SSL 4000 Series

Waves has sporadically released the SSL 4000 Series Channel Strip plugins independently and free from previous bundle restrictions. This is great news. What’s even better is their limited time pricing of $29.

On the surface both channel strips feature various equalization stages and dynamics processing modules. There are a few discernible differences between the E-Channel and G-Channel versions. Also, certain shared and/or unique parameters and features are worth discussing.

Equalization

The main difference between the two versions is how certain gain settings within two specific EQ modules affect bandwidth (aka “Q” values).

For instance, the E-Channel’s HMF and LMF module bandwidth remains constant at all gain levels. Conversely, the G-Channel’s HMF and LMF module bandwidth will vary based on the gain level settings. Specifically, as a filter’s gain level is increased or decreased, the bandwidth narrows and potentially becomes more surgical.

Both versions include a Split option within the High-Pass/Low-Pass filter modules. When activated, the filters are placed before the dynamics modules.

The E-Channel’s HF and LF eq modules are (by default) Shelving Filters. Pressing the BELL selector changes their attributes as described.

The G-Channel’s HF and LF eq modules feature fixed Shelving Filters. As well, the HMFx3 option multiples the HMF frequency by three. The LMF /3 option divides the LMF frequency by 3.

The E-Channel’s Dyn S-C option inserts the filters and EQ into the dynamics sidechain for frequency sensitive processing. The G-Channel’s FLT Dyn S-C option inserts the filters into the dynamics sidechain (Note: “filters” refers to high-pass/low-pass modules).

Dynamics

The Compressor features soft-knee processing with automatic makeup gain. The default attack time is slow and program dependent. Activating F.ATK sets the attack time to 1 ms. The Compressor will function as a limiter when it’s ratio is set to infinity (Note: attack time attributes are the same in the Expander/Gate module).

The following in-depth Compressor and Expander/Gate attributes are listed in the native SSL Duende Plugin documentation:

Both versions of the plugin include two DYN To options:

Bypass: This deactivates all dynamics modules
CH Out: This inserts the dynamics processing at the output (post EQ)

Additional Features

Both versions include a switchable Analog Emulation stage, Phase Reverse, Input Trim, and Output Fader. The Level Meters are switchable for Input and/or Output level monitoring.

The plugins are aligned as follows: -18 dBFS = 0dBu

-paul.

References:
Waves Audio Plugin User Guide
SSL Duende Documentation

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Aphex 320D Compellor

What is a Compellor? In short it is a Compressor-Leveler-Limiter. The device is specifically designed for the transparent control of audio levels.

It operates as a stereo processor or as a two-channel (mono) processor supporting independent channel control.

The device includes 3 interactive gain controllers:

– Frequency Discriminate Leveler
– Compressor
– Limiter

Additional features include a Dynamic Release Computer (DRC), Dynamic Verification Gate (DVG), and a Silence Gate.

The original device (model 300 Stereo Compellor) was released in 1984. The product line evolved and culminated in 2003 with the release of the 320D. Through the years the Compellor has been widely used in professional broadcast, post houses, recording studios, and live venues.

In 2004 I purchased a used model 320A from a radio station. The “A” reference indicates it’s analog circuitry. I’ve used the 320A for countless audio file and tape transfers, post production processing, Telephone/Skype recording sessions, and monitoring. The device provides three selectable Operating Levels … +8dBu, +4dBu, and -10dBV.

Recently the complex level and gain reduction metering for the right channel failed. I replaced the faulty 320A with a 320D. This version features digital and analog I/O with common selectable (analog) Operating Levels (+4dBu, and -10dBV).

At some point my faulty 320A will be shipped out to Burbank California for authorized service.

320D – Automatic Processing and Detection

As noted Aphex classifies the Compellor as a Frequency Discriminant Leveler. It responds slower and less aggressively to low frequencies. In essence low frequency energy will not initiate gain reduction.

A Dynamic Release Computer (DRC) instantiates program dependent compression release times.

The Dynamic Verification Gate (DVG) computes the historical average of peak values and verifies whether measured values exceed or are equal to the historical value. When the signal level is below the average, leveling and compression gain reduction is frozen.

Controls

The device Drive control sets the preprocessed VCA gain. Higher settings yield a higher level of gain reduction (VCA refers to Voltage Controlled Amplifier).

The Process Balance control allows the operator to fine tune the Leveling and/or Compression balance and weighting. Leveling is a slow method of gain reduction. It maintains transient retention and wider dynamics. The Compression stage works faster and acts more aggressively on inherent dynamics. The key is by combining both modes, the processed output will be very consistent

A Rate (speed) toggle option is provided: Fast, suitable for speech/voice, or Slow, suitable for program material such as produced TV and/or Radio programs.

The device Output control normalizes the processed audio to 0VU.

Silence Gate: Aphex stresses – this is not an audio gate! It is a user defined threshold parameter. When the signal drops below the threshold for 1 sec. or longer, the Silence Gate freezes the VCA gain. This prevents the buildup of noise during pauses and/or extended passages of silence.

The device Limiter features a very fast attack and high threshold. It is designed to prevent occasional high transient activity and overshoots.

A Stereo Enhance mode is available on the 320A and 320D models. When activated it widens the stereo image. It’s effect is dependent upon the amount of applied compression.

