Studio

Wednesday, November 20, 2013

Anymix Pro AAX Now Available

Acclaimed surround-mixing plug-in for music and post-production now compatible with ProTools 11.

IOSONO is pleased to announce the release of Anymix Pro AAX. The successful surround mixing plug-in is now supporting Avid’s new AAX Native Format, ensuring a smooth workflow with new time-saving features for users of Avid’s popular digital audio workstation Pro Tools 11.

Anymix Pro AAX now also features support for Avid control surfaces via EUCON (System 5 series), Avid C|24 and ICON. The update also includes new automatable parameters and shortcuts for a better workflow.

“We have received great feedback on Anymix Pro, our users have been especially happy with time saving tools like the distant-dependent feature as well as the high-quality upmixing functions. We’re always trying to incorporate user feedback into our software and so we’re happy to now offer the much awaited AAX support,” says Katja Lehmann, Marketing Manager for IOSONO.

Anymix Pro AAX 64-bit for Mac and Windows is now available for EUR 299,00 in the IOSONO online shop at http://www.iosono-sound.com as well as selected distributors. The update will be free for all existing Anymix Pro customers already using the VST/RTAS version.

IOSONO

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Posted by Julie Clark on 11/20 at 02:20 PM
RecordingNewsSound ReinforcementStudioPermalink

Germano Studios Acquires Two Red 1 500 Series Mic Pre’s Through GC Pro

Germano Studios is known for bridging state-of-the-art technology and classic analog sounds, and the Focusrite Red 1 500 Mic Pre’s fit comfortably into both categories

Guitar Center Professional (GC Pro) recently sold two Focusrite Red 1 500 Series Mic Pre’s to Germano Studios, one of the world’s premier recording facilities.

These are among the first Red 1 units to be purchased since they recently began shipping. The acquisition of what is considered by many discriminating ears to be among the finest microphone preamplifiers ever designed will help support Germano Studios as a leader in bridging cutting-edge technology and classic analog sound.

The sale to studio owner Troy Germano took place in October and was handled by New York-based GC Pro Account Manager Tony Donelson.

The original Focusrite Red Range gained legendary status immediately following their release in 1992. The Red mic preamps rapidly became highly sought-after in the world’s top studios, with their unique signature sound and ability to perform faultlessly on a wide range of instruments and with almost any microphone.

Made in England like its forebears, the new Red 1 500 Series module carries on the legacy of the Red range.

“The Focusrite Red 1 500 Series mic pre’s are a perfect addition to our Studio 1,” states Troy Germano, who is also a three-time recipient of the prestigious TEC Award, in the category of studio design. “These new, stellar-sounding 500 series modules give the engineers another great option when recording.”

The two Red 1 500 Series mic pre’s fit perfectly with Germano Studios’ philosophy of including both cutting-edge technology and classic analog sound.

“Troy has one of the best recording studios in New York City, and one of the best anywhere,” states Donelson. “The Red 1 500 Series mic pre’s are brand new, and we wanted to make sure that we made them available to Troy as soon as we could.

“There’s been tremendous interest in them and demand for them ever since they were announced. That’s not surprising, because the Red 1 Series 500 mic pre’s are that good.”

GC Pro

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Posted by Julie Clark on 11/20 at 01:44 PM
RecordingNewsEngineerMicrophoneProcessorStudioPermalink

Jason Hook Of Five Finger Death Punch At Home With KRK Systems

Jason Hook of Five Finger Death Punch utilizes KRK Systems.

Jason Hook, lead guitarist for heavy metal band Five Finger Death Punch, is currently using KRK Systems’ VXT8 monitors and 12sHO sub in his home studio, where he holds the additional roles of record producer, songwriter and solo artist.

The skilled musician relied on his hard work and life experiences to record the band’s latest album, The Wrong Side of Heaven and the Righteous Side of Hell, Volume 2, at the fully equipped recording studio that he literally gets to call “home.”

A long-time user of KRK gear, Hook chose the VXT8s for their low resonance, improved structural integrity, extended low-end and slotted ports for reduced port turbulence.

He credits the sleek curvature of the design for providing excellent imaging characteristics and a wider sweet spot.

“I love KRK and have used them for years,” says Hook. “It wasn’t until after I became a Gibson artist that I found out KRK was part of the brand. That’s why when I upgraded my home studio, which is always a work in progress, I had to get my monitors from KRK. Basically, I do all of my music in there.

“It has all of my amps, pedals, guitars and recording gear, such as my Pro Tools HD rig. The KRK VXT8s definitely have the muscle I need for rock music.”

Like all KRK products, Hook’s VXT8s feature the company’s trademark yellow Kevlar woofer and visually striking enclosure design. The KRK12sHO 400-watt powered sub offers a strengthened version of the woofer.

According to Hook, one of the standout features of the 12sHO is the Bypass Footswitch Control, which allows users to defeat the sub and provide full-range audio to their recording monitors for use with a standard latching ¼-inch mono footswitch.

“This is what I used all through the making of the band’s last record, which is set for release on November 19th,” says Hook. “I have a pro-level system right here at my house that allows me to work at home at a level that’s compatible with a professional studio.”

Hook’s musical career got its kick-start in Oakville, Ontario, when a friend introduced him to what would become a favorite band.

“My neighbor came by the house with a few of the band’s records and asked me if I wanted them, since he wasn’t allowed to keep them at his own home. I remember saying, ‘Sure, what is it? What’s this band KISS?’,” he recalls. “I was looking at the cover, thinking ‘oh my god―that’s crazy!’”

“And it grabbed me immediately. It was extremely exciting. Plus, I liked the songs—they were simple, catchy and had a high level of energy. I remember thinking that KISS, which captured people’s attention with all the makeup and photos and images, was so much more exciting than playing whatever video game was popular at the time. I was like, ‘This is awesome; I want to do this!’”

Hook’s private guitar lessons began when he was just six, but his formal music training also included drums, piano and violin.

According to Hook, he was a guitar player from day one, writing his own songs and putting together his own bands. In fact, he would multi-track himself with what he describes as a “ghetto blaster” and an external microphone.

“I would put the external mic on one side of the speaker and then play into the other mic, and so, I was playing along with a cassette and recording it into a separate cassette. That was crazy,” says Hook.

Eventually he moved on to multi-track recorders, such as the Tascam 246 Portastudio 4-track and then on to eight- and 16-track recorders. Eventually he dropped $5,000 for a computer, monitor and Pro Tools software.

While Hook’s interest in home recording grew, so did his career as a musician. While he may be best known for his role in Five Finger Death Punch, he is also a highly-skilled studio and touring musician who has played with the likes of such diverse artists as Alice Cooper, Vince Neil and Mandy Moore.

By the time he joined Five Finger Death Punch in early 2009, he had already released a debut hard rock/metal album, Safety Dunce, for which received a Best Instrumental Record award at the 2007 L.A. Music Awards, and had recorded a follow-up solo project that is currently awaiting release.

