Loudness Meter Descriptors …

In the recent article published on Current.org “Working Group Nears Standard for Audio Levels in PRSS Content”, the author states:

“Working group members believe that one solution may lie in promoting the use of Loudness Meters, which offer more precision by measuring audio levels numerically. Most shows are now mixed using peak meters, which are less exact.”

Peak Meters are exact – when they are used to display what they are designed to measure:Sample Peak Amplitude. They do not display an accurate representation of average, perceived loudness over time. They should only be used to monitor and ultimately prevent overload (clipping).

It’s great that the people in Public Radio are finally addressing distribution Loudness consistency and compliance. My hope is their initiative will carry over into their podcast distribution models. In my view before any success is achieved, a full understanding of all spec. descriptors and targets would be essential. I’m referring to Program (Integrated) Loudness, Short Term Loudness, Momentary Loudness, Loudness Range, and True Peak.

Loudness Meter

A Loudness Meter will display all delivery specification descriptors numerically and graphically. Meter descriptors will update in real time as audio passes through the meter.

Short Term Loudness values are often displayed from a graphical perspective as designed by the developer. For example TC Electronic’s set of meters (with the exception of the LM1n) display Short Term Loudness on a circular graph referred to as Radar. Nugen Audio’s VisLM meter displays Short Term Loudness on a grid based histogram. Both versions can be customized to suit your needs and work equally well.

meters-480

Loudness Meters also include True Peak Meters that display any occurrences of Intersample Peaks.

Descriptors

All Loudness standardization guidelines specify a Program Loudness or “Integrated Loudness” target. This time scaled descriptor indicates the average, perceived loudness of an entire segment or program from start to finish. It is displayed on an Absolute scale in LUFS (Loudness Units relative to Full Scale), or LKFS (Loudness Units K Weighted relative to Full Scale). Both are basically the same. LUFS is utilized in the EBU R128 spec. and LKFS is utilized in the ATSC A/85 spec. What is important is that a Loudness Meter can display Program Loudness in either LUFS or LKFS.

The Short Term Loudness (S) descriptor is measured within a time window of 3 seconds, and the Momentary Loudness (M) descriptor is measured within a time window of 400 ms.

The Loudness Range (LRA) descriptor can be associated with dynamic range and/or loudness distribution. It is the difference between average soft and average loud parts of an audio segment or program. This useful indicator can help operators decide whether dynamic range compression is necessary.

Gating

The specification Gate (G10) function temporarily pauses 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.

Absolute vs. Relative

I mentioned that LUFS and LKFS are displayed on an Absolute scale. For example the EBU R128 Program Loudness target is -23.0 LUFS. For Podcast/Internet/Mobile the Program Loudness target is -16.0 LUFS.

There is also a Relative scale that displays LU’s, or Loudness Units. A Relative LU scale corresponds to an Absolute LUFS/LKFS scale, where 0 LU would equal the specified Absolute target. In practice, -23 LUFS in EBU R128 is equal to 0 LU. For Podcast/Mobile -16.0 LUFS would also be equal to 0 LU. Note that the operator would need to set the proper Program Loudness target in the Meter’s Preferences in order to conform.

ab-rel

LU and dB Relationship

1 LU is equal to 1 dB. So for example you may have measured two programs: Program A checks in at -20 LUFS. Program B checks in at -15 LUFS. In this case program B is +5 LU louder than Program A.

Placement

Loudness Meter plugins mainly support online (Real Time) measurement of an audio signal. For an accurate measurement of Program Loudness of a clip or mixed segment the meter must be inserted in the DAW at the very end of a processing chain, preferably on the Master channel. If the inserts on the Master channel are post fader, any change in level using the Master Fader will result in a global gain offset to the entire mix. The meter would then (over time) display the altered Program Loudness.

If your DAW’s Master channel has pre fader inserts, the Loudness Meter should still be inserted on the Master Channel. However the operator would first need to route the mix through a Bus and use the Bus channel fader to apply global gain offset. The mix would then be routed to the Master channel where the Loudness Meter is inserted.

If your DAW totally lacks inserts on the Master channel, Buses would need to be used accordingly. Setup and routing would depend on whether the buses are pre or post fader.

Some Loudness Meter plugins are capable of performing offline measurements in certain DAW’s on selected regions and/or clips. In Pro Tools this would be an Audio Suite process. You can also accomplish this in Logic Pro X by initiating and completing an offline bounce through a Loudness Meter.

-paul.

Audition CC: Loudness Normalization Pt.2 …

In my previous article I discussed various aspects of the Match Volume Processor in Adobe Audition CC. I mentioned that the ITU Loudness processing option must be used with care due to the lack of support for a user defined True Peak Ceiling.

I also pointed to a video tutorial that I produced demonstrating a Loudness Normalization Processing Workflow recommended by Thomas Lund. It is the off-line variation of what I documented in this article.

Here’s how to implement the off-line processing version in Audition CC …

This is a snapshot of a stereo version of what may very well be the second most popular podcast in existence:

Amplitude Statistics in Audition:

Peak Amplitude:0dB
True Peak Amplitude:0.18dBTP
ITU Loudness:-15.04 LUFS

source-(480)

It appears the producer is Peak Normalizing to 0dBFS. In my opinion this is unacceptable. If I was handling post production for this program I would be much more comfortable with something like this at the source:

Amplitude Statistics in Audition:

Peak Amplitude:-0.81dB
True Peak Amplitude:-0.81dBTP
ITU Loudness:-15.88 LUFS

intermediate-(480)

We will be shooting for the Internet/Mobile/Podcast target of -16.0 LUFS Program Loudness with a suitable True Peak Ceiling.

The first step is to run Amplitude Statistics and determine the existing Program Loudness. In this case it’s -15.88 LUFS. Next we need to Loudness Normalize to -24.0 LUFS. We do this by simply calculating the difference (-8.1) and applying it as a Gain Offset to the source file.

The next step is to implement a static processing chain (True Peak Limiter and secondary Gain Offset) in the Audition Effects Rack. Since these processing instances are static, save the Effects Rack as a Preset for future use.

Set the Limiter’s True Peak Ceiling to -9.5dBTP. Set the secondary Gain Offset to +8dB. Note that the Limiter must be inserted before the secondary Gain Offset.

