little early compared to the measure marker. In the bottom view, changing the tempo to 79.03 BPM places the measure marker at the downbeat’s exact beginning.

If you need to change the tempo compared to the original file, then time-stretching becomes necessary. Sonar has a built-in time stretch function that’s very similar to the one in Acid; Cubase SX has a nifty ReCycle type feature that works particularly well with drum loops. For programs that don’t stretch, you have three options if you want to change tempo:

* Import the file into ReCycle, change the tempo as desired, then export back to WAV or AIF.
* Import the file into Acid, Sonar, or a recent version of Sound Forge, “acidize” the file, then export.
* If the tempo change is small, change the pitch withoutcompensating for duration. Tranposing pitch upward will speed up the tempo, transposing down will slow it. For small changes, the pitch difference may not be noticeable (and in some cases, may be desirable).

After tweaking the track mix and setting the tempo, render the file to a stereo loop. Import this into your hard disk recorder or a digital audio editor and set looped playback mode. If there’s a click when the loop jumps back to the beginning, add a 4 ms fade-out to eliminate clicks, and if absolutely necessary, a 2-4 ms fade-in. In drastic cases, I use Sonic Foundry’s Click Removal DirectX plug-in to remove clicks at transition points.

Now let’s take a closer look…

Groove Clip Tricks

* All programs that use slicing to do time-stretching work most efficiently when speeding up rather than slowing down. Therefore, if you want to create a loop that works well from, for example, 100 BPM to 120 BPM, you’re better off creating it at 100 BPM and speeding it up than starting at 120 BPM and slowing it down.
* Editing markers is usually mandatory for drum loops played by human drummers instead of machines, as editing can compensate for any timing variations that interfere with the stretching process.
* Before getting too much into editing, try adjusting the Basic Slices and Transient Detection sliders first. Often choosing different values will solve flamming and other problems, without the need for editing.
* Use the lowest Slice Rhythmic Value possible (e.g., 8th note instead of 16th note), consistent with good sound. Extraneous slices can cut off drum decays. This is particularly annoying with kicks, as you lose some of the fullness and “ring.”
* Sonar will endeavor to keep any markers that you’ve moved manually in their assigned positions, so you can experiment at any time with the Slicing and Transient Detect controls without losing the positions of your carefully-placed markers.
* When you save a Sonar song or bundle, it retains all the Groove Clip parameters. To save a Groove Clip in acidized format for use other programs, simply drag the file to the desktop; it will be copied and saved with its groove parameters intact. However, you will likely want to rename it, as Sonar generates the name automatically.

To read the full detailed article see: Multitrack Drum Libraries

December 13, 2010

Focal CMS 40 Review

Filed under: Monitors, Speakers — Tags: , , , , , , , , — audiofanzine @ 1:07 pm

These monitor speakers ought to be of interest to mobile home studio owners or people who have a very small room to play or mix music. Following the Focal CMS 65 and CMS 50, the CMS 40 is even smaller but not less appealing…

We all know Focal for their top-range speakers with undeniable qualities but, unfortunately, not affordable to everyone. That’s why the manufacturer decided to launch a more affordable series a couple of years ago. It included two models, the CMS 65 and the CMS 50 equipped with 6.5″ and 5″ woofers respectively. A subwoofer is also available for brown noise fans. Both models received a warm response from users, so now Focal decided to launch an even smaller and less expensive speaker that benefits from all the qualities of its big brothers. Did they succeed?

Small but Powerful

Focal CMS 40

When we were unpacking the speakers, the first thing that surprised us was the very compact size of the CMS 40: 9.39″ x 6.14″ x 6.10″ and about 12 lb. In other words, these speakers are very small and can be easily transported — which is good news, particularly considering they provide the same high manufacturing quality as their big brothers. On the other hand, they are also quite heavy — the price to pay for good quality manufacturing, I guess… You get the same reinforced and damped aluminum housing, black powdered paint and protection grills for both drivers: a 4″ woofer made out of a polyglass membrane and an aluminum/magnesium reversed-dome tweeter. Once you comfortably set up the speakers, you can remove the protection grills and fix the tweeter phase plugs. According to the manufacturer, this improves their response. Since Focal is generous with accessories, you’ll also find a decoupling table stand and four rubber feet in the box, as well as two height-adjustable spikes to tilt the speakers forward or backward, or even to the sides! It’s important to mention that all CMS are magnetically shielded so you can easily place them next to a cathode screen monitor.

