Simon wrote:

Hi Bob,

Simon Handley

Dear Simon:

Those kinds of correction systems still have to be set up by knowledgeable acousticians or they can easily over or under-correct. You have to be able to interpret the results and use some portion of them intelligently. So in the end it’s a mixed bag. I would suggest that you find a very good acoustician first and describe to him your desire to use one of these systems to “correct” your room (note the quotation marks, as any acoustician would tend to laugh at “correct” not in quotation marks. The best solution is a combination of acoustical (absorption/diffusion/room construction/proper choice of monitors) followed by possibly a touchup using one of these automatic systems. If called upon to do the whole job they can literally screw up a good day :-(.

So, bottom line: Use with caution and knowledge, not a panacea, possibly a help, sometimes only a bandaid.

Hope this helps,

Bob

Roberto Estrella wrote:

 I have a doubt.

I have an interface with a 24bit A/D convert
Currently, some DAWs can work at 32 bit float, 32 bit fixed or 64 bit float.

If I use a 24bit A/D convert, Is there a benefit or detrimental effect when I record at 32 bit float or higher ?

Hi, Roberto. My two books should help you answer that question in more detail. I’m adding your question and my answer to our FAQ. 

There is no advantage to recording 32 bit float directly from your ADC. It just takes up more file space. But there is an advantage later on to calculating and storing or using the 32 bit float capability of your DAW when you equalize or otherwise process. In other words, even if you originally captured to a 24 bit file, the next stage calculation will grow to a longer wordlength. You can use the 24 bit original file for later calculations without any problems. 

Hope this helps,

Bob

Fabian Köhler Posted the following question:

In your opinion,
Does it make sense to record with 2 ambient mics/ main mics when it comes to sessions with multiple overdubs? I tried so with a single omni-microfone of good quality (Neuman 183) in what i considder a room with ‘good accoustics’. Results where good and open sounds on single instruments (summing with the close up mics), but when it came to mix the signals, summing the ambient-mic traces resulted in a confused, ‘spacy’ sound- so i had to replace natural ambience with a convulsion-reverb plugin- but like to use natural ambience of the room i prepared as registration studio next time, as it sounds much better to me (at least on single instuments).
For you, does it make sense- with the same ambience microfone position? Does the overall ‘openness’ of a mix benefit of this summed overdub-ambience-traces (which, in every case, summ even background noises etc.)?
I thank you for your attencion,

Fabian Köhler

Dear Fabian: 

Good question.

I would definitely prefer a pair of ambience mikes, not  a single mono ambience mike. Did you test the overdubbing situation with the stereo pair of ambience mikes or just the mono mike? If you only did the mono mike you probably did not get great sounding results.

I assume in the mix you have full control over the direct and ambient mix level for each overdub.

Still, depending on the early reflection and reverberant character of the room, you may run into cumulative resonances when you add the ambience mikes in for each overdub. There is no yes or no answer to your question… you have to listen to the mix and decide what the consequences of mixing in these multiple ambient mikes could be. It could multiply the bad aspects (resonances) of the room. That sound fine with one recording, but not so good with multiple overdubs. 

So, yes, it is possible that the artificial approach may work better, because you can tune the reverb to the specific mix you are making with the specific instruments and create a perfectly complementary artificial room.

But there also can be problems with a convolution reverb as it does not create depth as well as real ambience mikes or the best algorithmic verbs. For this kind of thing artificially done, I prefer a combination of the hardware Bricasti Reverb with UAD’s Ocean Way. Or, TC’s VSS4. And/or sometimes, Leapwing Stage One. You’re making esthetic and musical decisions, so there is no guaranteed single formula.

Best wishes,

Bob

Martin wrote:

Dear Bob,

I am a recording/mix engineer.

I came across a section where you discuss digital vs. analog mixing where you speak about summing mixer vs. any analog processor and that your ears could tell that all is really needed was one single processor to give that “analog” sound , etc.

I believe your ears.

I just have a problem reconciling the following:

( I have mastering EQ’s, Comp, etc.)

For a given professional analog hardware generating a song say, tube guitars, optical processors, VCA processors, prof. mics, etc., mixed in the box. 

I know I am speaking in totally subjective terms, never the less coming back out (2-track) to the analog domain and running THAT digital mix though a compressor and viola its sound “warmer” or “analog” or whatever (like you say) you want to call it.

I ask what was wrong with all the analog hardware the song got created with.

Dear Martin, for the first part of your question:

There may be nothing whatsoever wrong with the analog hardware the original instruments in the song got created with! But each of the analog devices that was used on individual instruments was not used to help the depth or dimension or warmth of the overall sound, or to help the color or definition of each instrument in context with the rest of the instruments — they were used to help create the original sound of the individual instrument. And sometimes:

1) that original sound was “just perfect” or

2) sometimes that original sound became no longer perfect when heard in context with the rest of the instruments, there may have been frequency conflicts or other issues that required new processing needs for that instrument in context with the rest.

And why would ONE more conversion and process through a single analog processor

bring back “anything” that was not there from the original analog hardware that produced the song in the first place?

Good question. There’s no guarantee that one (or more) analog processors used in the mastering stage will enhance, bring back or help the sound of the mix you send through it. No surprise: You can totally ruin the sound of a great mix by passing it through any processor, or certain processors, or using the right processor in the wrong way! On the other hand, some all-digital mixes may lack depth, or definition, or don’t bring out the qualities of the individual instruments as much as could be useful. This could be because the mix engineer was not as skilled in use of delays, or distortion, or EQ or color as possible. Or it’s possible that the mix engineer did a superb job but has a question whether he got the most out of the mix possible. If it was an in-the-box mix, the mastering engineer (or the mix engineer) may find that the subtle addition of distortion of certain analog processors brings out more depth, dimension and clarity from the mix. Subtle amounts of the right kind of distortion enhances depth, dimension and clarity in a recording, bringing out inner details that may be desirable. Too much distortion or the wrong kind of distortion can ruin a recording. In the mastering stage we carefully judge whether the subjective improvement outweighs the objective degradation of going through another conversion. 

Hope this helps,

Bob

Jim P wrote

Jim Prendergast here. My question is: when I’m bringing 24-bit audio stems out of my DAW to a summing mixer (then returning the  resulting stereo mix to the DAW), should I dither the outgoing stem (even everything at this point is staying in 24-bit)?

Hi Jim

Yes, you should dither the outgoing stem to 24 bits IF you have performed any processing (including gain change) prior to sending that stem out. Get a free plugin called “Bitter” from Stillwell Audio and insert it at various places in your chain, manipulate the faders at various places and you’ll see what’s happening. The word length grows from 24 to 32 float when you make a change. BUT as you know, when feeding out to the summing mixer it needs to be 24 bits as there is only a 24 bit path. Hope this helps!

Typically if I’m using a plugin on the feed to an external analog device, I’ll put the 24 bit dither just after the plugin.

Bob

Jim wrote

This helps immeasurably. The answer is what I gathered from reading your (wonderful) book, but I needed to make sure I understood correctly. 

From: Aaron Anderson

I do not have an unlimited budget (who does?) so i’ve decided the best thing for me to do is to build a world-class (in theory) 2track signal path, and mix in the computer.. i also do a lot of processing of sounds with different amps, mics, synthesizers, an eventide, samplers, stuff like that… i really like to put sounds in odd spaces and create textures.. anyways, i love the sound of albums produced by Flood (mark ellis.. he’s always talking about 15ips in interviews. he’s done U2, NIN, depeche mode..) but I don’t have much experience with tape.. and i still haven’t quite gotten the tone i want in the digital realm.

i love compression and dark sounding recordings (achtung baby by U2, radiohead’s ok computer, etc the more color the better!) and to this end i use things like the cranesong hedd (great tube simulation), a joemeek comp (goofy box), and a pair of distressors (my swiss army knives).. i also have a couple of neve broadcast modules the sounds, but they still don’t really have the bottom and darkness i associate with “tape.” does the SPL box really do what it claims? and more importantly, do you think it might help me find what i’m looking for..

Dear Aaron:

My answer is POSSIBLY. I am concerned about recommending the use of analog tape simulators in the mixing process unless you have world-class monitoring which can tell you unequivocally when (if) you’ve gone “too far.” There’s nothing worse than the sound of oversaturated analog tape; turn the drive knob one step too far and the distortion you generate will turn from “good” distortion to “bad” distortion. Once any damage has been done, it cannot be undone. Furthermore, I’ve found that after good mastering, a little bit of analog tape simulation is enough; so using such a device in the mixing chain can be a problem, because you cannot anticipate its interaction with the mastering processors. As usual I recommend that you send two versions of a mix to the mastering house, one with and one without processing. This applies to any processor(s) you put on the mix bus.

In general I recommend waiting until the mastering process for such a box, if you are planning on putting it in the mix bus before your mixdown file. One of the artifacts of an analog tape simulator is that because of the addition of harmonics, it reduces the transient character of the music; this is like a second cousin of a compressor. All my usual considerations of wordlength and dither also apply, because the external Machine Head box outputs 24 useable bits.

Now, for the subjective sound of the Machine Head. I have found that if you start with a 16 bit source (low resolution medium, already dithered at the 16 bit level) and then process it through the Machine Head, you will not come close to the resolution and quality of a well-made 1/2″ 30 IPS tape. The Machine Head is not a perfect substitute for analog tape, but it is a very powerful simulation. It adds a bit of “digititus” onto its “analogitus.” That’s the tradeoff. Subjectively, it raises the perceived quality of certain musical sources that need “fatness,” but don’t expect the exact equal to mixing to the highest resolution media. Of course there is no single “magic” solution, and talent and ears count as much as the equipment used. However, by now (2012) I can recommend some things that are much better:

Recently (2012), analog tape simulators have gotten shockingly good. The UAD ATR-102 and Studer A800 make marvelous tools for mixing engineers or mastering engineers (with the same caveats about overdoing it). These are the first analog tape simulators I’ve found without any detectable digititus or harshness. They do add very similar depth, warmth, and “punch” to the real tape machine. And for mastering engineers (with the same caveats). I also have and use when desirable the Anamod ATS-1, an all analog box which is not subject to the alias distortions but finally I find it hard to hear the difference between the ATS-1 and the UAD products, which are very precise and satisfying replicas to my ears (minus the flutter, and good riddance).

I recently mastered a project that needed some sweetening using the UAD A800 set for 30 IPS, BASF 900 tape and turning down the input by 6 dB and turning up the output by 6 dB. THAT SUBTLE…. using the I/O gains as a form of compression control.

Abuse? Abounds! I’ve received some VERY SPONGEY mixes where the mix engineer got thoroughly carried away with an analog tape simulator.
Now, if you’re looking for A’s, you know what to do. If you are looking for A++, you also know what to do. Consider waiting till the mastering.

Sincerely,

Bob

From: Bill Hobson
 
Thanks for all the great information.
Your website actually got me started recording on wide track analog tape (16 track 2 inch) after first using ‘bad digital’ for about five years. I’m still running Alesis ADATs and all of my clients are still using that format because they are afraid of the tape costs.
I’m working on schemes to make it more affordable and I’d like to know, from your experience, how many passes is a reel of analog tape good for? Have there been any tests to determine this? What causes the sound quality to deteriorate?
 
Thank you,
 
Bill Hobson

Dear Bill:

Thanks for your comments! 16-track 2-inch is a great format.

You’ve asked a good question. I think that you should get a real good bulk eraser for 2″ tape first, because you can’t take chances on incomplete erasure or clients hearing a piece of something previously recorded.

The bulk eraser could set you back thousands of dollars. But even so, I wouldn’t fool around with re-recording on analog tape. It has a finite life and modern formulations seem to lose their brilliance just on multiple playbacks, which is not a function of xide shed, but of magnetic retentively. Which implies that re-recording new material should do all right, but personally I wouldn’t take the chance.

