This may have been answered in another thread, I couldn't find it....

I recorded an interview with my JVC HD100UA. We put a wireless mic on the interviewee, and plugged it into channel 1. We also set up a shot gun mic and plugged it into channel 2. I haven't touched any of the audio settings on the camera.

We had the volume meter on the camera, and we were using DVrack.

We cranked the volume on the camera all the way. That made it loud, but the meters on the camera and dvrack did not show any clipping.

When I play it back on my computer ( I bought decent speakers for it) it sounds fine. Maybe a few bits where the person gets a little emphatic it sounds a bit too loud for one syllable, but not bad. I play it back on Adobe PP2.0, and the volume meters in the NLE don't show clipping.

I burned it to a dvd, and someone watched it on their tv. They said they heard clipping.

I have to do a lot of research, of course. I'm wondering whether I really have clipping going on, or whether I just need to lower the gain on those bits in post? Do I need to change settings on my camera to get an accurate read of the volume?

Dennis,

you didn't say much about what steps you went through between the recording and burning it to a DVD. It is certainly possible to introduce audio gain in the sound processing there that could lead to clipping later.

I would suggest that while you prepare your audio for the DVD, look for peaks and make sure that there is nothing at all above -6dB in any audio channel. Some may suggest to go even lower, to -10dB. Then create your DVD.

Apart from clipping, this brings the audio level of your DVD in line with professional DVDs such as those of feature films. If your audio goes all the way up to 0dB - with our without any clipping - it'll sound much louder than what viewers are used to. Limit your peaks to -6dB, and viewers won't need to rush to the volume control when they play your DVD.

- Martin

Actually, as long as your peaks don't exceed 0 dB, there should be no clipping. I like to run with about -.1dB headroom. You have the option of putting a limiter on your audio track, either in the HD110, or in post processing. If you're already experienceing some minor clipping, you can probably fix it with a small amount of compression or limiting.

Another thing to check is the volume on the wireless reciever and/or transmitter itself. I've been an audio guy for almost 18 years. If your gain is set too high on your wireless receiver and it is clipping, it won't necessarily show in your camera monitor. If you're "throwing a distorted signal" from the beginning, it will be distorted until the end. With the HD-110, I have found that most recordings do pretty well with the audio set to "Auto", unless you're in an extremely controlled environment.

They said they heard clipping.

Who did, who heard it?

I think you have to be careful who and what people say about your work, the problem could be at their end.

Alot of good info here.I'll try to add a bit.Kits comment seems like it could be your situation.If the mic output is correct for the transmitter , the receiver level should be somewhere mid position to keep it from distortion before it hits your recording device control.Otherwise you may be cranking the level up at one end and down at the other.Kinda like driving your car with your foot to the floor on the gas and riding the brake to control speed.I try to center all controls when starting, then fine adjust from there.
Another thing I do is after mixing the audio , using decent monitors, I run it through a set of cheap speakers to make sure I haven't any distortion from excess freq. or dymacics that the good speakers handle but cheapies don't.
Some TV's have pretty crappy speakers/preamp.
If your building the video for TV broadcast or otherwise, you may want to roll off the low freq. at a minimum 20hz up to 80hz sometimes.
3 other things. Check it yourself,check it yourself,check it yourself.

I am also editing a documentary which will be finished to DVD.

My audio is OK from origin, being around -12db.

I am not sure how to finish my audio for DVD. There are interviews and bands playing. The interviews have the audio at a constant level, but the bands have a lot of peak variations.

I am not sure at how much dbs my final audio should be. I am also not sure if I should apply a compressor or a limiter in Soundtrack Pro so that all my audio is around the same level.

I think I should leave my audio levels at around -5 db, and limit it at -1db. But I really dont know if this is OK or not.

I appreciate suggestions.

Hi Dennis,

There are multiple possibilities.

1) They *think* they hear clipping. Solution: listen to the same tv set in their presence, and have them show you exactly where they hear clipping. Also, make sure you are using the same terminology. While you might know what clipping technically is (= audio levels go above 0dBfs and distort because they are flattened), they might have picked up the word somewhere and use it in the improper sense (don't laugh, I've seen this over and over...).