His first offering with the band, War Is The Answer, debuted as number seven on the Billboard Top 200 album chart.

When Hook isn’t on the road with the band, co-headlining music festivals such as the Rockstar Energy Drink Mayhem Festival with Rob Zombie, he is at his home studio or launching his own signature M4-Sherman guitar with Gibson.

KRK Systems

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Posted by Julie Clark on 11/20 at 01:31 PM
RecordingNewsLoudspeakerStudioPermalink

Tuesday, November 19, 2013

In The Studio: An Introduction To Audio Metering

Explaining some of the metering tools we have available in DAW
This article is provided by Audio Geek Zine.

 
A while ago James commented on the podcast and asked for a segment explaining some of the metering tools we have available in DAW. If you follow the podcast you may have already heard this info.

Before we get into the DAW let’s briefly go over the way analog equipment is calibrated.

VU
Mic preamps, converters, hardware effect processors are all designed to work optimally at 0 VU. They can usually handle more than that before distorting, but 0 VU is where the signal to noise is best.

VU stands for Volume Unit and is the oldest analog metering system. VU meters are relatively slow moving with at 300ms response time. This slow response of a VU meter better represents an averaged volume level close to how our hears work. 0VU is equal to +4dBu or professional line level.

dBu
The dBu scale measures the analog voltage level in our equipment with 0dBu calibrated to about 0.775 Volts. The u in dBu stands for ‘unloaded’ which means that the voltage is measured with a zero resistance load.

Again, 0VU or +4 dBu is the ideal constant voltage of all your analog components in the recording and monitoring chain.

Here’s an example chain – microphone, mic preamp, compressor, audio interface line input, Analog to digital converter, recording software.

The microphone signal gets boosted up to line level by the preamp. Line level goes into and out of the compressor into the audio interface. The analog to digital converter assigns bits representing the voltage coming in and sends the data to your DAW.

Digital Meters
Once it’s in your DAW the level you see will not be 0 on your track meters, it will actually be closer to -18 dBfs depending on the calibration. This may seem like a really low level but this is actually the optimal level that all the analog components that come before it.

Once you build up your song with several other tracks, you’ll be happy you have that extra headroom and lower noise floor.

0VU = +4 dBu = -18 dBFS

This is the only thing you need to remember

dBFS
The dBFS meters show Decibels relative to full scale. Instantaneous digital levels below the 0dBFS absolute peak. When 3 consecutive samples pass 0 the clip light will come on.

dB RMS
Now what’s left is RMS metering. Some DAWs have this in addition to Peak metering on the master. Similar to how VU meters work, RMS meters show an average level. The RMS value relates to how loud a sound is perceived.

These days all music is mastered to peak just below 0 dBFS, the unwritten standard is -0.3, but the song with the higher RMS level will appear to be louder.

There isn’t a widespread calibration standard for RMS metering so you’ll have to compare values from a few references to what you’re working on.

I recommend everyone get the free TT Dynamics Range meter. This is an extra large Peak and RMS meter in all plug-in formats. I put this meter on my master right away and keep it open so I can always keep an eye on levels as I build the mix.

If you’re interested in this stuff, dig into this site for much more detail.

Jon Tidey is a Producer/Engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog AudioGeekZine.com. To comment or ask questions about this article go here.

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Posted by Keith Clark on 11/19 at 07:01 PM
RecordingFeatureBlogStudy HallDigital Audio WorkstationsMeasurementStudioPermalink

Universal Audio Announces New MAAG EQ4 EQ Plug-In

For UAD Platform and Apollo Audio Interfaces

In partnership with direct developer Brainworx, Universal Audio has added the Maag EQ4 Plug-In to the UAD Powered Plug-Ins platform.

In a self-professed quest for “audio perfection,” Cliff Maag designed the EQ4 to provide exceptional transparency and top-end presence while maintaining a true, natural sound.

Based on his classic — and long-discontinued — NTI EQ3 from the ’80s, the EQ4 features Maag’s legendary Air Band control, a major component to the vocal mix chain on Madonna’s Ray of Light as well as Celine Dion’s Taking Chances.

Now owners of Apollo audio interfaces and UAD-2 DSP Accelerator hardware can record and mix with an exacting digital emulation of this unique EQ.

Available for purchase via the UA online store for $229, the Maag EQ4 plug-in is part of the new UAD Software v7.4 — which also includes the Fairchild Tube Limiter Plug-In Collection. Download UAD Software v7.4 here.

Specifics:
—Adds presence to vocals, acoustic guitars, and overheads
—Elevates treble frequencies without harshness or hiss
—Five-position Air Band control
—Adjusts overall EQ level with an added Trim control
—Requires UAD-2 DSP Accelerator Card or Apollo Interface, which are available from authorized dealers worldwide

Universal Audio

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Posted by Keith Clark on 11/19 at 06:41 PM
Live SoundRecordingNewsProductDigital Audio WorkstationsProcessorSignalSoftwareStudioPermalink

Tech Tip Of The Day: The Crest Factor In Mastering

What does "loudness" really mean in audio?
Provided by Sweetwater.

 
Q: I’ve heard a lot about the “crest factor” when people are discussing loudness. Does this relate at all to the issue of CD loudness in mastering?

A: It’s funny how often the term “loud” is used in the same sentence with “mastering” these days. “Real” mastering is of course much, much more than maximizing the loudness of a recording.

But loudness is an issue, regardless of whether you’re in the “more is better” camp or the “just enough” camp.

The first thing to understand about crest factor is that it is the ratio of the difference between peak level and RMS. With a pure sine wave there is a significant difference between the peak and RMS value (about 3 dB of power, but who’s counting).

So clearly more complex waveforms, such as music, are going to have differences between the RMS and peak.

For the sake of this discussion you can think of RMS as an “average” level. Though this is not entirely correct technically, it is close enough to serve our purposes.

Typically music has a crest factor of 10 to 20 dB (much more in some cases). That loosely means there is a 10 to 20 dB difference between the peak and the “average” electrical power of the music.

Now, think about what a compressor does: it decreases the dynamic range of an audio signal, usually by lowering the peaks. When this happens the crest factor is also lowered because the relative shape (amplitude) of those waveforms is necessarily changed.

When this happens there are some resulting phenomena we can observe. The most noticeable is that assuming we turn that compressed signal up to compensate for the compressor turning down those peaks the signal will now “sound” louder. This is because, on average, it is louder. The peaks are not louder, but most of the other material is.

All of the analog audio equipment in the chain is now working harder to produce this audio, and we hear the difference; the sound is much more in your face, but on our meters we may or may not notice much difference. Of course this depends on the meters.

Most of the meters most of us have access to do not make any significant distinction between compressed and uncompressed music. Given that peak reading meters are only concerned about peaks there will be no observable difference (depending of course on how far you turn things up).

Meters that attempt to give you some type of an average level will show some difference if you know what to look for, but many of those are still calibrated for “average” music, so if you push beyond this in terms of crest factor (compression) they will be inaccurate.