Take care,

Bob

Subject: RE: Back to Analog
Bob:Great article. If I may, I’d like to add my understanding of the problem. Linear PCM is at it’s best with full-scale signals. PCM achieves its maximum linearity, and thus sonic resolution, at high levels. At the same time, linear PCM has low resolution, and thus has its highest non-linearity, at low levels. Analog tape compresses, and thus distorts, at high levels, but is linear and has low distortion at low levels. If we can lower the noise floor of analog tape, and avoid the overload characteristics in our recordings, we achieve great performance. That’s where 1/2″ 30ips comes in, and even technology such as SR noise reduction.

These methods increase the dynamic range sufficiently so that the noise floor is low and the overload characteristics are high enough to be out of the way. 1/2″ 30ips also provides a large recording area which gives good linearity over a wide dynamic range. Thus, I argue that the reason why 1/2″ 30ips sounds better than 16bit PCM has less to do with dynamic range and more to do with superior low level resolution.

Another way of looking at it is in terms of bits of resolution. 16 bit PCM, at an operating level of 20 dB below full-scale, is really around 12 bits of audio. At 40 dB below operating level, there are only 6 bits describing the audio signal. To grasp what this means, a 1-bit error in the digital description of the signal gives a distortion of 1.5 percent at 16 bit resolution.

(Note, part of his argument is irrelevant, since dither linearizes the PCM system at low levels and gets rid of this type of distortion. But I have no argument that 16 bits are not enough, for many other reasons)

Even if the digital description of amplitude is dead-on, there is significant harmonic distortion in the audible frequency range going on at 6-bit resolution. And it just gets worse as thelevel drops on digital recordings. How can we possibly experience good sonic detail at low levels with 16 bit PCM? At 40dB below operating level, analog tape maintains its linearity, and thus reproduces with better clarity. I don’t have figures at hand, but I’m sure the harmonic distortion is very low.

The point is that what we’re hearing is understandable in simple terms, and can be measured with common instruments. It’s not rocket science. All of this just supports your point even more, that analog recording has not yet lost it’s edge… -Michael Karagosian President, Cinema Group, Ltd.
(once upon a time, an engineer with Dolby Laboratories)

Of course, it can be argued that with dither, a 16-bit system can perform as well as that analog tape. And more and more linear A/D converters are being developed with excellent linearity at low levels. But the ears seem to indicate that despite the fact that analog tape’s noise is higher than the dither noise of 16-bit, the apparent low level resolution of the analog tape is superior to 16-bit digital. Once you get to 24 bits, though, low resolution of the digital improves, and the subjective judgments get closer.

I think it’s because the “effective sample rate” of analog tape is higher… Or, to look at it another way: the filtering in the analog tape is gentle, with response to about 30 kHz, but severe filtering in the 44.1 ks/s digital system is required, and I think I hear the artifacts of that filtering… Or because of analog tape’s slightly greater 2nd and 3rd harmonic distortion, even at low levels, than digital, which warms up the sound without fuzzing it. As opposed to “naked digital” which can be extremely revealing of any of the harshness in a source. When you move to 24/96 rates and higher, many people find the digital sound to be as pure or purer than analog tape.

I replied to Michael:

Thanks very much for the comments and your perspective, Michael!

Of course, one thing neither you nor I discuss directly is why analog seems to have more space. However, the relationship between “more bits” and increased space (e.g., 24-bit has more space and depth than 16) seems to point out that the spatial and depth information in a recording are directly related to low level resolution as well.

By the way, I was the first official “user” of Dolby SR in the New York area and on the “first users” list!

Best wishes,

Bob

And Michael has the best last word:

Bob, glad you liked it. I appreciate that you’re keeping the topic of quality alive. I feel like the music industry has been overridden by the surge of technology, a lot of which is not music-friendly, and most of which is poorly understood by users.

Michael

From: Chris Sansom
Subject: Re: archiving
 
My comments are:
Hi. I was wondering if you could help us with a little advice. We are about to copy a large amount of analogue masters (1/4 and 1/2 inch) as their quality is deteriorating. What do you recommend to transfer onto? We are considering making DAT (for day to day reference) and Genex MO (24bit/192kHz)- for archive copies at the same time. Another possibility is making 24bit wav or aiff-files and archiving on DVD-R. Would making 1/2″ analogue copies be a good move?

Dear Chris:

Your choices are very delicate and many people face them every day. The bottom line at this time is: Any digital medium you choose to copy to will have to be recopied at periodic (recommended every 10 years) intervals! In that respect, the Genex MO is about as good as any medium.

The quality of the analog tape reproducer is EXTREMELY critical. If you are looking for audiophile quality, then you need an extraordinary analog tape reproducer as well as an extraordinary A/D converter. Do not skimp on either of those choices. As well asthe expertise to align and set up for the transfer. The transfer is the key.

24/192 is an option nowadays, since the costs of storage media and converters have come down.

If the recordings are 2 track, DVD-R is as good a choice in my opinion as the Genex MO, and less proprietary, more likelihood of being playable in 10 years! And easier to copy.

As for analog copies, 1/2″ 30 IPS analog copies are an excellent move if these tapes are extremely precious to you.

Another consideration is sampling rate. Would it be best to use a multiple or of 44.1-48kHz? Would making 24/96k archive copys produce a problem when converting to 16/44.1.

Don’t worry about the multiples of the sample rate. Today’s very best sample rate converters can do an equal job from integer or non-integer multiples of 44.1 or 48k. I would recommend 96K over 88.2 marginally at this point if you are thinking of making a DVD. If the DVD-A gets off the ground, then 88.2 will be supported. Another possibility is DSD or SACD. But you will not go wrong at 96/24.

Best wishes,
Bob Katz

From: Greg Bates
 
My comments are: Your articles are phenomenal! Maybe sometime you could comment on the use of aural enhancers, I use one as the final link between my mix and DAT. Is this a healthy practice, especially if I were to use your services for mastering?

Dear Greg:

Thank you very much for your incredible comments and for thinking of Digital Domain!

As a general rule I’m pretty down on using any processor between a mix board output and a final recording medium. If you think an entire mix needs “enhancement,” then I suggest you go back to the beginning and study what’s wrong with your mix in the first place, or maybe there was nothing wrong with your mix at all, and your monitors are telling you the wrong thing. If you’re not sure, the best solution is to send the mastering house two versions, one with the aural enhancer (or whatever processor) and one without. 8 times out of 10 we choose the version without the enhancer as we can often do much better with other techniques. Often times I find people use enhancers succumbing to the “brighter is better” argument, without judging the fatigue or recognizing the other tradeoffs when any processor is inserted.

Bob

I’m wondering why many high end stereo A/D’s have relatively low analog input levels. Most pro consoles and gear are clipping at >24 dBu. However take a look at some sample A/D default input levels referenced to 0 dBFS:

• Cranesong HEDD +16 dBm (manual doesn’t state if this is max)
• Mytek AD +19 dBu
• Apogee Rosetta +20 dBu (can adjust to +24)

Why the “missing” headroom? Is it because full program material generally requires less headroom? This would assume all audio is pop music! What am I missing, something simple I am guessing.

Well, the Cranesong is completely adjustable, I’ve been able to set it to +16 dBu max through at least +24.

We have a Mytek 192 and it’s adjusted to +24 dBu so it must be adjustable though I forget how we did it!

The answer I think is a combination of incomplete documentation and a bit of ignorance or assumptions on the part of the designers. Many of them assume people will be using the converters for mixing with consoles that have VUs and the mixes will have a crest factor of 14 dB or less and that the users will want to “slam” the converter meters. A lot of assumptions, eh? A “perfect” A/D converter in my mind should be able to accept up to (or beyond) +24 dBu = 0 dBFS as a maximum and as a minimum have sufficient gain to accept +10 dBu = 0 dBFS, but I’m willing to accept +15 dBu = 0 dBFS, which should cover a wide range of circumstances.

From: Barton Michael Chiate
Subject: Re: Question
My comments are:
Bob, Thanks for all your articles. Great stuff!! Required reading for my students. I have a question. Maybe you can help. I have several recordings of classical music that were recorded to DAT through an Apogee AD 1000 with UV22. I dumped them into my computer to edit at 24 bit.(Figured higher is better)

Hello, Barton. Thanks for writing and for your fine comments.
The only reason to “bump them” to 24 bit is if you’re processing the material or doing any editing or gain changing. Otherwise the 24 bit didn’t buy you anything, as I think you already know. In fact, most DAWs would let you keep the original files at 16 bit and calculate at a higher wordlength. This saves space, since “bumping” to 24 bit just adds zeros at the tail but doesn’t add any new information.

I have not done aything to them but edit. They are now ready to master but I don’t know the best way to get them back to 16 bit. I have an AD8000 with the D/A card and a Finalizer Plus. So far, I have been using the dither in the TC to a DAT to burn CD’s but I would like to avoid writing the file to disk again so I can burn the CD for the duplicators.

I see. Well, Murphy’s law says to get a bitscope (always have one around) and make sure you didn’t accidentily put in any extra bits. If the program did put in extra bits, then ask why and find out why. If the bits are due to legitimate processing, then you really have a 24-bit file. Then use the UV-22 (or something better!) to go to DAT and then load that back into the computer if you must. Or just loop out and then back in….our Sonic system can play back 24 and record 16 simultaneously, can yours?

Or, if it’s really 16 sitting in a 24-bit slot (8 bits zeros), and you’re stuck, then just copy it out to DAT or S/PDIF-AES CDR and then back into the computer. That’s probably the fastest way.

Don’t go through analog. Why do that? If you have the AD8000 you must have a DAT machine around somewhere; doesn’t the AD8000 go D-D?

Bob

Chas wrote:

Dear Chas:

A to D converters do not receive files. They receive analog input. So your question is not relevant to A to D. D to A converters do not automatically add dither and you should not feed a 32 bit float file directly to a D to A converter as it will truncate some low level information. That said, the difference is quite subtle. But I do recommend you dither the 32 bit float file to 24 bits prior to sending out to anyone, including cutting. And examine your levels very carefully!

And if they do not does that mean i should only send 24 bit dithered files to the cutting master so he doesn’t cut from an undithered 32 to 24 D to A conversion?

Yes, absolutely!

Hope this helps,

Bob

From: Mark Schiffelbein
 
My comments are:
 
Although I have read many articles in the past about baking old master tapes, I never need to use the technique up until now. Of course now I can’t find the information.
Do you know the time and temperature. Any other tips? Do’s and Don’t’s?
 
Any help would be appreciated.

Metal reels, fairly dependable oven. Recommend about 100 degrees F for 6 hours. Watch out–splices and leader tend to fall apart. Don’t try to bake tapes made prior to Scotch 206 as these do not need baking. If the tape is not squeaking or sticking on the machine, DO NOT BAKE.

Hope this helps,

Bob

From: Allen Minor

I just purchased a Sony wireless microphone system. I have a Sony DCR-TRV900 digital video camcorder. The system includes a WRT-805A body pack transmitter with ECM-44BMP microphone, and a WRR-805A tuner/receiver. The tuner/receiver instruction manual states the output connector is 3.5mm dia., balanced. The camcorder used stereo minijacks connectors.

My question is this: “Can I simply plug this unit into my camcorder and begin shooting? or do I really need to get an XLR adapter such as Studio One. The salesperson said I only need to just plug it in; however, I am very wary of the receiver instructional manual which states “balanced.” I’m wondering if this might be an error in the manual as that is a pretty small output jack to be balanced.