2) The peak meter on your NLE screen is never accurate. It's a fact of life. It is slower than the actual transients of your audio. You might consider investing in a good PPM meter (RTW or DK Audio e.g.) Now you will now why these are so expensive *grin*.
Try a brickwall limiter on the master fader, set at the level you don't want to exceed (e.g. -10 dBfs). Ensure your gain staging is correct.

3) You hear the clipping on their system too, so there is something wrong on your end. If you have a buddy who is a recording engineer professional, ask him to check out your rig. It could be very simple things.

Hope this helps a bit

Cheers
Arthur

Only one thing I can add here is that you should always monitor your audio while recording it with a good set of cans.

A meter will show you the level of your audio, but not the quality. If there is distortion due to improper gain staging before the volume knob, you'll be able to dial the knob down to where the meters look fine, but your audio will still be clipped/distorted from earlier in the signal chain.

-gb-

Alot of good info here.I'll try to add a bit.Kits comment seems like it could be your situation.If the mic output is correct for the transmitter , the receiver level should be somewhere mid position to keep it from distortion before it hits your recording device control.Otherwise you may be cranking the level up at one end and down at the other.Kinda like driving your car with your foot to the floor on the gas and riding the brake to control speed.I try to center all controls when starting, then fine adjust from there.
Another thing I do is after mixing the audio , using decent monitors, I run it through a set of cheap speakers to make sure I haven't any distortion from excess freq. or dymacics that the good speakers handle but cheapies don't.
Some TV's have pretty crappy speakers/preamp.

Good advice, I believe in the 10:30am to 3:00pm rule, if 12:00pm is the middle point for audio, anytime the audio levels have to be set lower than 10:30am or higher than 3:00pm there might be an audio mismatch somewhere along the way.

Good advice, I believe in the 10:30am to 3:00pm rule, if 12:00pm is the middle point for audio, anytime the audio levels have to be set lower than 10:30am or higher than 3:00pm there might be an audio mismatch somewhere along the way.

Whats that about 10:30am to 3:00pm???

Beats me, and while we're at it, what do you mean with an "audio mismatch"?

Cheers
Arthur

Beats me, and while we're at it, what do you mean with an "audio mismatch"?

Cheers
Arthur

If you imagine the mark on the knob of the level control as the hand on a clock, it's pointing at about 7o'clock when it's all the way off and about 5 o'clock when it's full up. The optimum setting is usually in the zone between 1/3 and 3/4 of the way up, or between 10:30 and 3:00 by the clock analogy. Audio mismatch would mean the output of one stage is too high or too low for the input of the next stage in line.

Only one thing I can add here is that you should always monitor your audio while recording it with a good set of cans.

A meter will show you the level of your audio, but not the quality. If there is distortion due to improper gain staging before the volume knob, you'll be able to dial the knob down to where the meters look fine, but your audio will still be clipped/distorted from earlier in the signal chain.

-gb-

Excellent point.

I've noticed over the years that if my audio dials have to be set ultra low or ultra high to get an acceptable reading, the odds are that the audio levels are not properly matched throughout the audio set-up. I call that the 10:30 to 3:00 rule (in which the audio dials have a full range of 7 o'clock to 5 o'clock with 12:00 o'clock being the center defaul).

Ok, so what you mean is gain staging mismatch. Now I get it, thanks!

Still, I don't subscribe to that theory. For correct gain staging, you would need to do a calibration of your system first, no? One amplifier's 10 o' clock might be too little, while on another one it might already cause distortion...

So most of you guys use consumer stereo amps? Sorry, if this sounds snobby, really not my intention, but I come from the studio world, so I am not familiar with what the average videographer uses.

Cheers
Arthur

Ok, so what you mean is gain staging mismatch. Now I get it, thanks!

Still, I don't subscribe to that theory. For correct gain staging, you would need to do a calibration of your system first, no? One amplifier's 10 o' clock might be too little, while on another one it might already cause distortion...