A true RMS meter, while expensive, will always give you an accurate reading of your levels. Anything else is a compromise.

For the most part the meters we have work adequately, however, we need to know how to interpret what they do and do not tell us.

In today’s world of digital we have become obsessed with peak levels and the 0 dBFS point on our digital equipment. This is understandable because going beyond this level has dramatic consequences, while being almost anywhere below it has relatively subtle consequences.

We do try to always push our levels right up to that limit. (A full diatribe on all the misunderstandings about levels in digital systems is beyond the scope of this writing, but suffice to say many of the “common sense” practices in use today are somewhat misguided.)

Regardless of the valid and invalid reasons for this, the result is that many engineers stay primarily focused on that peak level and don’t really think a whole lot about what is going on below that.

In the mastering world one of the prevailing trends has been to remove so much dynamic range from material that the average level is extraordinarily loud.

Everyone wants their song or CD to sound as loud or louder than the other material out there so mastering engineers are forced to push the envelope, which often results in most of the dynamic range being squashed out of the material.

A small part of the problem here is that the meters mostly used do not accurately show how high the overall energy level is getting, as explained in yesterday’s tip. Engineers don’t “see” what they are doing to the audio. Clearly the judgment should come down to what it sounds like anyway.

Ultimately this is among the aesthetic decisions that can be made for a given recording. So long as the material doesn’t go above 0 dBFS it’s a “legal” recording.

Anything beyond that is largely a matter of taste, so it’s not as if inaccurate metering is a real show stopper, but many engineers actually rely on meters more than they’d probably like to admit.

This is illustrated effectively in how their eyes are glued to those “over” indicators on their DAW or digital recorder.

As mentioned before, various meters react to this phenomenon differently. Peak meters will not show any difference because they, by definition, aren’t concerned with average levels.

VU style meters give some indication if you know what to look for. LED meters, which often are some attempt at a compromise between peak and average meters, will give similar clues.

Assuming you don’t have the disposable income to invest in the sophisticated metering systems many professional mastering engineers use, you can learn to better use what you already have.

Put in four or five different CDs of different styles of music you like and notice how the meters you normally use to set your levels react to them. Don’t just look at the top line. Look at how far down they go, and how quickly they seem to be moving.

This gives you some idea of how they are reacting to the dynamics of the music. You can use this knowledge as you watch the meters when your material is playing.

Keep in mind that in many cases these recordings were produced on state-of-the-art equipment by some of the best people in the business. Don’t expect to be able to duplicate what they do exactly.

Further, there’s almost surely a lot your meters don’t/can’t tell you. So just because you can make them look the same on your material as compared to another CD doesn’t necessarily mean it Is the same.

 
For more tech tips go to Sweetwater.com

 

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Posted by Keith Clark on 11/19 at 06:01 PM
RecordingFeatureBlogStudy HallDigitalDigital Audio WorkstationsMeasurementMixerMonitoringSignalStudioPermalink

Steinberg Pioneers Gesture Control For Cubase DAW

Freely available Cubase iC Air unleashes new level of control, freedom and flexibility within Cubase 7 and Cubase Artist 7 when interacting with Leap Motion Controller or Intel systems powered by Intel Perceptual Computing SDK

Steinberg Media Technologies GmbH today announced the release of Cubase iC Air, allowing users of Cubase 7 or Cubase Artist 7, together with Leap Motion Controller or Intel systems powered by the Intel Perceptual Computing SDK, to control Steinberg’s popular DAW intuitively by the command of gestures.

“With the advent of various gesture recognition technologies we’ve been looking into possibilities to control Cubase in a novel way by using simple hand gestures as an alternative control alongside the conventional mouse and keyboard — and Cubase iC Air is our first achievement in this field of research,” comments Shih Ming Law, key developer for Cubase iC Air.

Cubase iC Air embeds a control panel within Cubase 7 and Cubase Artist 7, giving visual feedback of the user’s hand gestures that are picked up by depth cameras from Leap Motion and Intel.

Pre-programmed gestural commands allow the user to perform transport controls, such as start, stop, forward and rewind, to jump to the next or previous track, to audition through sections of audio and zoom in and out the arrangement.

By implementing a virtual version of Steinberg’s AI Knob, users are also able to control many other parameters of Cubase as well as VST 2.4 and VST 3 plug-ins through Cubase iC Air.

Intel global account manager Jean-Pierre Navarro says: “The great collaboration with Steinberg’s developer team helps get the most of the Intel platform and the Intel Perceptual Computing SDK. Cubase iC Air provides users with a new and exciting way of interaction. This is an entirely new way of experiencing music.”

Cubase iC Air is a software component that requires Cubase 7 or Cubase Artist 7 as host application and a Leap Motion Controller or depth camera run by the Intel Perceptual Computing SDK.

Cubase iC Air is a freely downloadable component, available from the Steinberg website.

Features at a glance
• Control Cubase 7 and Cubase Artist 7 with hand gestures through supported hardware
• Use pre-defined gesture commands for transport control, navigation and more
• Control virtually any parameter through Advanced Integration

Steinberg

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Posted by Julie Clark on 11/19 at 02:44 PM
RecordingNewsDigitalDigital Audio WorkstationsStudioPermalink

Monday, November 18, 2013

In The Studio: Five Microphone Placement Techniques For A Bigger Sound

When you want to make it "larger than life"
This article is provided by Bobby Owsinski.

 
Sometimes when recording, microphone placement can seem either too difficult or way too easy. As with most things in life, it’s really somewhere in the middle, but sometimes it’s not very easy to get there.

Here’s an excerpt from the newly released Recording Engineer’s Handbook Third Edition that shows five simple miking techniques that will help you get a bigger and more accurate sound.

Before you start swapping gear, know that the three most important factors in getting the sound you want are mic position, mic position and mic position.

Get the instrument to make the sound you want to record first, then use the cover-your-ears technique to find the sweet spot, position the mic, then listen. Remember that if you can’t hear it, you can’t record it.

Don’t be afraid to repeat as much as necessary, or to experiment if you’re not getting the results you want.

That said, the following are some general issues and techniques to consider before placing a mic:

1. One of the reasons for close-miking is to avoid leakage into other mics, which means that the engineer can have more flexibility later in balancing the ensemble in the mix. That said, give the mic as much distance from the source as possible in order to let the sound develop, and be captured, naturally.

2. Mics can’t effectively be placed by sight until you have experience with the player, the room you’re recording in, the mics you’re using, and the signal path. If at least one of these elements is unknown, at least some experimentation is in order until the best placement is found. It’s okay to start from a place that you know has worked in the past, but be prepared to experiment with the placement a bit since each instrument and situation is different.

3. If the reflections of the room are important to the final sound, start with any mics that are used to pick up the room first, then add the mics that act as support to the room mics.

4. From 200 Hz to 600 Hz is where the proximity effect often shows up and is one reason why many engineers continually cut EQ in this range. If many directional microphones are being used in a close fashion, they will all be subject to proximity effect. and you should expect a buildup of this frequency range in the mix as a result.