Hi Allen. You are right to be wary about the output connector claiming to be balanced yet on a mini jack. Well, it is an inexpensive unit, cutting corners. In some ways you should be happy (I guess) you’re getting a balanced output, if you are really getting one.

Find a voltmeter. Even an inexpensive one. You can go to Radio Shack and get one for virtually nothing (under $10.00). Get a stereo mini-jack and plug it into the output of the unit. Measure between ground (the sleeve) and one of the inner connections. Note the voltage (AC) on an average passage. Play the same passage again, and measure the voltage between ground and the other inner connection. The voltage should be about the same, let’s say around or a little under a volt on forte passages. Then measure between the tip and the ring, the voltage should be around twice the previous. If not, then you do not have an (active) balanced line output. If you can’t measure any voltage there, or extremely low voltage (millivolts), then you probably have a balanced mike level output from the source, and an XLR adapter would be exactly what you
need, wired as follows.

If your Camcorder has a balanced line input on XLR, then you should make an adapter from the mini, tip to pin 2, ring top in 3, ground to pin 1. If your camcorder has an unbalanced line input on some other connector, then you should make an adapter from the mini, tip to hot (probably also tip, on a 1/4″ connector or RCA connector, I assume) and sleeve (ground) to cold (ground or sleeve on the camcorder). Ignore the ring of the mini in this case.

A qualified technician should understand what I’m writing and make the correct cable. If not, then don’t do the wrong thing. The result could be obvious distortion of your audio.

Hope this helps,

Bob

Why some mastering engineers insist on using unbalanced lines…

Do any of the engineers on the list use balanced interfaces at all?

>Balanced is for Mics and Telephones, wrote Dave Collins of A&M Mastering on the mastering list!

and I replied

It’s nice to meet with another engineer who subscribes to that. All other things being equal, unbalanced is better. Now just what does that mean? Well, basically, it boils down to the “less is more” philosophy.

Here are the caveats: In a small room, where all the power is coming from a central source, and all the analog gear is plugged into that power and no analog audio enters or leaves the room, and you have your signal to noise and headroom issues all straightened out, then unbalanced is almost always better sounding than balanced.

Exceptions: a) Equipment whose balanced stages are so-well-designed that it is impossible to design the same piece of gear with fewer stages unbalanced than the balanced version. b) Equipment which has completely balanced topology throughout, fully mirror imaged in its inherent design, and impeccably designed with the best components (example, Nelson Pass gear). But I’m not so sure it sounds better because it’s balanced or just because it’s better! I’ve removed 5 op amps from the front end of a certain well-known A/D converter, and removed 5 veils from the sound. Think about that. I’ve removed 3 tubes and an output transformer and several relays and resistor networks from the output stage of my Ampex MR-70 electronics, and been rewarded with “more tight,” definitely quieter and more transparent sound.

Deltaforce wrote:

hello, Bob. What I was taught in mastering, try to aim for a rolloff at 25 Hz or so and a rolloff at roughly 18 Khz. That will make your track sound louder and more bearable for the human ear.

It’s correct ?????? cut 25hz and 18000khz ?? eq in the master channel ?

thanks bob ! regards from spain.

Dear Deltaforce:

This is NOT a guarantee. You always have to listen carefully and compare. This recommendation probably came about with people trying to make a “louder” master as many times extreme low frequencies take up extra power without being a help to the sound. Or from older practices for LP masters, which are not relevant to CDs and DVDs. But many times, especially with powerful and deep bass drums and low bass notes, keeping the low bass range HELPS. Always listen and compare both ways to make sure. Never low cut without being sure it is a help.

Best wishes,
Bob

From: John Campoli

Hi Bob,

I’ll simplify my question regarding my mixes. Regarding bottom end uncertainty, which is easier to deal with on your end, bass heavy or bass light, if I were to err one way or the other? Which will yield better results, other than having it correct that is?

Thank you,

John

Dear John:

The answer is that neither too much nor too little bass is your best solution. The best way to deal with any uncertainty is to send me a mix of yours in-the-making for a listen/eval and I’ll try to make a determination of the best solution for you.

Otherwise, if you work in a vacuum of uncertainty, The results in mastering can be quite inferior either way. In general we’re talking about the bass instrument and to a lesser extent the bass drum, though often both are involved. I’ve gotten mixes from clients using too small loudspeakers in a bad room where the bass in the mix was offendingly loud and boomy. Finding exactly the frequencies that are problematic and dealing just with them without causing the bass drum to be lost or suffer or much worse the keyboards, guitars or midrange instruments to suffer can be an exhausting, time-consuming and costly venture to avoid in a mastering session. It’s much better to get it as right as possible before mastering. If the bass is too light then the sound can suffer by sounding thin, harsh, lose punch or be uninvolving. And sometimes I can help that situation in the mastering easier than the “too heavy”.  But I would never Recommend either way to you or any client. I recommend that you try to get it as right as possible, neither too light nor too hot. And also pay attention to the dynamics, clarity, tone and impact of the bass instrument and ensure that all (or most) of its notes are heard and that it works well (collaborates) with the bass drum.

It’s true that “bass is the last frontier” and can be extremely frustrating to mix. One excellent solution is to send me 3 or 4 stereo stems: vocals, “melody” instruments, bass, bass drum. With those stems I can take a mix of yours that might have been a B minus because of boomy bass and lumpy kick, and turn it into an A plus. With far less effort on your part and far less effort on my part than any other solution. And with far less compromise on the result. You should still strive to be as correct and consistent as possible in your levels before sending me a stems mix because the mastering engineer’s job is not to mix, and I’m stepping over the line here to help a mix engineer who’s having problems. So the more consistent your work then the less time I have to spend finding a different solution for each song. The more time I can spend making the best master for you and not “fixing a mix.” In situations like this I might spend an extra hour or more because of the stems but save a half hour or more that I would have spent trying to fix a full mix with bass problems. And as I said. The result of having stems like this can be fantastic.

Hope this clarifies the situation.

Best wishes,

Bob

Hey Bob!

Ok, so here is the breakdown.  Half of the songs (5 of 10) were recorded 24 bit.  Half are 16 Bit.  all varying sample rates (I was all over the place apparently).  We recorded into Cubase 5 using the RME Fireface preamps going into Cubase.  All Multi track. The 16 bit songs have 16 bit track sources in the session.  No external gear.  Mixed in the box.  I still have the sessions with all the tracks, and the ability to export the songs/sessions/tracks as 24 bit files. (96K too if you like).

I’ll look into plugins for 24 bit dithering on the master bus.  If you have any suggestions, that’d be great!

Keep in touch,

Brian

Hi Brian,

Since you are using Cubase, like the new Pro Tools 10, you can export your mixes as 32-bit float and no need to dither for the moment. Send the 32-bit float mixes to me for mastering. It is also important to realize that *all playback is fixed point so *when playing back a 32-bit float file you must dither to 24 bits to get the right resolution to the
DAC. Otherwise the DAC will truncate the lower information and you will hear a degraded sound. So you cannot compare a 24-bit file to a 32-bit file by listening. All DACs accept up to a 24-bit fixed point signal.

Hey Bob,

So, I listened to the album in 24 bit last night. It was like hearing it for the first time.  I can’t really explain why it sounded so much better, but it definitely was a noticeable improvement.  I think maybe clarity is a good way of describing it. I’m definitely a changed man when it comes to exporting files.  No longer shall I shackle my mixes to the binds of 16 bits.

Right now I’m waiting for the final go ahead from the rest of the band.  Once they give me the o.k., I’ll shoot the album your way.

I can’t wait to hear your opinions on the mixes!

Thanks again,
Have a great weekend,
Brian
The real key isn’t as much the improved resolution of 24-bit as then it continues to raise the question if the end medium is still 16-bit. Instead, the real key is that you had been truncating information and creating distortion! That’s why your mixes sound bigger, warmer and clearer. Even though your source tracks were 16-bit! I’ve been trying to get this point across to engineers for many years by the way.

Best wishes,
Bob

Dear Mr. Katz,

I read an article about mastering in a german recording magazine in which the author recommended to change incoming material with 16 bit into a 32 bit float file.The theory behind is that with these extra reserved bits you will get better results after processing the file through various plugins. As we know the lower bit information (where all the fine details of music are stored… ) gets lost if you have only 16 bits – this theory of creating these extrabits and use them for more accurate and exact processing sounds logical for me. What do you think about that?

Thanks in advance for your answer.

Respectfully yours

Mike Amon

Dear Mike: It’s not necessary to explicitly do this in any modern-day DAW. The DAW will do it for you. All calculations on any source file will be done at 32 bits float, even if the source file is 16 bit. The results have to be captured in 32 bit float to take advantage of this, or dithered to 24 bits fixed.

Hope this helps,
Bob

From: “Arksun”

 
Hiya Bob, How’s tings.
 
I was wondering whether you could settle a long running argument between myself and a producer friend of mine. My friend Andrew Longhurst believes that for A/D conversion there is absolutely no need to go greater than a 20-bit convertor, as the noise level is significantly high enough in an analog cable connection that a 24-bit convertor would offer no sound improvement what-so-ever.

Dear Laurence:

Things are great, very exciting, and very controversial. The future of the mastering industry keeps on shifting.

Anyway, to prove your point, you’d have to have an A/D converter with true 24 bit dynamic range, whose noise level is that low. Most so-called “24 bit” A/Ds have only approximately 20 bit dynamic range in the first place! But arguably, may have resolution below that number. And then, you must conduct a controlled experiment. It’s possible that your friend is right, but I leave no stone unturned,and only a controlled listening experiment would settle the issue. In theory, your friend may be right, but it has to be settled on the basis of masking theory, and masking theory demonstrates that you can hear signal quite far below the noise in certain frequency ranges. Maybe 21 bits, maybe 22… it’s hard to say, but I believe it is *marginally* greater than 20 in the case cited above. But in the same vein, in theory, it seemed that a 16-bit converter should be adequate to record an analog 1/2″ 2-track tape as the noise level of the analog 1/2 is far greater than 16-bits, but in practice, it takes a high quality converter of at least 20 bits to do justice to an analog tape, so as far as I’m concerned, all bets are off until they have been proved by listening tests. That goes for all assertions of theory versus practice, by the way!

I on the other hand am looking at it from the ‘real-wave’ representation point of view, and am arguing that it’s precisely because of this further wave form complication induced by noise that the highest possible bit conversion and sample rate are needed to ‘capture’ the maximum amount of detail in the analog signal being fed, especially with a full bandwidth complex mix.

Your statement may also be true. And the resolution between the two statements boils down directly to masking theory! I subscribe to the theory that there is inner detail in music or test signals which is audible below the noise floor of any system for a considerable distance. Of course, it is not an infinite distance. At some point it becomes academics your friend makes clear. For example, in digital calculations, it has been shown that you can hear a 24 bit truncation below a 18-bit noise floor. Why? Because the distortion has not been masked by the 18-bit noise. So, it boils down to how far down below the noise floor theory #1 is correct, versus how far below the noise floor theory #2 iscorrect! Quite simple, eh? At what point the masking of the system noise covers up the distortion caused by the reduced resolution. This can only be settled by psychoacoustic means, and ultimately by listening. In the meantime, I suggest caution and conservatism, that is “the more bits, the better–probably.”

So far, each time I increase the wordlength precision of my own work, I find an audible improvement. There are currently a few 24-bit A/Ds which are a pinch better to my ears, but I am not certain if that is because they are 24-bits or simply because their circuitry is more accurate, more euphonic, more detailed, less jitter, or any of the many other possible influences for better sound where at that low level where it becomes impossible to separate out the reasons for “better.”

And to repeat points I make elsewhere: this 20-bit question only comes up at the beginning and end of the chain, because when you start processing (DSP calculations), more bits are definitely better.

Hope this helps,

Bob

Hi!