Regarding speaker position, Focal advises the user to keep at least 1.3 ft from the CMS 40. The rear-panel fixing points allow you to mount them on a wall or any other support. The rear vertical connections allow you to mount the speaker directly against the wall, which is acoustically possible since the bass reflex housing is front ported. A very good feature for home studio owners who work in a “closet.” You can use the CMS vertically, horizontally or upside down in order to keep the tweeters at the same height as your ears.

In short, the small CMS is adaptable to almost any environment — a great asset. The manufacturer states that this speaker is not very sensible to the critical acoustic environments of non-optimized rooms!

Settings and Features

Focal CMS 40

Let’s start with some good points: the power on/off switch and the volume control (-66 dB to 0 dB) are conveniently placed on the front panel, where you also have power and clip LEDs. On the rear panel you’ll find a balanced XLR input (10 kOhm), an unbalanced RCA input (47 kOhm) and the power socket. You can set the input level to +4dBu, -10dBV or 0dB.

Adjusting the speakers’ response is very simple with two filters: a low shelve (0 Hz – 450 Hz) with -/+2 dB amplification/attenuation and a high shelve (4.5 kHz -20 kHz). The frequency response stated by the manufacturer is 60 Hz to 28 kHz (-/+3 dB). Two integrated amps of 25 watts each (one per transducer) deliver 97 dB as maximum SPL level (@ 1 m).

Unlike the CMS 50 and CMS 65, the CMS 40 has no real low-cut filter so you’ll have to set the cutoff directly on the subwoofer (the CMS Sub for instance!) at approx. 60 Hz. However, we tested the monitors without a subwoofer since we had already tested it with the CMS 50 earlier this year.

Let’s listen to the sound…

Conclusion

Focal introduces a very surprising compact speaker to extend their CMS range, whose previous models were very appealing. The CMS 40 is no exception with its irreproachable manufacturing quality, plentiful accessories and remarkably well-balanced sound. Considering its 4″ woofer, the CMS 40 delivers a clear and dry low-end and very present and analytic mids. The high-frequency response is also good, just like the CMS 65 and CMS 50. We noticed that the sound is less hollow than with other speakers and that the CMS 40 sound more linear than the ADAM A3X, even if the frequency response of the ADAMs is wider in the low and high ends of the spectrum.

The CMS 40 do a very good job when mixing and they reveal details you could miss with other speakers. We had no surprises listening to our mixes through other speaker systems, which is a very good point. Moreover, the CMS 40 has a wide sweet spot and can be used in a room with poor acoustic properties. At $800/pair, this monitor speakers are highly recommended for mobile home studio owners or people working in a very small room who want to buy a well-built and faithful speaker pair.

Advantages:

  • Well balanced sound
  • Accurate mids
  • Limited but precise low frequency response
  • Sturdiness and manufacturing quality
  • Adjustable spikes, removable grills and decoupling table stand
  • Very compact size
  • Affordable price
  • On/off switch and volume control on the front panel

Drawbacks:

  • Quite heavy
  • No 1/4″ jack input

To read the full detailed article see:  Focal CMS 40 Review

 

December 10, 2010

Making Equalization Work For You

Filed under: Equalizers — Tags: , — audiofanzine @ 11:31 am

Equalization is one of the most important and powerful tools in the recording enthusiast’s arsenal, yet too many people adjust equalization with their eyes – not their ears. For example, one time after doing a mix, I noticed the client writing down all the EQ settings I had made. When I asked why, he said it was because he liked the EQ and wanted to use the same settings on these instruments in future mixes.

Wrong! EQ is a part of the mixing process; just as levels, panning, and reverb are different for each mix, EQ should be custom-tailored for each mix as well. But to do that, you need to understand how to find the magic EQ frequencies for particular types of musical material, as well as what tool to use for what application.