Metering

The 320D Compellor features three, bi-color (red, green) LED metering modes: Input, Output, and Gain Reduction. For Input/Output metering – the red LED’s indicate VU/average. Green LED’s indicate peak level.

When the meter is set to display gain reduction (“GR”), the green LED’s indicate total gain reduction. Depending on the Process Balance control weighting – a floating red LED may appear within green LED instances. The floating red LED indicates Leveling gain reduction. If Leveling gain reduction is in fact occurring, the total gain reduction will be indicated by the subsequent green LED(s).

Below are 4 examples:

Example 1 displays Input or Output metering with an average (red) level of 0VU and a peak (green) level of +6dB. This translates to a +4dBu average level and a +10dB peak level (analog OL set to +4dBu).

Example 2 displays 4dB of Leveling Gain Reduction and 8dB of Total Gain Reduction.

Example 3 displays 12dB of Leveling Gain Reduction.

Example 4 displays 10dB of Compression Gain Reduction.

**Notice the position of the Process Balance control for examples 2, 3, and 4.

320D I/O

The 320D is essentially an analog processor utilizing standard XLR I/O jacks. The device also includes AES/EBU XLR jacks along with an internal DAC for digital I/O. The Input mode and/or Sample Rate is user selectable.

When implementing digital I/O – the Incoming audio is converted to analog as it passes through the device. The audio is then converted back to digital and output accordingly.

The digital input is calibrated internally and matches -20dBFS to 0VU on the Compellor’s meter. The +4dBu/-10dBV Operating Level options only affect the analog I/O.

Notes:

The Aphex Compellor is a long standing, highly regarded, and ubiquitous audio processor. It has been an integral multipurpose tool for me for 12+ years. My newly purchased (used) 320D is in near mint condition. In fact it looks and feels as if it was hardly used by the previous owner.

My system includes additional Aphex audio processors (651 Compressor, 109 EQ, 622 Expander/Gate, and a 720 Dominator II Multiband Peak Limiter). As well, a Mackie Onyx 1220i Mixer, Motu I/O, dbx 160A Compressor, dbx 286A Mic Processor, Marantz CF Recorder, and a Telos One Digital Hybrid. All components, with the exception of the 286A – are interfaced through a balanced Patchbay.

A typical processing/monitoring chain will pass system audio through the Compellor, followed by the 720 Peak Limiter. The processed audio is ultimately routed to the system’s Main Output(s). This chain optimizes playback of poorly produced Podcasts, VO’s, live streams, or videos. The routing is implemented via Patchbay.

A typical audio processing chain will route Pro Tools audio out via hardware insert (or bus, alternative output, etc.) through the Compellor (or a more complex chain) and returned in Pro Tools. In this scenario I use a set of assignable interface line inputs/outputs. The routing is implemented via Patchbay. I document the setup and use of hardware inserts here.

-paul.

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LevelView by Grimm Audio

LevelView by Grimm Audio is a highly functional and well designed real time Loudness Meter.

Here are the details:

LevelView features a unique multifaceted Rainbow Meter. Clicking the Rainbow display toggles the Meter scale (EBU +9 or EBU +18).

There are three compliance modes: EBU R128, ATSC A/85, and a custom User specification (Gated or Ungated). The Rainbow Meter displays a Relative Scale. Consequentially the defined target will be equivalent to 0 LU.

The upper blue Rainbow arc represents Short Term Loudness measured within a 3 sec. time frame. The inward blue arcs indicate slower time frame variances (10, 30, 90, and 270 seconds).

The arced needle meter located above the Rainbow Meter represents the Momentary Loudness measured within a 400ms time frame.

Visual dots displayed (and held) on both the Momentary and Short Term Loudness indictor plots represent the maximum values for each descriptor. Both indicators will shift to orange when their values exceed recognized guidelines (+8 max M, and +6 Max S).

The numerical descriptor table features a large Integrated Loudness value. This may display an Absolute Scale value in LUFS, or a Relative Scale value in LU’s. Clicking the descriptor text toggles it’s view.

Additional numerical descriptors include maximum Momentary Loudness (max M), maximum Short Term Loudness (max S), LRA (Loudness Range), PLR (Peak to Loudness Ratio), and maximum True Peak (max TP). Clicking the max TP descriptor text will toggle the measurement algorithm and display max TP or max SP (Sample Peak). Descriptors will shift to orange when a displayed value exceeds recognized or specification guidelines.

The graph located at the lower left is the Loudness Range histogram. It displays the distribution of the measured Loudness over time. The data will indicate whether further dynamic range compression may be necessary.

LevelView supports Manual start and stop measurements. Setting the meter to Auto will force it to follow the host DAW’s transport. In essence the meter will automatically start/stop and reset based on the status of the transport.

Link mode records and stores data continuously. This allows the operator to revert back in time and re-measure a passage without resetting the stored measurements. In the event a passage is skipped, a gap warning will appear in orange. Re-measurement of a skipped segment will clear the gap warning. The Stop button resets the memory. Note the LevelView documentation indicates that the host “must provide time code for the Link function to work.”

It is possible to run various connected (Host and Client) instances of LevelView on a network or over the Internet. I will be testing these options in the near future.

LevelView is available as an AU, VST, or AAX Plugin. The AU and VST versions support (5.1) Surround Sound measurement. The meter conforms to the SMPTE/ITU channel matrix standard (L-R-C-LFE-Ls-Rs).