You’re web site is a real gold mine of information. It can sound quite technical since I’m only starting out my little home DAW (on a powerful Pentium II). I see I have a lot to learn and everything about mixing and especially mastering is great to read. I was wondering what you thought about my plans (if you have the time to answer, even shortly). I see the new 24bits/96kHz revolution and want to wait to have everything on my PC, from A/D to sound card, to internal processing (sequencer software like Samplitude 2496 and also plugins) with this type of resolution. I have seen plugins from Waves that use 56 bits internally and dither down to 24 bits afterwards. I see that ADAT is creeping up slowly to 24bits, too. These are all the parts I want to have on my DAW. What do you think about the quality level could achieve on this setup? I know I’ll have to research a lot for each piece of equipment to be of very good quality but I have the time (about 2 or 3 years before final completion of what I want) although not that much money (in the thousands but not tens of thousands…). Any advicewould be appreciated. Thanks for your time and wonderful articles!

Hello. Thanks for your fine comments.

You know, it may seem funny coming from me, but my best advice here is don’t get too hung up on the numbers… The engineers who are most successful at using the toys you’re describing….have put in many years of dues recording and mixing music the old fashioned way. It’s up to your ears and talent; weighs as much as the technical knowledge around here.

Yes, by all means follow as much of my continuing technical advice as possible. That advice is designed to keep you out of trouble, but it’s not a magic road to great sounding music. All those soundcards, and all that resolution and all those tracks don’t amount to a hill of beans unless you know how to put the tracks together to make a whole piece of coherent, beautiful-sounding music.

In my opinion, the quality level someone can ultimately achieve by these numeric advancements will be more limited by talent and ears than by the equipment. “24” is not some magic number that makes everything automatically better.

Now for a few technical comments: on the question of “24” it’s also how the equipment works inside on its way to and from “24” that will determine the quality of the sound. High internal precision is important, much greater than 24.

But let me clue you into a little secret. On the conversion end (A/D and D/A side), most so-called 24-bit converters contain 4-6 bits or more of marketing hype. Mikes, preamps, converters, gain stages, and room noise contain enough energy to self-dither all but the most pristine sources to less than 20 bits! “20” bits done right on the input and output ends of the entire process is more than enough, as long as the processing in-between works with longer words.

24-bit signal to noise ratio on the input or output end, or just about anything greater than 19 bits, is equivalent to trying to detect a candle that’s a mile away, and in front of the sun, to people have no idea how small a magnitude the LSB of 20 thru 24 bits is.

So, watch out for the marketing hype, use your talent, and go by the numbers as we talk about it here at Digital Domain. Sort of a “More Bits Please” article with common sense attached!

Next point, the newer equipment you propose in your letter will actually present numerous challenges and problems, new ones that I have barely touched on, from software bugs to ergonomic nightmares. “It won’t be like this when we get the computer”–or will it?

After you find the computer-based studio equipment that you think meets your dreams, expect to spend two-three more years debugging it, and replacing half of it because the manufacturer didn’t anticipate some of your situation. Your talent may find work arounds for those problems, but can you wait that long? Consider working with the best you can now… make good music now using traditional-based high-resolution tools, some of them computer based, but long established. There are plenty of examples of “totally digitally mixed” albums currently released that sound absolutely horrid because someone “drove it all by the numbers”!

Best wishes,

Bob Katz

I know it’s possible to blow a channel on a mixer, but is it possible to blow a channel on a analog compressor by pushing the signals gain until it’s clipping? im planning on buying an Art dual channel tube compressor and want to know if it is possible to blow the channels or damage the unit by overloading it with signal?

It’s very difficult to blow or damage a channel on a mixer or a compressor with any normal signal that you could feed to it, including a reasonable overload. Don’t pin the mechanical meters, though, they can be damaged.

From: aart

 
My comments are: I was wandering if I have too use a 110ohm coax  cable (spdif) now that I am recording at 24bit ?? (24bit out of the Tascam tmd4000 to the RME hamefall 9652 spdif in)
 
Dear Aart…. you don’t have to worry about cable matching for the wordlength. Only for jitter considerations, and in this case, since you are transferring information digitally that has already been converted, jitter is not a problem. If the mismatched impedance works (for a short cable connection), then go ahead and use it. The Hammerfall card accepts 24 bit data in via its SPDIF input. In fact, you must confirm that the Tascam puts out 24 bits. I was told recently (and this may be wrong) that the Tascam only produces 24 bits on its bus outputs, not on the 2 mix. If this is correct, then you can only mix to the bus outputs if you want 24 bits.

Bob

From: Gorimaa

Message: Hello, I would like to know how to calibrate my beyerdynamic DT 880 PRO (or other headphone) to be able to use the K-system. Thank you so much.

Hi Gorimaa,

It is possible with a special calibrator to measure the SPL of headphones, but there are so many practical issues trying to mix or master on headphones that I don’t think it’s worth the trouble. If you want to get close (but you won’t be exact) you could play calibrated -20 dBFS uncorrelated pink noise on both speakers with your monitor control set to -10 dB to produce a sum of 76 dBC/slow on the speakers, and switch back and forth between the speakers and the headphones. But you must be using a calibrated monitor control for the headphones as well. And when you’re done, where are you? —- No where, because the purpose of the K-System is to judge transient degradation along with examining the position of your monitor control, using your ears. And headphones are deceiving, they exagerrate transients, as do near-field monitors, so it falls apart.
Best wishes,

Bob

From: Kirk James

 
My question is this: With the Mackie 1202 mixer showing analog levels holding nicely just below 0, how should I set the gain level for CakeWalk? Since Gina is handling AD conversion, Cakewalk is therefore receiving a digital audio signal, so it seems logical that I should be setting Cakewalk for about -20 dB to leave me enough headroom/cushion. This realization, probably obvious to you, had slipped by me.

I don’t trust the meters in the Mackie. I just don’t know what they’re reading, dynamics-wise; they have too few steps and don’t tell you enough. They have a characteristic which is neither VU nor peak. A high quality external meter is required. My conservative suggestion is to have the Mackie producing peak levels of about +18 to +20 dBu at most on the highest peaks. This is to keep the Mackie from being pushed too close to its clip point, which can make it sound ratty. This peak level should correspond with 0 dBFS in the Gina. Also, assuming you’re running the Mackie balanced out and the Gina balance in. Any differences and we have to use different numbers!

You can accomplish that calibration by confirming first with a calibrated volt meter that the Mackie is producing +4 dBu with a test tone, at its 0 meter mark (1.223 volts). At that point, drop the level of the Mackie 4 dB (as measured on the external voltmeter) (it now reads .775 volts, which is 0 dBu). Then adjust the Gina’s gain until that reads -20 dBFS in the digital world.  If you do not have a calibration mark at -20 dBFS in your software then you need a calibrated digital meter (Dorrough, Mytek, Sony, DK, etc.). You’re calling that analog level of .775 volts (0 dBu) to be -20 dBFS in the Gina. Now, your highest peak of the Mackie will be +20 dBu, well within its capability. Simple arithmetic, eh? 0+20 = +20!

That optimizes the system. The rest is up to you. At that point, ignore the Mackie meters and read the Gina for all A/D transfers. Or preferably, the external digital meter, which has an accurate over counter, and so on. I doubt that Cakewalk has the quality of digital metering to tell you what you’re really doing.

The other issues in my articles, about VU levels and compression and so on, should be dealt with by using both: calibrated monitoring gain and by using a combined VU and Peak meter, like the Dorrough or DK or the Pinguin, which you can integrate into your PC. The Pinguin incorporates my new K-system meters and it’s not expensive.

Hope this gets you started,

Bob

Jonathan Wrote:

I hope this finds you well and that it’s OK for me to email you with a question rather than a spot of business?
I’m reaching out to you as I’m new to the mastering world and have found your books and website super helpful – especially regarding wordlength management and dither. So many thanks for that.
My question is about dither and the gaps on CDs. I’ve got dither as the very last process from my DAW (Nuendo) after fades and SRC and then I import the resulting 1644 files into Sonoris DDP creator. All this works great but I realise that the gaps between tracks that DDP creator creates are at digital zero where of course the fades on my tracks fade into the dither noise, which then stops abruptly at the CD gap – i.e. there is no fade to digital zero on the dither.
I guess you may be chuckling (or sighing) by now as maybe not so many people listen loud enough to hear that dither just cut out (!) but it’s bugging me having taken the trouble to make sure everything else is right on.
Am I losing it? Thanks for your time Bob, and also all you have done to share your knowledge.
Best wishes, Jonathan.

Hi, Jonathan. Thanks for your kind words. 

Ideally, your DAW should be able to handle everything, including dither noise between tracks, and ideally the track marks as well, so by using Sonoris you’ve discovered a weakness in your procedure. 

Normally when mastering we use a DAW that integrates everything, your segues, dither noise between tracks, material that’s in the pause between tracks, and track marks. 

I don’t use an external program like Sonoris to deal with these issues for the very reasons you mention and more! For example, I’m about to make a master today that was recorded on analog tape and on purpose I’ve included tape hiss, continuous tape hiss between tracks…. which definitely is meaningful and audible. 

That said, if you never put any intentional material in the gaps and if you don’t think that the 16 bit dither noise between tracks matters (which it probably does not), then you can keep on using Sonoris in conjunction with Nuendo.

Hope this helps,

Bob

This message is getting long in the tooth, but I’m leaving it in the FAQ for historical reasons and there are still a few good pieces of advice below  —  BK

 
From: Mark Gemino

My comments are:

I found your site today extremely interesting. I am about to mix songs to end up on CD format. I would like your opinion on the best method to record for mastering afterwards…
My multitrack source tapes are (4) ADAT XT’s (16 or 20 bit @48Khz)Optical outs (3x optical ins & 1x 8ch analog in) into a TASCAM TMD4000 digital console AES out goes AES into (20-24bit? @ 48Khz) which then connects to a TC Finalizer 96K (24 ?bit @48Khz) . I use the first preset labeled CD MASTER on Finalizer(BTW).

I strongly suggest you skip the Finalizer if you are going to be sending material to a mastering house. As mastering engineers, we sometimes have to undo that which has been done by the Finalizer, and the Presets are designed to make a “generic demo”. In my article: Secret of the Mastering Engineer, actually published by TC Electronic, I recommend this approach, I think on page 18. You can download this article from our website.

For AOR rock, I suspect my opinions below are probably right, but I would like to hear your recording to be sure. It’s tough to recommend in the dark without hearing the recording. But based on experience:

my recording options are :

a) TASCAM ATR 60HS 1/2″ analog (W/456 Quantegy) @ 15IPS via xlr Analog output on Finalizer (could connect to TMD4000 analog output (possible?)

b) TASCAM DA30 DAT AES input from TC Finalizer using digital sync

c) TASCAM CD-RW5000 CD recorder AES input daisy chained to back AES out of DAT deck?

——-The TASCAM Digital mixer has dither capablity – i left it at what ever was preset

If you’re going to the 16 bit, you should set 16 bit output dither. If you’re going to the analog, turn the dither off.

What is the best way to hook all this up to send to you for final mastering?

Between the above choices, what you have is “pretty damn good analog” versus cheap digital, so it’s a no brainer: the analog. I suggest you rebias your Tascam for Quantegy GP9 tape at 400 nw/M equals 0 VU equals -14 dBFS with 1 kHz tone. Consider 30 IPS as well, depending on your music (listen and decide which speed sounds best to you). If you lay down tones and follow the tape preparation recommendations listed at our website you can’t go wrong. It sounds to me like your machine is set to the factory settings. You should find an excellent analog technician who knows how to align recorders and who will align this machine. In large studios, the analog recorders are individually aligned before each mix session, and if your machine is “out of the box” I can guarantee it needs to be tweaked.

