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Paul Figgiani
Audio Post Production and Loudness Compliance Support
Audio Post Production and Loudness Compliance Support
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Noise Reduction

For Broadband Noise Reduction I use the Denoise module included in the stand alone version of iZotope's RX4 Advanced Audio Repair and Enhancement suite.

When working in a DAW, it is possible to insert an instance of their Dialogue Denoiser plugin into the signal chain. The plugin works really well when used to apply a second pass of moderate noise reduction, where  audio was initially processed with the Broadband Denoise module.


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Recording Multiple Skype Clients On A Single Host System

It is possible to record two (or more) independently connected Skype clients on discrete tracks on a single computer in RT. The workflow requires independent Mix-Minus feeds configured in a supported DAW such as Pro Tools or Logic Pro.

Proper configuration will yield latency free results.

Let me know if you have any questions


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Real Time Print To Track

Logic and Audition users will be familiar with the term Bounce to Track. This process allows the user to perform an Off-line Mixdown of a selected group of Session Tracks without physically exporting. In most cases the Mixdown appears on a supplemental target Track.

Bouncing Off-line is a time saver. However it can be precarious. It would be irresponsible to submit a finished piece of audio to a client without 100% conformation the bounced delivery file (most likely slated for distribution) is glitch free. In essence it is imperative to throughly check your piece prior to submission.

Off-line Bounce (aka Bounce to Disk) was once notoriously absent from Pro Tools. Avid finally implemented support a few years ago.

In professional Post Production, engineers may perform a real time (On-line) Bounce of a mix Session. The process is commonly referred to as Printing. It requires the operator to sit through the Session in it’s entirety. Besides glitch detection capabilities, it is possible to edit clips before the playhead reaches their location. As well, you can edit clips and/or sub-segments within a previously completed Print and only re-Print the manipulated segment.

So how is this done? Simple - if the DAW or Interface supports it. For instance in Pro Tools the user can route Bus outputs to the input of a standard Audio Track. The key is you can ARM a standard Audio Track to record any signal that is passing through it. This would be the Print Track.

Adobe Audition CC does not support direct Bus Output —>> Audio Track routing. However, it is still possible to implement a Print workflow. You will need a supported Audio Interface with a Mix Return. Simply route all Session Tracks and Buses to the Main Output. Then add a supplemental Audio Track. Set it’s input to Mix Return. ARM the Track to record and fire away.


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Hard Limiting Avoidance

I believe the term Hard Limit should be avoided when discussing Spoken Word audio processing. It often implies the use of a Limiter to drive audio levels against a user defined Maximum Peak Ceiling to boost average, perceptual Loudness. This approach often leads to artifacts and distortion.

The ultimate goal for Spoken Word audio distribution is intelligibility. An extended dynamic range prior to compliance processing is not recommended. Optimizing Spoken Word audio dynamics prior to Loudness Normalization will alleviate the necessity for aggressive (“Hard”) Limiting. This is especially important when processing for Internet/Mobile/Podcast distribution, where significant added gain may be a necessity.

In essence an ITU compliant Limiter should function as a failsafe for Maximum True Peak adherence. If an inserted Limiter is applying significant gain reduction, the source audio most likely lacks proper optimization. If this is the case the source audio should be re-mastered. Or, producers should consider shooting for a slightly lower Integrated Loudness target within reasonable tolerance. Both options will reduce applied Limiting.

ToneBoosters recently announced a new version of their highly regarded TB Barricade EBU/ITU Peak Limiter, branded Barricade, V4. Besides it’s UI redesign, Barricade now features two stage serial processing with the inclusion of a customizable Dynamics Compressor located before the Limiting stage.

I believe this tool is well suited for Spoken Word audio processing. The Compression stage provides internal capabilities for Spoken Word optimization before failsafe/compliance Limiting. It is feature rich and very affordable.

Note the plugin is available in AU, VST, and VST3 formats. The developer does not distribute Pro Tools supported AAX plugins.



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Acon Digital Acoustica 7

With NAMM 2017 approaching - new product announcements are imminent. This morning Acon Digital announced their new editing, mastering, and restoration suite - Acoustica 7 Premium Edition.

The press releases discloses an interesting feature set, most notedly the inclusion of their highly regarded Digital Restoration Suite processing tools. I’m particularly fond of their DeNoise module. They’ve also integrated aspects of their well designed and highly capable Equalize, Verberate, and Multiply plugins.

What interests me most about Acon’s new product offering is support for EBU/ITU compliant Loudness Metering and Normalization with adjustable absolute and relative Gate thresholds. I think the engineers at Acon realize in todays’s world an audio toolset must include support for Loudness Normalization and compliance processing. Considering the quality of Acon’s current product offering I anticipate an impressive implementation.

Acoustica 7 Premium Edition runs on Mac and PC. It hosts third party VST, VST3, and AU plugins. Surround format support is available for single track or multitrack sessions.

Acoustica 7 Premium Edition will be available Q2 2017 for $199 USD. A limited Standard Edition will be priced at $59 USD.