The meter may also run in a stand-alone mode with no DAW dependency. I/O configuration options are provided.

My Assessment:

I like this meter and I appreciate it’s unique design and accuracy. The networking options, support for Surround Sound, and stand-alone capability make it highly flexible and well worth it’s reasonable cost ($70 U.S. at Don’tCrack). I’m happy to recommend it.

Improvements I’d like to see:

– Scaleable UI
– Option to define a custom Maximum True Peak in the User mode (currently it defaults to -1.0 dBTP)

-paul.

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Loudness Compliance Summarization

– I continue to endorse -16.0 LUFS for (stereo) Podcast distribution. If meeting this target requires an excessive amount of limiting, a slightly lower target is a viable option. However from my perspective a -20.0 LUFS spoken word piece consumed in a less than ideal environment on a mobile device would be problematic. I’m comfortable supporting upwards of a -2.0 LU deviation from the recommended -16.0 LUFS target (when applicable).

**Note mono files require a -3 LU offset to establish perceptual equivalence to stereo file targets.

– Loudness Range (LRA) represents the statistical distribution of Loudness. An LRA no higher than 8 LU will optimize intelligibility by restricting dynamics and/or wide variations in Loudness..

– Networks and Catalog based program sets managed by indie producers must institute Program Loudness consisctency across all distributed media. This will free listeners from making constant playback volume adjsutments when listening to several programs in succession. Up to 1.0 LU tolerance (+/-) is reasonable. However upside Program Loudness should never exceed -16.0 LUFS.

– Without sufficient headroom – lossy, low bitrate encoding may generate peak levels that exceed a compliance ceiling and/or introduce distortion. -1.5 dBTP is the favored maximum ceiling prior to lossy coding. Of course a lesser value (e.g -2.0 dBTP) is appropriate. However, a peak ceiling below -3.0 dBTP may indicate excessive limiting. This should be avoided.

-paul.

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Intelligibility Optimization

The attached image displays a processing workflow designed to optimize Spoken Word intelligibility. The workflow also demonstrates a realtime example of Integrated Loudness/Maximum True Peak compliance targeting.

There are 7 reference point Sections worth noting:

Section A includes the Adobe Audition Effects Rack Signal Level Meters indicating the source (Input) level and the (Output) level. The Output level reflects the results of the workflow’s inserted plugins. The chain includes a Compressor, a Limiter, and a Loudness Meter. Note the level meters indicate signal level. They do not indicate or represent perceptual Loudness.

Section B displays the gain reduction applied by the Compressor at the current position of the playhead. For the test/source audio I determined an average of 6dB of gain reduction would yield acceptable results. The purpose of this stage is to reduce the dynamic range and/or dynamic structure of the Spoken Word resulting in optimized intelligibility AND to prevent excessive down stream limiting. This is an important workflow element when preparing Spoken Word audio for Internet/Mobile, and Podcast distribution.

Section C includes my subjective limiting parameters. The Limiter will add the required amount of gain to achieve a -16.0 LUFS deliverable while adhering to a -1.5 dBTP (True Peak Max). If the client, platform, or workflow requires an alternative Loudness target and/or Maximum True Peak ceiling – the parameters and their mathematical relationship may be altered for customized targeting. Please note the Maximum True Peak referenced in any spec. is more of a ceiling as opposed to a target. In essence the measured signal level may be lower than the specified maximum.

Section D indicates the amount of limiting that is occurring at the current position of the playhead.

Section E displays the user defined Integrated Loudness target located above the circular Momentary Loudness LED (12 o’clock position). The defined Integrated Loudness target is also visually represented by the Radar’s second concentric circle. The Radar display indicates the Short Term Loudness measured over time within a 3 sec. window. The consistency of the Short Term Loudness is evident indicating optimized intelligibility.

Section F displays the unprocessed source audio that lacks optimization for Internet/Mobile, and Podcast distribution. Any attempt to consume the audio in it’s current state in a less than ideal listening environment will result in compromised intelligibility. Mobile device consumption in like environments will exacerbate compromised intelligibility.

Section G displays the processed/optimized audio suitable for the noted distribution platform. The Integrated Loudness, True Peak, and LRA descriptors now satisfy compliance targets. Notice there is no indication of excessive limiting.

-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 assign 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 assignments. However, it is still possible to implement a Print workflow (see attached image). You will need a supported Audio Interface with a Mix Return. Simply assign 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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Loudness Meter Scale Variations

I thought I’d revisit various aspects of Loudness Meter Absolute/Relative Scale correlation, and provide a visual representation of a real time processing Session with both Scales active.

Descriptors and Scales

Modern Loudness Meters display various descriptors including Program Loudness – also referred to as Integrated Loudness. There are two scales that can be used to display measured Program or Integrated Loudness over time …

The most common is an Absolute Scale, displayed in LUFS or LKFS. LUFS refers to Loudness Units relative to Full Scale. LKFS refers to Loudness Units K-Weighted relative to Full Scale. There is no difference in the perceptual measured loudness between both descriptor references.

It is also possible to measure and display Integrated/Program Loudness as Loudness Units (or LU’s) on a Relative Scale where 1LU == 1 dB.

When shifting to a Relative Scale, the 0 LU increment is always equivalent to the Meter’s user defined or spec. defined Absolute Loudness target.