5. One way to capture a larger than life sound is by recording a sound that is softer than the recording will most likely be played back. For electric guitars for instance, sometimes a small 5-watt amp into an 8-inch speaker can sound larger than a cranked full Marshall stack.

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blogs. Get the Recording Engineer’s Handbook Third Edition here.

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Posted by Keith Clark on 11/18 at 06:25 PM
RecordingFeatureBlogStudy HallEngineerMicrophoneStudioTechnicianPermalink

Universal Audio Announces Fairchild Tube Limiter Plug-In Collection

For UAD platform as well as Apollo audio interfaces

Universal Audio has captured Fairchild’s revered tube character with the new Fairchild Tube Limiter Plug-In Collection for the UAD Powered plug-ins platform and Apollo audio interfaces.

In 2004, Universal Audio released the Fairchild 670 Legacy plug-in, heralded as the best Fairchild 670 emulation available. Now UA has improved the original time constants and gain reduction curves while modeling — for the first time ever — the complete tube-powered amplifier and transformer sections of their hardware counterparts.

Far beyond other Fairchild emulations, only the new UAD Fairchild Limiter Collection is based on an accurate circuit model of Ocean Way’s “golden-reference” units.

The Fairchild Tube Limiter Collection also offers “digital only” features such as sidechain filtering, dry/wet parallel blend, and headroom controls, giving owners of Apollo audio interfaces and UAD-2 DSP Accelerator hardware modern workflow appointments.

Available for purchase via the UA online store for $299, the Fairchild Tube Limiter Collection is part of the new UAD Software v7.4— which also includes the Maag EQ4 plug-in.

Specifics:

—Includes the Fairchild mono 660 and stereo 670, each with their own sonic attributes and feature sets

—Models entire electronic path, including tube amplifiers and transformers for complete analog color and behavior

—Based on Ocean Way’s famous, meticulously maintained Fairchild “golden units”

—Lateral-Vertical mode for stereo imaging and balance control

—Includes new “digital only” features such as sidechain filtering, dry/wet parallel blend, and headroom controls

—Requires UAD-2 DSP Accelerator Card or Apollo Interface, available from authorized dealers worldwide

Download UAD Software v7.4 here.

Universal Audio

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Posted by Keith Clark on 11/18 at 06:08 PM
Live SoundRecordingNewsProductDigital Audio WorkstationsProcessorSoftwareStudioPermalink

In The Studio: Guerrilla Techniques To Maximize Compression

Cleaning up and taming the the initial peaks

Unlike an expander, which increases dynamic range, a compressor reduces dynamic range. In recording, running a signal through both a compressor and an expander can be very effective.

Why would you want to reduce and enlarge the dynamic range at the same time? Actually, you wouldn’t.

They don’t both come into play at the same time; an expander does its thing when signals are at their quietest (or nonexistent), and a compressor does its thing in the louder part of the dynamic range.

So if you’re recording a cymbal crash through both a compressor and an expander, the expander will be working before the sound begins; the expander’s gate opens up immediately when the cymbal is struck, and then the compressor takes over.

The compressor works perhaps for a few seconds while the cymbal decays (with the expander doing nothing, since the signal is over the expander’s threshold).

Then the signal enters a kind of no-man’s-land, between the compressor’s and the expander’s active ranges, where neither circuit does anything to the signal.

Finally, when the expander senses that the crash is decaying below its threshold, its gate begins to close again (Figure 1).

This process accomplishes two things: the expander cleans up the noise before and after the crash, and the compressor tames the initial peak and thereby allows the whole signal to be brought up in volume, allowing it to have more punch and presence in the mix.

If you were recording or sampling a series of cymbal crashes, one after another, the compressor would be even more beneficial: It would tend to even out the crashes in volume, which would make the quieter crashes less likely to get buried in the mix and the louder ones less likely to overwhelm the mix.

As a bonus, compression makes a sound less likely to overload stages downstream in the signal chain—which is particularly important if you’re recording digitally.

Figure 1: A crash cymbal (left), and the same crash cymbal through a compressor/expander (right). The compressor and the expander come into play at different points of the cymbal’s decay (click to enlarge)

Here’s a look at a compressor’s typical parameters and how to use them:

Threshold: To understand how compression works, it helps to imagine expansion upside-down. When a signal rises past the compressor’s threshold, the compression circuit begins to kick in, and when a signal falls below this threshold, the compressor stops working.

So compression happens only when the signal is above the threshold—just as expansion happens only when the signal is below the expander’s threshold.

Given a gradually rising signal, compression can kick in suddenly, which is called hard-knee compression, or the circuit can come into play gradually as the signal rises, which is called soft-knee compression.

Some compressors allow you to specify which kind it performs; soft-knee compression tends to sound more transparent and natural.

Ratio: This term is a little easier to understand regarding compression.

In an ordinary signal-chain stage, such as a mixing board’s channel fader, the gain is linear:

Any increase in level at the circuit’s input will be matched by an identical level increase at the output.

If it’s a unity-gain stage (meaning that no amplification is occurring), three more decibels going into the circuit will result in 3 dB coming out. This is a 1:1 ratio: what you put in is the same as what the circuit pumps out.

A compressor changes this ratio, but only in the region of the dynamic range that’s above the compression threshold.

If the compressor is set for a 2:1 ratio, that means that when the signal level is above the threshold, increasing the level going into the circuit by 2 dB will result in only 1 dB more amplitude at the output.

Likewise, pumping in an extra 10 dB will result in only 5 dB of output. But if the signal is below the compression threshold, pumping in an extra 10 dB will result in a 10 dB increase at the output—the compressor is unity-gain (1:1 ratio) below the threshold.

Figure 2: Above the compression threshold, at a 2:1 compression ratio (left), a 4 dB level increase at the compressor’s input results in only a 2 dB level increase at the output. With a high compression ratio of 10:1 (right), you need to put 10 dB more signal into the input to get 1 dB more output signal (click to enlarge)

Figure 2 shows how this works in graph form. It should be easy to see that if you set the compression ratio higher, you need to pump even more signal into the circuit to get the same rise in output: with a 10:1 ratio, a whopping 20 dB of extra signal level will cause the compressor’s output to rise by only 2 dB.

With an infinite compression ratio, you can’t get the output to rise over the threshold no matter how much signal you put into the circuit. Any compression stronger than about 20:1 is considered limiting.

A limiter is like the flipside of a noise gate—it’s kind of black-or-white, either doing its thing or doing nothing (depending on the signal level at the moment), without much gray area in between.

Figure 3: With both a compressor and an expander in line, the gain is unaffected only between the compressor’s and expander’s thresholds (click to enlarge)

Combining things, Figure 3 is a graph showing how having both an expander and a soft-knee compressor in your chain affects a signal’s dynamics.

Attack & Decay: These parameters are essentially the same as in an expander.

Attack specifies how fast the compressor gets to work when presented with a signal that’s above its threshold, and decay specifies how fast it returns to a 1:1 ratio when the signal falls below the threshold.