Process, and you are done.

In this snapshot the upper waveform is the Loudness Normalized source (-24.0 LUFS). The lower waveform in the Preview Editor is the processed audio after it was passed through the Effects Rack chain.

lund-method-(480)

In case you are wondering why the Limiter is before the secondary Gain instance – in a generic sense, if you start with -9.5 and add 8, the result will always be -1.5. This translates into the Limiter doing it’s job and never allowing the True Peaks in the audio to exceed -1.5dBTP. In essence this is the ultimate Ceiling. Of course it may be lower. It all depends on the state of the source file.

This last snapshot displays the processed audio that is fully compliant, followed by it’s Amplitude Statistics:

normalized-(480)

stats-audition

In Summary:

[– Determine Program Loudness of the source (Amplitude Statistics).

[– Loudness Normalize (Gain Offset) to -24.0 LUFS.

[– Run your saved Effects Rack chain that includes a True Peak Limiter (Ceiling set to -9.5dBTP) and a secondary +8dB Gain Offset.

Feel free to ping me with questions.

-paul.

Skype in the Box …

Scenario:

Studio Host and Skype participant to be recorded inside your DAW utilizing a slightly advanced configuration.

The session will require a proper mix-minus using your mixer’s Aux Send to feed the Skype Input – minus the Skype participant.

Objectives:

[– Two discrete mono Host/participant recordings with minimal or no processing.

[– Host Mic routed through a voice processing chain using plugins.

[– Incoming Skype routed through a compressor to tame levels, if necessary.

[– One fully processed stereo mix of the session with the Host audio on the left channel and the Skype participant on the right channel.

[– Real time recording and output.

There are certainly various ways to accomplish these objectives utilizing a Bounce to Track concept. The optional inserted plugins and even the routing decisions noted below are entirely subjective. And success with this implementation will depend on how resourceful your system is. I would recommend that you send the session audio out in real time to an external recorder for backup.

Configuration:

This particular example works well for me in Pro Tools. I tried to make this design as generic as possible. My guess is you will have no trouble applying these concepts in any professional DAW. (Click to enlarge)

Skype-NEW-480

Setup:

First I’ll mention that I’m using a Mackie Onyx 1220i Firewire Mixer. This device is defined as my default system I/O. The mixer has a sort nifty feature that allows the creation of a mix-minus just by the press of a button.

onyx-480

Pressing the Input button located on the mixer’s Line In 11-12 channel(s) sets the computer’s audio output as the channel’s input, passing the signal through Firewire 1-2. Disengaging this button will set the Input(s) to Line and the channels’s 1/4″ Input jacks would become active.

Skype recognizes the mixer as the default I/O. So I plug my mic into the mixer’s Channel 1 Input and hard-pan left. I then hard-pan Channel(s) 11-12 right. With the Input button pressed – I can hear Skype. In order to create a successful mix-minus you need to tell the mixer to prevent the Skype input from being inserted back into the Main Mix. These options are located in the mixer’s Source Matrix Control area.

This configuration translates into a Pro Tools session by setting the Track 1 Input (mono) to Onyx Channel 1 and the Track 2 Input (mono) to Onyx Channel 12. I now have discrete channels of audio coming into Pro Tools on independent tracks.

Typically I insert noise reduction plugins on the Mic Input Channel. A Gate basically mutes the channel when there is no signal, and iZotope’s Dialog DeNoiser handles problematic broadband noise in real time. At this stage the Skype Input is recorded with no processing.

Next, both Input Channels are bused out to independent mono Auxiliary Inputs that are hard-panned left + right respectively in preparation to route the passing audio to a Stereo Record bus. To process the mic signal passing through Aux 1 I usually insert something like Waves MaxxVolume, FabFilter’s Pro-DS, and Avid’s Impact Compressor.

For the Skype audio passing through Aux 2, I might insert a gain stage plugin and another instance of Avid’s Impact Compressor. This would keep the Skype audio in check in the event the guest’s delivery is problematic.

The last step is to bus out the processed audio to a Stereo Audio Track with it’s channels hard-panned left + right. This will maintain the channel separation that we established by hard-panning the Aux Inputs. On this track I may insert a Loudness Maximizer and a Peak Limiter. The processed and recorded stereo file will contain the Mic audio on the Left Channel and the Skype audio on the Right Channel.

Finally you’ll notice I have a Loudness Meter inserted on the Master in one of the Pro Tools Post Fader inserts. Once a session is completed I can disarm the “Record” track and monitor the stereo mixdown. Since the Loudness Meter will be operating Post Fader, I can apply a global gain offset using the Master Fader. Output measurements will be accurate. Of course at this point the channels that contain the original discrete mono recordings would need to be muted.

Notes

All the recording and processing steps in this session can be executed in real time. You simply define your Inputs, add Inserts, set up panning/routing, and finally arm your tracks to record. You will be able to converse with the Skype guest as you monitor the session through the mixer’s headphone output with no latency issues. When the session ends you will have access to independent mono recordings for both participants and a processed stereo mix with discrete channels.

Note that you can also implement this workflow as a two step process by first recording the Host/Skype session as discrete mono files. Then Bounce to Track (or Disk) to create the stereo mixdown.

Again the efficiency of this workflow will depend on how resourceful your system is. You might consider running Skype on a separate computer. And I reiterate: as you record in the box, consider sending the session audio out to an external recorder for backup.

-paul.

Podcast Loudness Processing Workflow …

Below is Elixir by Flux. This is an ITU-R BS.1770/EBU R128 compliant multichannel True Peak Limiter. It’s just one of the tools available that can be used in the workflow described below. In this post I also mention the ISL True Peak Limiter by Nugen Audio.

If you have any questions about these tools or Loudness Meters in general, ping me. In fact I think my next article will focus on the importance of learning how to use a Loudness Meter, so stay tuned …

elixir

In my previous post I made reference to an audio processing workflow recommended by Thomas Lund. The purpose of this workflow is to effectively process audio files targeting loudness specifications that are suitable for internet and mobile distribution. in other words – Podcasts.

My first exposure to this workflow was reading “Managing Audio Loudness Across Multiple Platforms” written by Mr. Lund and included in the January 2013 edition of Broadcast Engineering Magazine.