So most of you guys use consumer stereo amps? Sorry, if this sounds snobby, really not my intention, but I come from the studio world, so I am not familiar with what the average videographer uses.

Cheers
Arthur

I'm refering to audio clipping when one is doing camera acquisition work. In my studio my amp is actually set to 2 out of 10 for mono, and 3 out of 10 for stereo, so clearly the 10:30 to 3:00 rule does not apply in an edit room environment for purposes of speaker playback settings.

If the CAMERA default position for the audio dial is 12:00, default meaning the middle point, I've found the 10:30-3:00 rule works very well. The rule can possibly be stretched to 10:00 rather than 10:30. Once the sweep spot of 10:30-3:00 is missed, either loud sounds can distort (when the setting is over 3:00), or lower sounds can become inaudible (when the setting is lower than 10:00)

I think there's a major language problem happening here.

"Clipping" is a term used in ANALOG audio recording. There really is no similar phenomenon in digital audio recording. In analog, when you pushed a signal over the "optimal range" and into the red zone on a VU meter, the result was something like compression - up to the point that "s" sounds got splashy and things weren't pleasing to hear.

In the digital realm, if you exceed the input level to the point where you run out of bit depth to map the audio into, your audio CATESTROPHICALLY fails. To the point where you can't understand what someone's saying.

So, if you have clean audio at any stage after the recording, there's no actual audio "clipping" happening.

If the "client" is hearing something that sounds like old analog clipping, then it's probably either over-compression making the sibilent sounds splashy - or perhaps they have blown tweeters on their monitors - or it's one of other things that can "sound" like old style "clipping" - but again, if you're recording digitally, its certainly not "CLIPPING" in any traditional sense.

(just trying to keep the language straight in this analog to digital transition world!)

Hope this helps.

Bill, as far as I know, clipping is also used in the digital realm, i.e. if the amplitude of the waveform tries to go over 0 dBfs and thus the waveform is 'clipped' or chopped off, since indeed the amplitude cannot go over 0 dBfs. So you get a trapezoidal waveform instead of the normal sinusoidal waveform, hence the term clipped.

BTW, bit depth has nothing to do with the amplitude or strength of your signal, but with the resolution of the sampling process.

Just thought I should add this.

Cheers
Arthur

"Clipping" is, and will continue to be used in the digital realm. Whether you're talking about truncating, brickwalling, squaring, digital clipping, whatever....once the signal reaches 111111111111111111111 (sixteen ones), no greater level may be achieved and it's cut off, clipped, truncated, smashed, whatever term you want to use may be.
Many, if not most, still refer to this as "clipping" or truncating.

I don't know that it matters what word is used, so long as we know when we're discussing digital vs analog.
Sony, Apple, Digidesign, Avid, Cakewalk, Echo, M-Audio, etc all refer to the term "clipping" in their digital product user guides.

However, it is true that the nomenclature from the analog to dig world can get pretty weird when crossing terms.

Ok, so what you mean is gain staging mismatch. Now I get it, thanks!

Still, I don't subscribe to that theory. For correct gain staging, you would need to do a calibration of your system first, no? One amplifier's 10 o' clock might be too little, while on another one it might already cause distortion...

So most of you guys use consumer stereo amps? Sorry, if this sounds snobby, really not my intention, but I come from the studio world, so I am not familiar with what the average videographer uses.

Cheers
Arthur

The "10-3" rule is more of a guideline than a way of setting levels, as least as I interpret it. If you need to turn the recording level control down under about 10 o'clock to prevent the meter from going too high you're probably sending too hot a signal and are in danger of clipping at the input on peaks while if you have to turn it up past 3 to get an adequate recording level on the meters you're presenting too low a signal at the inputs and are likely to suffer a reduced S/N ratio. Like Goldilocks and her porridge, we want to get the signal level presented to the camera's audio inputs to be just right, high enough for good S/N but low enough to avoid the risk of clipping.

I don't know how many use consumer amps but I doubt if many professional videographers do, certainly not many who are shooting for broadcast or theatrical distribution. Very similar considerations apply in a professional video studio as do in a music studio and while there are differences in the specs of some of the tools, such as the optimal monitor crossover points for dialog editing versus for music mastering, for example, the level of quality required for professional results in just as demanding.

