Article

Wednesday, February 08, 2012

How To Archive Multitrack DAW Recordings

The archived recordings must be prepared to weather obsolescence

 
Multitrack DAW recordings are dependent on a complex system of primary and secondary technologies.

As discussed in An Introduction to Archiving Music Recordings, each of these technologies represents an obstacle to the long-term viability of a multitrack archive.

Simply put, if the various software and hardware products you’re using today aren’t going to be around in their current versions for the useful life of the sound recordings you’re creating (i.e. the copyright term), the archived recording must be prepared to weather that obsolescence.

The goal of preparing multitrack DAW data for archive is to minimize the layers of technology necessary to completely reconstruct the master recording in the future.

This article will introduce some basic techniques for creating both Consolidated and Flat Multitracks for archival purposes.

What Is A Consolidated Multitrack?

A Consolidated Multitrack is a digital audio fileset that completely expresses the EDL (Edit Decision List) information from a multitrack master recording. Specifically:

—Each DAW track is expressed as a single, continuous Broadcast Wave file (BWF);
—All of the consolidated audio files share the same start times and durations;
—All of the consolidated audio files share the same digital audio precisions, i.e. sample rate and bit depth;
—All of the consolidated audio files share the same descriptive naming convention, e.g. trackname_songtitle_artistname.wav.

If all of the above specifications are met, a folder containing the consolidated audio files could be used to perfectly reconstruct the multitrack recording as far into the future as the Broadcast Wave file format remains viable.

Since the Broadcast Wave file is a widely accepted standard file format for media producers, its long-term viability (and eventual uniform migration) is virtually guaranteed.

Creating a Consolidated Multitrack:

1. From your last active session/project file, ‘Save As’ to create a discrete file from which you will create a Consolidated Multitrack.
2. Hide or delete any auxiliary signal path to simplify the working environment.
3. If additional Takes or Playlists are to be included in the Consolidated Multitrack, create new tracks to allow all of the source audio to be simultaneously visible/accessible.
4. Using session boundaries, location markers, or some other timeline tool, establish a repeatable global timeline selection that includes all audio from the earliest drop-in to beyond the longest running audio file.
5. Once your global selection is made, use the Consolidate or Merge functions to create a single continuous audio file that expresses the EDL information for each track.
6. Carefully, consistently label all of the newly consolidated audio files to reflect enough information that they could completely identify themselves by name, e.g. bassamp_take2_ohbabybaby_jimmysingsalot.wav

Once the above steps have been followed, a choice has to be made about how to present these consolidated audio files as a discrete multitrack recording for archive.

Minimally, a folder that follows the same naming convention as the consolidated audio files should be created to contain all of the associated audio files and metadata (like screen shots, rtf files containing session notes, credits, etc.). This method works fine, but will always require the multitrack to be reconstructed in a DAW for playback.

Alternately, a facility like Pro Tools’ ‘Save Session Copy’ could be used to create a new, independent playback session for only the archival material.

Using this method one would need to be careful to remove any non-archival audio and metadata from the source session before saving the copy.

This approach would facilitate more convenient short-term use of the archive, but doesn’t actually provide any additional content.

What Is A Flat Multitrack?

A Flat Multitrack is a digital audio fileset that completely expresses the EDL information from a multitrack master recording, but also expresses some subset of DAW metadata. What metadata is ‘flattened’ into the archive is up to you, your client, or contractual obligations, but it could include:

—Plug-in processing like amp simulation, ‘printed’ effects from auxiliary channels, or automated processing;
—Automation data, like the fader rides on a lead vocal track;
—Bounced submixes that would otherwise require reconstructing both complex routing and plugin processing.

It is critically important to note that a Flat Multitrack should never be archived instead of a Consolidated Multitrack, but only in addition. The Consolidated Multitrack is the master recording; the Flat Multitrack (when applicable) is an extension of that master.

Once a Consolidated Multitrack has been created, a Flat Multitrack can be created by repeating the process with a few additional steps:

1. From your last active session/project file, ‘Save As’ to create a discrete file from which you will create a Flat Multitrack.
2. Hide or delete all auxiliary signal path and metadata that is not going to be flattened.
3. If additional Takes or Playlists are to be included in the Flat Multitrack, create new tracks to allow all of the source audio to be simultaneously visible/accessible.
4. To flatten real-time processes like automation, time-based effects, or submixing, bounce/re-record the appropriate track outputs to new tracks, and remove the source tracks from the session. Note what metadata has been flattened.
5. Flatten additional metadata by processing audio files with offline versions of real-time plug-ins. Note what metadata has been flattened.
6. Make a global timeline selection, and use the Consolidate or Merge functions to create a single continuous audio file that expresses the EDL information for each track (including whatever metadata has been flattened into them).
7. Carefully, consistently label all of the newly consolidated audio files to reflect enough information that they could completely identify themselves by name, e.g. bassamp_take2_flatcompression_ohbabybaby_jimmysingsalot.wav

Since it would be unlikely that every track within a DAW project would have metadata worth flattening, there will likely be some tracks that remain in their consolidated form. I would caution that it would be both redundant and confusing to include these audio files in a Flat Multitrack archive.

Preferably, an additional folder of flattened audio files can be clearly labeled, and organized with the Consolidated Multitrack data. Future users can then reconstruct the Consolidated archive, and opt-in to any of the available, clearly labeled, flat content.

Contents Versus Carrier

It should be noted that this tutorial only addresses the form of the contents of a multitrack archive. The question of how to effectively store this information is an entirely additional- though related- matter.

Anybody who is serious about the subject should examine the Producer and Engineers Wings’ “Recommendation for Delivery of Recorded Music Projects” (pdf). It contains an example of a widely-adopted approach to redundant archival storage.

Rob Schlette is chief mastering engineer and owner of Anthem Mastering (anthemmastering.com) in St. Louis, MO, which provides trusted specialized mastering services to music clients across North America.

Be sure to visit the Pro Audio Files for more great recording content.

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Engineer/Producer Matthew Noble Utilizes Metric Halo ChannelStrip On Recent Projects

For more than three decades, Matthew Noble has been at the forefront of pop music as a session guitarist, programmer, songwriter, engineer, and producer, with an engineering client list that includes Rihanna, Diana King, Southside Johnny, and Rod Stewart, among many.