Mr. Lund states:

“Mobile and computer devices have a different gain structure and make use of different codecs than domestic AV devices such as television. Tests have been performed to determine the standard operating level on Apple devices.

Based on 1250 music tracks and 210 broadcast programs, the Apple normalization number comes out as -16.2 LKFS (Loudness, K-weighted, relative to Full Scale) on a BS.1770-3 scale.

It is, therefore, suggested that when distributing Podcast or Mobile TV, to use a target level no lower than -16 LKFS. The easiest and best-sounding way to accomplish this is to:

[– Normalize to target level (-24 LKFS)

[– Limit peaks to -9 dBTP (Units for measurement of true peak audio level, relative to full scale)

[– Apply a gain change of +8 dB

Following this procedure, the distinction between foreground and background isn’t blurred, even on low-headroom platforms.”

Here is my interpretation of the steps referenced in the described workflow:

Step 1 – Normalize to target level -24.0 LUFS. (Notice Mr. Lund refers to LKFS instead of LUFS. No worries. Both are the same. LKFS translates to Loudness Units K-Weighted relative to Full Scale).

So how do we accomplish this? Simple – the source file needs to be measured and the existing Program Loudness needs to be established. Once you have this descriptor, it’s simple math. You calculate the difference between the existing Program Loudness and -24.0. The result will give you the initial gain offset that you need to apply.

I’ll point to a few off-line measurement utilities at the end of this post. Of course you can also measure in real time (on-line). In this case you would need to measure the source in it’s entirety in order to arrive upon an accurate Program Loudness measurement.

Keep in mind since random Program Loudness descriptors at the source will vary on a file to file basis, the necessary gain offset to normalize will always be different. In essence this particular step is variable. Conversely steps 2 and 3 in the workflow are static processes. They will never change. The Limiter Ceiling will always be -9.0 dBTP, and the final gain stage will always be + 8dB. The -16.0 LUFS target “math” will only work if the Program Loudness is -24.0 LUFS at the very beginning from file to file.

Think about it – with the Limiter and final gain stage never changing, – if you have two source files where file A checks in at -19.0 LUFS and File B checks in at -21.0 LUFS, the processed outputs will not be the same. On the other hand if you always begin with a measured Program Loudness of -24.0 LUFS, you will be good to go.

Examples:

[– If your source file checks in at -20.0 LUFS … with -24.0 as the target, the gain offset would be -4.0 dB.

gain

[– If your source file checks in at -15.6 LUFS … with -24.0 as the target, the gain offset would be -8.4 dB.

[– If your source file checks in at -26.0 LUFS … with -24.0 as the target, the gain offset would be +2.0 dB.

[– If your source file checks in at -27.3 LUFS … with -24.0 as the target, the gain offset would be +3.3 dB

In order to maintain accuracy, make sure you use the float values in the calculation. Also – it’s important to properly optimize the source file (see example below) before performing Step 1. I’m referring to dynamics processing, equalization, noise reduction, etc. These options are for the most part subjective. For example if you prefer less compression resulting in wider dynamics, that’s fine. Handle it accordingly.

Moving forward we’ve established how to calculate and apply the necessary gain offset to Loudness Normalize the source audio to -24.0 LUFS. On to the next step …

Step 2 – Pass the processed audio through a True Peak Limiter with it’s Peak Ceiling set to -9.0 dBTP. Typically I set the Channel or “Stereo” Link to 100%, limiting Look Ahead to 1.5ms and Release Time to 150ms.

Step 3 – Apply +8dB of gain.

You’re done.

You can set this up as an on-line process in a DAW, like this:

Lund-480

I’m using the gain adjustment feature in two instances of the Avid Time Adjuster plugin for the initial and final gain offsets. The source file on the track was first measured for Program Loudness. The necessary offset to meet the initial -24.0 LUFS target was -4 dB.

The audio then passes through the Nugen ISL True Peak Limiter with it’s Peak Ceiling set to -9.0 dBTP. Finally the audio is routed through the second instance of the Adjuster plugin adding +8 dB of gain. The Loudness meter displays the Program Loudness after 5 minutes of playback and will accurately display variations in Program Loudness throughout. Bouncing this session will output to the Normalized targets.

Note that you can also apply the initial gain offset, the limiting, and the final gain offset as independent off-line processes. The preliminary measurement of the audio file and gain offset are still required.

Example Workflow

Review the file attributes:

measurements-480
source_480

The audio is fairly dynamic. So I apply an initial stage of compression:

Intermediate-480

Next I apply additional processing options that I feel are necessary to create a suitable intermediate. I reiterate these processing options are entirely subjective. Your desire may be to retain the Loudness Range and/or dynamic attributes present in the original file. If so you will need to process the audio accordingly.

Here is the intermediate:

processed-stats-480
Processed-480

The Program Loudness for this intermediate file is -20.2 LUFS. The initial gain offset required would be -3.8 dB before proceeding.

After applying the initial gain offset, pass the audio through the limiter, and then apply the final gain stage.

This is the resulting output:

normalized-specs-480
new-loudness-normalized

That’s about it. We’re at -16.0 LUFS with a suitable True Peak Max.

I’ve experimented with this workflow countless times and I’ve found the results to be perfectly acceptable. As I previously stated – preparation of your source or intermediate file prior to implementing this three step process is subjective and totally up to you. The key is your output will always be in spec..

Offline Measuring Tools

I can recommend the following tools to measure files “off-line.” I’m sure there are many other options:

[– The new Loudness Meters by TC Electronic support off-line measurements of selected audio clips in Pro Tools (Audio Suite).

[– Auphonic Leveler Batch Processor. I don’t want to discount the availability and effectiveness of the products and services offered by Auphonic. It’s a highly recommended web service and the standalone application that includes high quality audio processing algorithms including Loudness Normalization.

[– Using FFmpeg from the command line.

Example syntax:

ffmpeg -nostats -i yourSourceFile.wav -filter_complex ebur128=peak=true -f null –

[– Using r128x from the command line.

Example syntax:

r128x yourSourceFile.wav

Note there is a Mac only front end (GUI) version of r128x available as well.