OK, now I am really confused, feel like on the tower of babylon *grin*

So you are talking about the input levels (gain) and others are talking about volume levels (suggesting to me the output levels on the amp towards the speakers) Hence my reference to the consumer speakers.

If we're talking input levels, then I gotcha.

As a guideline, when recording digital, I mostly record around -10dBfs to -6 dBfs at 24bit bit depth and around -8 dBfs to -3 dBfs at 16 bit bit depth. So my 16 bit signals will be hotter, since I am more prone to loose detail at 16 bit with poorer s/n ratio (-98 dB) than at 24 bit (-144 dB).

Still, when recording audio on location, I make sure I got a good mixer with clean preamps that have a good headroom and as accurate metering as possible.
Like I stated before, I come from the audio world and video is new to me, so I joined to learn more about my new Canon XH A1, and while at it, I would like to share some of the knowledge I gathered from being 20 years professional in the music recording and producing business.

Cheers and have a smashing weekend y'all (hey, I got a lot of texan friends, and they taught me well *grin*)


Arthur

You didn't say what type of wireless. Folks also mentioning limiting. I got burned once using a wireless transmitter to send a signal to my receiver bodypack which I had plugged ito my Z-1. I watched the signal on the transmitter off the board and everything looked good. The problem I had was I had the auto limiter selected to the on position and whenever the signal got too hot, the signal got limited. On playback it sounded like a lot of pops when these spot happened.

If I would have done what Greg suggests byt listening to the signal with phones, I would have heard the problem and could have turned off the limiter.

I actually blamed it on the sound guys board limiter. When I got home, I had three body packs with no batteries in them and the 4th one that did (the one I used) had the limiter on.

Better to have it off and manage your signal.

Jeff

Geez, another practical Q&A thread devolving into theory - how can I resist jumping in? (apparently not)

Anyway, to begin with, I was a bit confused with the original post:

"We cranked the volume on the camera all the way. That made it loud, but the meters on the camera and dvrack did not show any clipping."

Something isn't quite right here, is it? Aside from the camera's amplifier certainly not operating in the linear "Class A" range, I didn't think adjusting the camera amp would affect the DV Rack audio via the firewire port. My laptop is in for repair this week so I can't test this out. Anyway, I seriously doubt the JVC camera mic amp will perform well running maximum gain. We don't know about Dennis' transceivers and cabling so it's unclear if there's a possible impedence mismatch, bad cable/connector, etc. Who knows, it might even be a mic problem - assumedly the listener who complained of distortion was listening to BOTH channels, not knowing the left-right used different mics (twice the opportunity for cockpit and equipment errors). Or perhaps Dennis only used on channel in Premiere and converted the best channel back to stereo in post - we can only guess. And who knows what the DVD sounds like in player/TV.

I also am suspicious along the lines mentioned by Kit and some others. For instance, on my Sennheiser G2s if I'm not careful I could easily overmodulate the transmitter by forgetting to adjust the sensitivity (say going from a dynamic to condensor mic) - the receiver setting could be be spot-on and the camera, DV Rack, MP3/WAV recorders all are happy with their VU meters seemingly perfect (garbage in - garbage out).

Anyway, aside from the troublesome JVC amp cranked all the way up, I'd put on a good set of headphones and carefully listen to both the right and left channels of the PP2 timeline sequence. And if that sounds peach, burn another DVD and crank it up.

Finally, be extra careful with the mix-n-match scenarios. Using a lav on one channel and shotgun on the other is great insurance, but if you're not certain both channels are pure, either could come up and bite you if something goes amiss (and in fact, may have here). Your client won't be happy if you didn't listen carefully to minute 33 and the wireless receiver gets a noise burst, the mic cable is loose, etc. In audio engineering, the more open mics the more opportunities for gotcha's. Gad, these theoretic discussions are fun! Okay, back to cutting In/Out points...

Good luck, Michael

PS - Jeff Mack also gets my thumbs up with the possibility of a limiter compensation problem (ugg I hope that wasn't it)

....