These days, he performs most of his work out of the Loft Studios in Bronxville, NY and in the newly renovated Riverworks Recording in Dobbs Ferry, NY. Recent work with the musical Big River and gospel artist Rell Holland & Experience have put Noble’s new favorite plug-in, the Metric Halo ChannelStrip, through its paces.

“I tried Metric Halo’s ChannelStrip because some other people that I respect were using it,” explains Noble. “My friend Keith Brown, who is a well-known Nashville songwriter, was working on a project with Billie Decker, who is one of the hottest mix engineers in country music. Keith’s enthusiasm for the plug-in, together with his revelation that Billie uses it ‘all over the place,’ was enough to motivate me to check it out.”

Riverworks Recording boasts a huge, luscious acoustical space, which has changed the way both Noble and the producers and artists he works with approach the recording process.

“So much of my work there has involved tracking live instruments, as opposed to the ‘virtual players’ that live inside our modern computers,” he says. “While it’s been a refreshing change, it has also brought with it challenges. For example, getting a great drum sound and a great overall mix with the new expectations for how long things take these days is not easy.

“ChannelStrip has been very helpful because all the functions that I need to access quickly are all in one plug-in. These include the less ‘sexy’ functions, such as phase reverse and multiple trims, in addition to full-blown and flexible dynamics and equalization. Having everything in one plug-in has greatly improved my workflow.”

Noble often puts Metric Halo’s well-crafted presets to use: “The ChannelStrip presets are a great starting point. They’re especially useful in a time crunch, when the client is breathing down your neck. The acoustic guitar and drum presets are often spot on, right out of the gate. When I tweak, the informative GUI lets me know exactly what I’m doing.”

Of course, the best GUI in the world is useless if the algorithms behind it don’t cut the mustard. It’s here that Noble finds it really shines. “ChannelStrip has a great sound,” he said. “Like an SSL, it can be very aggressive and not at all subtle. Despite all its flexibility and sonic muscle, it has remarkably low CPU drain, which means I can use it whenever I need it.”

Metric Halo

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Tuesday, February 07, 2012

Revolabs Enhances HD Control Panel For Entire HD Line Of Wireless Microphone Systems

Revolabs has announced that the company’s Windows-based HD Control Panel software has been enhanced to support the entire HD line of wireless microphone systems, bringing the monitoring and configuration tools found on the Executive HD to the HD Single/Dual Channel and the HD Venue systems.

In addition, based upon customer feedback, Revolabs has created several new features for the HD line, including a DIP switch display, mute groups for Executive HD systems, and an expanded control system API.

With the HD Control Panel, users can monitor and control networked HD wireless microphone systems from a single PC software program with an intuitive graphical user interface.

The HD Control Panel allows users to control the mute status and gain of each microphone, and to lock out presenters from using the mute button.

The software also provides the ability to monitor each microphone closely for its real-time status, such as battery level.

The monitor tab of the HD Control Panel has been enhanced to provide the DIP switch status for each system, eliminating the need to look on the back of the system to see which switches are active.

Revolabs has also added several commands to the HD systems’ API, allowing A/V control systems to send global commands, turn off microphones, and even initiate pairing, all from the convenience of a room’s touch panel.

Finally, Revolabs has bolstered the Executive HD with the ability to assign systems to mute groups. This allows all systems in a building to be bussed together without muting each other, unless they are assigned to the same group.

“We are pleased to bring the capabilities of the HD Control Panel to users of our HD Single/Dual Channel and HD Venue systems, in addition to offering powerful new features across our entire HD line,” says JP Carney, CEO of Revolabs. “We take pride in listening to our customers as we continually strive to meet their evolving needs. New features, such as those released today, are a direct result of customer feedback.”

The enhanced HD Control Panel and new features are available through a firmware update (version 2.6.1) to both the base station and microphones. The update is available now at www.revolabs.com/downloads.

New feature enhancements require a Gold unlock code provided as part of a Revolabs service plan. Any system that has previously been unlocked will automatically receive the new features upon completion of the firmware upgrade.. 

Revolabs

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Monday, February 06, 2012

Lexicon Offering Individual Plug-Ins From PCM Native Effects & PCM Native Reverb Bundles

Lexicon has announced the availability of the individual plug-ins from its PCM Native Effects and PCM Native Reverb Bundles, with a total of 14 plug-ins available, including Pitch Shift, MultiVoice Pitch, Chorus, Resonant Chords, Random Delay, Dual Delay, Stringbox, Vintage Plate, Plate, Hall, Room, Random Hall, Concert Hall and Chamber.

“Offering the individual plug-ins from our PCM Native Effects and PCM Native Reverb Bundles represents our commitment to provide Lexicon users with greater flexibility and ease to obtain exactly the sound quality they are looking for from the specific plug-in(s) they need for any project,” says Rob Urry, vice president Harman Professional Division & GM of Signal Processing and Amplifier Business Units.

The PC- and Macintosh-compatible plug-ins are designed to work with popular DAWs like Pro Tools, Logic and Nuendo, as well as with any other VST, Audio Unit or RTAS-compatible host.

Each plug-in can be run in mono, stereo or mono in/stereo out, and on-screen input and output meters are provided for precise level setting.

All Lexicon plug-ins are Native only, and require iLok2 authorization. The individual plug-ins will be available in February 2012.

Lexicon
Harman Pro

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PreSonus Adds New Control Options To StudioLive Mixers

PreSonus has announced new updates to its StudioLive Series digital mixers, including a number of features not found on any other digital mixer from any manufacturer. 

New features include:

QMix. Up to 10 musicians can simultaneously control their PreSonus StudioLive monitor (aux) mixes using an iPhone or iPod touch and PreSonus’ QMix app, a free download from the Apple App Store. QMix/VSL is the only solution that allows multiple users to each control their own aux from separate iPhones.

Smaart Engine Technology. PreSonus has begun incorporating Rational Acoustics Smaart Measurement Technology for sound-system analysis and optimization directly into PreSonus Virtual StudioLive remote-control/editor/librarian software.

With Smaart technology and VSL, you’ll be able to precisely identify nasty feedback frequencies and get your loudspeakers to play nicer with the room-all without having a degree in acoustical engineering.

