Normalizing the audio

In addition to the previous I'm experiencing an issue I hope I would get help for and it would probably help other users as well:  1. I'm using my Tascam Dr-70D audio recorded with XLr input mic lavs and although the audio recorder is set for stereo, on Adobe Audition the audio comes only from one speaker. Is the easiest and best way to get both speakers working without further complications is the Fill from left or right effect dragged to selected multiple audio clips which have left and right channels but the other channel is empty ?

Thanks

First, let's make sure Loudness and the measurement nomenclature is accurately referenced:

LU -> Loudness Unit. 1 LU is a direct correlation to 1 dB. So yes, 1 LU == 1 dB.

LUFS -> Loudness Units Relative to Full Scale. The Integrated or Program loudness descriptor (referenced in LUFS) represents the average, perceived loudness of a measured piece in it's entirety. It's similar to (but not the same as ) RMS. It does NOT represent peak amplitude or a peak ceiling, which by the way does not in any way indicate perceptual loudness. Peak amplitude is a representation of proportional voltage, aka signal level.

When any spec. asks for something like -23.0 LUFS, the average loudness of the entire piece is mixed in RT to that descriptor. Or, the piece is Loudness Normalized after the fact using an off-line process.

It's important to note all specs. require True Peak (as opposed to Sample Peak) compliance as well. Sample Peaks or Intersample Peak values are referenced using dBFS or dBTP respectively on a discrete meter. In essence, Intersample Peaks must be recognized and prevented. The sole purpose of this requirement (or restriction) is to prevent clipping, distortion, codec artifacts, etc. A capable Limiter inserted at the end of a processing chain is used to maintain the user or spec. defined peak ceiling.

Loudness Meters require a user defined or spec. defined Integrated Loudness target. The measurement scale can be set to display Absolute values in LUFS. Or - Relative values in LU's. Their correlation is as follows:

0 LU (Relative) == the LUFS (Absolute) Integrated Loudness Target

For example:

Integrated Loudness target -23.0 LUFS (Absolute scale) == 0 LU (Relative scale)

Integrated Loudness target -16.0 LUFS (Absolute scale) == 0 LU (Relative scale)

Integrated Loudness target -14.0 LUFS (Absolute scale) == 0 LU (Relative scale)

Integrated Loudness target -24.0 LUFS (Absolute scale) == 0 LU (Relative scale)

If you prefer to mix/master using the Absolute scale, you're referencing LUFS. If you prefer to mix/master using the Relative scale - just remember that 0 LU is always the equivalent to the defined Absolute target.

Again this descriptor has no correlation to Peak Amplitude and/or signal level!

As far as specification compliance, you simply mix/master the average, perceptual loudness to the specified Integrated Loudness target and adhere to the specified amplitude ceiling. How you do this is of course subjective.

Lastly, the delivery targets, most notably the Integrated Loudness target - will vary based on the institution or distribution platform. For example you wouldn't target -23.0 LUFS for a highly dynamic Podcast that would most likely be consumed on a mobile device in a less than ideal environment. It would be difficult to comfortably consume under those circumstances. It's simply not loud enough. Add wide dynamics and these problematic consumption issues will be intensified. -16.0 LUFS (Stereo) is recognized as a suitable target for this particular platform.

Bottom line - recognize the difference between referenced average/perceived loudness and peak amplitude. Use your ears accordingly and mix/master based on how (and where) you intend to distribute your work.

-paul.

@produceNewMedia

This is actually quite a complicated issue. I think that the place to come at it from is to assume that viewers don't want to have to keep adjusting the volume control whilst they're watching a programme. Where it gets more complicated is that, obviously, not all dialogue comes in at the same level, and also some speakers manage to use a much greater dynamic range than others - so the soft parts of their speech are much harder to hear.

This is where compression comes into play, especially with mixed sources. It's perfectly possible to reduce the peaks of spoken dialogue (aka 'limiting') by anything up to 9dB without anybody seriously noticing, and that on its own can make level setting a whole heap easier, sometimes obviating the need for any other compression at all.

As far as levels are concerned, there's a difference between the level you make the original recording at, and the level you process it at. Normally you'd make the original recording with peaks up to around -12dB, just to give you enough 'headroom' if somebody shouts, or whatever. But you wouldn't process it at this level - that's far too low.

All compressors only work at their correct setting levels when they're presented with signals that peak at 0dB, so the first thing you do with any dialogue recorded at -12dB is normalize it to just under 0dB, and then attend to whatever you're going to do with it (like run it through Audition's Dynamics Processing). Rather than me going through all of the Dynamics Processing options, I can point you at a video which explains all this pretty well - Dynamics processing - YouTube

As far as the mix itself is concerned, almost certainly you want all these sources as mono anyway, and you just pan them where you need to in the stereo field.

In strict terms, particularly in Audition, Normalisation does not make all your peaks the same level. Provided all your audio is in one file then Normalisation to -3dB will only amplify the audio to make the loudest peaks equal -3dB. It won't squash up lesser peaks to be the same level. Normalisation measures the highest peak within a particular audio file and adjusts the gain of all the audio by the same amount to bring just that peak up to the selected level.

If the audio that you are editing in Audition is recorded to separate takes, some of which are loud dialogue and some are quieter, then you certainly don't want to individually Normalise all the files to a fixed level because, as you rightly say, you will lose the dynamics of the performances. So providing you are recording at a fixed level and you don't alter the recording levels between scenes then you can just apply a fixed Amplification level to all the audio files to bring the highest peaks up to around -3dB.

After you have finished editing your video you may find that there is too much level difference between the quiet and loud parts. If under normal listening conditions you can't hear some of the dialogue under your added effects or music or if the loudest bits are too loud then you can apply some Compression to the whole Dialogue track to slightly reduce the dynamic range.