If we're talking input levels, then I gotcha.

As a guideline, when recording digital, I mostly record around -10dBfs to -6 dBfs at 24bit bit depth and around -8 dBfs to -3 dBfs at 16 bit bit depth. So my 16 bit signals will be hotter, since I am more prone to loose detail at 16 bit with poorer s/n ratio (-98 dB) than at 24 bit (-144 dB).

Still, when recording audio on location, I make sure I got a good mixer with clean preamps that have a good headroom and as accurate metering as possible.
Like I stated before, I come from the audio world and video is new to me, so I joined to learn more about my new Canon XH A1, and while at it, I would like to share some of the knowledge I gathered from being 20 years professional in the music recording and producing business.

...

What sort of material are you recording and on what sort of equipment when you see those meter readings, also what kind of metering are you using? I ask because they seem a bit hot for in-camera recording levels for DV. Standard practice for gain-staging is to adjust the camera's recording levels so that with your mixer's output set to unity gain, a 0VU reference tone from the mixer reads -20dBFS (I see you're in Europe and the EBU calls for -18dBFS) on the camera's meters. The mixer output level is left to unity gain, the camera's level controls aren't touched after that, and level adjustments during the shot are made with the mixer's input faders. With the gain staging setup that way, source material adjusted to average around 0VU on the mixer's meters will show average levels around -20dBFS with peaks hitting around -10 to -8 dBFS on the camera's meters. But when you do your final master for release, nothing should ever peak over -10dBFS.

But when you do your final master for release, nothing should ever peak over -10dBFS.

Steve, is that what you really meant to say? Peaks never beyond -10dBFS? That level would be rejected by most of the bureaus that we service.

OK, I'll happily acknowledge that the term is used in all sorts of manuals - and is pretty popular. It's just that I still have trouble feeling comfortable with it relative to my years in radio and analog recording.

The point I was trying to make (ham handedly and obviously not well!) was that clipped audio in an analog relm left you with a usable signal. Heck, lots of analog circuit designers touted "soft clipping" as a desirable feature to maximize S/N in their machines.

Re-purposed into the digital relm, people are getting the term confused, as evidenced by the OP.

"Clipped" audio (or whatever other term you want to employ) in the digital relm is pretty easy so spot as it's totally unintelligible.

The fact that the OP described his original source as "clipped" - but was able to play it back and hear it correctly off the camera master - indicated to me that it could never have actually been "clipped" in the first place - and led me to think he was confusing "clipping" in the analog sense with it in the digital sense.

Of course, maybe I just read things wrong. (Wouldn't be the first time, nor the last!)

But the conversion from analog to digital terminology seems to be messing up a lot of folks like the OP who are trying their best to learn both the language and the techniques of good recording.

For what it's worth.

I know what you meant, Bill...didn't mean to sound like I was picking on you. I think we all have our pet-peeve words or terms. "Over modulate" when it's not transmission related is one of mine, and then there are the odd-ball descriptives.
Then again...there are those here that wouldn't know a 7.5 ips from a 15ips either... :-) For many, the analog world is an alien concept

The "10-3" rule is more of a guideline than a way of setting levels, as least as I interpret it. If you need to turn the recording level control down under about 10 o'clock to prevent the meter from going too high you're probably sending too hot a signal and are in danger of clipping at the input on peaks while if you have to turn it up past 3 to get an adequate recording level on the meters you're presenting too low a signal at the inputs and are likely to suffer a reduced S/N ratio. Like Goldilocks and her porridge, we want to get the signal level presented to the camera's audio inputs to be just right, high enough for good S/N but low enough to avoid the risk of clipping.


Thanks for explaining better than I said it.

Steve, is that what you really meant to say? Peaks never beyond -10dBFS? That level would be rejected by most of the bureaus that we service.

Average program material at -20dBFS, absolute 'never exceed' peak of -10 is what I was given to understand to be the normal specs for both broadcast and theatrical release, although theatrical might push the max a bit higher while keeping the same average. Are you saying that's too high or too low?