The first version of VSL to incorporate Smaart technology will be part of PreSonus Universal Control 1.6, which is expected to be available later this spring.

Universal Control 1.5.3 and StudioLive Remote 1.2. Universal Control 1.5.3 features an improved version of Virtual StudioLive that supports the new QMix iPhone app, including QMix permissions (so that each user controls only one specified aux mix) and the ability to name aux buses.

Universal Control 1.5.3 also adds VSL features that work with PreSonus StudioLive Remote 1.2 for iPad to enable SL Remote permissions so that iPad users can only control front-of-house mixer features or a specified aux. Tap tempo has been added to both VSL and StudioLive Remote.

VSL adds the ability to copy and load channels, copy main mix to aux mix (and aux to aux), link channel faders so that they can move together, and make your StudioLive mixer default to Fader Locate Mode once a fader has been adjusted in VSL or in StudioLive Remote for iPad.

PreSonus

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Friday, January 27, 2012

Everything You Wanted To Know About Sound Level Meters (SLMs)

The primer: what, how, why, what's available, techniques, applications and more

A sound level meter (SLM) is a device used to make frequency-weighted sound pressure level measurements displayed in dB-SPL.

0.0 dB-SPL is the threshold of hearing, and is equal to 20uPa (microPascals). This correlates to what one would aurally perceive when in a deep cave or in a large anechoic chamber.

Packaged as a single-function handheld test device, SLMs are intended to be held at arm’s length during measurements (to reduce the effects of the body on the measurements) or secured to a tripod stand for more stability.

All SLMs feature an omnidirectional measurement quality condenser microphone, a mic preamp, frequency weighting networks, an RMS detector circuit, averaging circuits, the meter display, AC and DC outputs used to feed other measurement devices or for recording (see Figure 1, below).

Most SLMs have the same set of user adjustments, including SPL range selection, A and C weighting filters, slow and fast detector response, and minimum or maximum SPL.

The SPL range switch provides a balance between minimizing the preamp noise level and measuring a wide range of sound pressure levels.

Most of the commonly available SLMs measure from about 30 to 130 dB-SPL and do this in 3-4 ranges.

 
The more advanced and expensive SLMs feature removable microphones, 1-octave and/or 1/3-octave filter sets, additional weighting filters including B, D and Flat or Linear (no filter), additional detector response options (Impulse and Peak), averaging (over time) and data logging or storage (either on-board, as computer files or both).

Almost all SLMs are designed and specified to perform to one of four internationally standardized levels of accuracy:

 
Note that these tolerances are at 1 kHz, the standard calibration frequency for SPL measurement. In order to ensure the flatness of the SLM there are additional tolerances specified for various frequency bands and microphone classes as well.

Class-0 SLMs are employed primarily to calibrate other SLMs and may be used for very high precision noise measurement in controlled spaces and/or for academic research.

Class-1 and Class-2 SLMs are most widely used by acousticians, sound system professionals, industrial designers/ manufacturers and researchers in academia and government. Measurements made with these levels of accuracy are generally acceptable as evidence in the resolution of legal disputes.

Class-3 SLMs are restricted to noise survey meters and dosimeters.

Microphone Sizes
Most general purpose SLMs are provided with a 1/2-inch free field microphone, permanently attached to the SLM body.

Higher quality SLMs with removable microphones may be outfitted with capsule sizes ranging from 1/8-in to 1-in.

The smaller microphone capsules (1/8-inch and 1/4-inch) have three primary advantages:
—They are capable of higher SPL measurements;
—They remain omnidirectional up to higher frequencies;
—And they present less disturbance of the air due to their reduced physical size (the mic itself has less impact on the measurement).

Larger 1-inch capsules exhibit less self-noise and are therefore best suited to measurement in very quiet spaces, of very low-level noise sources, and of low-frequency energy.

Microphone Classes
One can also choose between three different classes of measurement microphones: Random Incidence, Free Field, and Pressure.

In reality, these three classes of microphone are all fundamentally “pressure response” microphones and are designed to sense pressure - not pressure gradients or particle velocity.

The difference is primarily in how they pick up sound at high frequencies, and this relates to how they are positioned relative to the sound source.

At higher frequencies (where wavelengths become so small as to equal, or be smaller than, the size of the microphone diaphragm) the direction in which the microphone is aimed - relative to the direction of the sound being measured - will have an effect on the accuracy of the measurement.

Random Incidence microphones are intended for use in a diffuse sound field where the sound is arriving from all directions due to a high level of reflections.

This type of microphone must be angled approximately 70 degrees to the sound source in order to yield accurate measurements.

Free Field microphones are designed for use in an open space (free of reflections) and should be held at 0 degrees to (pointed directly at) the sound source.

Pressure microphones are designed specifically for measurements in couplers and closed cavities such as plane wave tubes, ear cavities in dummy heads and for noise measurements in air ducts.

Some SLMs provide a switched correction filter allowing one type of microphone to be used in place of the other.

Note: In general, a “measurement quality” microphone differs from a recording or sound reinforcement microphone in its flat response, the absence of an onboard preamp, and its less rugged construction.

Detector Averaging Time
The standard averaging time choices provided by the majority of SLMs are Fast and Slow.

The Fast setting has a 125milliseconds (mS) averaging time (developed in 1950 to match the ballistics of the moving coil meters used at that time) and is useful for steady state (not dynamic) sound.

For measurement of more dynamic sound (when the Fast setting results in fluctuations of 4 dB or more), the Slow setting provides a 1 second averaging time resulting in much more consistent and stable readings.

It is this setting (along with an A-weighting curve) that is most frequently used for measurement of pop music SPLs.

Some higher priced SLMs also provide Impulse response time (four times faster than the Fast setting) used for extremely dynamic and short term (impulse) sound and peak response time, which accurately responds to signal pulses as short as 50 microseconds.

Frequency Weighting Filters
Because we humans do not hear frequencies in a linear manner (as illustrated in the often referred to Fletcher-Munson curves, aka: equal loudness contours), sound level measurements made with a flat response do not accurately reflect how we perceive sound.

When measuring pop music sound reinforcement systems (with the program signal containing generous amounts of bass energy), broadband readings would be artificially high without the frequency shaping provided by the A-weighting filter.

C-weighting, with its more gradual low-and high shelving filters, is used for very high SPL sound system and noise measurements.