Of course, how you get to the analog recorder can be problematic, because you have to have an excellent D/A converter to do it. And the converters in the Tascam console are only fair. If you’re really on a budget but have even $400 to $1000 or more, you can get a consumer D/A converter that, in conjunction with the 1/2″ analog machine, will produce a superior product.

If you do not have a superior D/A converter, it’s difficult to guess, but I still suspect the 1/2″ recorder will do better than the 16-bit choices you have. Especially since neither the Tascam console nor the TC Finalizer provides a great-sounding 16-bit dither, and especially since once it gets to us we would have to take that squeezed-down 16-bit product and try to expand it again, then squeeze it down to 16 bit again. We’re not making sausages here, we’re trying to make music, so let’s avoid that.

Bottom line: the best alternative to the analog tape would be a 24-bit digital recorder or to record to a 24-bit file OR, if your workstation supports it, a 32-bit float file. This is a good replacement for your aging 1/2″ machine if it has not been maintained recently.

If you’re thinking of mastering with us, you’re also welcome to send me a one song sample of your music in progress and I’ll be happy to tell you how you’re doing and whether the mix is as good as it can be, that is “ready for mastering”. If your mixing environment is problematic, you can be thrown off by the acoustics of your control room or the performance of your monitors.

Hope this helps!

Bob

From: J Zuckerman

My questions are:
Is there a difference between a PMCD and a CD-R burned with all the specs using pro software like Masterlist or Jam?

At one time, the difference between a PMCD and an orange-book CDR was important, but not anymore. There were once plants equipped with specialized Sonic Solutions workstations, that could only take a PMCD, but at this time, all plants can accept standard format CDRs. However, I do not necessarily recommend Masterlist or Jam as the definitive, be-all programs capable of producing satisfactory masters. Professional mastering engineers prefer to format their masters using dedicated “all-in-one” programs such as Sonic Solutions, SADiE or Sequoia.

Also, does the timing information get automatically burned into the P and Q channels when recording a disc-at-once CD?

Using Jam or Masterlist? The answer is yes!

Bob

From: michael lynch

My comments are:
I love this site and I find it very helpful!
What I would like is some info or a website about CD Manufacturing Equipment
(not CDR)
Thanks

Michael

Dear Michael:

Thanks a lot for your comments. I don’t know any sites (so far) about CD Manufacturing equipment. If you mean for replication of large quantities of CDs (in the thousands); you should be reading One to One Magazine, or Replication News.

Good luck,

Bob Katz

From: Denis Kelly

My comments are: We have bought a classical CD which runs 78 min. will not play the end of the last track. What is the manufacturing specification on the running time of a C.D.?

Dear Denis:

The maximum running time which a plant will press varies from plant to plant. One plant I work with has a maximum permissible length of 79:38. But some plants have pressed full 80 minute CDs, which stretch the Redbook specifications (tighter space between the lines of pits).

Any time over about 76-77 minutes exceeds the Redbook specifications and invites potential problems. The CD you have in hand is a perfect example of the risks that are taken. Over about 76 minutes, we always caution our clients that they may get some returns and some of the same problems that your CD has. However, if your extra-long CD was made by a reputable plant, it probably just gets by. Try the CD on several players and see if most of them can play the last track. In that case, you’ve just met up with an extra-long CD that is “barely acceptable.” We expect that with extra-long CDs, some small percentage will be returned as defective. That’s the cost, and most of our clients are willing to accept those odds.

Sincerely,

Bob Katz

From: tal trivish

My comments are:

i would like to know what is the best speed to write a cd – audio and data.
also i wanted to ask you, what is the cdr known for the best results.
i am very happy with your web site (it’s very “hot” here in israel)

tall trivish,
Israel

Thanks for your comments. I used to be able to say that the best speed to write a CDR for audio is probably 1X. But there are no current writers that can write at 1X, so the issue is academic. I generally these days choose around 16X as it’s not too fast and practical. I get very low error rate with my Taiyo Yuden blanks using this speed with the Plextor writers I’ve got. Though Plextor no longer manufactures their own writers even if the brand name says so.

The situation is in flux. Check for errors. If you’re really into this “jitter” thing, concentrate on getting a great jitter-immune DAC and stop worrying about writing speed, because it’s no longer anything you can really control objectively.

For data writing, pick a speed that is at the center of the BLER curve, neither too high nor too low, for the lowest errors. For the vast majority of writers and media, that is approaching 16X at this date. It started at 1X and slowly went up as writers and blanks became optimized for higher speeds.

Hope this helps,

Bob

From: John Pglynn

Date: Mon, 25 Jan 1999

I record Christmas CDs for local young musicians who want to record their own xmas CDs. I have had great success and reliability with my Yamaha 2260 accept for one client. The CDs I burned for her would not play on the CD player in her Dad’s jeep cherokee or in her Sharp boombox. They worked fine elsewhere. I am lost as to what to do.

Returning her cash was all I could do. Do you have any suggestions? I burned the disks using Adaptecs easy CD pro. The disks tested fine in the 3 cd players in my home. Great web site, thanks for being there!

All the best,

John Glynn

Dear John:

You’ve done the best you can. Many CD players, portables and car players are not capable of reading CDRs. If you found they play well on most home CD players, then that’s the best you can do.

My comments are: reading your opinion on modern compression, specifically the ‘four cherry bombs and an m50’ pertaining to rap music, i instinctively felt this was.. wrong 🙂 had to think about it, but here’s why: if you are in your car playing loud rap music and you want to scare the bejeezus out of some old lady at the light next to you, every second of your track has got to be an m50. otherwise, you miss it. and it would sound lame if you had to fast forward to that critical break.

I imagine with other genres the concept is similar. music is, like it or not, often used as a form of communication on this level. making every second have maximum impact is critical in a society where seconds can make or break.

Makes sense. The faster society moves, the shorter the time you have to make your impact. But I think there is still some room to breathe and applying more dynamic range, even if it’s microdynamics, may improve the sound, in both hip hop and rap. In my article on compression I also point out that overcompression may be suitable on the time-scale of a single, but very boring and undramatic for an album. So I recognize the duality of this affair.

Only time will tell (pun intended) whether a (slightly) increased dynamic approach will be appropriate for hip hop and rap. Whether it will sound more interesting, or, as you say, in contrary, take away its impact. I’m voting for the former because I think we’ve reached the limit of squashing and beyond, especially in R&B, rock, country, jazz, etc., and it’s time to back off—big time! But you’re right, hip hop and rap are (usually) trying for a very different effect than even the most “punchy” R&B (Bell Biv Devoe, Cherelle, Janet Jackson….).

But… the ebb and flow within the natural rhythm of the music, what I call “microdynamics”, within even 2 to 4 bars… can create more impact. Consider the musical phrase…

shooby dooo WOOOOP shoooby dooo WOOOOP

versus

SHOOOBY DOOOOO WOOOOPPP SHOOOOBYYY DOOOO WOOOOPPPPP

The first version is two cherry bombs and an M80 in short succession instead of all M80s… makes MEEEEE jump more. I don’t know about you.

(see how the emphasis in caps in the above sentence grabs your ATTENTION)

Why not send me one of your hip hop tracks for mastering with us and we’ll see which approach sounds better! I’m betting I can make you dance more with my mastered version. Fair enough?

Best wishes,

Bob Katz

From: John Watts

Subject: Thanks for the mastering lesson

Hi Bob,

Sometimes, just a few important words can change the world (or at least the course of a mix.) After further reading on your Web Site about compression as well as the “loudness” race , I feel I have a much better handle on how to go about my mix/mastering. Thanks so much for a sane presentation of what’s really important– the music! And yes, I won’t finalyze too much– promise.

All the best

John Watts
 
That’s not an FAQ per se, but I can’t resist leaving John’s comments in. Thank you, John.

Bob

From: Cook, Brian BG

Hello Bob,

Thanks for the very informative Web page on mastering, I’m hoping I could take up a little of your time to ask you some questions.

I recently had some rock/pop recordings from my 24track ADAT studio mastered at [SNIP] by the so called top guy and I wasn’t really impressed. While there are definite improvements to the bottom end, the reference CD seems very compressed and quite top heavy which sounded fine in the studio with big speakers etc but in my home studio (JBL 4208, NS10’s) it sounds quite harsh. I noticed that the CD sounds very loud when compared to similar material and that the meters on my DA30 DAT show that the levels are constantly tickling -1 and 0db. Am I a victim of the “everyone want’s to be the loudest syndrome”?

Just for the record, the mastering chain was:

Sony 7030 > Apogee > Compressors/Eq etc > Anolog 1/2 Tape > Apogee >> Sonic Solutions

During the mastering session I noticed that the Soft Knee light on the Apogee converter was constantly flickering which leads me to believe that the way these guys work is to just push everything right up and let the apogee soft knee stop anything going over the edge. Does this sound right, is this normal practice?

One interesting thing that I did was to look at the levels of a lot of music that I like (usually mastered in the US), the album version of a track will sit at around the -4db mark and appears fairly dynamic where as the single version of a track tickles 0db most of the time and sounds more compressed. This leads me to conclude that it is pretty common to create a more compressed “louder” version of a track for a single. Am I right? Using this theory my whole album sounds like a single.

I realize that I should be bringing this up with the mastering house responsible but I thought I’d get another opinion before I commit to any changes. Because most of the processing was done in the anolog domain I may have to remaster it and these guys are expensive and 600 miles away from where I live.

Regards,

Brian Cook

(Melbourne, Australia)

I responded to Brian, telling him I was sorry to hear about his troubles. In essence I suggested to let his ears be his guide. Without hearing his CD, I can’t say for sure, but it looks like his mastering engineer was more interested in getting a high average level, but sacrificing sound quality. Good thing some clients have ears! As for Brian’s comments on measured levels, the peak meter is a very difficult tool to use as a judge of “hot hot is it?”. I suggest using an averaging meter such as a VU meter or the other meter discussed in my compression article .

As for Brian’s question whether special compressed radio singles are made, yes, this is often the practice. There is no justification for this practice, other than to attract impressionable radio program directors by the instantly loud material. In fact, it can be demonstrated that CDs with too high an average level (VU measured levels exceeding about -8 dBFS in my estimation) will sound worse on the radio because the averaging processors used for radio try to create a consistent loudness, and they’ll bring things down if they’re too hot, and up if they’re too low!

Barry Keel wrote

Bob,
I just purchased your book Mastering Audio and look forward to reading it. I also noticed some FAQs on your website in which people have emailed you with a question or two. Hope you don’t mind that I do the same. The two questions I have concern clocking. FYI, I have been working in audio quite a while and started with digital audio in the 90s. I do follow the same principles as close I can with your views on digital audio.

The first scenario I have concerns locking a RADAR with Pro Tools using an Apollo 16 MKII. I know that you say if you have a quality interface with a good clock, using the internal clock of the A/D is best. In this scenario we lock Pro Tools to the RADAR and the transport locks fine. Since the RADAR is only 24 channel we use Pro Tools for any overs. We are recording analog into both the RADAR and through the Apollo to Pro Tools. Here I have been using WC clocking the Apollo from the RADAR WC. It works fine. But, since they are really two separate systems, would WC be best or is it best to use the internal clock on both systems?

Barry,

If you are in the process of recording into both the RADAR and using the Pro Tools for “overs” then at that moment Pro Tools has to be locked to the same clock or things won’t work right. So at that time personally I would select the unit with the most tracks, make its ADCs run on internal and then the secondary unit runs from wordclock that came from the first unit. 

