All these talk about setting proper gain settings for recording, appropriate listening levels for playback/mixing and levels for broadcast/theatrical release raises some interesing questions, solutions and tips about audio work.

I've been doing audio for awhile and I dont consider myself highly experienced in the field, but experienced enough to at least do the job. Now, I need to know a couple of things to set my mind straight before I take on another audio project.

Audio mismatch. I understood this completely since I had a very bad personal experience with it before. The problem could be because the microphone output to the recorder was too hot, a case of a bad cable, improper audio connections, or the recording gain setting was too low/too high.

Usually this is preventive as long as you pay attention to your cans and the levels of the recording. During monitoring, headphones should not be giving out its own distortions. If it does, either tone down the listening level or get a better pair of cans.

So am I right to say that you are monitoring the analog signal thus analog clipping (in your headphones; not the levels) are soft transient distortions which are fine, digital clipping is unintelligible and unusable?

Next is the issue of reference tone. I do my work on Pro Tools and as usual, finish it on DAT for layback. So, with my reference tone at -20dB in Pro Tools, at which level should I set the input level of the recorder to if I'm sending the signal through analog connections (-20dB as well or 0dB)?

Lastly, you guys mentioned peaking at -2db, -6db, -10db etc.....what is the usable range for broadcast/theatrical releases so that my loudest sound is safe and my softest sound is still audible?

Interesting philosophy used by video vs audio people. Any music audio recordist in the industry knows that the push is to maximise volume on playback. This means peaks not to exceed -0.1 dB and RMS values around -12dB. These numbers are after compression/limiting. Curious that the video broadcasters use a -20dB reference. At -20dB, playback requires cranking up the volume control MUCH more than musician is used to.

Not being critical, just noting that beauty is in the eye of the beholder.

-20, of course, is for average (rms) levels with peaks usually up to -12dB.

That's the network, NPR and PBS spec. If you're not submitting to them, you can do what you want unless your client wants something else.

Regards,

Ty

It is indeed the spec. Then there is the concept of "pushing the envelope."
I can count on one hand, the number of projects we've had sent back down due to pushing the required levels by a few dB. New York, Vernon Johns Story, Last Queen, The War that Made America, The Way West...all PBS shows, all Emmy, DuPont, or Peabody winners, and all with pushed audio. There is a difference in acquisition, which is where 99% of videographers screw up due to misunderstood threads like this one, and delivery, which is screwed up also due to misunderstanding the differences between averages and peaks.
Acquisition, you want as HOT as possible without exceeding 0. It's common practice, and common sense.
Delivery, you can do whatever you need, but at least you've got the bit depth to have anything to do with, what you wish.

Very well said Douglas. You can't put back what wasn't there to begin with, but, you sure can take something out you don't want. Brings to mind an old axiom, "you can't take it with you if you don't have it when you go." There's only one rule of digital audio acquisition....don't exceed 0 dBFS. And it's a fundamental concept to grasp the difference between acquisition of source material and delivery of a product to a customer.

Which is, BTW, precisely why you want a mixer with a VERY GOOD sounding limiter;
so you can record hot without fear of overs. Also be advised that not all computer systems treat internal audio busses the same.

Some folks have "normalized" all tracks to within a half a DB of clipping only to find that things sound a bit grainey during the mix. They don't realize that the internal buss has to combine all of thse VERY LOUD tracks and may, at some point, not be able to handle that much combined level.

Keeping it real in audio,

Ty Ford

EXACTLY why I think 24 bit(192kHz) recording beats the pants off of 16 bit recorders.

Ty brings up another excellent point, many folks start with normalizing. Big mistake, IMO. Are they normalizing RMS and not peak? Are they normalizing to reduce dynamic range or bring the audio to a louder point? It can create serious havoc once you start mixing if you don't know why you're normalizing beyond "I heard it was good to do."
We don't normalize here; it's the first thing new editors are taught. Leave the levels alone, unless you have reason to believe it wasn't acquired correctly.

Just my opinion, but, "normalizing" is a useless and destructive process. If done at all, it's done at the mastering stage, after the mix has been finished. Even then, normalizing can be done at the RMS level, provided that the peaks are compressed at some level below 0dBFS and not clipped. But, if compression has been applied during mastering, there's not much point in a further compression step.

It is a pleasure to read the contributions from pros like Spot, Ty and Bill. Thank you gentlemen for contributing your decades of experience in a public forum. Readers, in case you are missing it, this is the real stuff.Zulkifli, you are on the right track, and have somewhat answered your own question on gain structure. Here are the basics. Overload can occur at any stage. The mic diaphram, the (condensor) mic preamp, the input stage of the mixer, the output stage of the mixer, the input stage of the camera or recorder, etc.