On SLMs without a Flat setting, the C-weighted setting may be used to feed a spectrum analyzer such as Rational Acoustics Smaart for frequency analysis of the broadband sound.

Table 2 below, and then Figure 2 below that, show the frequency response for A and C weighting filters, plus two other weighting filters employed for more exacting and specific noise measurements.

 
All weighting curves are equal in level at 1,000 Hz, the standard reference frequency that SLMs are calibrated to using acoustic calibrators.

Note On SPL Annotation: When communicating or recording (speaking or writing) SPL values, it is necessary to annotate what - if any - weighting curve has been employed. A-weighted SPL is written or stated as “XX.X dBA” and C-weighted measurements as “XX.X dBC”. Such annotation eliminates the need to provide the “SPL” suffix.

The absence of weighting notation implies that a Flat setting (no weighting filter) was used and in this case the suffix “SPL” must be used to differentiate from other, non-acoustic decibel values.

Historical Note: The A, C and B frequency weighting curves were first developed for specific sound levels based on equal-loudness contours.

Because equal-loudness contours were developed from research utilizing pure tones, it later became evident that the weighting curves chosen for SLM measurements at certain levels do not correlate very well to how we actually hear complex sounds.

 
More recently it has been found that the A-weighting curve provides a realistic approximation of how we hear over a fairly wide range of sound level measurements.

Display
With very few exceptions, modern SLM’s are provided with LCD displays with 3- or 4-digit readouts. Most of these provide 1/10-dB resolution except in the case of 3-digit displays when they go over 100 dB-SPL.

Prior to LCD technology, analog meters were employed and these suffered from all the inherent shortcomings of this method of display: slow rise and settle time, susceptibility to damage from pinging (over loading) and overall mechanical fragility.

Another shortcoming of analog meters is the difficulty in establishing stable readings due to the range selecting that is common in most SLMs. One must take the range that is selected and then add or subtract from the meter position accordingly.

Measurements made when the sound pressure level has changed and is out of range are prone to be inaccurate, as well. Some feel that fast impulse noise measurements are easier to read with an analog SLM than with a “digital” SLM.

Both styles of display are difficult to read in dim lighting but some SLM’s feature backlit LCD’s for this condition. And either type of display may be quite difficult to read while at arm’s length if the display size is too small.

Equivalent Continuous Sound Level
Although we live sound folk generally believe that we are able to accurately determine the overall sound pressure level when we view the displayed real time values on our SLM’s, these readings are actually no more than a glimpse, or snapshot, of the varying sound pressure levels that we are exposed to.

Due to the dynamic content of the sounds we normally work with (this applies to both speech and music) and our own human shortcomings, we are unable to mentally integrate the frequently changing SPL’s and determine an accurate mean SPL value.

Research into hearing safety and annoyance and the standards that have been developed from this research are based on the accumulative exposure to sound energy over periods of time.

To this end, integrating SLMs have been developed to provide measurements that are conducted over a selected time period and averaged so that they provide an accurate equivalent energy value, designated as Leq (LAeq or LCeq when weighting is employed).

Integrating SLMs also provide Lmax and Lmin values, and the user can choose the time period that the Leq averaging takes place (normally from several seconds to 24 hours).

More and more organizations and governments are adopting this as a more accurate means for establishing risk of hearing damage, community noise standards, annoyance issues, etc.

Although stand-alone integrating SLMs are considerably more expensive than basic SLM’s, some software is available to provide this statistical averaging from the AC output of any SLM.  and both the TerraSonde Audio Toolbox and Neutrik Minilyzer provide this function along with other SPL measurements and other useful audio system measurements.

Relative Vs Absolute Measurement
Absolute measurements are those made with calibrated SLMs that provide actual sound pressure levels referenced to 20uPa (0 dB-SPL).

Such absolute SPL measurements may be used with and accurately compared to those provided by other calibrated and equal quality SLMs.

Conversely, measurements made with un-calibrated SLMs or those with of low quality (including less then linear frequency response) do not accurately indicate actual sound pressure levels.

SLMs that have been incorrectly user-calibrated or have simply lost their calibration over time also do not provide accurate absolute SPL values.

Relative measurements are simply comparisons between sound pressure levels and do not require absolute calibration nor other “precision” performance characteristics such as a flat frequency response.

Non-calibrated and lower-quality SLMs do provide meaningful comparative measurements as long as the same SLM is employed for these measurements and the measurements are made in the same manner.

Example: A Radio Shack SLM, which is specified as providing +/-2.0 dB-SPL accuracy and which has a less-than linear frequency response, is still able to provide realistic indication of the changes in relative sound pressure levels even though it does not provide accurate measurement of absolute sound pressure levels.

Likewise, the SLM function contained in Smaart may be used “as is” (without calibration) for accurate relative SPL measurement.

One relevant and valid relative SPL measurement that can be made in sound reinforcement work is determining the even-ness of coverage provided by a loudspeaker system.

In this case virtually any SLM is able to indicate any deviations that may exist in the sound pressure levels as the SLM is moved about the coverage area of the loudspeaker system.

In a similar manner we can observe changes in relative SPL as the gain of a sound system is brought up and down and we can accurately determine changes in SPL as a noise source (machinery, for example) is turned on and off.

Available SLMs
As stated previously, SPL measurements made with Class-2 and Class-3 SLM’s provide sufficiently accurate (within +/- 1.0 dB) readings for what we require in sound reinforcement work.

Until recently, SLM devices that met this level of accuracy were prohibitively expensive ($1,200 and up).

Most of the old classic SLM’s are cumbersome to set up and, due to their inherently fragile construction, are seldom used for touring sound or on-site installation work.

Without question the most common SLMs used by live sound folks have been those made by Radio Shack. These appear to represent a very good value, are easy to use and are readily available almost anywhere in North America.

In researching for this article I have found that, along with all of the sound system operators and contractors who own a Radio Shack SLM (or two), there are large numbers of other users including university physics/acoustics and biology departments, municipalities, hi-fi enthusiasts, and theater system installers and property management companies.

And there are at least a few websites devoted to using the Radio Shack SLM for tweaking hi-fi, project studio and sound reinforcement loudspeaker systems and for simple noise measurement.