For broadcast, we're hitting peaks @-6dB, for replication/non broadcast, peaks are hitting -2dB, and as time goes on and oversampling becomes less common, we'll start pushing to fractions.
Our local bureau won't take anything less than -10dB at peak.
If they're stems to be mixed later, they go out at -1dB so that they're huge, and will be mixed to lower levels later. We keep things hot in house to preserve res, and then allow it to go down when it's ready for mix.
Kind of a hold over from the days of 1630, I guess.

Ok, so what you mean is gain staging mismatch. Now I get it, thanks!
<clip>
So most of you guys use consumer stereo amps? Sorry, if this sounds snobby, really not my intention, but I come from the studio world, so I am not familiar with what the average videographer uses.
Cheers
Arthur

Arthur,

So in your studio world, where do you/they set the sliders on the mixers?(let's assume the audio engineer could also adjust a pre-amp)

A. Above unity
B. Below unity
C. Dead-on unity gain whever possible

Michael

Hi Micheal,

Not sure I understand what you mean. Maybe I'll try to explain how I record on location, so you get a picture - pardon the pun - on how I do stuff.
If you mean when recording the audio on location, I never record the audio on camera, but seperate (used to use a Nagra, now I use a portable PT rig - macbook pro and 002, connected to a Apogee stereo pre/converter). Maybe not the most efficient to haul around, but it is in my comfort zone (it is what I know to use) and it works *grin*

As far as gain staging on location, the PT is at unity, the preamp gain is set whereever needed, the Apogee converter is calibrated at 0VU = -18 dBFS.
The preamp gain on the Apogee is set to peak anywhere between -12 and -6. MEtering is done with a very good plugin on ProTools from Roger Nichols Digital called Inspector XL. There I use two meters, one VU and one PPM.
So in fact, I record pretty hot for broadcast terms. The final level on the deliverables is done in post. I will aim for an average of -18 dBFS (EBU) and the stations I deliver to allow peaks of -10, if not too many.

In the studio (audio production) the signal chain is as follows (at least where I work - btw, my work place can be seen on www.icpstudios.com).

- Mic > Preamp > compressor > eq > ProTools HD

At mixdown : PT HD > console > Master deck (most of the time a Studer 2 track)

This is pretty simplified though, in reality, depending on the audio source, I will use different things in the chaiin if the source asks for it.

So in audio, I will set the preamp to average at -12 to -10 and peak at -6 to -4. Since I record direct to disk, PT HD is at unity and the gain is set on the preamp.
The final master that goes to the mastering house peaks at -3 to leave some headroom for the mastering engineer.

As far as calibration of the converters is concerned, it depends on the console I am using. In the SSL room (a SSL 4000G) the converters are set to -14 (they are modded, since normally the 192 i/o doesn't go higher than -15 - but we found the signal chain on an SSL sounds better when the converter is set at -14).
On the Neve 88R in our studio B, the converters are set to -16, again, we found that to be the optimum soundwise for the Neve.
On the vintage EMI Neve in room C, the converters are calibrated to -18.

Sorry if this post is a bit long, but I thought I give you a bit of background info, so you might better understand where I come from, and since English is still a foreign language to me, I want to avoid misunderstandings.
I found it is easy on a BBS to be misunderstood, since you cannot hear my voice or see my facial expression when you read my ramblings *grin*.


Have a great weekend

Cheers
Arthur

For broadcast, we're hitting peaks @-6dB, for replication/non broadcast, peaks are hitting -2dB, and as time goes on and oversampling becomes less common, we'll start pushing to fractions.
Our local bureau won't take anything less than -10dB at peak.
If they're stems to be mixed later, they go out at -1dB so that they're huge, and will be mixed to lower levels later. We keep things hot in house to preserve res, and then allow it to go down when it's ready for mix.
Kind of a hold over from the days of 1630, I guess.