With a soft-knee compressor and a signal that slowly rises and falls in level, attack and decay may not come into play at all—but when presented with things like sudden transients, they can have a definite effect on a sound.

As I mentioned, in a combined compressor/expander unit there may be just one set of attack and decay knobs; I tend to leave mine set to a very fast attack and a medium-length decay, which seems to work just fine for most sounds.

Indicator LEDs: A “compressor active” LED is a critical feature on a compressor, particularly if you don’t have much experience working with compression.

Since you can’t always hear when compression is happening, particularly with a low ratio, it really helps to have an LED that lights immediately when the compressor’s threshold has been exceeded.

It provides great visual feedback as to how the compressor is operating with regard to the dynamic range of the performance you’re recording.

A compressor without this LED is much harder to use effectively, requiring more guesswork and listening skill. Some compressors have additional “gain reduction” LEDs or a gain reduction VU meter; these are nice to have, but they aren’t as important.

If you’re shopping for a compressor, by all means get one with at least a compressor-active indicator LED.

For the following exercise I’ll assume you’re using a compressor with ratio and threshold controls as well as a compressor-active LED.

Using A Compressor
Set up a microphone for vocals and run it into an input channel on your board.

Put the signal through your compressor by way of the channel insert jack, and put on a pair of headphones. If you have an integrated compressor/expander, set up the expander section as described above—get it to clean up your studio’s background noise, but don’t let it chop off any final consonants or slowly decaying vowel sounds.

If you find that the expander’s gate is fluttering open and closed (which can happen if the threshold is right around the background noise level), try raising the threshold a bit, increasing the decay time a bit, or both.

Now you can go to work setting up the compressor section.

Set the compressor’s ratio knob to about 3:1 and stand in front of the mic, about where you’d be when you’re singing. While you watch the indicator LED, make vowel sounds that start soft and increase in volume, and notice when the LED comes on. (If it doesn’t come on at all, turn down the compressor’s threshold knob and try again.)

A good starting position for the threshold is the point where your vocal starts to get loud—in musical terms, somewhere in the mezzo-forte range.

If your compressor has a gain-reduction meter or LEDs as well, watch what happens when you sing even louder above the threshold. (When this kind of meter says that 6 dB of gain reduction is occurring, it means at that particular moment, the output would be 6 dB hotter if the compressor weren’t there.)

Now try turning the compressor’s ratio knob up or down and repeat the exercise, and try to hear a difference. You may notice that with a higher ratio setting (like 6:1), as you sing louder and louder above the threshold, the sound of your voice in the headphones may seem to get quieter.

What’s actually happening is that the sound of your vocal cords being conducted through the bones of your skull is overtaking the headphone sound, because the latter is being compressed while the former isn’t.

That’s okay. If you want to hear what the compression really sounds like, record yourself.

Record your vocals getting louder and louder fi rst at 2:1, then 4:1, and then 8:1, and listen to the difference. The best compression—and this applies not only to vocals but pretty much to every instrument—should do its thing without calling any attention to itself.

If you recorded yourself in the above exercise, you may have noticed that the sound can start to get “squashed” near the top of the dynamic range, particularly with lower compression thresholds and higher ratios.

How much “squashing” you can get away with depends largely on the context of the track you’re recording. If it’s a loud, rocking song with loud, rocking vocals, you can get away with a more “squashed” sound—in fact, very heavy compression is a cool effect frequently used on vocals, drums, and other sounds.

On the other hand, if you’re recording a soft tune with an intimate vocal, you need to be more careful about how the compression is sounding. If the compression in this kind of setting is noticeable, it can ruin a performance; it sounds unnatural and detracts from the performance’s intimacy.

Unless you’re using heavy compression as an effect, try to make the compression as transparent as possible—it should sound like it’s not even there.

If you can hear the compression kick in during playback, try using a lower ratio as well as a slightly lower threshold. The lower ratio will make the compression sound less heavy-handed, while the lower threshold will bring down the hottest peaks’ levels to about where they were with the previous settings.

If you end up being the only person in the world who knows that a track was compressed, then you know you’ve done a good job applying compression.

Why Compression Is So Important
It might be a good idea to compress just about every track you record, before you record it.

An exception is commercially sampled sounds (including drums):

—First, because sampled sounds are often already compressed somewhat;

—Second, because you may be bringing these into the mix by way of MIDI and therefore aren’t actually recording them during the tracking phase;

—Third, because sampled sounds have more even, predictable dynamics than live acoustic sounds.

For everything else, adding compression while tracking just makes everything much easier.

Here’s why: Acoustic sounds, as well as many electronic sounds, tend to vary quite a bit in level.

If you’re recording a shaker percussion part without any compression, the loudest shakes may be a good 9 dB louder than the softer shakes.

Meanwhile, on your uncompressed rhythm-guitar track, some chords may be right up near the top of your system’s dynamic range (0 dB), while others are at –4 dB.

Figure 4: Uncompressed tracks are hard to mix because their levels are constantly changing (click to enlarge)

This kind of variation could be occurring on most of your song’s tracks, including the all-important lead vocal, throughout the whole song (Figure 4). It’s extremely difficult to create a consistently good mix of a song under these circumstances.

Even if you could get the mix perfect for one moment in time, a half-second later the mix could be totally different—the rhythm guitar could suddenly get much louder and suffocate the lead vocal for a moment, which may have gotten suddenly quiet at that moment anyway, making matters worse.

You have no real control over the blend, and the end result could be a random pastiche of obnoxiously loud moments and inaudibly buried moments.

Compression solves this by getting levels under a certain amount of control. If the levels of each instrument vary by only 2 dB or 3 dB, you can achieve a much smoother mix than if each instrument is varying by 9 dB.

From the song’s beginning to its end, the mix will be much more consistent than a similar mix of uncompressed tracks (Figure 5).

Figure 5: The same tracks with compression can be mixed in a much more smooth, consistent way (click to enlarge)

Simply stated, it makes things easier in the studio.

And since we Guerrilla recordists mix as we go, we need to have the tracks compressed as they go down. Million-dollar facilities may be able to compress each track separately during the mix; we can’t.

De-essing
The process known as “de-essing” is a type of compression. If you record a vocal track very bright, adding some top-end EQ, you’ll get a very present, airy sound—two good qualities to have on a vocal track.

However, moments where words have “s” sounds can cause an explosion of noisy sibilance. Sibilance refers to excessive high end associated with the sound of the consonants S, Z, etc.

Sibilance can mess up an otherwise well-recorded track, and if you have an effective way to tame a track’s sibilance, you should use it.

The most common way to do this is by de-essing the track (Figure 6).

Figure 6:  In de-essing, the signal is split, and compression is applied based on its high frequencies only. This is done by sending a high-pass-filtered signal into the compressor’s side-chain input (click to enlarge)

De-essing refers to compression based only on the signal’s high-end frequency content.

