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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.

-paul.
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Noise and Downward Expanders

The attached graphic includes three frequency analysis plots. At the far left the analyzer reflects high level (isolated) broadband noise. The most prominent aspect is a loud, persistent 120Hz hum. This noise is constant throughout active and inactive speech.

The middle frequency plot displays full Downward Expansion attenuation of the noise. The attenuation occurs during prolonged passages of inactive speech. In essence the combined signal level (noise + speech) has dropped below a defined Threshold thus initiating signal attenuation (Gate closed).

The frequency plot located at the far right displays active speech with it's signal level above the defined Threshold (Gate open). Notice how the inherent noise is once again persistent (and audible).

Please note a Downward Expander is incapable of distinguishing speech from noise. The process solely relies on combined signal level and it's relevance to a defined Threshold. Basicilly in this example passages of active speech will (for the most part) include audible noise.

Keep in mind the source signal level of inherent noise will determine the seamlessness of the transition between persistent noise and attenuation. If the level of the noise floor is very low, transitions will be less prominent and therefore acceptable to the listener. However if the signal level of the noise floor is elevated - transitions between audibility and attenuation will be jarring. This is certainly not recommended.

-paul.

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Spoken Word Intelligibility Optimization

Can you interpret each of the 7 reference points?

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Dual Mix-Minus in Audio Hijack

On May 16th I documented how to record Multiple Skype Clients on a Single Host System using Pro Tools or Logic Pro X. The workflow requires running two independent (simultaneous) instances of Skype on a single machine. Each participant (+ Host) may be recorded in RT on a discrete track. Two DAW implemented Mix-Minus feeds make it all work with no dependency on an external mixer’s Aux Sends.

http://www.producenewmedia.com/recording-multiple-skype-clients-on-a-single-host-system/

A few people asked if it was possible to implement a similar workflow in Audio Hijack. It is. The attached image displays the necessary Audio Hijack configuration.

You will notice I use the same virtual (“Aggregate”) Audio Device designed for the Pro Tools/Logic implementation. Also notice I am not using Skype Application modules in Audio Hijack. Instead, Input Device modules are used for source audio assignments and Mix-Minus signal routing.

Please note concerns regarding problematic latency have been mentioned. The fact is monitoring the session through a supported Audio Interface’s headphone bus (via hardware) will eliminate latency. Of course any in the box software processing applied to the host mic signal will be absent from the monitored audio. If you prefer to monitor a processed signal - use external gear such as the dbx 286s or the Wheatstone/Vorsis M1. Additional processing may be applied in post.

-paul.
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Dual Mix-Minus in Audio Hijack

On May 16th I documented how to record Multiple Skype Clients on a Single Host System using Pro Tools or Logic Pro X. The workflow requires running two independent (simultaneous) instances of Skype on a single machine. Each participant (+ Host) may be recorded in RT on a discrete track. Two DAW implemented Mix-Minus feeds make it all work with no dependency on an external mixer’s Aux Sends.

http://www.producenewmedia.com/recording-multiple-skype-clients-on-a-single-host-system/

A few people asked if it was possible to implement a similar workflow in Audio Hijack. It is. The attached image displays the necessary Audio Hijack configuration.

You will notice I use the same virtual (“Aggregate”) Audio Device designed for the Pro Tools/Logic implementation. Also notice I am not using Skype Application modules in Audio Hijack. Instead, Input Device modules are used for source audio assignments and Mix-Minus signal routing.

Please note concerns regarding problematic latency have been mentioned. The fact is monitoring the session through a supported Audio Interface’s headphone bus (via hardware) will eliminate latency. Of course any in the box software processing applied to the host mic signal will be absent from the monitored audio. If you prefer to monitor a processed signal - use external gear such as the dbx 286s or the Wheatstone/Vorsis M1. Additional processing may be applied in post.

-paul.
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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

-paul.



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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.

-paul.
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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.

http://www.toneboosters.com/tb-barricade-v4/

-paul.
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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.

http://links.acondigital.com/m/10985752/
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