There are three main applications for EQ:

  1. Problem-solving
  2. Emphasizing or de-emphasizing an instrument in a mix
  3. Altering a sound’s personality

Each application requires specialized techniques and approaches.

Problem Solving

EQ can fix a variety of problems, and the tool of choice is usually a parametric equalizer, which consists of a limited number (typically 1-8) of frequency bands (Fig. 1). For each band, you can change not just the degree of boost or cut, but also the frequency at which this boosting or cutting occurs, as well as how wide a range of frequencies is affected – from very sharp to very broad. (“Pseudo-parametric” equalizers omit the bandwidth control, and the lack of this can seriously hamper the experienced EQ aficionado. Arguably, manufacturers err on the side of too narrow a fixed bandwidth, which makes it difficult to do subtle changes.)

Fig. 1: The PSP Audioware MasterQ is a high-quality equalizer plug-in for digital audio workstations with seven frequency-altering stages. Going from left to right, these are a highpass filter, low frequency shelving EQ, three parametric stages, high frequency shelving EQ, and a lowpass filter.

As one example of how you’d use a parametric EQ, slicing a sharp notch at 60Hz (50Hz in Europe) can knock hum out of a signal; trimming the high frequencies can remove hiss. Another common problem is an instrument with a resonance or peak that interferes with other instruments, or causes level-setting difficulties. Following is a procedure that takes care of this situation.

Several years ago I produced an album by classical guitarist Linda Cohen (Angel Alley, which was re-released on CD). She had a beautiful instrument with a full, rich sound that projected very well on stage, thanks to a strong body resonance in the lower midrange that caused a major level peak. However, recording was a different matter from playing live. If levels were set so the peaky, low frequency notes didn’t overload the recording media, the higher guitar notes sounded weak by comparison.

Although compression/limiting was always an option, it altered the guitar’s attack; while this effect might have gotten lost in an ensemble, it stuck out with a solo instrument. A more natural-sounding answer was to use EQ to apply a frequency cut equal and opposite to the natural boost, thus leveling out the response. But there’s a trick to finding problem frequencies so you can alter them; the following works like a charm.

  1. Turn down the monitor volume – the sound might get nasty and distorted during the following steps.
  2. Set the EQ for lots of boost (10-12dB) and fairly narrow bandwidth (around a quarter-octave or so).
  3. As the instrument plays, slowly sweep the frequency control. Any peaks will jump out due to the boosting and narrow bandwidth. Some peaks may even distort.
  4. Find the loudest peak and cut the amplitude until the peak falls into balance with the rest of the instrument sound. You may need to widen the bandwidth a bit if the peak is broad, or use narrow bandwidth for single-frequency problems such as hum.

This technique of boost/find the peak/cut can help remove midrange “honking,” strident resonances in wind instruments, and much more. Of course, sometimes you want to preserve these resonances so the instrument stands out, but many times applying EQ to reduce peaks allows instruments to sit more gracefully in the track.

Digital workstation EQ, as found in hard disk recording systems, can be particularly effective due to its precision. In one of my more unusual projects, I needed to remove boat motor noise from some whale samples. Motor noise is not broadband, but exists at multiple frequencies. Applying several extremely sharp and narrow notches at different frequencies took out each component of the noise, one layer at a time, until the motor noise was completely gone.

This type of problem-solving also underscores a key principle of EQ: it’s often better to cut than boost. Boosting uses up headroom; cutting opens up headroom. In the example of solving the classical guitar resonance problem, cutting the peak allowed for bringing up the overall gain to record a much higher overall level.

Let’s take a closer look…

Other EQ Tips

Problem-solving and character-altering EQ should be applied early on in the mixing process, as they will influence how the mix develops. But wait to apply most EQ until the process of setting levels begins; remember, EQ is all about changing levels – albeit in specific frequency ranges. Any EQ changes you make will alter the overall instrumental balance.

Another reason for waiting a bit is that instruments EQ’ed in isolation to sound great may not sound all that wonderful when combined. If every track is equalized to leap out at you, there’s no room left for a track to “breathe.” Also, you will probably want to alter EQ on some instruments so that they take on more supportive roles. For example, during vocals consider cutting the midrange a bit on supporting instruments (e.g., rhythm guitar) to open up more space in the audio spectrum for vocals.