For example, in an R128 -23.0 LUFS Absolute Scale workflow, setting the Meter to display a Relative Scale changes the target to 0 LU.

So – if a piece of measured audio checks in at -23.0 LUFS on an Absolute Scale, it would be perceptually equal to measured audio checking in at 0 LU on a Relative Scale.

Likewise if the Meter’s Absolute Scale target is set to -16.0 LUFS, it will correlate to 0 LU on a Relative Scale. Again both would reflect perceptual equivalence.

All broadcast delivery specifications suggest Absolute Scale Integrated Loudness targets. However, for any number of subjective reasons – many operators prefer to use the alternative Relative Scale and “mix or master to 0 LU.”

Please note Loudness Units are also the proper way in which to describe Loudness differentials between two programs. For instance, “Program (A) is +2 LU louder than Program (B).” One might also describe gain offsets in LU’s as opposed to dB’s.

LU Meter

Hornet Plugins recently released Hornet LU Meter. This tool is a Loudness Meter plugin designed to measure and display Integrated/Program Loudness within a 400ms time window. This measurement represents the Momentary Loudness descriptor.

The Meter is indeed nifty and affordable. However there is one sort of caveat worth noting: As the name suggests, it is an LU Meter. In essence Integrated (Momentary) Loudness measurements are solely displayed on a Relative Scale.

Session

The displayed Session (image) consists of a single mono VO clip. The objective is to print a processed stereo version in RT checking in at -16.0 LUFS with a maximum True Peak no higher than -2.0 dBTP.

The output of the mono VO track is routed to a mono Auxiliary Input track titled Normalize. If you are not familiar with Pro Tools, an Auxiliary Input track is not the same as an Auxiliary Send. Auxiliary Input tracks allow the user to pass signal using buses, insert plugins, and adjust level. They are commonly used to create sub-mixes.

I’ve inserted a Compressor and a Limiter on the Normalize Auxiliary Input track. The processed audio is passing through at -19.0 LUFS (mono).

The audio is then routed to a second (now stereo) Auxiliary Input track titled Offset. I use the track fader to apply a +3 dB gain offset, This will reconstitute the loss of gain that occurs on center panned mono tracks. The attenuation is a direct result of the Pro Tools Pan Depth setting.

The signal flow/output is now passing -16.0 LUFS audio. It is routed to a standard audio track titled Print. When this track is armed to record, it is possible to initiate a realtime bounce of the processed/routed audio.

The Meters

Notice the instances of the Hornet LU Meter and TC Electronics Loudness Radar. Both Meters are inserted on the Master Bus and are measuring the session’s Master Output.

I set the Reference (target) on the Hornet LU Meter to -16.0 LUFS. In essence 0 LU on it’s Relative Scale represents -16.0 LUFS.

Conversely the TC Electronic Meter is configured to display Absolute Scale measurements. The circular LED that borders the Radar area indicates Momentary Loudness. The defined Integrated Loudness target is displayed under the arrow at the 12 o’clock position.

Remember the Hornet LU Meter solely displays Momentary Loudness. If you compare it’s current reading to the indication of Momentary Loudness on the TC Electronic Meter, the relationship between Relative Scale and Absolute Scale measurement is clearly indicated. Basically the Hornet Meter registers just below 0 LU. The TC Electronic Meter registers just below -16.0 LUFS.

I will say if you are comfortable monitoring real time Momentary Loudness and understand Relative/Absolute Scale correlation, the Hornet tool is quite useful. In fact it contains additional features such as Grouping, auto/manual Gain Compensation, and auto-Maximum Peak protection.

Additional insight on the K-weighting Curve or K-weighted filtering:

K-weighting suggests de-emphasized low frequencies by way of a high-pass filter. A high-shelving filter is applied to the upper frequency range, and the measured data is averaged.

TC Electronic describes applied K-weighting on audio channels as a “method to build a bridge between subjective impression and objective measurement.”

-paul.

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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.

Preparation

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.

Note:

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.

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Programmatic Ads and Loudness Standardization

This is a re-post of an article that I published in October, 2015 …

In a recent Midroll article titled “Why Programmatic Ads Aren’t Necessarily Great for Podcasting,” the staff writer states:

“A number of players in the Podcasting and advertising industries are making bets on programmatic Ad delivery — dynamically inserting Ads into a Podcast as the episode is downloaded. It’s an understandable temptation, but we at Midroll see some tradeoffs.”

I wonder how networks will handle potential perceived Loudness inconsistencies between produced Ads and new or preexisting programs?

minus-sixteen-small

I’ve mentioned my past affiliation with IT Conversations and The Conversations Network, where I was the lead post audio engineer from 2005-2012. Executive Director Doug Kaye built a proprietary content management system and infrastructure that included an automated component based Show Assembly System. Audio components were essentially audio clips (Intros, Outros, Ads, Credits. etc.) combined server side into Podcasts in preparation for distribution.

One key element in this implementation was the establishment of perceived Loudness consistency across all submitted audio components. This was accomplished by standardizing an average Loudness Target using a proprietary software RMS Normalizer to process all server side audio components prior to assembly. (Loudness Normalization is now the recommended process for Integrated Loudness targeting and consistency).