The original Radio Shack model 33-2050 utilizes an analog meter display and has been in production for the past few decades, and several years ago, the model 33-2055 was introduced, providing a “digital” (alphanumeric) LCD display, recessed and better-protected switches and a maximum/minimum measurement function.

Otherwise, these two models employ the same microphone, base circuitry and battery complement.

 
Alternative SPL Measurement Tools
One alternative to the stand-alone SLM’s described in this article is the SLM function included with some of the better portable RTA’s (real time analyzers) that are often employed for live sound and acoustic measurement.

Models from Ivie, Audio Control, and GoldLine all provide accurate broadband sound level measurement (Class-2 or Class-1, depending on the microphone used) plus octave and 1/3-octave spectrum analysis.

This combination of measurements is very useful for determining both the SPL and the frequency components of the sound/noise under con sideration.

 
Most, if not all, of the current crop of computer-based electroacoustic measurement system provide an SPL function along with their frequency and time-related measurements. Of these, SIM from Meyer Sound and TEF from Goldline are sold with microphones and are calibrated at the factory.

Smaart (any version), SpectraFoo, SysSid, PC-RTA and other similar systems utilize third-party microphones and preamps that must be calibrated by the end-user, using an acoustic calibrator or piston.

Once calibrated, these systems provide very good accuracy and feature a number of variables (weighting filters, etc).

Unfortunately, they are not really “handheld” nor nearly as portable as stand-alone SLMs. And (most importantly) they do not retain their calibration from site to site and therefore require an acoustic calibrator for continued use in measuring absolute SPL’s.

Another product that is a variation of a SLM deserves mention here. SPL “management” systems have been available for perhaps a decade and are directly related to the issues of sound system volume levels.

These simply employ microphones and a software-based SLM system that provides either an obnoxious visual warning or more extreme reactive functions (such as muting of the sound system signal!) when the predetermined SPL limit has been reached or surpassed.

Calibration
In situations where the accuracy of sound level measurements must be recorded and/or verified, it is necessary to calibrate the measurement system both before and after the measurements are made.

Software-based electroacoustic measurement systems that feature an SLM function also need to be calibrated if used for absolute SPL measurements.

An acoustic calibrator provides the standard 94 dB or 114 dB (or both) 1 kHz test tone and should also include the proper size opening or adaptor rings for the microphone(s) you intend to employ.

As is the case with SLMs, calibration devices are manufactured to several levels of tolerance (see table near beginning of article) and they have historically been prohibitively expensive.

Recently there have been several acoustic calibrators introduced that provide very good accuracy for sound system and acoustics measurement.

SLMs may also be calibrated with a pistonphone, but the cost of this device is prohibitively high and the high level of accuracy it provides is not practical for most general sound system related measurements.

Why We Use Them
The simple and snappy answer as to why we use SLMs: so we can know what’s going on.

The need for accurate and repeatable sound level measurement is due simply to the tendency for humans to aurally perceive sound in a very subjective manner. This applies to all aspects of sound perception in the audio field: frequency content, mix “correctness”, system EQ, dynamics, etc.

At some point or another we will disagree on the relative loudness being produced plus these other subjective sound qualities.

Whereas other electroacoustic measurement tools remain very subjective in how we interpret the data they provide, a reasonably accurate SLM provides an irrefutable means to objectively verify how loud the sound is (or is not).

SLM measurements may also be used to verify exactly how much the sound level has been changed (as noted in the section on relative versus absolute measurement).

Anyone involved in live sound system work is aware that there is an ongoing and increasing problem with the issue of volume levels produced in public performance, and specifically, in pop music reinforcement.

It’s wise for anyone involved in this field to become familiar with the actual measured numbers, not to mention the need to continually monitor what we are doing. Issues of personal health and legal liability are an ongoing and increasing concern.

Measurement of unintended noise levels and/or the effectiveness of acoustic treatment are additional exercises that most live sound folks eventually will become involved in.

How We Use Them
For basic every day sound system level measurement, using an SLM is fairly straightforward.

The standard SLM employs a free field microphone and is intended to be held at arm’s length and tipped up slightly towards the sound source.

As you move away from the source and into the reverberant or diffuse field, or when measuring in a multiple source sound field such as on stage, the angle of incidence of the microphone becomes much less of a concern.

When measuring noise levels at installation sites, some care should be taken. If measuring the noise level of a rack room, for example, be sure that the SLM is positioned away from walls, floors and other large boundaries.

Signal-to-noise levels are seldom an issue for loudspeaker system measurement because we can simply turn the system up, more than far enough above the noise floor.

But when measuring the noise from a loudspeaker system (hum, hiss and/or the fan noise that may exist in self-powered systems), or from a mixing console and its power supply, etc., or from other common noises such as HVAC systems, vacuum cleaners, power tools, etc., the unintended noise must be at least 10 dB-SPL lower than the noise being measured.

When measuring and logging levels to help resolve a dispute, both A and C weighted measurements should be provided, as should the distance from the source and any other detailed notes.

Also don’t forget that a non-annotated measurement (“00.0 dB-SPL”) will be interpreted to be flat (unweighted).

General Measurement Guidelines
When conducting sound pressure level measurements there are a few basic rules that apply:

• Keep the SLM at arm’s length, aimed in the direction of the sound source(s) and positioned away from room boundaries. By the way, the mixing surface on a large (40-plus channel) mixing console will effect SPL measurement when the SLM is laid down on the mixing surface or angled against the meter bridge.

• For almost all sound system measurements, use the A-weighting filter and Slow response setting.

• Be sure that you’re measuring within the range that you have selected. SLMs with multiple range positions and those with analog displays, in particular, may provide incorrect readings when the SPL is over/under the selected range.

• Prior to making SPL measurements, ensure that the SLM has been turned on for a few minutes and has reached room temperature.

• Wind and air-blowers will effect SPL measurements.

• Be aware of where you are taking your measurements. For example: if you measure from within the area where the bass energy from the sound system has combined (aka “power alley”) or within a standing wave null, the readings will be non-representative of the SPLs for the majority of the audience.

Tom Young is principle consultant for Electroacoustic Design Services in Connecticut, and he has designed hundreds of systems for churches and similar spaces. Tom’s also the moderator of the Church Sound Community Forum here on ProSoundWeb.