-paul.

Fresh Air Podcast: Audio Analysis …

In my No Free Pass for Podcasts post I talked about why the Broadcast Loudness specs. are not necessarily suitable for Podcasts. I noted that the Program Loudness targets for EBU R128 and ATSC A/85 are simply too low for internet and mobile audio distribution. Add excessively dynamic audio to the mix and it will complicate matters further, especially when listeners use mobile devices to consume their media in less than ideal ambient spaces.

fa-processed

Earlier today I was discussing this issue with someone who is well versed in all aspects audio production and loudness processing. He noted that ” … the consensus of it all is, that it is a bad idea to take a really nice standard that leaves plenty of headroom and then start creating new standards with different reference values.” The fix would be to “keep production and storage at -23.0 LUFS and then adjust levels in distribution.” Valid points indeed. However in the real world this mindset is unrealistic, especially in the internet/mobile/Podcasting space.

The fact of the matter is there is no way to avoid the necessity to revise the standards that simply do not work on a platform that consists of unique variables.

And so considering these variables, the implementation of thoughtful, revised, best practices that include platform specific targets for Program Loudness, Loudness Range, and True Peak are unavoidable. Independent Podcasters and network driven Podcasts using arbitrary production techniques and delivery methods simply need direction and guidance in order to comply. In the end it’s all about presenting well produced media to the listener.

Recently I came across a tweet where someone stated “I love the show but it is consistently too quiet to listen to on my phone.” They were referring to the NPR program Fresh Air. I’m not exactly sure if this person was referring to the radio broadcast stream or the distributed Podcast. Either way it’s an interesting assertion that I can directly relate to.

I subscribe to the Fresh Air Podcast. This will probably not surprise you – I refuse to listen to the Podcast right out of the box. When a new show pops up in Instacast, I download the file, decode to WAV, convert to stereo, and then reprocess the audio. I tweak the dynamic range and address show participant audio level variations using various plugins. I then bump things up to -16.0 LUFS (using what I like to refer to as “The Lund Method”) while supplying enough headroom to comply with -1.0 dBTP as my ultimate ceiling. I’ll get into the specifics in a future post.

According to the leading expert Mr. Thomas Lund:

“Mobile and computer devices have a different gain structure and make use of different codecs than domestic AV devices such as television. Tests have been performed to determine the standard operating level on Apple devices. Based on 1250 music tracks and 210 broadcast programs, the Apple normalization number comes out as -16.2LKFS (Loudness, K-weighted, relative to Full Scale) on a BS.1770-3 scale.

It is, therefore, suggested that when distributing podcast or Mobile TV, to use a target level no lower than -16LKFS. The easiest and best-sounding way to accomplish this is to: 1) Normalize to target level (-24LKFS); 2) Limit peaks to -9dBTP (Units for measurement of true peak audio level, relative to full scale); and 3) Apply a gain change of +8dB. Following this procedure, the distinction between foreground and background isn’t blurred, even on low-headroom platforms.”

In this snapshot I demonstrate the described workflow. I’m using two independent instances of the bx_control plugin to apply the gain offsets at various stages of the signal flow. After the initial calculated offset is applied, the audio is routed through the Elixr True Peak Limiter and then out through the second instance of bx_control applying +8dB of static gain. You can also replicate this workflow on an off-line basis. Note that I’ve slightly altered the limiting recommendation.

Lund-small

So why do I feel the need to do this?

Podcast Source

These are the specs. and the waveform overview of a recently published Fresh Air Podcast in it’s entirety:

raw-specs
fa-source-complete

Next is a 3 min. audio segment lifted from the published Podcast. The stats. display measurements of the attached 3 min. segment:

source_revised
source-1

Podcast Optimized for Internet/Mobile

Below is the same 3 min. segment. I reprocessed the audio to make it suitable for Podcast distribution. The stats. display measurements of the attached audio segment:

web-specs-2
source-2

The difference between the published source audio and the reprocessed version is quite obvious. The Loudness Normalized audio is so much more intelligible and easier to listen to. In my view the published audio is simply out of spec. and unsuitable for a Podcast.

Bear in mind the condition of the source audio is not uncommon. The problems that persist are not exclusive to podcasts distributed by NPR or by any of their affiliates. Networks with global reach need to recognize their Podcast distribution platforms as important mechanisms to expand their mass appeal.

It has been noted that the Public Radio community in general is exploring ways to enhance the way in which they produce their programs with focus on loudness standardization. My hope hope is this carries over to their Podcast platforms as well.

-paul.

For more information please refer to “Managing Audio Loudness Across Multiple Platforms” by Thomas Lund at TVTechnology.com.

Avid Impact …

Since upgrading to Pro Tools 11 – I lost access to one of my favorite plugins – The Glue by Cytomic. The Glue is an analog modeled console-style Mix Bus Compressor that supports Side-Chaining and features a classic needle type gain reduction meter. This plugin gets high marks in the music production community. In my work I find it very useful on mix buses and to tame dynamics in individual clips. At this time there is no AAX Native version available, although I’ve read a release may be imminent.

After using The Glue for about a year – I grew very fond of the form factor and ease of use. And, the analog gain reduction meter is just too cool. Here’s a video that demonstrates how The Glue can be used as a Limiter to tame transients.

I have a bunch of Compressors that I use in Pro Tools including C1 by Waves and Pro-C by FabFilter. I also use the Compressors included in the Dynamics modules in iZotope’s Ozone and Alloy plugins.

I decided to look around for a suitable replacement for The Glue that would work well in my Pro Tools environment. I was surprised when I stumbled upon something offered by Avid … Impact Mix Bus Compressor.

impact_blog

Before shelling out $300 for this plugin, I decided to check eBay. Sure enough I found a reliable reseller who was accepting offers for this previously registered plugin by way of an iLok license transfer. I secured the license for $80. I’m hoping this is legit

Regardless, I’m looking forward to adding this new tool to my Pro Tools rig. We’ll see how well it stacks up against The Glue.

-paul.

Update:The license transfer worked out fine and from what I’ve heard the process is totally legit …

Waves WLM Plus Loudness Meter …

Waves has just released a stellar update to their critically acclaimed WLM Loudness Meter. The new WLM Plus version, available for free to those who are eligible – includes a few new and very useful features.