Would never debate with your level of experience, Spot, but I thought I understood it yet now I'm getting confused so if you could clarify I'd appreciate it. Where is reference tone/average program material & dialog running when your peaks are that high? I understood that it's common for some of the analog stages in the broadcast chain such as conventional videotape as well as traditional theatrical houses only have about 8-10dB headroom and a master set up with a -20dBFS=0VU reference-level that has an average to peak ratio exceeding 10dB risks driving something down the line into clipping when it finally airs, although digital video, HD, DVD, and some theatrical material can sustain up to a 20dB ratio. That's kind of born out by Arthur's post above where he writes "I will aim for an average of -18 dBFS (EBU) and the stations I deliver to allow peaks of -10, if not too many." Those figures were also quoted in a thread on delivery specs in another venue a few weeks ago where Jay Rose cites -20dBFS/-10dBFS in the network contract requirements he usually works under and it's also echoed in my reference material from Holman ("Sound for Digital Video"). So what accounts for the discrepencies in the numbers - are we talking different meter calibrations (dBVU/dBU versus dBFS), peak versus averaging metering, different delivery destinations, or what?

Spot,
just a minor correction. 0dBFS is actually 0, not 2^16 or 2^24. In the digital realm the quieter the sample value the bigger the number. That makes bit depth conversion very simple and explains why you cannot exceed 0dBFS. Meters can sometimes display over 0dBFS due to successive values of 0. You can as you mentioned also get clipping due to intersample interpolation pushing an interpolated value over the top.

Steve,
the confussion relates back to the old days of analogue and VU meters, typically the meters were set for 20dB headroom but they're average reading meters whereas today typically digital audio is measured with peak reading meters. The problem with peak reading meters is they tell you nothing about how loud something sounds. The problem with VU meters and modern digital audio is they can be thumping the stops when nothing is actually clipping. If your peak reading meters on digital audio are only hitting -10 or -20 you're wasting a lot of headroom.

Back to the original problem. Even though the meters on the camera were not showing clipping mics can be overloaded and wireless transmitters can be overloaded too. I've come accross the latter quite a bit, it pays to setup a wireless mic transmitter carefully. One trap is you can set it up for reasonable levesl at the transmitter with the normal speech levels and then the guy stand if front of a foldback monitor. Nothing you can do at the camera end will help, in fact turning it down can make it even more impossible to fix in post.

To see what's happened look at the waveforms. Digital clipping if things stay digital will always have a flat top, a nice straight line. Analogue clipping has some rounding due to saturation. Well unless it's really extreme clipping but in general you can easily see the difference.

And lets not overlook that sometimes what you're recording can be clipped before it hits your mics.

........

We had the volume meter on the camera, and we were using DVrack.

What's DVrack? OK I googled. Does the system have a different adjustment for RMS (average) and peak? If so you could have been reading rms and clipping peaks.

We cranked the volume on the camera all the way. That made it loud, but the meters on the camera and dvrack did not show any clipping.

You cranked the volume all the way? Do you mean the record levels?

When I play it back on my computer ( I bought decent speakers for it) it sounds fine. Maybe a few bits where the person gets a little emphatic it sounds a bit too loud for one syllable, but not bad. I play it back on Adobe PP2.0, and the volume meters in the NLE don't show clipping.

Playback from what?


I burned it to a dvd, and someone watched it on their tv. They said they heard clipping.


Can you hear clipping when you play the DVD on your TV?

[QUOTE=Dennis Stevens]
I have to do a lot of research, of course. I'm wondering whether I really have clipping going on, or whether I just need to lower the gain on those bits in post? Do I need to change settings on my camera to get an accurate read of the volume?[/QUOT

Depends on where the clipping ocurred. Listen to the DVD your client has a problem with on your own DVD system (not just the computer) and see if you hear it.

You may not be as sensitive to clipping as your client. If the clipping occurred at the wireless mic during excited moments, there's not a lot you can do. Yoc can clip he wireless and not see overly loud levels downstream, but if you zoom into the waveforms you see the flat topping.

I've had this happen to me. Set the levels and then the person on camera unesxpectedly gets a little nutty and crashes the wireless transmitter. The take away message is to consider lowering the input sensitivity of the wireless transmitter. You'll have more system noise, but more headroom.