More articles by Tom Young on PSW:
Time Is On Your Side (If You Want It To Be) With Sound System Alignment
Microphone Techniques For Drums In Contemporary Worship Environments
The Aux-Fed Subwoofer Technique Explained

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Mojave Audio Announces New MA301fet Condenser Microphone

Mojave Audio has introduced the MA-301fet, a solid-state, large-diaphragm condenser microphone with a hand-selected, 3-micron thick, 1-inch (diameter) gold sputtered capsule.

Based upon the MA-201fet designed by company founder David Royer, the MA-301fet’s multi-pattern capabilities - cardioid, omnidiectional, and figure-eight (bi-directional) - makes it well suited for numerous applications, including vocals, voice over and broadcast, electric guitar, piano, acoustic instruments, both drum overheads and room mics (drum ambience), and high SPL sources such as kick drums and bass guitar amps.

It also includes a 3-position pickup pattern selector, a 15 dB pad, and a switchable bass roll-off.

The microphone’s warm, military grade FET circuitry, Jensen transformer, and custom designed low-noise resistors combine to deliver a clean signal path with low noise and performance reminiscent of classic condenser microphones.

“The new MA-301fet draws upon the success of our MA-201fet and adds a number of features that many recording engineers have been asking for,” states Mojave Audio president Dusty Wakeman. “With the addition of multi-pattern capability as well as the switchable 15 dB pad and switchable bass roll-off, this new microphone is an incredibly versatile recording tool that, I’m confident, will find a home in a wide range of environments.”

With a MSRP of $895, the Mojave Audio MA-301fet is expected to be available in February 2012.

Mojave Audio

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Thursday, January 26, 2012

PreSonus StudioLive Mixers Now Outfitted With Smaart System Analysis Tools

PreSonus has begun incorporating Rational Acoustics Smaart Measurement Technology directly into the software used to control its StudioLive digital mixers.

PreSonus Virtual StudioLive (VSL) remote-control/editor/librarian software will now incorporate Smaart Spectra and Smaart Locator, powerful tools for sound-system analysis and optimization, as part of PreSonus Universal Control 1.6, expected to be available later this spring.

Smaart is not a single technology but an evolved collection of audio measurement tools and techniques. Using Smaart technology, users can tap into the power of the StudioLive mixer’s EQ to improve the sound of their system.

With Smaart-enhanced VSL users can view the spectral content of their mix in real time, and easily make changes.

Clicking on the Graphic Equalizer button in Universal Control 1.6, Smaart Spectra’s Real Time Analyzer activates Spectrograph’s algorithms, displaying the spectral content of whatever is routed though a particular graphic EQ.

Users can also activate a Real Time Analyzer, much like the plug-in used in PreSonus’ Studio One 2.

In addition, the Smaart Spectra Spectrograph display can help to precisely identify feedback frequencies, enabling even less experienced users to more easily tune their loudspeakers to the room.

Smaart Spectra graphs a continuous series of spectrum measurements, showing frequency on one axis, time on another, and level indicated by colors - making it particularly useful for quickly identifying feedback frequencies, which can be easily addressed using StudioLive GEQs.

“This is just the start of a beautiful relationship - PreSonus products are about to get a lot Smaarter,” says PreSonus chief technology officer Bob Tudor. “This is the real thing, trusted by acousticians and live sound engineers the world over. We’re thrilled to be incorporating Smaart Spectra and Smaart Locator into Virtual StudioLive.”

PreSonus

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Wednesday, January 25, 2012

In The Studio: Audio Engineering Faux Pas – Part 1

Audio engineering faux pas and ways to embrace the tools you already have, while pushing them to their fullest potential.

 
This article is about a shift in perspective mostly. It’s about really embracing the tools you already have and using them to their fullest potential.

For example, I have a 12-channel Mackie mixer that I double as an analog external summing amp. It contains FX sends and returns, allowing me to utilize any number of effects units (not to mention plug ins), which can be brought back in to the stereo bus or run parallel.

I also use this mixer to control every piece of audio/visual equipment in my apartment from my bedside.

Thus rises the essence of my point. If you look, there is most likely more at your fingertips than you would realize.

Here are a few other ideas:

Digital Distortion
Exceeding 0 dBFS is not necessarily always a bad thing. In fact, it can be used in many recurring situations. I often use it as an effect to gain an overload distortion quality which may sometimes work inherently with guitar tracks, certain vocal tracks, and definitely synthesizers.

It can also be used to achieve a distorted lo-fi effect or an over-compressed quality without using an actual compressor.

Elastic Audio
One of the main goals when using elastic audio is to mask the fact that you are using elastic audio. Instead of taking this approach, try embracing the imperfections inherent to the recalculation of waveforms. Stretching or compressing waveforms drastically can produce extremely unique processing errors, using a monophonic algorithm for a polyphonic instrument can produce interesting aliases, and using the varispeed algorithm on anything can produce seemingly infinite outcomes.

Buffer Size
For a unique glitch effect, drop your playback engine’s buffer size to its lowest playable setting and record the output to a separate track. This allows you to practically “audiosuite” a decrease in buffer size for specific instruments (or across the board if you’re feeling frisky).

Delay Compensation
Delay compensation can be used similarly to the buffer size. Additionally, when combined with precise time adjustments between different tracks, rhythmic feels can be solidified as well as totally phased out of control. There are many effects that can be attained, either in plug-in form, or using the “audiosuite” method.

A word of advice on opinions… f*ck everyone else.. try things for yourself.

Precautions
First of all, nothing is the end of the world (except the apocalypse) and there will always be more work (until the apocalypse). The music will go on (until the apocalypse).

That being said, there are very good reasons for a lot of the precautions taken within a DAW recording environment.

Now, I’m not necessarily saying go out and get the trashiest plug-ins that you can and try to figure out how to make them work for you (unless you want to, which would be kick-ass), but keeping this perspective allows you to work better with the tools you already have, and allows for your DAW to be a creative landscape, as a canvas on which to draw, rather than a harsh digital program to which you must adhere.

You and the DAW are just as flawed as each other in that your existence and the becoming of that existence are of imperfect origin. Therefore, just as you have to overcome imperfections in yourself, you must overcome the imperfections inherent to any piece of equipment. Besides, didn’t you get in this to push buttons, turn knobs and have total control? So take it.