Here’s how it traditionally works using hardware compression: An audio track is split into two parts, and one of the parts goes through a high-pass filter, which removes everything but the highs—say, everything below 8 kHz.

This highs-only signal is then fed into a compressor’s side-chain input. This is an input that allows the compressor to react to a signal different from the one it’s actually compressing.

Meanwhile, the non-filtered portion of the signal goes into the compressor’s normal input.

When there’s a burst of high frequencies from an excessive “s” sound, it easily passes through the high-pass filter, goes into the side-chain input, and causes the compressor to attenuate the full-range signal going into the input jack.

The result: the track is compressed slightly at the moment where the excessive sibilance occurred, thereby taming the sibilance.

Theoretically, you could achieve the same result by manually pulling down the fader whenever the harsh “s” sounds occurred, and then immediately returning the fader to its original position—but this would be difficult or impossible (particularly if you’re singing at the same time!).

De-essing, like ordinary compression, makes this process automatic, so you don’t have to worry about it.

In a digital studio, the simplest way to de-ess a track is to put a de-essing plug-in on it.

The plug in will probably allow you to specify a threshold for the de-essing compression circuit, and perhaps also a cutoff frequency, which specifies how high the highs must be to trigger the de-esser (Figure 7).

Figure 7: A de-esser plug-in. Level indicators are always helpful; this plug-in has one for gain reduction (GR) of the sibilance (click to enlarge)

If you don’t have a de-essing plug-in or you’re recording analog, you could use an outboard compressor to do ordinary compression while tracking, and then run the track through a de-essing configuration during mixdown. Or, you could buy a second compressor.

Some compressors have a built-in “de-ess” button which, when pressed, turns the compressor into a de-esser (mid- and low-frequency dynamics remain unaffected). No matter how you employ de-essing, be sure to experiment with various settings to see what sounds best.

Of course, it’s better to avoid excessive sibilance on a track in the first place. With many condenser mics, singing too closely to the mic, or too directly into the mic, can enhance sibilance in an unnatural way.

By backing off the mic and singing a little off-axis—meaning you’re kind of singing past the mic rather than directly into it (Figure 8)—you can achieve a more natural sound with fewer sibilance problems.

Figure 8: Singing slightly past a microphone (white arrow), rather than straight into it (gray arrow), reduces problems with breath noise and also softens sibilance. This mic has a pop filter installed (click to enlarge)

P-Pops
“P-pops” aren’t really related to controlling dynamics, but in terms of the frequency spectrum they are the polar opposite of sibilance, so I’ll mention them here.

A P-pop results when plosive consonants—the sounds of the letters P and B—produce a puff of air that hits the mic diaphragm, causing a low-frequency thump on the track.

At worst, this thump can have so much energy it causes the signal to distort one of the downstream stages—but usually it just sounds annoying and unprofessional, and often it causes a compressor to make the signal “duck” for a moment, which sounds unnatural.

The simplest way to prevent P-pops is to install a pop filter in front of the microphone. These are available commercially, either as a cover that slips over the mic itself, or as a two-layer fine-mesh screen stretched over a ring, which attaches to the mic stand and can be positioned with a flexible assembly.

For a while I used a piece of silkscreen stretched on a wooden frame with good results. And similar to taming sibilance, you can go a long way toward preventing P-pops just by singing a little off-axis.

There’s really no reason you need to sing directly and closely into a mic, unless you’re recording background vocals consisting of pure vowel sounds and you want to make them sound as airy and intimate as possible.

You’ll get a more natural, balanced sound if you back off the mic a bit and sing over it or past it a little.

Click to enlarge book cover

This article is excerpted from Guerrilla Home Recording (Second Edition) by Karl Coryat, click over to musicdispatch.com. NOTE: ProSoundWeb readers can enter promotional code NY9 when checking out to receive an additional 20% off the retail price plus free shipping (offer valid to U.S. residents, applies only to media mail shipping, additional charges may apply for expedited mailing services).

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Posted by Keith Clark on 11/18 at 02:46 PM
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JoeCo Appoints Professional Audio Technology for Australia

Professional Audio Technology is the new distributor for BlackBox range of recorders and players in Australia

JoeCo has appointed Professional Audio Technology Pty Ltd (PAT) to distribute the BlackBox range of multi-channel live audio recorders and players in Australia. Based in Hornsby, New South Wales, PAT adds JoeCo to a comprehensive portfolio of pro audio brands that cover a range of markets including Broadcast, Live Sound, Conferencing and Hire.

Since launching the first BlackBox Recorder in 2009, JoeCo has established an award-winning series of computer-free 24- and 64-channel 1U recording and playback systems that are compatible with virtually all analogue and digital consoles and other equipment.

Various I/O options are available including MADI, DANTE, AES, Lightpipe and analogue. Audio is recorded directly to an external USB2 drive in Broadcast WAV format and can be instantly re-purposed for archiving, re-mixing or post production applications.

“We are very pleased to welcome Professional Audio Technology to our international distributor network” says JoeCo’s Joe Bull. “With BlackBox users now active in a wide range of markets, it was important for us to find the right representative to support the varying needs of these diverse customer groups.

“We are therefore delighted to be working with Patrick Salloch and team and look forward to taking the brand forward in Australia.”

“We are delighted to have been chosen by JoeCo as their Australian distributor and are looking forward to a long and prosperous relationship between our companies” says Patrick Salloch, Managing Director of PAT. “We feel that the BlackBox recorders and players are a fantastic addition to our existing product range and cater for specific needs within our customer base.”

JoeCo demo stock is now available at Professional Audio Technology and customer enquiries are welcome.

JoeCo

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Posted by Julie Clark on 11/18 at 02:42 PM
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Hosa Technology Debuts SuperSpeed USB 3.0 Cables

Hosa USB-300 Series SuperSpeed USB 3.0 cables feature data transfer rates up to 10x faster than USB 2.0

Hosa Technology is pleased to announce the introduction of the USB-300 Series SuperSpeed USB 3.0 cables.

Featuring data transfer rates up to ten times faster than USB 2.0, new SuperSpeed USB cables are ideal for DJ’s, musicians, and audio professionals seeking to maximize the performance of their equipment, including portable disk drives, recording and playback systems, cameras, and more.

Available in 3-, 6-, and 10-foot lengths, Hosa’s new SuperSpeed USB cables offer two configurations: (1) the traditional Type A to B option commonly used to connect a PC to an audio interface, external hard drive, or similar device and (2) the Type A to Micro-B connector that can be found on DSLR cameras, smart phones, tablets, and portable drives.

Regardless of the connector type, Hosa SuperSpeed USB cables are fully compliant with USB 3.0 specifications, providing transfer rates of up to 5 Gbps and increased power handling capabilities.

All Hosa USB-300 Series SuperSpeed USB cables utilize Nickel-plated plugs for enhanced signal transfer and an Aluminum-Mylar shield for superior EMI and RFI rejection, making these cables an outstanding choice for connecting an audio interface, USB microphone or instrument, or computer peripherals to newer generation PCs.