Finally, remember that EQ often works best when applied subtly. Even one or two dB of change can make a significant difference. However, inexperienced engineers often do something such as increase the bass too much, which makes the sound too muddy, so they increase the treble, and now the midrange sounds weak, so that gets turned up…you get the idea. One of your best “reality checks” is an equalizer’s bypass switch. Use it often to make sure you haven’t lost control of the original sound!

To read the full detailed article:  Making Equalization Work For You

December 8, 2010

RME Babyface Review

Most manufacturers have been adding compact audio interfaces to their product range for several years, and now is time for RME and its Babyface. Many mobile musicians and sound engineers have been eagerly waiting for this new USB2 compatible interface…

This end of year is full of new launches at RME: the high-end Fireface UFX (already reviewed by AudioFanzine) and the Babyface, which belongs to the affordable line of RME products. The word “affordable” is relative, of course, considering that the Babyface’s price tag is nearly $750… However, the Babyface is the German manufacturer’s most compact and affordable external interface and it will surely appeal to mobile home-studio owners searching for quality.

Inside the box you’ll find the user’s manual, a breakout cable and an extension cable to add inputs and outputs to the Babyface (see below), a USB2 cable, a nice transport bag to carry the interface, the cables and a mic (for example), and the Babyface itself with its blue and gray finish. The interface is quite compact (3.9″ x 1″ x 6.3″) but it is heavy enough (1.1 lb.) to sit stably on your desk — it feels sturdy. This impression is reinforced by the metal housing with the typical RME blue finish. Only the knobs and the jog wheel are made out of plastic. The wheel doesn’t seem to be too tough; the first few months of intensive use will show if it has what is takes…

Plug-in Baby

RME Audio Babyface

In spite of its compact size, the Babyface offers comprehensive connections: two mic inputs on XLR connectors, line outputs (on XLR connectors as well), MIDI in/out on 5-pin DIN connectors, and a headphones minijack output (which can also be used as line out). All connections are routed through the breakout cable, linked to the Babyface via a 25-pin D-Sub connector, similar to the ones on VGA graphic cards. On the interface itself you have an instrument input, which replaces the second mic input when activated via the TotalMix FX software, and a second phones out which is electrically linked to the first one. This means that the maximum output volume decreases when two headphones are connected at the same time, and also that both outputs deliver the same audio signal. In other words, you can’t send different mixes to the headphones. You’ll also find an ADAT Toslink input and output, which is a rather nice surprise considering the size and price of the interface. The ADAT option allows the user to connect an external converter and add 8 in/out channels. Nice! Finally, the interface features a connector for an external PSU (not included) and a USB cable with two connectors, in case the USB bus of your computer doesn’t provide enough current (the manufacturer states that the Babyface requires 300 mA).

RME Audio Babyface

On the top panel you’ll find some LEDs and buttons to control certain parameters without having to use the TotalMix FX software. The jog wheel allows you to control the gain of both analog inputs (simultaneously or separately), the volume of the main line outputs or the phones out level. You can select the mode (In, Out or Phone) using the select buttons underneath the jog wheel. A simple click on the jog wheel allows you to activate the dim function (temporary volume reduction) while in Out or Phone mode. The last LED shows the sync status of the digital clock. The source of the clock can be internal or external (via ADAT and S/PDIF).

Two 10-segment LED meters show the level at the inputs or outputs, which is a very valuable feature considering the size of the interface. Usually, manufacturers use only one or two LEDs for similar products… Well done RME!