Samuel O’Sullivan has been playing various instruments and composing within the bounds and mixtures of multiple genres for more than 10 years. He first established as a drummer/percussionist, has made his mark as a guitarist, vocalist, pianist, violinist, composer, and recording engineer. In addition to producing albums for various bands, O’Sullivan produces his own music under the name “A Mess of a Mind”.

Be sure to visit the Pro Audio Files for more great recording content.

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Behringer Introduces FIREPOWER FCA610 & FCA1616 Recording Interfaces

At the NAMM 2012 Show, Behringer introduced the FIREPOWER USB/FireWire audio interfaces, the FCA610 and FCA1616.

The FCA610 and FCA1616 incorporate 24-bit/96 kHz A/D-D/A converters, support Windows XP/Vista/7 plus Mac OS X, and provide onboard phantom power for use with condenser microphones. 

Due to its small size and low-latency operation, 6-input/10-output architecture, and two XENYX mic preamps, the FCA610 is a tool for traveling musicians who record and edit on their laptops. The portable FCA610 can receive power from a computer’s 6-pin FireWire bus or via the included external power supply.

Built-in MIDI I/O allows for easy connectivity with keyboards and other outboard MIDI hardware. All standard I/O formats are supported, including analog and S/PDIF (both coaxial and optical).

The half-rack-space FIREPOWER FCA610, which stows easily in a travel kit, can also be used as a premium 2-channel mic preamp and A/D-D/A converter.

With an expanded 16-channel I/O, four XENYX mic preamps and ADA8000 ADAT connectivity, the FCA1616 is more suitable for permanent applications as well as live performance multi-track recording rigs.

All standard I/O formats are supported; including analog, S/PDIF (coaxial and optical), ADAT and S/MUX, and a built-in MIDI I/O allows the user to connect keyboards and other outboard MIDI hardware.

The single rack-space FCA1616 also features eight analog Inserts for use of external effects such as compressors, gates and EQs. A dedicated power supply comes with the unit.

Included with both FIREPOWER interfaces is a massive software download at behringer.com that includes the widely popular Audacity audio editor, as well as a selection of audio software such as Podifier, Juice, Podnova and Golden Ear. Also included are more than 100 virtual instruments and 50 FX plug-ins.

FIREPOWER Features:
• Low-noise, high-headroom audio interface with 24-bit/96 kHz resolution
• Operates as multi-channel audio and MIDI interface via FireWire and USB2.0
• XENYX mic preamps with individual switches for Phantom Power, Pad, Low Cut and Hi-Z
• Direct Monitoring and Main Volume control on hardware front
• Two headphone outputs with individual volume control, mono and source signal select for flexible monitoring purposes
• Level control of stereo or 7.1 active loudspeaker systems with a single knob turn
• Smooth cross-fading between inputs and DAW playback signals
• Status and signal presence indication for all analog and digital I/O
• Standard port for Kensington security lock

Behringer

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d&b audiotechnik Announces Several Updates On Software Workflow Tools

d&b audiotechnik has announced several new updates for its ArrayCalc simulation software, R1 Remote control software and the R10 Service software, as well as a firmware update for the E-PAC amplifier.

The new d&b ArrayCalc V6.7 software, available in March 2012, will contain the d&b R1 Remote control software’s export function, which removes the original copy and pasting process of the amplifier settings from the simulation software ArrayCalc to the R1 Remote control software.

It also automatically generates a workspace in an R1 project file as well as exporting the amplifier settings to an R1 control settings file. This workflow sequence replaces the manual process entirely.

From virtual optimization in the ArrayCalc simulation software to the rapid verification of the simulated settings and the fine adjustment in real life on site, using this d&b software offers an effective workflow for users from their laptops.

The service workflow has also been streamlined so that the d&b R10 Service software from V2 onwards automatically and on demand searches for the latest available amplifier firmware versions and Remote software updates on the d&b server via the internet and downloads them. This removes the need to manually look for any updates on the website.

Even though the E-PAC amplifier was discontinued in 2007 it is still being supported with a new firmware update V4.17.

To download the update, go to www.dbaudio.com.

d&b audiotechnik

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Tuesday, January 24, 2012

L-Acoustics U.S. Sets Training Dates for KARA, KUDO & SOUNDVISION

L-Acoustics U.S. has announced its first two product training sessions for 2012.

The first three-day training is set for February 20 to 22 in Red Hook, NY and will specifically focus on the new KARA modular line source system and SOUNDVISION version 1.9.

The second session, hosted in Oxnard, CA exactly one month later from March 20 to 22, will cover the large-format KUDO line source system and SOUNDVISION.

“We’re particularly looking forward to our KARA and SOUNDVISION session in Red Hook as it marks our first official East Coast training,” says L-Acoustics head of U.S. touring support Scott Sugden. “We’ve had a lot of interest in a regional event like this from our eastern customer base and we’re very happy to now make it a reality for them.”

Primarily designed for technicians, mix engineers and sound designers referred by L-Acoustics Rental Network agents and clients, the first two days of each training will offer a blend of theoretical knowledge and field procedures focusing on operating and optimizing either KARA or KUDO in a safe and controlled environment.

A third day, which can be attended separately or in conjunction with the KARA/KUDO training, will be dedicated to covering the manufacturer’s SOUNDVISION 3D acoustical modeling software.

Upon completion of these seminars, attendees will receive a certificate of attendance.

The number of participants for both the Red Hook and Oxnard training sessions is limited to 12 people and priority will be given to L-Acoustics Rental Network agents and system owners.

For additional details on the training seminars and their related costs, click on the Support tab at www.l-acoustics.com or contact .(JavaScript must be enabled to view this email address).

L-Acoustics

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Public Beta Version Of JBL HiQnet Performance Manager Software Available

JBL Professional HiQnet Performance Manager sound reinforcement system design software is now available to users as a public beta version.

Previously it had been available to a closed group of 60 worldwide beta testers.

HiQnet Performance Manager is a highly refined user interface that facilitates the design of touring and live performance venue sound reinforcement systems.

Created especially for touring and theatrical sound engineering, Performance Manager is an application-specific iteration of the HARMAN HiQnet System Architect configuration and control software application for professional-grade audio system integration.