That’s really also the most practical solution. I don’t think it’s worth the trouble switching clocks around when you want to add even more tracks to Pro Tools. But you can, if the Apollo performs poorly on external clock. What do you do for mixdown? Do you transfer all tracks over to Pro Tools or run the Radar and Pro Tools in sync? In the former case, then make the Apollo be the master clock (internal clock). In the latter case, keep RADAR as the master clock, UNLESS the Apollo performs poorly on external sync and the RADAR performs better on external clock than the Apollo. 

How does one make a determination what is the better source? Well, remember, “warmer, wider, deeper” is usually the situation with less jitter. So just play back two identical high quality stereo tracks on the RADAR and on Pro Tools, both of them feeding an identical DAC (not different DACs). Evaluate the different clocking situations. There are more definitive testing methods for this but they do require test gear.

The second question concerns using an RME HammerFall DSP card. It is a digital only card and I use it at one of the studios I work at.  I have asked for some more information on the RME SteadyClock technology from RME but have not received it as of yet. I did read some documentation on their website I found confusing. From what I remember it was stating that their interfaces have 2 clocks, one that locks the incoming signal and then re-clocks it with another internal clock.

It’s just a two-stage PLL (phase locked loop), which helps to reduce jitter. There is really only one clock, an interface can’t operate on two simultaneous clocks. Basically, the first stage clock feeds the second. 

Because of this they stated it would be best to set the card to Internal for the clock source.

Unless it is receiving external data from another source, then either it has to slave to the external source, or if you want the RME to remain on internal, it has to feed word clock to the source machine. 

When transferring over SPDIF or AES I usually set the clock source to the device I am receiving from. What are your thoughts on the RME clocking and would you be able to shed any light on this? They do make a WC add-on. I wonder if it would be best to use the WC add-on and not send the clock with the data stream. If so, would you let the RME be Master and the source device be slave or vice versa? Or, should I set it to clock from the receiving device without using WC?

What are the sources that you are feeding into the RME? Since it is a digital only card, the most critical thing is for it to be well locked to any source that you are recording into/through the RME interface. It doesn’t matter if that’s the lowest jitter situation since you are only transferring data. What is the DAC in this situation? What is the ADC?

To be continued! 

Hope this helps?

Bob

Jefferson wrote:

Hi Bob, thanks for the book and your signature. The book is invaluable to me and the section on dithering reminded me of stuff I had forgotten and got caught out with? All fixed now thanks. There is reference to a blog page but I can’t seem to find it so I have a question here.

The clocking issues from source to DAC and how you resolve them (and any other problems effecting audio quality) with streaming to you listening environment. For example, my home listening is CD player (slave clock) to Mytek mastering DAC to Pre-amp to Power amp to speakers.

Dear Jefferson:

So your CD player is slaved to the DAC? That’s actually an interesting and potentially better sounding way to go, depending on the DAC’s architecture. You’d have to ask Michal if that’s a more valid way to go, jitter-wise.

What would this look like when using some streaming device as the source?

Good question. Your best bet in that case is the new Mytek Brooklyn or the less expensive Liberty, or another DAC that’s got asynchronous USB. I’m sorry to be advocating getting another DAC, but that’s the way to deal with streaming, asynchronous USB. In that case the crystal clock inside the DAC continues to be the master clock and there is a buffer between the stream and the DAC. It’s kind of like the stream slaving to the device, not the other way around. You want to have the master clock in the DAC, not have the DAC slave to the jittery stream. There is no clock in the stream, actually, it’s just data that moves on command.

Hope this helps,

Bob–

From: Geoff Goacher 

 
Dear Bob:

I’ve really enjoyed reading your enlightening articles on the web site and Mix magazine. I have a great deal of respect for your opinions. > > I’m writing you with a couple of compression questions. First of all, is it true in mastering (or mixing for that matter) that signal compression/limiting should not exceed a gain reduction of 6 dB, unless you are going for a “noticeable compression effect?”

Hi, Geoff. Thanks for your comments.

Compression and limiting are so different that a generalization of “6 dB” in one sentence for both, and also for both mastering and for mixing and for all kinds of instruments is just that, too general. 6dB gain reduction on a vocal solo with a nicely-designed tube opto compressor may be barely noticeable, but 1 dB on an entire mix with the wrong unit can be severely degrading! So I stay away from such generalizations. Use your ears…

Second, is it true that pumping and breathing effects are really only a problem when a compressor/limiter is making gain reductions greater than about 10 to 12 dB?

Wrong. Pumping and breathing effects are matters of the attack and release times of the unit and how they relate to the tempo and frequency makeup of the particular music, as well as the signal to noise ratio of the music, for if there is no noise (e.g., tape or pre amp hiss) to “breathe,” you can get away with more compression. It is true that the more gain reduction, the more you will notice breathing and pumping, but in many instances, only a couple or 3 dB of compression will cause audible pumping or breathing effects. Multiband compression can reduce pumping by, for example, keeping the bass fraction from modulating the vocal, but I’m not a fan of multiband compression as it can produce a very unnatural sound, “balls to the wall,” but with no dynamics. It’s much too tempting… Read my article Secrets of the Mastering Engineer for more information on that.

Lastly, is it true that RMS level limiting (brick wall) provides the most natural-sounding mix compression? (Due to affecting only the highest peaks of the signal and making the majority of the signal louder, but uncompressed)

Not RMS, but peak limiting. Very short duration, very quick release time, very quick attack time, high ratio, 1 to 3 dB or so gain reduction can be very invisible. But invisible is not the same as “natural.” When you are talking about mix compression, are you solely talking about

a) trying to increase the loudness character by reducing the peak to average ratio but not affecting the sound, if that’s what you mean by “natural” or
b) are you talking about modifying the mix characterin a “pseudo-natural” manner?

In a), then limiting is the key, for it will be “invisible,” and what can be more “natural” than “invisible”

in b) some sort of compression rather than limiting may be the key, for you are looking to alter the sound in a “natural” way.

These are things that I’ve heard from various people but I wanted to get a more qualified opinion before I incorporated them into my working theory on using compression (or why you shouldn’t use it). Thanks for your help!

Geoff Goacher

Hope this helps!

Best wishes,

Bob Katz

From: Kyle Song

Subject: Re: The real advantages of limiting and compression?

Kyle quoted me…

“TRUE TRUE! No matter how long you’ve been doing it – this is true. Each signal hits the gear differently, and the only way to know if it’s better is to match the volume and listen. Experience can make your instincts right a higher percentage of the time, but using your ears is the only way to know for sure.

Of course an electric bass through an 1176 (or some other well defined cliche) is a no brainer, and will always (usually) work, but you still have to listen.”

Unless the limiter was “part of the sound” too. If its only there for volume -then I might agree with you in many cases, but if the purpose was musical, it oughtta stay. I suspect you’ll agree given your other comment.

Nice post – thanks.

Kyle

No disagreement, whatsoever.

Hello Bob,

First of all, Thank you for sharing your wisdom and for your help to this music world.

My Question:

Sometimes when i design sound effects that have a lot of low freqencies, the low freq sound confused, not densed or straight.

So, first question, do you know what i’m talking about? This phenomena has a name?

Yes, Tom, it’s called “too dense and confused”. But seriously, I suggest you get as accurate and wide-range a monitor system as possible. It will help you settle out what’s the problem in your mix. Probably you have two or more effects both working at the same frequency range, which causes masking and muddiness. The solution is often to equalize out the competing frequencies, so only one instrument or effect works in the same range. This is a problem with arrangement, fundamentally your concept needs work. Similarly when a bass and bass drum work together, they complement each other and when they work in the same frequency range, they can conflict and add up. 

I noticed that the only thing for me that solved it, was compressing almost limiting, above 10 ratio, with very very little amount of threshold. Usually i’m trying not to effect the dynamics over time of the sample with the attack and release. It seems the compressor kind of make the low freq more Fundamental.

Many times for example I used it on very processed kinda noise that i used as if it was my kick sound, so on this very short sample the compressor probably isnt working as the typical “fader riding” usually used in longer sounds.

What am i doing? is this the way to do it?

Honestly I don’t know. I’ve not encountered this solution, but if it works for you, go for it. Send me some samples and I’ll study them. 

If i’m making a kick from a sine wave, usually i dont even require this compression because the low freq sounds perfect for me.

Also, When i hear low freq in reality it never sound neither confused nor compressed, just real and natural 🙂

Welcome to the world of artificial reality.

I used to work near machines that sounded with a lot of low freq. I think i never heard a recorded audio material that captured that low freq behavior.

You should hear my shuttle launch recording. I need about 50,000 more watts to sound realistic. The ear is insensitive to low frequency sounds, so they require much more power than many systems can deliver.

Would love your knowledge on this,

Thank you very much

Tom

My pleasure. I don’t have all the answers, but I do have some educated guesses 🙂

Bob

Message: Hi Bob, I have your book and am a fan of your work. Question: What is a compression disc? My understanding its the name given to a disc between the track recording and mixdown steps, for evaluation purposes and pre-overdubs. Is this correct? If so what processing would you expect to be used?

Thanks,

Pete

Dear Pete:

Thanks for your kind words.

I’ve never heard the name “compression disc”. Honestly!  What might be sent to artists/producers between tracking and overdubbing and before mixing is a rough mix. Sometimes they might try to “help” the rough mix by feeding it through a compressor so the producer can hear low level stuff, but it is a dangerous path to take and should be done with skill and experience.

In addition, I’ve been fighting against a certain practice currently practiced by a lot of mixing engineers. They create a “pseudo mastered” product by feeding their mix through (often heavy) limiting and then sending that to their clients. If they aren’t careful, it can paint the mastering engineer into a corner.

I always suggest that mixing engineers send their decent mix to their clients and explain it has not been mastered. There are other factors to discuss, such as including pressures on the mixing engineers, which tend to push them to do things which they don’t want to do. But this is the basic answer.

Hope this helps,
Bob

Hello Bob,

Thank you for all the amazing information, and everything you do.
Sorry for my english

Recently i went to two mastering studios with my mixes for hearing the point of view of the engineers and to listen in better rooms than what i have.

They noted some problems in the mix, and showed me how to solve them and to listen to them.

So, i took it back and fixed the problem in my mix, 
then, they said the mixes were fine, and now the only thing missing is compressor & limiter.
Both of them practice the loudness war beacouse its their job, mostly Dance and Rock music, and i respect them for that,
BUT, I dont want to compress my music.

When i set the mixes to peak at -0.1 dbfs, the RMS was -13 on one track, and even -17 in another.

Can i release it like that??
Is that ok, if it sounds good and the mix doesnt have problems??
Thank you very much!

Hi from Bob. RMS level is only one measure of the song. How it sounds and how it dances is just as important. Knowing the RMS level without also knowing the crest factor tells you nothing. Apparently the crest factor (measured using flat RMS) of your tracks is between 17 and 13 dB, which measurably is excellent. But of course crest factor just gives you an indication that you’ve got some snappy transients, but it doesn’t tell you about the punch and the sound. You need to listen, to evaluate the liveliness and movement of the track to your ear, its balance, rhythm, and the arrangement of the song itself. How it sounds on the dance floor and translates to different types of speakers. I suggest you get a true loudness meter that conforms with the EBU standard, like the TC Electronic Radar or the Grimm LevelView, or, when it is released, a K-system meter that is weighted to the ITU weighting.

In answer to your question, in these days of the loudness race, you probably cannot successfully release a dance track whose average program loudness (measured on an EBU meter) is -17 LUFS, simply because it will confuse the hell out of the DJs, who don’t understand the issues much at all. They’ll think it’s way too low. And competitively speaking, it doesn’t have a ghost of a chance in today’s competitive environment. But all is not lost! First of all, if it doesn’t dance because it’s so hot that there is not enough bounce or crest factor, it’s even worse than being too low. A good mastering engineer should be able to make a “reasonably competitive” dance track with an average loudness of around (I estimate) -12 LUFS that won’t be rejected and dances and bounces better than anything hotter than that. Let me know if you find a DJ who rejects a great-sounding track with -12 LUFS average loudness, I still think there’s plenty of room for that in the dance field. I hope I’m not wrong, I’ve had success with it (so far).

