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.
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.
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 input 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 input 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.
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 suites 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, keeping peak amplitude 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 of 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 this claim. 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. Bear in mind 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.
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.
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.
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.
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 …
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.
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.
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.
If you look in the FCPX Titles Effects Browser under the Lower Thirds Category you will notice an Information Bar Lower Thirds. The is a bundled FCPX Title. The title itself is actually quite stylish. It’s subtle, with a semitransparent black bar and customizable text positioned on two lines.
A few days ago I was sifting through a forum and noticed a post by a member who uses this title regularly. He was asking for help regarding the opacity of the “bar.” Basically it’s opacity was not customizable. It was preset to somewhere around 50%. The forum member politely asked if someone could possibly load up the title in Motion, tweak in an Opacity slider for the bar, and make it available. I knew this would be easy, especially if the default Title supported the “Open a copy in Motion” option. It did and the rest is history.
If you review the settings snapshot below you will notice I added additional options that makes my version much more useful, at least for me. I added support for Global Y Positioning (more on this below), Fade In/Out Frames, Bar Opacity, Bar Left Indent, and Bar Roundness.
By default the Title places the text within the 1.78:1 Title Safe Area located at the bottom left of the zone. The Global Y Position setting allows the operator to cumulatively move all Title elements up on the Y axis to 2.35:1 Title Safe positioning.
The Original version of the Title has two check boxes that control whether all elements fade in and/or out. I added Fade in and Fade Out sliders that support frame by frame customization. Setting the sliders to zero results in no fading.
Bar Opacity is now supported. I believe I set this up to default to 50% Opacity. Regardless – it’s now fully customizable.
Bar Left Indent is an interesting setting. Notice there is also a Bar Roundness setting that will change the shape of the bar. Since by default the bar is anchored to the left of the image frame, applying roundness to it results in a partially obstructed left edge. The Bar Left Indent setting moves the bar’s left edge in a few pixels to the right to compensate. In fact It can be used without any roundness applied as well for creative purposes.
There have been some reports of font change instability. In fact this behavior is also present in the original version of the Title. I found this to be not that big of a deal.
The Installer will place the Title in the FCPX Titles Browser under the Custom Lower Thirds Category/Information Bar Theme.
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.
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 Limiter and setting the digital ceiling accordingly. During source file playback this type of limiter will detect when ISP’s may occur in the encoded MP3. This allows the operator to set the digital ceiling for the source as high as possible prior to encoding.
For podcast and internet audio a limiter set to a standardized ceiling of -1.0/-1.5 dBFS works well and is recommended.
Let’s assume you are finishing up a rough edit for client review consisting of multiple clips and sound. The client requests visible timecode in the review movie. Final Cut Pro includes a Timecode Generator Filter located in Effects/Video Filters/Video. Since this is in fact a filter, it must be applied to each individual clip. The problem with this implementation? The TC Generator will reset on a clip to clip basis as the playhead moves through the sequence.
Our objective is to have the TC Generator display a continuous representation of the entire sequence timecode. I have two suggestions …
The image below represents a Nested sequence:
The original sequence consisted of multiple independent clips. Nesting a selection of timeline assets creates a new self contained sequence without any reference to previous edit points.
To create a Nested Sequence, select the timeline assets. Head up to the FCP Sequence Menu and select Nest Item(s). You can also use the keyboard shortcut ⌥ C. Apply the FCP Timecode Generator Filter to the Nest. The filter will display the RT playback timecode in the Canvas. The Timecode will be visible in the output movie.
Andy’s Timecode Generator
There is another way to do this using a (free) third party generator. Andy’s TC Generator allows you to add a TC Generator directly to your existing sequence as an overlay on a upper video track. The developer notes that you can adjust the offset to match your sequence, or use it as it’s own free running reference. Very cool.
One final note about one of Final Cut Pro’s newest features: The Timecode Viewer HUD.
The resizable Timecode Viewer (Tools/Timecode Viewer or press Control-T) makes reading current timecode very easy. The Timecode Viewer displays the timecode for either the Timeline/Canvas or the Viewer as well as the corresponding sequence name or clip name. You can customize what is displayed by right-clicking either the upper or lower display areas of the HUD.
Tip: for easy access, add a Timecode Viewer Shortcut Button to a Button Bar in the FCP window of choice – (Tools/Button List/Timecode Viewer).
Confused by the term Pulldown, or Telecine?
Here are the facts:
24p = 23.98 fps (Progressive)
29.97 fps = 59.94 interlaced fields per second, aka 60i
• Interlaced video displays 60 half frames per second
• Progressive video takes entire video frames on the go
• Progressive video requires 2x the bandwidth of interlaced video
This is the conversion process: 24p (film or video) — 29.97 (video).
• 2:3, or 3:2 (aka 2:3:2:3): 60 fields / 24 = 2.5. So each frame of 24p material needs to last for 2.5 frames of video
• 2:3:3:2 is referred to as Advanced Pulldown
Here’s how it works: we are transferring 24p to 60i, which means we are converting 24 frames per second into 60 fields per second. The first frame of film is transferred to the first two fields of video and the next frame of film is transferred to the next three fields – 2:3. This results in some frames of film spanning two different frames of video or, to put it another way, some frames of video that are composed of fields from two different frames of film.
Here is a glimpse of what I have planned for the next release of aspectRatio:
At this point I’ve implemented a suggested dimensions method that displays values evenly divisible by 16. The results are triggered by the Target Width and returned Output Height calculation.
Select MPEG formats are based on 16×16 macro-blocks. Evenly divisible (by 16) output dimensions will maximize the efficiency of the encoder and yield optimum results. For example: a purist would prefer a small 16:9 distribution video to be 480×272 instead of the common 480×270
Also included in this release: a user defined output font color preference setting [orange/red], and a Menu option that re-opens the main UI window if the user inadvertently closes it while the application is still running.
A release date has yet to be determined …
Audioarts Engineering, a division of Wheatstone Broadcasting recently debuted their attractive small footprint Air 1 professional audio broadcast/production console. The company states the Air 1 was “specifically designed to meet the needs of on-air, production, news applications, remotes, and the emerging podcasting market.” Features include Dual program Buses, Cueing support, Long Throw Faders, Switchable PGM meters, 2 Monitor Outs, 2 Mic Preamps, Headphone Amp, Solid State Illumination on all switches along with a useful On-Air Indicator light.
Additional features include balanced 1/4″ I/O, external power supply for cool – hum free operation, and bottom mounted Dipswitches designed for easy programing. Lastly, the mic inputs can be programmed to automatically MUTE the Monitor Output when activated. The Air 1 is 2.5″ high, 15.25″ wide, 11.5″ front to back.
No doubt this is a slick device. My guess is professional fans of the Audioarts product line will find this console very attractive. It’s perfect for small scale operations and remote productions. However due to its $1800 price tag, I don’t anticipate wide adaptation within the new media/podcasting space. Standard, sub $1K audio mixers seem to be satisfying the needs of *most* new media producers.