Now, let’s take a look at the software package included…

Conclusion

RME succeeded in launching a compact and rugged interface with remarkable sound quality. At about $750, this baby provides two quality mic preamps and converters, ADAT in/out, a jog wheel, a transport bag, and a pair of nice-looking VU-meters. Add TotalMix FX —the virtual mixer that allows you to manage all 22 channels and process the signals (EQ, filter, reverb, and echo)— to the package and you get the best mobile audio interface on the market. It obviously has some drawbacks, like the poor precision of the gain controls, the fact that the two headphones outputs are not independent and the sturdiness of the jog wheel, but nothing is perfect in this world…

Advantages:

  • Quality of the preamps and converters
  • ADAT input and output
  • TotalMix FX with EQ, reverb and echo
  • 10-segment LED level meters
  • Size (it does matter!)
  • Metal housing
  • USB powered
  • Convenient jog wheel and buttons
  • Nice transport bag
  • Xmas is coming soon

Drawbacks:

  • Input gain control in 3 dB steps
  • Will the plastic jog wheel survive over the years?
  • The two headphone outputs are not independent
  • I have to send it back

To read the full detailed article see:  RME Babyface Review

December 6, 2010

Korg PS60 Review

Filed under: keyboards, Synthesizers — Tags: , , , , , , , , , — audiofanzine @ 1:44 pm

While the market of low-budget synthesizers has never been so flourishing, Korg launches a performance synth conceived for live applications that require spontaneity and real-time options. Let’s step into the details…

3 pm on a gray autumn Saturday. Thick smoke fills the dark rehearsal studio when a hoarse voice raises from behind the drum kit…

“Hey, let me know when you are done turning knobs and playing with your touchscreen so you can finally give us some Rhodes and finish your synth solo!”

- I’m almost done, I just have to insert a program into the 2nd channel, edit a keyboard split and adjust the FX sends because I need an overdrive for the piano and a subtle delay for the solo part…

- What? A short delay? You’ve been setting your gadget there for hours. We’ve played only five songs and we still have 25 to go! We must pack our gear in two hours, drive 60 miles, mount again, make the soundcheck and start the show at 9 pm tonight…

- OK, I’m ready. Three, four, dzoiiiiiiing!!!

- What’s that chord you’re playing? Don’t you know “All by myself” is in A sharp?

- Darn, I forgot to transpose! I hate A sharp: too many black keys! Wait a minute guys, I just have to push the edit button, browse the transposition page… hm, wait, where is it? On the MkII, it was the 8th on the third-level to the right, but with the MkIII, it’s…

- I’m gonna kill you! I can’t stand your black and white keys, your cables hanging around, your twisted keyboard stands that keep on ripping my car’s leatherette seat covers…

 

Many keyboard players have experienced this when they still don’t master their brand new workstations yet, in spite of several weeks dedicated to getting to know their instrument. Complexity overcomes spontaneity! But what options do we have left, except for stacking several synths to have everything at our fingertips and edit splits and layers faster than Keith Emerson can play a fill over a five-octave keyboard or Jim Morrison can drink five bourbons… The Korg PS60 aims to be the answer: compact, quick, affordable, fully packed with ready-to-use sounds, and very editable. Let’s see if it holds true!

Double Six

Korg PS60

The PS60 is a very compact five-octave keyboard with the Korg-typical pitchbend/modulation joystick placed above the keys. Not very long nor heavy at all, due to the fact that it’s made out of plastic with a very nice and glossy finish. On the other hand, you’ll have to protect the device to take it on the road because it’s no tank… On the front panel there are many controls spread over a quite unusual layout. From left to right, you’ll find the joystick with a Hold key that allows you to hold the value corresponding to its position on the modulation axis, i.e. you can lock the return spring that brings the joystick back to the center position. You’ll also find a volume control and a key dedicated to Leslie simulations for organ sounds. But it has a fixed assignation… There’s also a row of keys for octave and half-tone transposition. Well done! Just above that, you’ll find a control section to store/recall performances pushing only one or two buttons. In the middle, a small 2×16 character, gray-blue LCD is placed above the selection keys for mode and performance selection.

But the most original section is clearly the control field’s right section. It allows you to select, turn on/off and mix on the fly two sets of six separate sound layers. In order to do that you get six rotary controls, 12 program-change keys, six channel on/off keys, a quick-edit selector for four parameters (volume, octave and two FX sends), and a split control section. You can quickly stack six program layers. When you activate the split key, you get two sets of six layers on both sides of the split point. The six parts are sorted by category: acoustic piano, electric piano, organ, strings, brass, and synth. Further on to the right, you’ll find nine controls and two keys that allow you to edit directly the two master effects and the global EQ to adapt the sound to the music. Once you are satisfied with the results you can save everything in no time. There’s no need to say that the handling is very easy and practical. But as you will see later on, the PS60 is not only a spontaneous stage keyboard but also a really comprehensive synth.