Due to this consistency, all distributed Podcasts were perceptually equal with regard to Integrated or Program Loudness upon playback. This was for the benefit of the listener, removing the potential need to make constant playback volume adjustments within a single program and throughout all programs distributed on the network.

Regarding Programmatic Ad insertion, I have yet to come across a Podcast Network that clearly states a set Integrated Loudness Target for submitted programs. (A Maximum True Peak requirement is equally important. However this descriptor has no effect on perceptual Loudness consistency).

Due to the absence of any suggested internal network guidelines or any form of standardized Loudness Normalization, dynamic Ad insertion has the potential to ruin the perceptual consistency within single programs and throughout the contents of an entire network.

Many conscientious independent producers have embraced the credible -16.0 LUFS Integrated Loudness Target for stereo Internet/ Mobile/Podcast audio distribution (the perceptual equivalent for mono distribution is -19.0 LUFS). It’s far from a requirement, and nothing more than a suggested guideline.

My hope is Podcast Networks will begin to recognize the advantages of standardization and consider the adoption of the -16.0 LUFS Integrated Loudness Target. Dynamically inserted Ads must be perceptually equal to the parent program. Without a standardized and pre-disclosed Integrated Loudness Target, it will be near impossible to establish any level of distribution consistency.

-paul.

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Adobe Audition CC Productivity

Below I’ve listed a few Adobe Audition CC (ver.2015.2.1) features/options that may be obscure and perhaps underutilized.

aud_small

Usability

1- Maximize Active Frame (⌘↓). This command toggles full screen display accessibility of the active (blue outlined) UI Panel.

2- Lock In Time (Multitrack). When activated, selected clips are pinned to their current location. I mapped ⌥⌘L for this function.

3- Group (⌘G) (Multitrack). Multiple clips will be congregated and may be repositioned cumulatively.

4- Suspend Groups (⏎⌘G) (Multitrack). This function temporarily deactivates the Group. Actually, this command toggles the behavior between deactivate and activate. There are also options to Remove Focus Clip from Group and Ungroup Selected Clips. They both support custom shortcut mapping,

5- Right + Click on any Clip’s Fade Handle (Multitrack) to display the following customization menu:

– No Fade
– Fade In/Out
– Crossfade
– Symmetrical
– Asymmetrical
– Linear
– Cosine
– Automatic Crossfade Enabled

6- Bounce to New Track (Multitrack). This feature will process and combine multiple clips located on a single track or multiple tracks. This will free up system resources. The following options support custom shortcut mapping:

– Selected Track
– Time Selection
– Selected Clips In Time Selection
– Selected Clips Only

7- Convert To Unique Copy (Multitrack). This function creates a sub clip derived from the original trimmed source clip. Media Handles are no longer accessible in the converted copy (Multitrack and/or Waveform Editor environments). I mapped ⌥⌘C for this function.

Editing

1- Time Selection in all Tracks (Multitrack). This is a Ripple Delete variation (⏎⌘⌦) that will retain clip relevant Marker position(s).

2- Split All Clips Under Playhead (Multitrack). I mapped ⌥⌘R for this function.

3- Merge Clips (remove thru edits) (Multitrack). I mapped ⌥⌘J for this function.

Mixer/Track Inserts and Sends

1- Individual Track supplied buttons will designate Sends and Inserts as Pre or Post Fader.

Markers

1- Markers implemented in the Waveform Editor may be Merged thus allowing easy selection of encapsulated audio.

2- Selected Range Markers present in the Waveform Editor may be exported as individual clips.

3- Selected Range Markers present in the Waveform Editor may be added to a Playlist where they may be reordered for auditioning.

Exporting

1- The (Multitrack) Session Export Dialog includes user defined Mixdown options:

– Master: Stereo, Mono, or 5.1
– Signal present on individual Tracks
– Signal present on individual Busses

2- Export with Adobe Media Encoder (Multitrack). This Export option runs Media Encoder and requires the user to select a predefined Media Encoder preset. Routing options are available as well.

-paul.

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CNN and Program Loudness Tolerance

I recently analyzed a few of the internal Podcasts produced by CNN. One particular installment is yet another example of a major media outlet distributing audio that is in my view unsuitable for this particular platform.

Let’s discuss file attributes and measured specs. for one of CNN’s distributed Podcasts:

The distributed audio is mono, 64kbps, with music elements. I’ve stated how I feel about this. I’m not a proponent of 64 kbps MP3 audio PERIOD (mono or stereo). In general audio in this format sounds horrible. Feel free to disagree.

Secondly, the Integrated (Program) Loudness for this particular program is just about -23.0 LUFS with a Maximum True Peak of +0.40 dBTP. From my perspective the perceptual Loudness misses the mark. And, the audio is clipped.

Lastly, the produced audio is way too dynamic for spoken word. The perceptual inconsistency of the delivery by the participants is inadequate when considering how (for the most part) this program will be consumed (mobile devices, problematic ambient spaces, etc.).

I decided to sort of showcase this particular program because it is a good candidate for flexible Target considerations. What do I mean by “flexible Target considerations?” Let me explain …

Again, the distributed file is mono. The recommended Integrated Loudness Target for mono Podcasts is -19.0 LUFS. This is the perceptual equivalent of -16.0 LUFS stereo. If I were to apply a +4 db gain offset to Loudness Normalize this audio to -19.0 LUFS, there would be very little change in the original dynamic structure of the audio. However without some form of aggressive limiting, the maximum amplitude or Peak Ceiling would be driven into oblivion. In fact audible distortion may occur with or without limiting. This is obviously not recommended.