Jose Gonzalez, Hosa Technology’s Product Manager, commented on the company’s new SuperSpeed USB 3.0 cables, “Our new SuperSpeed USB cables provide DJ’s, musicians, and audio professionals with a valuable means of making connections with the latest equipment to capitalize on the increased bandwidth provided by the USB 3.0 specification.

“USB 3.0 grants users the ability to transfer data at faster rates than ever before—at speeds up to ten times faster than USB 2.0—with significantly improved power capabilities so your devices are always fully charged when needed.

“With a good selection of cable lengths and connector options designed to fit any USB 3.0 device, I’m confident these new USB cables will be well received by our industry.”

The new Hosa USB-300 Series SuperSpeed USB 3.0 cables are expected to become available in December 2013. MSRP pricing is as follows:

USB-303AB SuperSpeed USB Cable, Type A to Type B, 3 ft $11.10
USB-306AB SuperSpeed USB Cable, Type A to Type B, 6 ft $14.10
USB-310AB SuperSpeed USB Cable, Type A to Type B, 10 ft $17.25
USB-303AC SuperSpeed USB Cable, Type A to Micro-B, 3 ft $22.95
USB-306AC SuperSpeed USB Cable, Type A to Micro-B, 6 ft $25.05
USB-310AC SuperSpeed USB Cable, Type A to Micro-B, 10 ft $27.00

Hosa Technology

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Posted by Julie Clark on 11/18 at 02:26 PM
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Introducing The New M5 ‘Pencil’ Condenser From RØDE

the new M5 compact condenser microphone from Rode exhibits low noise and full frequency response.

RØDE Microphones is proud to announce the release of the new M5 compact condenser microphone.

Inspired by the company’s flagship small diaphragm NT5, the M5 features a ½-inch cardioid electret condenser capsule and is available in acoustically matched pairs.

Designed and made in Australia, the M5 exhibits low noise and has a full frequency response that makes it ideal for studio recording and live performance.

The M5 is equally at home on a range of acoustic instruments, choirs, or anywhere you would employ a small diaphragm condenser microphone, either individually or as a stereo array.

A result of many years’ experience building the award-winning NT5 microphone, RØDE has succeeded in making a high-quality ‘pencil’ style permanently polarized condenser that will impress even the most demanding of artists and engineers.

This matched pair has been carefully selected to ensure a variation of no more than 1dB sensitivity between the microphones. A premium foiled certificate is supplied to verify the authenticity of the pair.

The M5 is finished with RØDE’s proprietary ceramic coating which offers a sleek matt black finish, and is supplied with WS5 windshields and RM5 stand mounts.

RØDE’s Global Marketing & Sales Director, Damien Wilson, commented: “The NT5 has consistently been one of our most praised and best-selling microphones since its release almost ten years ago. As pencil condensers go it is in my opinion one of the very best in the world, however we recognized that a lot of artists want to use a RØDE small diaphragm condenser microphone but doesn’t require the superlative dynamics and versatile capsule interchange found on the NT5. The new M5 offers them an incredible sounding microphone that represents great value.”

RØDE

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Posted by Julie Clark on 11/18 at 01:49 PM
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Friday, November 15, 2013

Remastering Three Jazz Classics: The Dave Brubeck Quartet, Art Pepper, and Sonny Rollins

Studying how they did it so well, decades ago.

(This article co-authored by Jenny Bartlett.)

It’s a joy to hear old recordings of a favorite musician remastered on CD. Especially when the remastering is done so well that you can enjoy the music more than ever.

What’s more, a great remastering job clearly reveals the recording techniques used in the past.

There’s a lot to be learned about recording by studying how they did it so well, decades ago.

A stellar example is the four-CD Dave Brubeck reissue, Time Signatures—A Career Retrospective (Columbia/Legacy C4K 52945), produced by Russell Gloyd and Amy Herot.

It’s astounding how good some of those 30-year-old tapes sound!

The mix is just right. Cymbals are bright and crisp; acoustic bass sounds full; piano and sax are warm rather than thin. And there’s very little tape hiss.

The mastering job was ably handled by Mark Wilder, a recording engineer with Sony Music Studios (formerly Columbia Records). Since the 1970s, he’s worked at Vanguard Records, Polygram, and finally Sony.

Bruce Bartlett: What was your philosophy in remastering the Brubeck tapes: to transfer them as is, or to clean them up?

Mark Wilder: In making the Brubeck compilation, we wanted the CDs to remain true to the original master tapes. Around 1984 to 1987, however, many CDs were remastered with a high-frequency rolloff (treble cut) to reduce tape hiss.

That was the philosophy of the time: to make remastered CDs hiss-free. Unfortunately, it was a destructive process.

Dave Brubeck complained about the sound of his early remastered CDs. Fortunately, we’ve learned a lot about remastering since then. I wanted to give him better treatment than that; I wanted to do it right. Our current philosophy is to adhere to what was on the originals. We don’t add equalization or reverb.

Sony’s new 20-bit A/D system helped to ensure a clean transfer from the original tapes to CD. It did the 20-to-16 bit transfer very well. Another current 20-bit Brubeck reissue is the Master Sound CD of Time Out. It’s so clear, you can hear bad edits onTake Five.

Bartlett: How does the sound of those early recordings compare with jazz recordings today?

Wilder: It’s amazing how well-recorded the group was back then. The sound is so three-dimensional, bigger than life.

Yet it’s amazing how little the engineers did to get that sound. They just put one mic a few feet from each instrument, and mixed live to 3-track—for left, center, and right. Then they edited the tape and mixed down to 2-track.

The old stuff sounds better than what we’re doing now. We’ve been going in the wrong direction sound-wise for many years. The layout of the stereo stage was more realistic then, too. Drums were on the left, piano on the right, sax and bass in the middle.

It’s easy to hear what each musician was playing because they were separated spatially. These days, you hear each instrument in stereo, on top of each other. The drums spread all the way between the speakers, and so does the piano.

In one Brubeck recording (Castilian Drums, not on this set), the stereo perspective changes radically within the recording. It starts with drums hard left and piano hard right. But when the drum solo starts, there’s an edit and suddenly you hear the drum set spread out in stereo.

At the end of the solo, you’re back to drums left and piano right. These effects are on Brubeck’s albums Countdown (Columbia CS 8575) and Time Further Out (Columbia CS 8490), .

Bartlett: How was the Quartet miked?

Figure 1: Mic techniques used on The Dave Brubeck Quartet (click to enlarge)

Wilder: Columbia was heavy into Neumann M49 and AKG C12 mics back then. Both are large tube-type condenser mics.

On the Brubeck groups, the engineers used one mic per instrument, and each mic was at a very respectful distance, about 1 1/2 to 3 feet away (Figure 1)...

You still hear all the air off the sax reed, and still hear all the tone. They got the placement quickly. Distant miking like this sounds great; I think we overdo close miking.