Korg PS60

Now, let’s take a quick look at the rather spartan rear panel: external PSU connector (normal for a low-budget product), on/off switch, stereo analog out, MIDI in/out, and a pair of multifunction foot controllers. Nothing revolutionary for today’s standards… The minijack 1/8″ headphones out is on the front panel. Nice! Let’s close this short overview by noting that the five-octave keyboard is velocity sensitive but it doesn’t support aftertouch, and it sports half-weighted keys with better quality than its competitors in the same price range.

Sound Set

Korg PS60

The PS60 uses a sound synthesis based on samples taken from the M3/M50 series in a compressed PCM ROM equivalent to 49 MB at 16 bits/48 kHz. You get 120 voices of polyphony and 12 simultaneous channels of multitimbrality. The unit always works in performance mode, which means that it always uses an arrangement of one or two sets with six sound layers. Each layer includes one of the 512 internal programs, including 440 factory-loaded ones. Each program includes a small demo to be chosen from 383 audition riffs which cannot be programmed. The sound samples provided with this review use these riffs to allow you to get a quick overview of the pop/rock oriented sound possibilities.

You’ll find some multisample acoustic pianos in different stereo variations (with or without sustain pedal and different tempered tunings) and a piano from the M1: typical sounds of older Korg workstation generations that cannot come close to the level of multisamples used by modern workstations. The multisample electric pianos sound much better, especially two Fender and one Wurlitzer sampled with three velocity steps. The Clavinet sounds are ok, especially thanks to the FX section. You get eight electric organs, which cover most music styles from smoky jazz to spellbinding gospel and distorted rock.

Korg PS60

You’ll also find two strings sections from previous Korg workstations: a very wide stereo ensemble and a small, slightly aggressive section. Choirs are well represented with four pop and classic multisamples provided in three variations. Brass sounds do not have an homogeneous quality. On the one hand you have the very nice, stereo pop section, the trumpet, trombone, French horn, flute, and clarinet sounds, but you also have three miserable saxophones. Even though the guitar & bass category doesn’t belong to the six instrument families on the front panel, you’ll find acoustic/electric bass and guitar sounds all the same. Bass guitars sound pretty good but guitars are disappointing: dead attacks, short held notes, audible loop points. However, the excellent amp simulation effects save the day… You also get about 50 different waveforms in different variations (sawtooth, sinus, impulse and DWGS & VS waves) — tradition is not a meaningless word at Korg. On the other hand, you won’t find any drum kits; it’s a pity since they are sometimes very convenient…

Now let’s take a closer look…

Conclusion

The PS60 offers an interesting concept at a very affordable price. You get a rather good pop/rock sound selection that, honestly speaking, cannot compete with big workstations or high-class stage keyboards. The same applies to the sound synthesis parameter set that requires an external piece of software (which is provided, luckily). One thing that sets the PS60 apart from all those high-end, sophisticated products is that it is clearly superior when it comes to quickly stacking, splitting, mixing and editing different sound layers during live performances. This will appeal to nomad musicians who want to avoid damaging their budget and their back!

Advantages:

  • Short learning curve
  • Well though-out direct-access controls
  • Good sound quality
  • A real synth with multimode filters and modulation matrix
  • FX section with one insert per voice (except for strings)
  • Editor/library manager included
  • Quality standard dynamic keyboard
  • Very easily readable LCD
  • Compact size and light weight
  • Rewritable OS
  • Reasonable price

Drawbacks:

  • Limited direct access to some sound synthesis parameters
  • Rather annoying menu browsing
  • Only 20 performance memories
  • No sequencer nor arpeggiator
  • Keyboard without aftertouch
  • No drum sounds nor kits
  • Construction a bit fragile

To read the full detailed article see:  Korg PS60 Review

Older Posts »

The Shocking Blue Green Theme Blog at WordPress.com.

Follow

Get every new post delivered to your Inbox.