A poor mastering engineer would ruin the track no matter what the RMS if you get my drift. Hope this helps,

Bob

My comments are: kudos to you for your compression page, most fascinating reading, i really enjoyed it. helped me a lot. i’m new at all of this and am wondering what you think of James Horner’s titanic soundtrack overall, like in terms of compression and stuff. do you think it’s too soft (is it classical or pop produced, i can’t quite tell)? i saw his zorro soundtrack listed as a recommended album but i didn’t see anything about titanic so i’m just wondering. any comments? is this an album i should be comparing my finished film soundtrack mixes to? please let me know, thanks in advance for your response. your help is greatly appreciated.

Hi, many thanks.

As I mention in my articles, it’s hard to compare any mix to a finished, mastered CD, more on this below.

Honestly, I haven’t listened to the Titanic soundtrack album, which is the only reason it is not on the list. But given that everything that passes through James’s hands in recent years gets loving care, I would suspect it sounds as good as Zorro. The Zorro soundtrack is on Sony Classical, I don’t know what company has Titanic. That can make a difference. But there is a “pop vocal” on the Zorro Soundtrack, which kind of sets the stage for how it should be mastered, and on the Titanic Soundtrack, there are pop songs and ethnic Irish music. What pleases me very much about the Zorro soundtrack is that you can listen all the way through and get to the pop vocal and not have to jump your volume control…it is properly and beautifully mastered. If Titanic is properly mastered, I have to assume that the pop vocal, the “Irish Stage band” and all the classical music are proportionalized just right.

I suspect just a small but nice amount of compression/limiting was done to the Zorro soundtrack to make this trick work, and similarly with the Titanic. I have done a lot of work of this type, and it is a real art to integrate the two genres. I often have to hype the classical a bit, compressing it a touch so as to make it fit with the pop cuts on the same CD. The best description I can give of this type of mastering is “midway between pop and classical”? The classical is hyped a little bit, but not so much as to disgust a classical ear, and the pop is perhaps a little less hyped than typical pop, but not so much as to disgust a pop producer’s ear.

Does this make any sense?

So, for you to listen to the Zorro (or Titanic) album and compare your own work, which is unmastered… to it…could be difficult. What I would do is assume the mastering engineer (make it be me…nudge nudge, wink, wink) will properly hype what is necessary. I would listen to classical albums that sound good and compare them with your “classical” arrangements, and to pop albums (as in the ones I cite) and compare them with your pop arrangements, and then assume that the mastering engineer’s capable hands will help to marry the two elements. I would not do anything special in the mixing to make them fit together, other than to mix everything on the same set of reliable loudspeakers.

Best wishes,

Bob

From: “Jerry Gerber”

Dear Bob:

Thanks for putting together a really informative and well-thought out web site. I have been perusing the various articles about mastering, digital audio, etc, and find your candor and point of view refreshing. I especially like what you have to say about compression and the “loudness war” which, obviously, no one can really win.

I myself have noticed the decibels creeping up over the past 5 years or so in film soundtrack work especially. I swore to myself that the next time I go see an action or science fiction movie I am bringing ear plugs!

(especially for the trailers).

Thanks again for the information.

Jerry Gerber

Personally I find a well-engineered “effects movie” when reproduced at the Dolby standard in theatres, has a very satisfying loudness, but I thoroughly agree the current practice of overcompressing trailers to get attention has gotten way out of hand. Do they mix those trailers on 6 inch speakers to avoid blowing out their ears? Unlike commercials played on a little TV, a movie theatre sound system can do some damage… As of about January 1999, movie trailers have calmed down quite a lot, thanks to the efort of Dolby labs and the SMPTE. Trailers are advertising, but films are art.

From: Richard Webb

Greetings,

I’m quite impressed with your revised article on compression, Bob. Quite a few examples which should help educate the public. I thought it was a good piece before, but the revisions have made it even better. Examples from current films help make the point to those who aren’t familiar. You wouldn’t believe how many times I have to explain to a layman that the commercials aren’t really louder than the shows.

During the last week, I’ve read your article and a piece from Fletcher’s (Mercenary Audio) web page which was entitled “What have they done to my art?” and one of the answers to his questions in that article has to do with over compression directly. When the buying public knows better, they’ll demmand better, and we can’t blame radio for all of the problem, as many folks don’t listen to radio these days, except for news and other information. STill, we mix for radio, and that’s where the education needs to start.

Do we mix for radio? I’ve found you can get a very good sound on the radio if you just mix for it to sound good at home. This has been true for many many years. People seem to have forgotten the sound of great records from the 50’s through the 80’s. Are we growing a new generation that’s somehow gotten used to overcompressed, squashed sound? There are, of course, certain considerations that one should think about to obtain hit quality on the radio. Musical arrangements that are less “dense” tend to sound better on radio and on smaller systems. This is only because the sound tends to get “confused” in a smaller box.

I’ve already had some musicians from a band I’ll be working with next week read it. I think we’ve changed their views. Little by little, we’ll educate our clients, one at a time.

A must read for everyone in the business.

I’m sure copies will come off my printer and fall into the hands of more than one musician who doesn’t have web access.

Thanks again.

Regards,

Richard Webb

Electric Spider Productions

Fabulous! Taking it one day at a time, also, here. I just mastered a “heavy metal” album that’s heavily compressed, no question about that, but the transients have a lot more impact than many metal albums I’ve heard. Even giving it one or two dB more peak to average ratio gives even this extreme genre a little more room to breathe. (Yes, the musicians and client were happy).

Don’t grow asparagus, Bob, please! (Some great audio encouragement, in response to a facetious threat I made on a day that I was depressed with the state of the industry, and that I would do better quitting the audio business and grow asparagus)

Date: Wed, 16 Sep 1998 14:41:58 +1000

From: Jodie Sharp

Subject: Don’t grow asparagus!!

Hi Bob,

Greg Simmons here, who STILL hasn’t sent you copies of AudioTechnology Issue #2 with the Chesky story in it. I’m really really sorry about that, we’re finding it very tough financially at the moment here in Australia, and every cent counts, but I will get some copies off to you ASAP. Not very professional of me, I suppose, but that’s how it goes if I want to keep the quality up. Talking of keeping the quality up, this brings me to your ‘depression’.

Bob, I listen to lots of recordings by lots of engineers, pick them apart and so on. I’ve got some really good monitors (a pair of ATC active SCM20s with Super Linear technology). They’re kind of portable, I often use them to demonstrate good recording and sound quality to my sound recording students when I lecture. Guess what recordings I play more than any others in these demonstrations? Older Chesky Recordings which you have engineered and mastered. And they always produce jaw drops from the students (especially when I tell them they’re probably listening to a recording done with a single stereo microphone and perhaps one or two spot mics, no compression, no EQ, nothing). This stuff INSPIRES students. You’re a bit of a legend in my classes! In a nutshell, Bob, you have become my reference for good engineering and mastering. FWIW, I reckon you’re one of the best around.

This compression thing will go it’s own way and find it’s own place, for better or worse. Perhaps you could install a ‘radio station simulator’ in your studio, something that emulates the type of compression broadcasters use, so you can say “THIS is how it will sound when played on the radio”.

But whatever you do, don’t give up. Keep fighting the good fight, and keep publishing your insightful and educational ‘papers’ on your website and in magazines. Oh, and please cheer up.

– Greg Simmons

Editor, AudioTechnology magazine

BTW, if you DO decide to grow asparagus, put me on the mailing list. I’m sure they’ll be the best asparagus I’ve ever tasted. But you’ll be wasting an awful lot of audio talent in the process, and that would be a shame.

With encouragement like the above letter and the following, I stopped threatening to grow asparagus and went back into my education campaign…

From: John W Deacon

Subject: Depression and stuff

Hey bob, just read your note, (via dave martin), that was posted to rec.audio.pro. I guess what you’ve gotta decide these days is whether you just blindly master the things for record labels, or master something with your own personal preferences. I have many arguments with the idiots here in austin, who always advertise things like, (“we can make your cd 3db hotter than anyone else). This seems completely absurd, i have taken mastered tracks from local studios, and in 90 percent of the cases i am finding after listening that i will not accept the job. I have been in the music industry for about 25 years now, and you’re right, the musical side of things seem to have disappeared. Its now loud loud loud and louder, no dynamic range, i look at the files of these things and they look like a straight line across the screen, go figure.

I guess thats one of the reasons i only do mastering now for things i am producing/playing on, and bands who appreciate the musicality of their recording. I just finished a song for polydor, that has a very minimal compression on the master, (not quite even 2:1 on the low bands, with 4 band comp), and it sounds so much more punchy and alive than any other recordings from here. The band were completely happy, another band echo juliet, who i have just finished tracking their 2nd album, will maybe not even have that. I think the art of listening to music may have disappeared. We were taught music appreciation when i was learning, (arrangements etc etc), and how to listen to music, this is not done now, it seems one big technical exercise, based on computers, in fact many “engineers” i talk to here only work in studios so thay can play with the software. This is CRAP……. I dont have a dmaned computer in the studio until i have to write to the cd, and the whole quality is so much better its frightening,

We all feel like you Bob, (well us guys who are a little more seasoned, So keep up the good work, we hear what you do, and appreciate your work

All the best

John Deacon

Granite Records

Austin Tx

From: Florian Camerer

My comments are: Congratulations!!
I just read Bob Katz’s article concerning Compression in Mastering and I find it one of the most comprehensive and informative sources on the subject around!!

In broadcasting (I am a sound engineer for the Austrian Broadcasting Corporation ORF) we have similar problems: programmes with a high percentage of spoken word (e.g. news etc.) and commercials make our lives very difficult, because they are sometimes heavily compressed to the uppermost level. In the transmission chain we have at least two or three additional peak-limiters which make themselves prominently recognizable in many cases. Also, there existed the unwritten rule in drama and documentary that dialog or commentary should be at peak-level (measured with VU-metering!!), because otherwise it would sound weak after the news or a longer commercial-block. When I started in television I soon rejected this “rule” in favor of more dynamics also in documentaries. I simply refuse to sacrifice any of the carefully crafted dynamics during the transmission. It takes some time to convince directors, but there are some tricks:

First I try to sit in the master control room 30 minutes before “my” program is being transmitted.

I would then – depending on the program – gradually lower the level modestly (to say a figure – 3 dB e.g.when it¥s news), and then, immediately before my program starts, I would raise the level back to the original! That’s sort of guerilla tactics but that is something that I have almost got used to – being an audio-partisan… If you know the guys from master control well, this is possible!! (In the past we had a dedicated person supervising overall audio quality and consistency of level in master control – today we – of course – have automation, and one single person must handle everything; the more high-techy we get, the more low-techy the audio-quality seems to be – speaking of bradcasting!!).

If you are not able to attend your transmission in master control, then you can do something if there is some sort of signation before your program. Try to get hold of it during post-pro and cut it to your program. Then lower the level of the signation on purpose, so that the consumer at home has to raise the volume to experience his familiar sensations.

(Hopefully, the person in master control doesn’t spoil your efforts by raising the level of the signation and sending all the rest into the transmission peak-limiter!!) You then have a similar average dialog or commentary-level as the news-program, but your transients, your explosions, gun shots or musical climaxes have all the headroom and naturalness they deserve!