There are two options to consider: 1) apply Dynamic Range Compression before Loudness Normalization, or 2) shoot for a lower Integrated Loudness target. For this particular example I chose to implement both options.

First, in my view optimizing the dynamics in this program for Podcast distribution is unavoidable. It’s just way too choppy and it lacks delivery consistency for spoken word. Also, by lowering the L.Normalized Target, the necessary added gain offset will be reduced resulting in less aggressive limiting. In addition, the reduced amount of added gain will curtail noise floor elevation and other variables such as exaggerated breaths.

As noted the distributed Podcast (displayed in the attached upper waveform example) checks in at -23.0 LUFS and it is clipped. My optimized version (displayed in the lower waveform example) checks in at -20.2 LUFS with a Maximum True Peak of -1.23 dBTP. It is well within a reasonable level of Program Loudness tolerance for Podcast L.Normalization. In fact the perceptual difference between the processed -20.0 LUFS audio and a -19.0 LUFS version would be pretty much undetectable. In essence the audio has been optimized and it exhibits improved intelligibility. It is now well suited for Podcast distribution.

cnn_small

(If you are interested in the tools that I use, they are listed under Available Services).

It is no secret that I am a staunch proponent of the -16.0 LUFS/-19.0 LUFS recommendations for Podcasts. However, in certain situations – tolerance for slightly reduced Program Loudness Targets is acceptable.

For the record – my remaster is much easier to listen to. CNN can do better.

-paul.

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Loudness Measurements and Silence

Consider this: Two extended segments of audio, Loudness Normalized (or mixed in real time) to the same Integrated Loudness Target.

Segment (A) is fairly consistent, with a very limited amount of intermittent silence gaps.

Segment (B) is far less consistent, due to a multitude of intermittent silence gaps.

When passing both segments through a Loudness Meter (or measuring the segments offline), and recognizing Integrated Loudness is a reflection of the average perceptual Loudness of an entire segment – how will inherent silence affect the accuracy of the cumulative measurements?

In theory the silence gaps in Segment (B) should affect the overall measurement by returning a lower representation of average Integrated Loudness. If additional gain is added to compensate, Segment (B) would be perceptually louder than Segment (A).

Basically without some sort of active measurement threshold, the algorithms would factor in silence gaps and return an inaccurate representation of Integrated Loudness.

The Fix

In order to establish perceptual accuracy silence gaps must be removed from active measurements. Loudness Meters and their algorithms are designed to ignore silence gaps. The omission of silence is based on the relationship between the average signal level and a predefined threshold.

Loudness Meter (G10) Gate

The specification Gate (G10) is an aspect of the ITU Loudness Measurement algorithms included in compliant Loudness Meters. It’s function is to temporarily pause Loudness measurements when the signal drops below a relative threshold, thus allowing only prominent foreground sound to be measured.

The relative threshold is -10 LU below ungated LUFS. Momentary and Short Term measurements are not gated. There is also a -70 LUFS Absolute Gate that will force metering to ignore extreme low level noise.

Most Loudness Meters reveal a visual indication of active gating (see attached image) and confirm the accuracy of displayed measurements.

Gate-(480)

Additional Gate Generalizations and Nomenclature

Common Noise Gate

A Downward Expander and it’s applied attenuation is dependent on signal level when the signal drops below a user defined threshold. The Ratio dictates the amount of attenuation. Alternatively a Noise Gate functions independent of signal level. When the level drops below the defined threshold, hard muting is applied.

Silence Gate

This is a somewhat proprietary term. It is a parameter setting available on the Aphex 320A and 320D Compellor hardware Leveler/Compressor.

Compellor

When a passing signal level drops below the user defined Silence Gate threshold for 1 second or longer, the device’s VCA (Voltage Controlled Amplifier) gain is frozen. The Silence Gate will prevent the Leveling and Compression processing from releasing and inadvertently increasing the audibility of background noise.

-paul.

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Understanding Pan Mode Options

Adobe Audition and Logic Pro X include Pan Mode preference options that determine track output gain for center panned mono clips included in stereo sessions. These options are often the source of confusion when working with a combination of mono and stereo clips, especially when clips are pre-Loudness Normalized prior to importing.

In Audition, the Left/Right Cut (Logarithmic) option retains center panned mono clip gain. The -3.0 dB Center option, which by the way is customizable – will attenuate center panned mono clip gain by the specified dB value.

For example if you were targeting -16.0 LUFS in a stereo session using a combination of pre-Loudness Normalized clips, and all channel faders were set to unity – the imported mono clips need to be -19.0 LUFS (Integrated). The stereo clips need to be -16.0 LUFS (Integrated). The Left/Right Cut Pan Mode option will not alter the gain of the center panned mono clips. This would result in a -16.0 LUFS stereo mixdown.

Conversely the -3.0 dB Center Pan Mode option will apply a -3 dB gain offset (it will subtract 3 dB of gain) to center panned mono clips resulting in a -19.0 LUFS stereo mixdown. In most cases this -3 LU discrepancy is not the desired target for a stereo mixdown. Note 1 LU == 1 dB.