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Elixir ITU True Peak Limiter

Certain ISP/True Peak Limiters provide added compliance processing flexibility. Case in point: Elixir by Flux.


Before processing or Loudness Normalizing, execute an offline measurement on an optimized source clip.

An optimized audio clip may exhibit the benefits of various stages of enhancement processing such as noise reduction and dynamic range compression.

The displayed clip (see attached image) checks in at -19.6 LUFS. It requires +3.6 dB of gain to meet a -16.0 LUFS Integrated Loudness target. Based on the pre-existing peak ceiling approximately 1.5 dB of limiting will be necessary to establish a -2.0 True Peak maximum.

Processing Example

We use the Limiter’s Input Gain setting to take the clip down to -24.0 LUFS (-4.4 dB for the measured displayed clip).

The initial -24.0 LUFS target will restore headroom and establish a consistent starting point for downstream limiting accuracy. This will allow the Threshold and Output Gain settings to be recognized and implemented as static parameters for all -16.0 LUFS/-2.0 dBTP (stereo) processing. The Input Gain setting however will be variable based on the measured attributes of the optimized source.

Set the Threshold to -10 dB(TP) and the Output Gain to +8dB. The processing may be implemented offline or in real time. The output audio will reflect accurate targets (-16.0 LUFS/-2.0 dBTP) and the applied limiting will be transparent.


The proprietary functional parameters included on the Elixir Limiter are not necessarily included on Limiters designed by competing developers. In essence the described workflow may need to be customized based on the attributes of the Limiter.

The key is the “math” and static parameters never change, unless of course you decide to alter the referenced targets.

Let me know if you have questions …

-paul. [no affiliation with Flux].

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Motu I/O - Sierra Compatibility and DSP

Motu quietly announced device support for Mac OSX Sierra (10.12.x). I own an Audio Express. It is integrated with a Mackie Onyx 1220i Mixer and a 48 point Patchbay.

Motu’s latest device offering is the UltraLite Mk4 USB Interface. Besides extensive I/O capabilities (18 in, 22 out) - the device features robust 32-bit Floating Point DSP. The processing options are wrapped into the bundled Mixing and Routing software that closely resembles a professional DAW.

When reviewing the specs. I discovered the inclusion of two DSP Compression options: Conventional and Leveling. The latter is made possible by the software modeling of the legendary LA-2A Optical Compressor. It is well suited for session activated AGC. Additional DSP options include EQ, HPF, Gating, and Reverb.

Effects can be applied when operating the device as an audio interface or as a stand-alone mixer without a computer. Web App. and Mobile Device control is supported as well.

My particular device (Audio Express) features a Mix 1 Return Includes Computer Output option. This makes it super easy to implement a Skype Mix-Minus via USB I/O when using a compatible DAW such as Pro Tools, Logic Pro X, or Audition. I’m certain the UltraLite Mk4 is capable of this as well. It is also possible to easily implement an analogue mix-minus for Telephone Hybrid integration.

Motu devices have a certain cutting-edge feel with regards to build quality and control. The preamps are quiet and the visual displays are stellar.

Link to the UltraLite Mk4 Manual:

The UltraLite Mk4 retails for $599.

***Additional information on the history of the LA-2A: Please note the plugin discount that I refer to in the article is no longer applicable.


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ART TPatch - $40 Balanced Patchbay

Expanding on a short twitter conversation with +Ray Ortega

The displayed implementation allows the operator to route a processed Mic signal to a solid state Audio Recorder as well as an Audio Interface. The key is the signal routing to both target devices is balanced. TRS patch cables will be required.

Patchbays, In General

In basic terms a Patchbay provides convenient and flexible access to your component’s inputs and outputs, all from a central location. Connected component outputs are mirrored on the upper frontal row of the Patchbay’s jacks. Connected component inputs are mirrored on the lower frontal row of the Patchbay’s jacks. Each jack is referred to as a patch point. I use a Samson Patchbay with 48 balanced 1/4” jacks (24 inputs + 24 outputs). It is commonly referred to as a 48 point Patchbay.

Rear Jacks

The upper row of jacks on the back of a Patchbay are inputs. They accept the output signal from any supported connected source. Output signals will be available on the corresponding upper frontal Patchbay jacks and will be active for further patching.

The lower row of jacks on the back of a Patchbay are outputs. They accept the input signal from any supported connected source. Input signal routing will be available on the corresponding lower frontal Patchbay jacks and will be active for further patching.

Lifted from the ART TPatch Manual:

"Each vertical grouping of two jacks on the front of a Patchbay and the corresponding two jacks and switch on the rear of the Patchbay comprise a module and provides one channel of linked input/output audio connections. For example, the jacks labeled 1 and 5 on the front panel and the jacks labeled 1 and 5 on the rear panel, along with the associated switch, are one module. All four modules are identical and each may be individually configured for Half-Normalled or Normal operation".


In this documented example vertical Patchbay jack pair 1 and 5 is set to Half-Normalled. This creates an internal signal path or patch between jack 1 (the source output) and jack 5 (the target input).