The plugin now acts as both a Loudness Meter and a Loudness Processor. New controls (Gain/Trim) are located in the Processing Panel and are designed to apply loudness normalization and correction. There is also a new switchable True Peak Limiter that adheres to the True Peak parameter defined in the selected running preset.

Here’s how it works:

Notice below I am running WLM Plus using my own custom preset (figg -16 LUFS). Besides the obvious Integrated Loudness target (-16 LUFS), I’ve defined -1.0 dBTP as my True Peak ceiling.

wlm-blog

What you need to do is insert the plugin at the end of your chain. Turn on the True Peak Limiter. Now play through the entire segment that you wish to measure and correct. During playback the textField value located on the WLM Plus Trim button will update in realtime, displaying the proper amount of gain compensation that is necessary to meet the Integrated Loudness target (it’s +2.1 dB in this example).

When measurement is complete, simply press the Trim button. This will set the Gain slider to the proper value for accurate compensation. Finish up by bouncing the segment through WLM Plus, much the same as any processing plugin. The processed audio will now match the Integrated Loudness Preset target and True Peaks will be limited accordingly.

I haven’t tested this in Pro Tools but my guess is this also works when using WLM Plus as an Audio Suite process on individual clips.

Of course you can make a manual adjustment to the Gain slider as well. In this case you would use the displayed Trim Value to properly set the necessary amount of gain compensation.

Great update to this well designed Loudness Meter.

-paul.

Adobe Loudness Radar Up and Running …

With the release of the Adobe “CC” versions of Audition and Premiere Pro, users now have access to a customized version of the tc electronic Loudness Radar Meter.

LR-Banner

In this video from NAB 2013, an attendee asks an Adobe Rep: “So I’ve heard about Loudness Radar … but I don’t really understand how it works.”

I thought it would be a good idea to discuss the basics of Loudness Radar, targeting those who may not be too familiar with it’s design and function. Before doing so, there are a few key elements of loudness meters and measurement that must be understood before using Loudness Radar proficiently.

Loudness Measurement Specifications:

Program “Integrated” Loudness (I): The measured average loudness of an entire segment of audio.

Loudness Range (LRA): The difference between average soft and average loud parts of a segment.

True Peak (dBTP): The maximum electrical amplitude with focus on intersample peaks.

Meter Time Scales:

• Momentary (M) – time window:400ms
• Short Term (S) – time window:3sec.
• Integrated (I) – start to stop

Program Loudness Scales

Program Loudness is displayed in LUFS (Loudness Units Relative to Full Scale), or LKFS (Loudness K-Weighted Relative To Full Scale). Both are exactly the same and reference an Absolute Scale. The corresponding Relative Scale is displayed in LU’s (Loudness Units). 0 LU will equal the LUFS/LKFS Loudness Target. For more information please refer to this post.

LU’s can also be used to describe the difference in Program Loudness between two segments. For example: “My program is +3 LU louder than yours.” Note that 1 LU = 1 dB.

Meter Ranges (Mode/Scale)

Two examples of this would be EBU +9 and EBU +18. They refer to EBU R128 Meter Specifications. The stated number for each scale can be viewed as the amount of displayed loudness units that exceed the meter’s Loudness Target.

From the EBU R128 Doc:

1. (Range) -18.0 LU to +9.0 LU (-41.0 LUFS to -14.0 LUFS), named “EBU +9 scale”

2. (Range) -36.0 LU to +18.0 LU (-59.0 LUFS to -5.0 LUFS), named “EBU +18 scale”

The EBU +9 Range is well suited for broadcast and spoken word. EBU +18 works well for music, film, and cinema.

Loudness Compliance: Standardized vs. Custom

As you probably know two ubiquitous Loudness Compliance Standards are EBU R128 and ATSC A/85. In short, the Target Loudness for R128 is -23.0 LUFS with peaks not exceeding -1.0 dBTP. For ATSC A/85 it’s -24.0 LKFS, -2.0 dBTP. Compliant loudness meters include presets for these standards.

Setting up a loudness meter with a custom Loudness Target and True Peak is often supported. For example I advocate -16.0 LUFS, -1.5 dBTP for audio distributed on the internet. This is +7 or 8 LU hotter than the R128 and/or ATSC A/85 guidelines (refer to this document). Loudness Radar supports full customization options to suit your needs.

Pause/Reset

Loudness meters have “On and Off” switches, as well as a Reset function. For Loudness Radar – the Pause button temporarily halts metering and measurement. Reset clears all measurements and sets the radar needle back to the 12 o’clock position. Adobe Loudness Radar is mapped to the play/pause transport control of the host application.

Gating

The Loudness Standard options available in the Loudness Radar Settings designate Measurement Gating. In general, the Gate pauses the loudness measurement when a signal drops below a predefined threshold, thus allowing only prominent foreground sounds to be measured. This results in an accurate representation of Program Loudness. For EBU R128 the relative threshold is -10 LU below ungated LUFS. Momentary and Short Term measurements are not gated.

• ITU BS.1770-2 (G10) implements a Relative Gate at -10 LU and a low level Gate at -70 LU.

• Leq(K) implements a -70 LU low level Gate to avoid metering bias during 100% silent passages. This setting is part of the ATSC A/85 Specification.

Loudness Radar In Use

In Audition CC you will find Loudness Radar located in Effects/Special/Loudness Radar Meter. It is also available in the Effects Rack and in the Audio Mixer as an Insert. Likewise it is available in Premiere Pro CC as an Insert in the Audio Track Mixer and in the Audio Effects Panel. In both host applications Loudness Radar can be used to measure individual clips or an entire mix/submix. Please note when measuring an audio mix – Loudness Radar must be placed at the very end of the processing chain. This includes routing your mix to a Bus in a multitrack project.

Most loudness meters use a horizontal graph to display Short Term Loudness over time. In the image below we are simulating 4 minutes of audio output. The red horizontal line is the Loudness Target. Since the simulated audio used in this example was not very dynamic, the playback loudness is fairly consistent relative to the Loudness Target. Program Loudness that exceeds the Loudness Target is displayed in yellow. Low level audio is represented in blue.