Did you end up using the lav track? Compare the wireless and shotgun tracks and see if the shotgun does better.

Another page in the "Good audio is not trivial" case book. :)

Regards,

Ty Ford

Would never debate with your level of experience, Spot, but I thought I understood it yet now I'm getting confused so if you could clarify I'd appreciate it. Where is reference tone/average program material & dialog running when your peaks are that high? I understood that it's common for some of the analog stages in the broadcast chain such as conventional videotape as well as traditional theatrical houses only have about 8-10dB headroom and a master set up with a -20dBFS=0VU reference-level that has an average to peak ratio exceeding 10dB risks driving something down the line into clipping when it finally airs, although digital video, HD, DVD, and some theatrical material can sustain up to a 20dB ratio. That's kind of born out by Arthur's post above where he writes "I will aim for an average of -18 dBFS (EBU) and the stations I deliver to allow peaks of -10, if not too many." Those figures were also quoted in a thread on delivery specs in another venue a few weeks ago where Jay Rose cites -20dBFS/-10dBFS in the network contract requirements he usually works under and it's also echoed in my reference material from Holman ("Sound for Digital Video"). So what accounts for the discrepencies in the numbers - are we talking different meter calibrations (dBVU/dBU versus dBFS), peak versus averaging metering, different delivery destinations, or what?

Ref tone/print tone is always -20dBFS, FWIW, I've written about the asurl=http://www.creativemac.com/2003/02_feb/tutorials/analog_dv_levels.htm] spec[/url] and dialog averages -10dB. Our pieces tend to be much more than dialog, however.
As far as meters, everything is measured on Dorrough PPM meters, which are pretty standard on the broadcast side.

Well, this is a great discussion.A lot of excellent info.
I liked my analogy
" Kinda like driving your car with your foot to the floor on the gas and riding the brake to control speed."
Dennis, did you determine what your situation problem is(was)?

I'm no expert on much of anything, but I'm pretty sure that the clipping involved in the recording had nothing to do with the digital world. I suspect an analog origin or, at worst, origination from the analog portion of the analog to digital conversion somewhere.

I do not know anything about the camera involved, but I'll bet that the camera's meter was showing the DIGITAL RECORDING signal level. Since the analog inputs were maxed out, I suspect that the "clipping" was probably just transistor preamp distortion at the front end. As we all know, tubes distort gently over a wide range whereas transistors or the circuits in ICs (also, of course, transistors) clip catastrophically ONLY when the headroom of the circuit is exhausted. If the meters were showing the RECORDING level rather than the output of the preamps, then it would be very easy to max the transients on the preamps (which were set to their highest level) without ever knowing it.

I've seen this very same thing happen in my own audio studio. I heard clipping all over the place, but I NEVER saw a single red-line event. It turned out the clipping was resulting from me overdriving the analog circuitry of my Lucid 9624AD converter (which did not have any associated metering) on quick transients from drum hits. The digital meters showed fine levels, but that's because they were only watching the digital realm. Except by listening, there was no way for me to "see" the clipping that was taking place in the analog circuits of the converter.

Anyway, that's how I see it.

Stephen

I know what you meant, Bill...didn't mean to sound like I was picking on you. I think we all have our pet-peeve words or terms. "Over modulate" when it's not transmission related is one of mine, and then there are the odd-ball descriptives.
Then again...there are those here that wouldn't know a 7.5 ips from a 15ips either... :-) For many, the analog world is an alien concept

Just because I know you'll appreciate this...

Several years ago I shared a cab to the airport from NAB with a couple of 20 something audio software engineers from Silicon Valley.

Don't know how it came up but I mentioned "cutting tape" - their eyes got big as saucers, their jaws dropped, and one of them said something like "you actually used to physically cut tape???!!"

I wanted to tell them that the first TV station I hung out in still had a splicing block for QUAD videotape. It was obsolete even back then, but how quickly the stuff of editing passes - leaving only the skills behind.

Nice reminder of what I teach over and over in my training products, - "editing" is doesn't really take place in oxide or silicone - it takes place a couple of inches behind an editors eyes.

FWIW

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....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?...
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.