Bartlett: Some excellent photos of the studio layout and mic placement are in the liner notes of the CDs Time Out (Columbia Legacy CK 65122) and Jazz Impressions of Japan (Columbia Legacy CK 65726), and on the LP The Riddle (Columbia CL 1454).

Studio layout: Drum set on the left, bass a few feet away in the center, piano a few feet away on the right with the open lid on the long stick facing toward the other players. Sax faced the bass player a few feet away. The bass and piano were baffled. Baffles were placed behind the group as well.

Miking: Drum set—C12 about 5 feet up and 1.5 feet in front of the set, angled down toward the snare drum. Upright bass—U 47 about 1 foot from the bridge. Sax—M 49 up fairly high about 1 to 1.5 feet away. Piano—M 49 looking at the strings, about even with the curved edge of the piano, halfway between the soundboard and the lid. (This tends to emphasize frequencies around 500 Hz).

Bartlett: What was the typical recording and editing procedure in those days?

Wilder: Recording back then was almost a factory process. They recorded live to 3-track in three-hour sessions.

Then the tape went to the editor, then to the mixer, who mixed three tracks to two. Little or no compression was used.

We compared the master tapes with early pressings, and they were very similar. It’s amazing how little the mastering engineers did to the sound.

By the way, Columbia’s acetates were quiet as digital, even though theory says they aren’t supposed to be.

Studio engineers worked so fast back then. They might record Duke Ellington in the morning, Doris Day in the afternoon, and Brubeck the next day. There was only one hour between sessions, and each session had a totally different setup.

In spite of the speed of these sessions, you never hear a blown solo or a blown fader move. And there’s never a dramatic sound change at an edit point.

Bartlett: Who were the engineers on the Brubeck sessions?

Wilder: Fred Plaut and Frank Laico were two of Brubeck’s recording engineers. Plaut is a true balance engineer; he’s my idol. I don’t know how he could pull off what he did in three hours.

Besides Brubeck, another artist whom Plaut recorded was Michael Olatunji. He was an African drummer who made ethnic cultural records. When I listen to them, I’ve never heard drums sound so beautiful in my life—such color and harmonics. Why can’t I pick up a record today and hear that?

Plaut did a lot of Broadway recordings, and he worked so fast. Here’s an example. Friday night after a Broadway theater performance, the actors would come over and record all night. Then the artists would go back to the theater for the next performance. By Monday morning, the records would be edited, mixed, cut, and on the shelf ready to be trucked out.

Eight-track recording was introduced in the late ‘60s. Plaut backed up his 8-track recordings on a 3-track recorder. He always ran a 3-track as a safety.

I listened to some of his 8-track recordings done in the late ‘60s. The 8-track is a mess, but the live-mixed 3-track has perfect balance and blend.

One Reissue, Two Mastering Engineers
We’ve seen how one engineer remastered the work of Dave Brubeck. Let’s turn to another set of reissues with an interesting twist: they give you a choice of remastering engineer!

In an unprecedented move, Analogue Productions released two CD versions of Art Pepper’s 50’s classic, Art Pepper meets The Rhythm Section. One version was mastered by Doug Sax; the other by Bernie Grundman—two of the most respected engineers in the business.

Doug Sax masters all of Analogue Productions’ reissues, so why was Grundman chosen?  It’s a story of who knows who.

The original recordings were done on the Contemporary Records label. Grundman was once an engineer at Contemporary; he knew Pepper’s album intimately as a fan and as an LP mastering engineer.

Since the album was one of his favorites, Grundman had always wanted to master the LP reissue.

Lester Koenig produced the original session, and his son, John Koenig, recommended Grundman to Chad Kassem, president of Analogue Productions.

Kassem decided to release two versions, one mastered by Sax, the other by Grundman, and both under John Koenig’s supervision. Kassem is letting audiophiles make up their own minds about which sounds better!

From the same company, another reissue with a choice of remastering engineers is Sonny Rollins’ 50’s jazz classic, Way Out West. Both Pepper’s and Rollins’ reissues are on 24 karat gold, limited-edition CDs, pressed in Japan by Superior.

Legendary engineer Roy DuNann recorded the original 1957 sessions at Contemporary’s studio in Los Angeles. DuNann used AKG C-12 and Neumann U 47 condenser microphones, which fed an Ampex 350 2-track tape recorder running at 15 ips.

In remastering, Grundman used his own custom electronics in the mastering console and in the Studer tape transport. He chose an Apogee A/D converter with a Harmonica Mundi redithering module. Grundman added reverberation to the dry master tapes using an EMT 250 plate and Ocean Way’s live chamber.

In his version, Sax used a Mastering Labs (TML) console, MCI tape machine with TML tube electroincs, and a custom TML A/D converter. Sax added reverb with a Lexicon 480L.
 
Thanks to the careful work of Wilder, Sax and Grundman, jazz lovers and audio technophiles alike can savor the quality of these pioneering recordings, whose musicality has yet to be surpassed.

Acknowledgements: The author gratefully acknowledges Mark Wilder, Joanne Sloane, Iola Brubeck and Dave Brubeck for their help with the Brubeck part of this article.

Discography
Sonny Rollins LP, Way Out West was originally Contemporary Stereo S7530. Sax’s remastered reissue is CAPJS 008, Grundman’s version is CAPJG 008.
Art Pepper meets the Rhythm Section was originally Contemporary Stereo S7532. Sax’s remastered reissue is CAPJS 010, Grundman’s version is CAPJS 010.

(This is a re-issue of an article originally written for the July, 1994 issue of Audio magazine.)

AES and SynAudCon member Bruce Bartlett is a recording engineer, microphone engineer and audio journalist. His latest books are Practical Recording Techniques (5th Ed.) and Recording Music On Location.

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Posted by Keith Clark on 11/15 at 05:03 PM
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Paul White’s “Producer’s Manual” Now Available In iPad Edition

All of the original content across 320-plus pages along with over an hour’s worth of video walkthroughs and more

Sample Magic has announced the release of “Producer’s Manual,” an essential recording and production resource, in a new, fully interactive iPad Edition App.

“Producer’s Manual” was first published in 2011, authored by Sound On Sound editor-in-chief Paul White. The team behind the book are presenting this all-new and expanded iPad edition featuring all of the original content across 320-plus pages along with over an hour’s worth of video walkthroughs direct from White’s studio, fully enlargeable photos, diagrams and an expansive glossary.

Every paragraph from the original book has been ported into the attractive, easy-to-use app, a scrollable reference manual that covers all aspects of recording and production music, including:

—What is needed to achieve great recordings, from vocals and drums to guitars and bands

—In-depth guides to dynamics, reverb, studio acoustics, monitoring and more

—Taking a mix to the next level. Essential jargon-free theory backed by practical insights on everything from EQ through mixdown approaches to classic hardware

—How to master your own material when the budget doesn’t stretch to professional mastering.

“Producer’s Manual” is available to download for £14.90 from the Apple Store here.

Further information about the book and app is available here.

Sample Magic

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Posted by Keith Clark on 11/15 at 03:14 PM
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