Each horizontal colored row represents 6 LU of audio output. This is the meter’s resolution.

histrogram

Loudness Radar (click image below for high-res view) uses a circular graphic to display Short Term Loudness. A rotating needle, similar to a playhead tracks the audio output at a user defined speed anywhere from 1 minute to 24 hours for one complete rotation.

LM-480

The circular LED meter on the perimeter of the Radar displays Momentary Loudness, with the user defined Loudness Target (or specification target) visible at the 12 o’clock position. The Momentary Range of the LED meter reflects what is selected in the Settings popup. The user can also customize the shift between green and blue colors by adjusting the Low Level Below setting.

The numerical displays for Program Loudness and Loudness Range will update in real time when metering is active. The meter’s Loudness Unit may be displayed as LUFS, LFKS, or LU. The Time display below the Loudness Unit display represents how long the meter is/was performing an active measurement (time since reset). Lastly the Peak Indicator LED will flash when audio peaks exceed the Peak Indicator setting.

If this is your first attempt to measure audio loudness using a loudness meter, focus on the main aspects of measurement:Program, Short Term, and Momentary Loudness. Also, pay close attention to the possible occurrence of True Peak overs.

In most cases the EBU R128 and ATSC A/85 presets will be suitable for the vast majority of producers. Setup is pretty straightforward:select the standardization preset that displays your preferred Loudness Unit (LUFS, LKFS, or LU’s) and fire away. My guess is you will find Loudness Radar offers clear and concise loudness measurements with very little fuss.

Notes:

You may have noticed the Loudness Target used in the above graphic is -16.0 LUFS. This is a custom target that I use in my studio for internet audio loudness measurements.

-paul.

Articles and Documentation used as Reference:

tc electronic LM2 Plugin Manual

ITU-R BS.1770-3 Algorithms to measure audio programme loudness and true peak audio level

EBU R128 Loudness Recommendation

EBU-Tech 3341 Loudness Metering

Loudness Meters: Absolute/Relative Scales …

Professional audio Loudness Meters measure Program (Integrated) Loudness using an Absolute scale displayed in LUFS (or LKFS). For example the EBU R128 Program Loudness target is -23.0 LUFS (Loudness Units Relative to Full Scale).

When the ITU defined new audio loudness measurement guidelines, the general consensus was that many audio engineers would prefer to mix to the familiar “0” level on a Loudness Meter for compliance targeting. A Relative scale option was implemented. It references Loudness Units (LU), where 0 LU equals the corresponding LUFS/LKFS compliance target.

So for EBU R128 … 0 LU == -23.0 LUFS.

In the snapshot below you can see my Nugen VisLM Loudness Meter set to display Absolute scale (left) and Relative scale (right).

scale-blog

Of course in most cases the scales and corresponding targets are customizable. For example I advocate -16.0 LUFS as the loudness target for audio distributed on the internet. By defining -16.0 LUFS as my Absolute scale compliance target in a meter’s setup options, 0 LU (Relative scale) would be equivalent to -16.0 LUFS.

Below is a basic side by side comparison of EBU R128 Absolute and Relative scales:

figg-scale

-paul.

Internet Audio: True Peak Compliance …

Wide variations in average (Program/Integrated) Loudness are common across all forms of audio distributed on the internet. This includes audio Podcasts, Videocasts, and Streaming Media. This is due to the total lack of any standardized guidelines in the space. Need proof? Head over to Twit.tv and listen to a few minutes of any one of their programs. Use headphones, and set your playback volume to a comfortable level.

Now head over to PodcastAnswerMan.com, and without making any change to your playback volume – listen to the latest program.

I rest my case.

In fact, there is a 10 LU difference in average loudness between the two. Twit.tv programs check in at approximately -22 LUFS. PodcastAnswerMan checks in at approximately -12 LUFS. I find this astonishing, but I am not surprised. I’m not signaling them out for any lack of quality issues or anything like that. In my view both networks do a great job, and my guess is they have sizable audiences. Both shows are well produced and it simply makes sense to compare them in this case study.

With all this in mind let me stress that at this particular time I am not going to focus on discussing Program Loudness variations or any potential suggested standard. I can assure you this is coming! I will say that I advocate -16.0 LUFS (Program/Integrated Loudness) for all media formats distributed on the internet. Stay tuned for more on this. For now I would like to discuss True Peak compliance that will be a vital part of any recommended distribution standard.

What surprises me more than Program Loudness inconsistency is just how many producers are pushing files with clipped, distorted audio. In many cases Intersample Peaks are present in audio files that have been normalized to 0 dBFS. (For more information on Intersample Peaks please refer to this brief explanation). Producers need to correct this problem before their audio is distributed.

The Tools

One of the most useful features included in Adobe Audition is the Match Volume Processor. This tool includes various options that allow the operator to “dial in” specific average loudness and peak amplitude targets. After processing, the operator can examine the results by using Audition’s Amplitude Statistics analysis to check for accuracy.

mvp-1

Notice in the snapshot above I set the processor to Match To: Total RMS, with a -18.50 dB RMS average target. I’ve also selected the Use Limiting option. I’m able to dial in custom Look-Ahead and Release Time parameters as I see fit. Is there something missing? Indeed there is. Any time you push average levels you run the risk of clipping the source. In Audition the Match Volume/Use Limiting option lacks the capability for the operator to set a specific Peak Amplitude Ceiling. I’ve determined that in certain situations Peak Amplitudes reach a -0.1 dB ceiling resulting in possible clipped samples and True Peak levels that exceeded 0dBFS. Keep in mind this is not always the case. The results depend on the Dynamic Range and available Headroom of any source.

So how do we handle it?

Notice above the Match Volume Processor offers two Peak Amplitude options: Peak Amplitude and True Peak Amplitude. The European Broadcasting Union’s EBU R128 spec. dictates -1.0 dBTP (True Peak) as the ultimate ceiling to meet compliance. Here in the states ATSC A/85 dictates -2.0 dBTP. Since most, if not all audio formats distributed on the internet are delivered in lossy formats, it is important to pay close attention to True Peak Amplitude for both source (lossless) and distribution (lossy) files.

fgm

I advocate -1.0 dBTP as the standard for internet based audio file delivery. True Peak Limiters are able to detect and alleviate the possibility of Intersample Peaks from occurring. It is recommended to pass audio through a True Peak compliant limiter after loudness normalization and prior to lossy encoding. Options include ISL by Nugen Audio, Elixir by Flux, and (the best kept secret out there) TB Barricade by ToneBoosters. If you are running Audition, Match To: True Peak Amplitude and you should be all set.