As well, a Half-Normalled configuration allows the user to tap into the output signal path on the front of the Patchbay (upper jack 1) and patch the signal to the input of a secondary target. Note the upper frontal output patching will not interrupt the Patchbay’s internal Half-Normalled signal flow.

(Half-Normalled configuration is documented in the side view graphic at the lower right of the attached image).


The Mic is connected to a dbx 286s. The dbx 1/4” output signal passes internally through and out of the Patchbay. It is ultimately routed to the Recorder’s input. Upper jack 1 on the front of the Patchbay also contains the source output. This can be patched to any secondary target input - in this case an Audio Interface.


”300K Listeners?”, “Mixed By?”, and Pro Tools Requirements

A bit of personal reflection …

I just listened to the first installment of Gimlet’s “Season 4” Startup Podcast. In this particular episode the host makes reference to a recent launch of a new internal program, touting it as a major success:“300k listeners”, “Straight to #1 on iTunes.” My question is - how does Gimlet know in fact that 300K people actually listened? Do they have the Download vs. Consumption puzzle that plagues the space solved?

By the way, in general I think the Podcast Season model is frivolous. I see no benefit - unless of course a particular program set is solely based on a specific ongoing storyline. If you are an independent producer talking about tech for example, the fragmentation of Seasons makes very little sense.

Also in this episode, the staff discusses a recent inquiry by the U.S. Department of Defense. Apparently the DOD offered to pay Gimlet $500K for the creation of a branded Podcast. I was not surprised in the least when they disclosed their decision to pass on the opportunity. Apparently they were concerned about the high profile alliance and the inevitable public perception. Personally, I think their decision was ridiculous. They are a production company. I’m sure the DOD had zero plans to allow Gimlet to expose their political views. Is it possible Gimlet’s concern was based on whether certain members of their staff would have issues with the underlying association?

Sorry - but to me this was/is a bad business decision … (have they reconsidered?)


Many of the Public Radio rooted Podcasts with high production value include post roll and/or out going credits, making references to various aspects of the production such as “Edited By" and “Mixed By.”

I think it’s important to note that the majority of these programs are highly scripted. I have no problem with that. However I believe the so called Editors are making all the decisions in regards to the assembly of audio components (Beds, Cuts, intricate Dialogue Edits, etc.) well before the audio engineers enter the mixing stage. No doubt the mix engineers are guided by elaborate EDL’s (Edit Decision Lists). The point is let’s not confuse the differences between contextual editorial and audio editing/ mixing.


I’m always interested in developing relationships with talented people who create compelling content. I often browse Job postings in the space specifically related to Audio Engineering and Audio Post. The vast majority of job solicitations require Pro Tools expertise and proficiency.

I’m perfectly comfortable with the requirement. However I think it’s safe to say that Pro Tools is not the tool of choice for the vast majority of audio savvy Podcast Producers. Is it capable? Of course it is. Regardless of what you think it is still the industry standard for Motion Picture Sound, Broadcast, and Music Production. And yes, it does require the use of an iLok key/license. The model is ubiquitous in the professional space, and should not be viewed upon as some sort of burden.

Logic Pro X has a faithful user base in the world of music creation/recording. And let’s not forget Audition. Adobe continues to listen, resulting in the availability of a viable and powerful tool.


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TC Electronic Clarity M Presets

It’s been about a week since I integrated TC Electronic’s Clarity M Loudness Meter into my rig. It’s a nice piece of kit. Besides the impressive overall performance and design - the build quality is surprisingly robust. The screen/graphics are excellent as well.

One of the main benefits of this device is the off-loading of resources when mixing. I think it’s safe to say it’s probably less taxing on the host system compared to running the software DAW plugin version. As well, having all measurement descriptors displayed simultaneously makes the meter’s use very efficient.

Note the device is not flawless. For instance I’ve noticed a few minor anomalies when running it in Adobe Audition. Also, the device sometimes locks up in a continuous measurement mode when performing an offline bounce in Pro Tools. Firmware updates are easy to install. I’m guessing all issues will be addressed in a future update.

Clarity M ships with multiple banks of Presets. I thought it would be interesting to expose TC Electronic’s obvious support for the -16.0 LUFS Integrated Loudness Target for the Internet and Mobile Distribution Platform. As you can see in the attached graphic - Podcasting, Music Mastered for iTunes, and Web Video presets all adhere to the -16.0 LUFS Target recommendation. This is no surprise considering extensive research by TC Electronic and referenced in (and not limited to) “Audio for Mobile TV, iPad, and iPod” by Thomas Lund.

Note the -16.0 LUFS Target is for stereo files. For equal perception in mono, you must apply a -3.0 LU gain offset resulting in a -19.0 LUFS Target.

In terms of distribution Targets, there is always room for a reasonable amount of subjective deviation. This is especially true when the suggested Targets require aggressive limiting. In this case it is acceptable to shoot for slightly lower Targets. Be advised this should be handled with care. Wide Target variations (at least in my view) will potentially negate the establishment of standardization in the space.


Clarity M:

“Audio for Mobile TV, iPad, and iPod” by Thomas Lund
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