“JBL HiQnet Performance Manager is a powerful tool that makes the design, setup and tuning of a JBL VERTEC loudspeaker system a lot faster and more efficient,” notes Adam Holladay, market manager, HARMAN System Development and Integration Group.

“The response from our beta testers has been extremely positive,” continues Holladay. “In venue after venue, JBL VERTEC system engineers have been able to achieve consistently higher levels of performance than before. They note that they are also able to set up a system more quickly, saving time and money. Now that we are making it available in a public beta version, many more users will have the opportunity to take advantage of the benefits of Performance Manager as a free download.”

The public beta version of Performance Manager is available at http://hiqnet.harmanpro.com.

The beta version is functional except that users will need to apply for a license key in order to go online and access the online operate modes of certain system devices. An in-depth series of training videos can be viewed at http://hiqnet.harmanpro.com/training/.

The full version of JBL HiQnet Performance Manager will be available in early 2012 at a suggested retail price of $399.

JBL HiQnet Performance Manager provides a comprehensive, step-by-step workflow that directly corresponds to real-world system configuration, taking the workflow paradigm introduced in System Architect 2 to a higher level of functionality for any live performance audio application. It is fully integrated with JBL’s Line Array Calculator II loudspeaker configuration and acoustic modeling software.

The user begins by loading templates of the speaker arrays used in the system, and then runs Line Array Calculator II for each array as part of the initial sound design task of determining how many and which type of loudspeakers are required to cover a given venue.

For each array, Performance Manager automatically loads the passive VERTEC or powered VERTEC DrivePack DPDA line array configuration into the main application workspace – the first of many automated design processes native to the software application. Loudspeakers can also be manually loaded into the templates if desired.

Once the user defines the required amplifier parameters for the passive loudspeakers within the arrays, Performance Manager automatically loads the correct number of Crown Audio VRACK or other user-determined amplifier racks into the audio system.

The software then associates the amplifier outputs with the bandpass crossover inputs for the selected array and programs the amplifiers with the correct JBL preset data, as well as gain shading and JBL Line Array Control Panel equalization parameters that are determined in JBL’s Line Array Calculator II as part of the modeling process to optimize sound pressure level and frequency response over the defined audience geometry.

Representations of the bandpass inputs for each loudspeaker section are overlaid onto the arrays, enabling the user to easily visualize the array configuration, whether JBL DrivePack-powered or driven by external Crown power amplifiers. Performance Manager software also significantly simplifies system-networking configuration – the user can simply drag and drop devices discovered on the network onto the pre-configured devices within the Performance Manager workspace to synchronize all addressing and parameter values.

In addition, the Performance Manager graphical interface provides embedded control panels for array calibration, time alignment and system EQ which utilize input section digital signal processing resources available in either Crown I-Tech HD power amplifiers or JBL DrivePack-powered loudspeakers with DPDA digital audio input modules. Input attenuation, equalization, delay settings and bandpass controls are all readily accessible directly within the main application workspace along with flexible grouping and comprehensive solo/mute functionality for system testing.

Once system tuning is complete, Performance Manager’s Show Mode display is optimized for the actual live performance, offering appropriate adjustment control ballistics for equalization and dedicated monitoring interfaces for levels, speaker loads, thermal conditions and AC power requirements.

The workspace for all stages of Performance Manager’s workflow has a common design motif, with monitoring functions overlaid on top of the same loudspeaker bandpass representations within the workspace, making visualization easier and more consistent across various workflow screens.

Harman Professional
JBL Professional

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Waves Audio Unveils NLS Non-Linear Summing Plug-In

At Winter NAMM 2012, Waves Audio unveiled the new NLS Non-Linear Summing plug-in.

NLS brings users the analog summing sound of three legendary consoles.

Created in association with three of today’s leading producers/engineers – Mark “Spike” Stent, Mike Hedges and Yoad Nevo, NLS captures the sound of:

—The SSL 4000G console belonging to Mark “Spike”’ Stent (Radiohead, Björk, Muse, Maroon 5, Madonna)

—The EMI TG12345 Mk 4 desk owned by Mike Hedges (The Cure, Siouxsie and the Banshees, Dido, Faithless, Manic Street Preachers, U2),  heard on such timeless recordings as Pink Floyd’s The Dark Side of the Moon

—The Neve 5116 console custom-made for Yoad Nevo (Bryan Adams, Pet Shop Boys, Sugababes, Goldfrapp, Air)

In the creation of this plugin, Waves meticulously modeled over 100 individual channels in all, painstakingly analyzing and recreating the distinctive color, character, and behavior of each and every input and summing bus amp.

Waves NLS delivers to the digital realm the depth and richness that has long been associated exclusively with analog gear.

The Waves NLS Plugin will be available Q1 2012 with pricing to be announced.

Waves

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iZotope Introduces Mastering Essentials For Acoustica Mixcraft Pro Studio 6

iZotope has announced Mastering Essentials, a new tool designed specifically for Acoustica Mixcraft Pro Studio 6.

By licensing elements from Ozone, iZotope’s flagship mastering suite, Acoustica is delivering a basic mastering solution approachable for musicians of any level. 

“Mastering Essentials is the perfect intro to mastering,” explains Alex Westner, iZotope director of business development. “With over 80 presets to choose from, users can get going quickly with great results. We take it a step further, though, by letting musicians experiment with customizing their sound.

“The three additional EQ, Reverb, and Tube Amplifier modules allow users to grow and expand their capabilities as they become more comfortable with mastering their own music.” 

iZotope Mastering Essentials will make its debut as part of Acoustica’s Mixcraft Pro Studio 6, the latest edition of the company’s Windows-based DAW.

“Everyone trusts the iZotope name when it comes to mastering,” says Dan Goldstein, chief technology officer of Acoustica. “Mastering Essentials enhances our Pro Studio bundle with a top-notch tool for polishing any project.” 

Mastering Essentials Key Features:

—Valve EQ module: high-quality 4-band EQ
—Room Simulation (Reverb) module: controls for reverb and stereo widening, including a vectorscope for visualizing the stereo spread of the audio
—Tube amplifier module: Bass Compression, a Tube Limiter, and Tube Saturation components
—More than 80 presets, including presets for general use, special FX, audio enhancement, restoration, and over 40 different genre presets

iZotope
Acoustica

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