The plugin developers mentioned above as well as Waves, MeterPlugs, tc electronic, Grimm Audio, and iZotope supply Loudness Meters and toolsets that display all aspects of loudness specifications including True Peak alerts. Visit this page for a list of supported Loudness Meters.

If True Peak detection and compliance is not within your reach due to the lack of capable tools, a slightly more aggressive ceiling (-1.5 dBFS) is recommended for Peak Normalization. The additional .5 dB acts as a sort of safety net, insuring maximum peak amplitude remains at or below -1.0 dBFS. One thing to keep in mind … performing Peak Amplitude Normalization after Loudness Normalization may very well result in a reduction in average, program loudness. Once again changes to the processed audio will depend on the audio attributes prior to Peak Normalizing.

Below I’ve supplied data that supports what I noted above. The table displays three iterations of a test file: Input, Loudness Normalized Intermediate, and final Output. For this test I used the ITU-R BS.1770-2 “Match To” option in Audition’s Match Volume Processor. I pushed the average target to -16.0 LUFS. As noted, this is the target that I advocate for internet and/or mobile audio. This target is +7 LU hotter than R128 and +8 LU hotter than ATSC A/85.

After processing the Input file, the average target was met in the Intermediate file, but True Peak overs occurred. The Intermediate file was then passed through a compliant True Peak Limiter with it’s ceiling set to -1.0 dBTP. Compliance was met in the Output with a minimal reduction in Program Loudness.

data-480

Producers: there is absolutely no excuse if your audio contains distortion due to clipping! At the very least you should Peak Normalize to -1.5 dBFS prior to encoding your lossy MP3. Every audio application on the planet offers the option to Peak Normalize, including GarageBand and Audacity. Best case scenario is to adopt True Peak compliance and learn how to use the tools that are necessary to get it done. If you are an experienced producer or professional, and you come across content that does not comply – reach out and offer guidance.

-paul.

Waves MaxxVolume Revisited …

Back in October of 2012 I wrote about my purchase and initial impression of MaxxVolume by Waves. Let me first say I’m so glad I bought this tool. Secondly, my timing was impeccable. I was under the impression (when I purchased it) that the price of this plugin was significantly reduced on a permanent basis from $400 to $149 for the “Native” single version. Not the case. It is currently selling for $350 and discounted to $320. Like I said – my timing was impeccable.

waves-mv-478

Anyway, I’ve spent many hours working with this tool. Before I discuss one instance of my workflow, let me also mention that I recently purchased a license for their Renaissance Vox Dynamics Processor. This is yet another stellar tool by Waves. It features three slider “faders”: Gate, Compressor, and Gain. The Gate (Downward Expander) is very impressive. It works well when it may be necessary to tame an elevated noise floor in something like a voice over. The Compression algorithm is what really makes this plugin shine. As expected this setting controls the amount of Dynamic Range Compression applied to the source. At the same time it applies automatic makeup gain. What’s special is as the output gain potentially increases, the plugin will automatically prevent clipping by applying peak limiting. It’s all handled by a single slider setting. It turns out the High Level Compressor included in MaxxVolume is similar to the Compression stage in Renaissance Vox …

I’ve settled in on an order in which I set up MaxxVolume to act as a leveler when processing spoken word. I load the plugin with all controls in the OFF state. First I turn on the Low Level Compressor. This is essentially an Upward Expander that increases the level of softer passages. It doesn’t take much of an increase in gain to achieve acceptable results. At this point I rely solely on my ears for the desired effect.

Next I turn on the Gate (Downward Expander) and listen for any problems with the noise floor that may have resulted from the gain I picked up with the Low Level Compressor. Since I pass all my files through iZotope RX2 before introducing them to MaxxVolume – they are pretty quiet. In most cases the Gate’s Threshold is set somewhere between -60 and -70 dB. By the way the processor is set to the LOUD mode. This setting uses a more aggressive release resulting in a slightly “louder” output signal.

Now that I’ve dealt with low level signals and any potential noise floor issues – I set the Global Gain to -1.0dB. If I am dealing with a previously (loudness) normalized file with a set average target, I almost never deviate from this -1.0dB setting.

The last stage of the processor setup affects the aggression of the Leveler and handles Dynamic Range Compression. As previously stated – the High Level Compressor also applies automatic makeup gain as it’s Threshold is decreased. What’s interesting is it also applies gain compensation to the signal where aggressive leveling may result in heavy attenuation. Here once again if I am dealing with a segment with a set average loudness target, I need to maintain it. So I turn on the Leveler and set it’s Threshold to apply the desired amount of leveling. When the audio passes (goes above) the threshold, leveling is active. The main Energy Meter displays the audio level after the leveler and before any additional dynamics processing functions.

I finish up by turning on the High Level Compressor, setting it’s Threshold to apply the necessary amount of gain compensation to maintain my average (Program/Integrated) Loudness target. I use Nugen’s VisLM Loudness Meter to monitor loudness. Finally I fine tune the Low Level Compressor and Gate.

logic-480

This particular workflow is just one example of how I use MaxxVolume. The processor does an excellent job when setup to function as a speech volume leveler. In other instances I use it to attenuate playback of audio segments, programs, etc. that have been normalized to a much higher average loudness target than I see fit. With the proper settings MaxxVolume provides a highly customized method of gain attenuation that sounds so much better than just reducing output levels with channel faders in a DAW.

MaxxVolume is now an indispensable tool in my audio processing kit …

-paul.

LoudMax and Final Cut Pro X

One of the great features of Final Cut Pro X is the availability of Apple’s 64bit Logic audio processing plugins (aka Filters). In fact FCPX supports all 64bit Audio Units developed by third parties.