As stated Logic Pro X provides a similar level of Pan Mode flexibility. I’ve also tested Reaper, and it’s options are equally flexible.

Pro Tools

Pro Tools Pan Mode support (they call it Pan Depth) is somewhat restricted. The preference is limited to Center Pan Mode, with selectable dB compensation options (-2.5 dB, -3.0 dB, -4.5 dB, and -6.0 dB).

There are several ways to reconstitute the loss of gain that occurs in Pro Tools when working with center panned mono clips in stereo sessions. One option would be to duplicate a mono clip and place each instance of it on hard-panned discrete mono tracks (L+R respectively). Routing the mono tracks to a stereo output will reconstitute the loss of gain.

A second and much more efficient method is to route all individual instances of mono session clips to a stereo Auxiliary Input, and use it to apply the necessary compensating gain offset before the signal reaches the stereo Master Output. The gain offset can be applied using the Aux Input channel fader or by using an inserted gain trim plugin. Stereo clips included in the session can bypass this Aux and should be directly routed to the stereo Master Output. In essence stereo clips do not require compensation.

Example Session

Have a look at the attached Pro Tools session snapshot. In order to clearly display the signal path relative to it’s gain, I purposely implemented Pre-Fader Metering.

pt-pan_small

Notice how the mono spoken word clip included on track 1 is routed (by way of stereo Bus 1-2) to a stereo Auxiliary Input track (named to Stereo). Also notice how the stereo signal level displayed by the meters on the Stereo Auxiliary Input track is lower than the mono source that is feeding it. The level variation is clear due to Pre-Fader Metering. It is the direct result of the session’s Pan Depth setting that is subtracting -3dB of gain on this center panned mono track.

Next, notice how the signal level on the Master Output has been reconstituted and is in fact equal to the original mono source. We’ve effectively added +3dB of gain to compensate for the attenuation of the original center panned mono clip. The +3dB gain compensation was applied to the signal on the Auxiliary Input track (via fader) before routing it’s output to the stereo Master Output.

So it’s: Center Panned mono resulting in a -3dB gain attenuation —>> to a stereo Aux Input with +3dB of gain compensation —>> to stereo Master Output at unity.

In case you are wondering – why not add +3dB of gain to the mono clip and bypass all the fluff? By doing so you would be altering the native inherent gain structure of the mono source clip, possibly resulting in clipping. My described workflow simply reconstitutes the attenuated gain after it occurs on center panned mono clips. It is all necessary due to Pro Tool’s Pan Depth methods and implementation.

-paul.

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Utilizing Multiple Outputs for Recording

The vast majority of audio industry professionals use DAWS running on proficient computer systems to record audio directly to secondary hard disks. For some reason direct to disk recording is not widely endorsed in the Podcasting space. Many consultants (for various reasons) advise against this recording method. Instead, they recommend the use of inexpensive hand-held solid state Recorders.

For instance I’ve heard a few people state “computers cause ground loops”, hence the widespread Portable Recorder recommendation. In my opinion that is a half-baked assertion. In fact, ANY electronic component in a signal chain (including your electrical system) is capable of producing inherent noise. Often the replacement of cheaply manufactured components (interfaces, mixers, processors, cables, etc.) will solve audible noise problems. The key is to isolate the source and correct or replace it.

Portable Recorders are well suited for location interviews and video shoots. For in-studio sessions I feel direct to disk recording on a proficient system is much more flexible compared to the use of an external device. More so, the sole use of a Portable Recorder without a proper backup strategy is flat out risky.

That being said I thought I would document a basic Skype Recording session that I implemented in Pro Tools using a multi-output Motu Audio Interface. The incoming audio will be recorded on a secondary hard disk installed (or interfaced) on the host system. The real time session audio will also be routed to an alternate Interface Output, feeding an external Recorder for backup purposes.

Recording_Session_small

Note a multi-output Mixer can be used in place of an Audio Interface. As far as software you can use any modern DAW to replicate the described session. If you are using a Mac, Rogue Amoeba’s distinctive Audio Hijack application is also highly capable.

Objectives:

1-Record Studio Host and Skype Participant on discrete mono tracks in real time.

2-Combine the discrete recordings and create a split-stereo clip with independent dynamics processing applied to each channel, all in real time.

3-Use a Pre-Fader Send to independently control the level of the split-stereo discrete recording, and patch the real time signal to the Interface S/PDIF Output. This will feed the external Recorder’s S/PDIF Input.

4-Monitor the session through Headphones and play out through Desktop near-field Monitors.

Please review the displayed Pro Tools session snapshot.

• The Input for the mono Host track is the Interface connected mic. The Input for the mono Skype track is “Mix 1 Return.” This is an Interface supported feature, allowing the operator to route the computer’s Output (in this case Skype) to an available DAW Input. This configuration effectively creates a mix-minus with discrete, unprocessed recordings on individual mono tracks.

• The mono recording tracks are routed to individual mono Aux Input tracks using Buses. The Aux Input tracks are hard-panned L+R and contain various inserted processing options, including a Gain Trim, Expander, and Compressor.

The processing applied in this session is not intended to replace what would normally occur in post. The Compressors are there just to tame dynamics in the event either participant exceeds nominal input levels. The Expander is set up to apply mild attenuation when the host is not speaking.

• The Aux Input tracks have their Outputs set to a common stereo Bus.