Analog distortion in the modern production chain is not acceptable (unless as an intended effect). Digital may be crunchier, but analog is still bad. In ancient times, with an analog magnetic tape recorder we tended to consider hotter peaks to the recording as (analog!) tape was very forgiving of minor overload, and tended to still sound "musical".

Even with those technologies, overloading the mic, preamp, recorder electronics, etc. sounded bad and was avoided.

So, the answer to your question about soft analog transient distortions is: Don't overload at any stage, period. Get to know the sound of your headphones, and monitor everything you can, all the time, with headphones. Monitor (spot check) the camera's headphone output frequently. Listen to tape playback, especially if/when you're unsure.

Bill,I understand 24 bits, but I don't understand how 192kHz helps you with this (assuming you'll eventually deliver to, say, 48 kHz). Can you please explain what you meant by this?

Thanks,
Martin

Somebody down the chain might be turning it down. Or your shows might be compressed by a Master Limiter when broadcast. It's the Master Controllers job to make sure that your program is no louder than anyone else's. So it's pointless to try to push your levels hotter than others. If you send them a hot master they're going to turn it down. I'd rather send out something I know is not going to be limited when broadcast. You never know how they have their limiters set.

I agree with you about acquisition. Especially since most acquisition formats are 16-bit. Got to maximize that dynamic range. I don't like having a mixer in the chain when I'm shooting by myself, so I tend to be a little conservative with levels. Noise can be cleaned up (especially with dialogue), clipping can't. When I'm recording an interview, I'll record one channel at 3 to 5 db less than another, so if the loud channel clips, I have the quieter one.

Now again, we're getting into a different realm. We're pushing peaks based on dynamic range, not pushing the final output. My goal is, and always has been, to shoot for the most wide dynamic I can get away with. Both my partner and I were scoffed at for years for pushing dynamic range, and then one day Walter Murch wrote about how he wrings every tiny bit of amplitude from every relative portion, and bam....we were no longer scoffed at.
Master Controller or not, PBS is by FAR the most difficult broadcast organization that we may submit to. They don't allow a submission to get as far as a master controller turning levels; they simply reject the submission whether it's for audio or video challenges.
My experience has been that there is no give and take with PBS. YMMV, but I'd be surprised.
The Food Channel and Court TV are just the opposite. Give em' anything and they'll ship it downstream.

I see what you mean. I suppose as long as your averaged level is around -20db, your peaks are probably less of a red flag. I can see pushing to -6db for peaks.

I've only submitted to PBS, so I didn't want to push things too much. Better to have it go through with 4 db less of headroom than to be rejected is my philosophy. Plus, it looks bad for client work if your master gets rejected. But if it was completely self-produced and there's no significant cost to making a new master, I can see pushing the limits.

Some might say 192kHz sounds more accurate or richer than 48khz. But it has absolutely no effect on dynamic range. If you're recording voice, I would seriously question any perceived difference between 192khz and 48khz. Music, maybe there might be some advantage to capturing at a higher frequency then processing (EQing, Compressing) at a higher frequency and only at the end bringing it down to 48khz. Notice I said maybe, I'm not convinced there would be any audible difference and certainly there are more pressing things to worry about in your production.

you're quite right, Brett....more pressing things all over the place. But, if the capability is there, why not? I suppose that's more the question.

Disk space, CPU processing overhead, workflow issues. If those things aren't an issue than absolutely go ahead and record 192khz.

Not much of a problem with any of these. Biggest one is to dither back down. And even that is NBD.

Don't get me wrong, I can see advantages of 192 kHz over 48 kHz. But if your final mix/product is going to have 48 kHz audio, then I am wondering if there is any advantage of going higher than that earlier in the recording chain. I have the equipment to to that, but never bothered to because I didn't see the point (and, yes, disk space, cpu overhead etc. are real considerations). So I am curious if anyone can convince me otherwise.

Unlike, of course, the choice of 24 bit vs. 16 bit acquisition, where it's a no brainer to go with 24 bits if you can.

- Martin

Not unless or until the playback systems can reproduce the bandwidth recorded at 192 kHz.

There is an argument to be made about Nyquist filters at 19 kHz (for 44.1 and 48 kHz) and the in-band damage they can cause. If you want an argument about it, try rec.audio.opinion.

You won't get one from me.

Regards,

Ty Ford

levels
 

The manual on my XH-A1 says the audio levels on the camera can look fine but the audio can still be overmodulated.

Larry, et al.

You can send line level to a mic level input on most any camera and set a "proper" level, but the audio will still be distorted. (it won't soundd good, but the level will look right.) That info is in my book, btw.

All video cmaeras I've run into share that mystery. I mean, why not have them set up so if you send line level and the input is set to mic, the meter goes WAY off scale?

Check to make sure your gozouta and gozinta are set porperly.

Regards,

Ty Ford