Let me first point out I’ve tested a fair amount of 64bit Audio Units in FCPX. Results have been mixed. Some work flawlessly. A few result in sluggish performance. Others totally crash the application. I can report that Nugen’s ISL True-Peak Limiter and Wave Arts Final Plug work very well in the FCPX environment.

ISL is a Broadcast Compliant True-Peak Limiter that uses standardized ITU-R B.S 1770 algorithms. Settings include Input Gain and True-Peak Limit. ISL fully supports Inter-Sample Peak detection.

Final Plug allows the operator to set a limiting Threshold as well as a Peak Ceiling. Decreasing the Threshold will result in an increase of average loudness without the audio output ever exceeding the Ceiling.

Recently Flux released a 64bit version of Elixir, their ITU/EBU compliant True-Peak Peak Limiter. Currently (at least on my MacPro) the plugin is not usable. Applying Elixir to a clip located in the FCPX storyline causes an immediate crash. I’ve reported this to the developer and have yet to hear back from them.

The plugins noted above range in price from $149 to $249.

One recommendation that often appears on discussion forums and blogs is the use of the Logic AU Peak Limiter to boost audio loudness while maintaining brick-wall limiting. This process is especially important when a distribution outlet or broadcast facility defines a specific submission target. A few audio pro’s have taken this a step further and recommended the use of the Logic Compressor instead of the Peak Limiter. In my view both are good. However proper setup can be daunting, especially for the novice user.

These days picture editors need to know how to color correct, create effects, and handle various aspects of audio processing. If you are looking for a straight forward audio tool that will brick-wall limit and (if necessary) maximize loudness, I think I found a viable solution.

lM.jpg LoudMax is an easy to use Peak Limiter and Loudness Maximizer. Operators can use this plugin to drive audio levels and to set a brick-wall Output Ceiling.

The LoudMax Output Slider sets the output Ceiling. So if you are operating in the “just to be safe mode”, or if you need to limit output based on a target spec., set this accordingly. If you need to increase the average loudness of a clip – decrease the Threshold setting until you reach the desired level. The Output Ceiling will remain intact.

LoudMax also includes a useful Gain Reduction Meter. If viewing this meter is not important to you – there’s no need to run the plugin GUI. The Threshold and Output parameters are available as sliders, much the same as any other FCPX Filter or Template. You can also set parameter Keyframes and save slider settings as Presets.

lM-H

Using the Logic Peak Limiter and/or Compressor is definitely a viable option. Keep in mind that achieving acceptable results takes practice. Proper usage does require a bit more ingenuity due to the complexity of the settings. I’ll be addressing the concepts of audio dynamics Compression in the future. For now I urge you to take a look at LoudMax. It’s 32/64bit and available in both VST and AU formats. The AU Version works fine in FCPX. I found the processed audio results to be perfectly acceptable.

At the time of this writing LoudMax is available as Freeware.

-paul.

Waves and MaxxVolume

The latest addition to my audio processing toolset is MaxxVolume by Waves. This dynamics processor has been on my radar for the past few years. I was always under the impression that Waves plugins required an iLok account/key. It was for this reason I never bothered to pull down the demo and test it.

A few days ago I noticed that a few online plugin resellers were advertising a price drop for MaxxVolume. I believe the original price was $300. Sweetwater and DontCrack are currently selling it for $149. I decided to purchase a license. By the way prior to doing so – I realized Waves has moved away from the iLok requirement. They now provide a standalone “Waves License Center” (WLC) application that can be used to manage both purchased and demo licenses. Licenses can be transferred to a host machine and/or a standard (FAT32 formatted) USB Flash Drive. You can then move and manage licenses via the Flash Drive or within their proprietary License Cloud.

After making a purchase you simply register the new product on the Waves site, run WLC, login to your Waves account – and move your license(s) from the cloud to your target destination. I must say the process was easy and seamless.

So what is MaxxVolume? The plugin is a four module dynamics processor: Low Level Compressor, Gate, Leveler, and High Level Compressor. All four processing stages run in parallel.

The Low Level Compressor is essentially an expander. So any signal that falls below the set threshold gets compressed upward. It’s controlled by a Threshold fader and Gain fader. The Gate feature is controlled by a single Threshold fader that applies gentile downward expansion affecting any signal that drops below the threshold setting. The Leveler is essentially an AGC (Automatic Gain Control) controlled by a single Threshold fader. Lastly the High Level Compressor is controlled by a Threshold fader and a Gain fader. This compressor functions just like any standard compressor – when the input signal exceeds the threshold it is attenuated. The Gain setting compensates for the attenuated signal.

Waves notes “It’s a Broadcast tool, bringing any program to a fixed destination level; ideal for radio and TV, podcasting, internet streaming, and more.” It took me some time to get a feel for how the four processing stages interact. So far I like what I’m hearing. The AGC is pretty impressive. I’m using Adobe Audition CS6 as my host. The processor works fine in the Adobe environment.

I will say this tool is not your sort of cut and dry loudness maximizer. It may not be suitable for less advanced or novice users. In my view a clear understanding of upward/downward expansion, AGC, and compression is a necessity.

-paul.

Intersample Peaks, Normalization, and Limiting …

When preparing to encode MP3 files we need to be aware of the possibility of Intersample Peaks (ISP) that may be introduced in the output, especially when targeting low bit rates. This results from the filtering present in lossy encoding. We alleviate this risk by leaving at least 1 dB of headroom below 0dBFS.

Producers should peak normalize source files slated for MP3 encoding to nothing higher than -1.0 dBFS. In fact I may suggest lowering your ceiling further to -1.5 dBFS sometime in the future. Let me stress that I’m referring to Peak Normalization and not Loudness Normalization. Peak Normalizing to a specific level will limit audio peaks when and if the signal reaches a user defined ceiling. It is possible to set a digital ceiling when performing Loudness Normalization as well. This is a topic for a future blog post.

Notice the ISP in this image lifted from an MP3 wave form. The original source file was peak normalized to -0.1 dBFS and exhibited no signs of clipping.

You can also avoid ISP’s by using a compliant ITU Limiter and setting the ceiling accordingly. During source file encoding this type of limiter will detect when ISP’s may occur in the encoded MP3.

For podcast and internet audio, a limiter set to a standardized ceiling of -1.0/-1.5 dBFS works well and is recommended.

-paul.