• Finally a third standard stereo audio track (Rec-Sum) uses the stereo Bus Output(s) as it’s Inputs. By hard panning the channels L+R we are able to maintain discrete channel separation within any printed stereo clip.

To record the discrete raw audio and the processed split-stereo audio in real time, we simply arm all session Audio tracks to record and fire away. The session can be monitored through Headphones and played out through near fields via the Main Output.

Secondary Output

The Motu Interface used for this session has a total of 8 Outputs, including a stereo S/PDIF option. I implemented Pre-Fader Send on the session’s Rec-Sum channel with it’s Output set to S/PDIF. This will route the track’s split-stereo audio to the S/PDIF stereo Input of an external Marantz CF Recorder. With the Send designated as Pre-Fader, it’s level control will be independent of the parent (Rec-Sum) channel fader, thus allowing discrete control of the real time signal being fed to the Recorder.

Note in the displayed Pro Tools session snapshot – the floating fader positioned to the left of the mixer is a user friendly and easily accessible copy of the much smaller Send fader displayed in the parent (Rec-Sum) track.

In summary, we can successfully initialize and capture 4 recordings in a single pass: the raw Host audio, the raw Skype participant audio, a split-stereo processed version of the Skype session, and a split-stereo copy of the processed Skype session stored on the Recorder.

The image below displays the completed session with the split-stereo clip playing through the Main Outputs.

Mix_small

My general recommendation:when it is feasible, use direct to disk and Portable recording options in unison on a proficient system to capture in-studio multitrack and single participant Podcast sessions.

-paul.

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Bit Depth and Dithering

In a professional environment Dither will be applied to audio clips (or mixes) when reducing word length. This process will mask errors that occur due to the removal of digital audio bits. I thought I’d cover the basics.

Dither_small

Digital Audio

Digital Audio incorporates individual samples consisting of bits created by the process of Quantization. This is essentially the conversion of a continuous, linear range of values present in analog audio into a fixed range of discrete values. Bit Depth (a.k.a. Word Length or Resolution) represents the number of bits stored in a sample’s measure of amplitude. It indicates the extent of inherent vertical precision. Higher bit depths (or bits per sample) encompass improved vertical dynamic resolution resulting in an extended Dynamic Range.

1 bit = 6dB of Dynamic Range. Theoretically 16bit audio has a quantified Dynamic Range of 96 dB. 24 bit audio has a quantified Dynamic Range of 144 dB. However, in order to accurately assess Dynamic Range we must also recognize the amplitude of the highest spectral component of the inherent noise floor. Specifically, where it resides relative to the maximum Peak value that a system is capable of reproducing. Dynamic Range is the measurement of this ratio or range.

Signal to Noise Ratio (SNR) is the quantified range between the nominal average signal level and the average level of the noise floor. Audio with an extended Dynamic Range will exhibit a higher SNR compared to audio with a reduced Dynamic Range. In essence 24 bit audio will allow you to work with additional headroom without any increase in noise compared to 16 bit audio.

Word Length Reduction

Truncation is the removal of bits with no compensating replacement. The repositioning of samples after converting to a lower resolution creates Quantization Errors resulting in audible artifacts and distortion. Dithering is technology that adds minimal perceived noise to audio before word length reduction. This noise will minimize and/or mask the audibility of distortion caused by Quantization Errors. It will also help preserve the sound quality and Dynamic Range of a higher resolution clip when converting or exporting to a lower bit depth.

There is a trade off: you are replacing bad noise with alternative “good” noise that is smoother, less audible, and much more consistent.

Noise Shaping is a supplemental feature that pushes noise into frequency ranges that are less audible to humans, thus allowing greater Dither with reduced perceptual noise.

Take a look at the Noise Shaped frequency response curve in the attached image. There is a clear visual indication of increased gain at higher frequencies.

Podcasting

So what does this all mean for the typical Podcast Producer? Is Dithering just another obscure aspect of professional Audio Mastering and/or Post Production that can be safely ignored?

Consider the following variables:

If you are recording spoken word in a well suited environment that is reasonably quiet, and you are using capable (and trouble free) gear that is properly configured, there is really no reason to record 24 bit audio. In my honest opinion with proper handling 16 bit audio from acquisition to distribution will be perfectly acceptable.

Remember, I’m specifically referring to spoken word audio slated for Podcast distribution. If you are tracking music, well then by all means make full use of the advantages of higher resolution audio recording.

If you elect to record 24 bit audio, and you are not properly implementing the word length reduction to 16 bit, you are essentially nulling the advantages of the original higher resolution audio. When down-converting, you will be unknowingly degrading the sound quality by introducing artifacts and distortion. That’s not my opinion – it is a fact.

Consider this: The stand-alone version of iZotope’s Ozone 7 Mastering Suite processes all imported audio to 32 bit word length. The manual specifically states:

“If you select a bit depth other than 32-bit, you may want to apply dither to your export. Ozone processes files at 32-bit so dither is desirable for files being exported to values lower than 32-bit.”

Most DAWS include Dithering options. In some cases it’s by way of a plugin. You may also notice Dithering options included in application Preferences or Export dialogs. Hopefully after reading this article you will understand what it all means and whether you should consider implementing it. Please note that Dither must be applied at the very last stage of any processing chain.

-paul.

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