I usually set my average commentary-level up to 5 or 6 dB lower than 0 dBU rel. That doesn’t seem to be very much, but dynamics in television is a bit more restricted than e.g. dedicated listening to classical music. Initially directors complained “Why do I have to turn up the volume control when I hear our program?” but now they more or less understand my arguments – by the way, I wouldn’t do it the other way. Fight overcompression!!!!!

I also agree with Mr.Katz 100 percent on the general reluctance of using compressors as an automatic means of solving flaws. In the days of moving faders, I almost all of the time am my own, human compressor, I do very careful gain-riding (at the end of words etc.) and together with slight compression when recording (commentary e.g.) it pays off for a natural, open-sounding voice.

Light at the end of the tunnel will be the meta-data capabilities in AC3-authoring, where the producer, better the mastering engineer can provide a set of compression settings like car, living room and “audiophile” room (no compression at all) leaving the choice to the consumer depending on the surroundings he’s in. That can also mean changing eq-settings for listening during the day and at 2 o’clock in the morning – the ominous “loudness” function…

I apologize for my overly long writing – I can get pretty upset when listening to current practices in television and radio…

Anyway, thank you very much for the wonderful resource – it should be read by everyone involved in audio.

Some points I would love to read your comments on: EQ, compatibility (mono-stereo-surround, tube revival, audio restoration;

All the best from Vienna,

Florian

From: John Glynn

Dear Bob,

By the way, are you the author of that inspiring essay on Compression? It would make a worthy additional book to the bible, maybe just after revelations. I have clients who need to read it NOW. I need to read it AGAIN. Actually I have much reading and recording to do, my stuff still sucks but it is improving. I want one good recording in my lifetime. I will consider anymore then that to be pure gravy. (Give me a break, I just saw Schindler’s list…)

Speaking of movies, I just went and rented Fugitive to listen to the overly compressed loud stuff you mentioned at the begining. My ears are not as tuned in as your own so I would not have picked that out on my own (yet) but in retrospect, if when the bus turned over the sound of the crash were noticeably flat and lifeless, even hauntingly quiet (just dry bending metal, no echo) then the sound of the impact of the train smacking that bus as an audible apocolypse would have been so much more thrilling. It would seem I possibly have more listening to do than reading or recording…

Ok, on to the Titanic. Think the dynamic viriaty will have survived the move to video?

Again, thank you for the quick response.

All the best, John Glynn
 
Dear John: Thanks for your nice comments. I listened to the other day (and watched) the THX-sound Blu-Ray of Titanic on my 5.1 super system and it was as impressive or more impressive than I recall the theatre. Wonderful dynamic range and clarity. I haven’t checked out the Fugitive, I fear the worst.

Take care,

Bob

From: “George D. Graham”

Dear Bob:

Thanks for the message. I just spent a while reading your excellent piece on compression from a mastering engineer’s view. Very well put! If you would not mind, I would like to put a link to your site from my compression article.

It really is discouraging hearing the state that CDs have gotten to, in terms of ruined dynamic range and “fatigue” factor. As a Public Radio music director (as well as a recording engineer/producer), because of the glut of relatively worthwhile new releases competing for airplay, I have the luxury of taking audio quality into consideration when choosing material to program on WVIA-FM. If given a choice between two CDs of roughly equal musical merit, I’ll go for the more “open” sounding, less compressed one any day, even over a record that might be a bit better musically. Even with the Orban Optimod in our on-air program chain (set with light-to-moderate compression, by broadcast standards), a relatively uncompressed CD really does sound better on the air, and it more in keeping with the station’s “sound.”

A good example is the recent Kate Rusby CD on Compass Records. It’s a wonderful Celtic/English record with acoustic instrumentation and almost no dynamic range, no ebb and flow, all squashed and wimpy. I like the music a lot and it’s the kind of thing our Public Radio listeners really go for, but the sound of it is very unpleasant, so I have pretty much given up on airplay long before it would have run its normal course.

That’s an example you can tell your customers.

I also often raise the dynamic range issue in my album reviews.

Best regards,

George

George D. Graham

People, I suggest you take a visit to George’s site to see his point of view and essays on the subject:

World Wide Web: http://georgegraham.com  
 
Where he has an essay, a plea for dynamic range on CDs.

Best wishes,

Bob

From: Mark Gemino

Bob,
Finished reading more of the information on your web (Boy so much there to comprehend!).
And it raised a few more questions. I will primarily mix to my 1/2″ Analog deck after I find some GP9 tape and rebias as you suggested.

I think that’s an excellent idea.

But i would like to back that up with a digital copy as well. I have access to a Protools V Mix system with a 24 bit (new) Adat bridge . It has a AES in/out and stereo monitor jacks as well. Would this be better than using the 96Khz finalizer @48Khz and stereo dithering to 16 bits on the TASCAM DA30-DAT, that way i could keep the audio files in @48Khz /24 bit on a hard drive stereo AIFF file type? (in theory).

Definitely store at 24 bits for archiving and retaining the resolution.

I think that in your case I would like to receive 48 kHz/24 bit Pro Tools (preferably stereo interleaved AIFF or WAV) files as well as the analog tape. The Pro Tools files represent a pristine exact digital output of the board, and the analog tape represents a colored (possibly more “beautiful”) output. One or the other will sound better through the mastering process, and we will make the best decision which is best.

I do not have a bitscope or oscilloscope to measure what Protools does and to compare each i have to power down reconnect/swap and then listen?

If you connect as I described, you should be in pretty good shape. Keep all faders in pro Tools at 0 dB, turn off the dither in Pro Tools (and if you have Pro Tools HD, use the Surround dithered mixer). Make sure if you do use the Finalizer for any interconnections, that it be in total bypass.

A) ADAT MDM’s out into Optical Digital in of TASCAM TMD4000 console (set to 24 bit word / no Dither) AES out to Digidesign 24 bit ADAT BRIDGE AES input and out the monitor outputs to the 1/2″ analog deck.

I think the D/A converters in the Finalizer are superior to those in the Digidesign monitor outputs. If possible, for the analog tape, go out the 24 bit AES mix output of the console into the Finalizer (set to bypass) and the Finalizer’s analog output to the 1/2″. The Finalizer also has tone generators. I don’t know if it does more than 1 kHz, I’ve forgotten. But at least it can do 1kHz at -14 dBFS to get you started. For aligning the 1/2″. You still need a technician with a phase oscilloscope and ideally a distortion analyzer for aligning the 1/2″.

And feed the 24 bit AES output of the Finalizer into Pro Tools. Not too bad, eh?
(only issue here is i don’t know much about setting word length or Dither but from
what i read you might suggest word length of 24 or more if available and no Dither.

Correct!

D) Rent a super D/A converter which is well regarded and forget about the Adat Bridge /Finalizer?

If possible… Something like a Prism, Weiss, or even in the consumer domain, a Mark Levinson, or Muse.

I understand your 83 SPL reference for monitoring but electrically i am still confused… you said -14dBFS (console 1Khz) equals 0 Vu on ATR 60 1/2″ deck

This is for the level recording onto the 1/2″ and is secondary to the monitoring consideration.

(the TMD4000 can be set internally to +24dBu /+20dBu default/ +15dBu for Analog CR/Stereo outs —would this effect AES levels in any way?)

Not at all. And since you’re going to avoid the analog outputs of the console they become irrelevant, too.

a) Select 24 bit Word length & No Dither on TMD4000 @48Khz sample rate Digital console output.
b) Turn on tones/ Internal to TMD4000 console select (1Khz).
c) Set the level on the output buss level meters to read -14dB with main fader at the 0 (zero) position. (AES output).

An accurate meter is very critical. If you want, you can make me a CD-ROM of this test tone as set by the console… feed it through your chain and into Pro Tools, then send me a CD-ROM of this test tone from the console and I’ll let you know if it made it.

d) On Finalizer select 48Khz /16 bit and stereo Dither check the meter reads -14dB (AES input) & L/R Analog out to 1/2″ deck.

DO NOT DITHER THE OUTPUT OF THE FINALIZER! SET THE FINALIZER TO 24 BITS (AND BYPASS IT, THAT IS, SUSPENDERS AND A BELT) AND FEED ITS ANALOG OUTPUT TO THE 1/2″ DECK.

e) Set ATR60 L/R input controls to read 0 VU on 1/2″ deck ( use Quantegy GP9 tape at 400 nw/M equals 0 VU).

Correct. You’re getting it! Let your technician give me a call if he has any questions about what I mean by 400 nW/M = 0 VU. Aligning an analog deck is what separates the men from the boys 🙂

f) Check DA30 DAT reads -14dB (AES input). AES connects out to CD input

I see why you want to dither the outputs of the Finalizer… to feed the DAT. No, don’t do that. Make two mix passes, one with the Finalizer set to 16 bit dither, to feed the DAT, and the second pass with the Finalizer in Bypass for its analog outputs to feed the analog tape and its digital outputs to feed Pro Tools. I don’t even think you need to have the DAT backup if you have Pro Tools.

g) Check CD RW5000 reads -14 dB (AES input).

Digital is “absolute.” If you are feeding -14 dBFS out of the Console, it will automatically be -14 on all the digital recorders and Pro Tools. Provided the console has an accurate test tone generator.

Clock concerns: i normally use Adat sync to digital console @48Khz sample rate , should i be using word clock and tie this to finalizer/adat bridge if used?

Good question. You really are thinking! In this case, if the Finalizer locks well to the console under the following circumstances, then you will have the LOWEST jitter if the Finalizer D/A ***IS*** the master clock FOR THE ENTIRE LASHUP:

The Finalizer has the ability to be the master clock (internal). If the Finalizer has wordclock outputs, then feed its wordclock to the console and let the console slave to it. If the Finalizer has a spare SPDIF output and the Console can slave to it, then slave the console to the Finalizer’s SPDIF, so you can feed the Finalizer’s AES output to Pro Tools and slave PT to the Finalizer.

This is a complex patch and you have to make sure there are no glitches or clicks, but it is the absolute best patch from the point of view of getting the best sounding analog output from your D/A converter (in the Finalizer).

next connect cd player to playback pink noise and set console CR /PB for 83dB spl w/ one speaker at a time (not sure if pink noise is narrow band or not?) using Radio shack level meter no weighting if this is a comfortable level to mix in… begin to mix

I said that? Depending on how much compression you are using in the mix, this will be an uncomfortable level for mixing. Likely you will have to turn down the monitor about 6 dB. If your monitor is turned down more than about 6 dB from that reference, then you are likely over compressing.

listen back at different PB levels
w/ speakers:
A) HS824
B) NS10’s
C) Urei 809’s
D) AKG headphones

Good. The reference speakers, the merrier, but know which one is your standard and determine if it translates to the rest. In your list above you don’t have anything I would consider a “standard”, so try to include some wider range, accurate speakers in your listening.

play Cd on:
A) Sony player in kitchen
B) Sony Disc man w/ Sony MDR headphones
C) Apple disc player on computer
D) CD player connected to Console

Even more important, a real good, big hi fi system in someone else’s house, someone you know and trust.

Hope this helps,

Bob

From: Franklin Kiermyer

Hi Bob, Please tell me, If I cange the suffix of a 96/24 stereo ,caf to .aiff am I altering any audio data or should it sound exactly the same? Thanks for taking the time to answer this question for me. 

All the best, 

Franklin Kiermyer

Dear Franklin:

A CAF file is Apple’s “core audio format” file and you would have to inspect if it is compressed or not. I believe both options are possible. It probably will NOT work if you change the extension. Try a converter like Barbabatch.

Good luck,

Bob

Hey Bob, what happened to your couch?
The couch turned out to be a Helmholtz resonator. I wanted to donate it to the museum of acoustic failures, but it’s now resting quietly (and functionally) in the office.

From: Nick Watt

My comments are: Dear Bob, I’m a newbie to DAW recording, so I hope you’ll bear with me. I’ve read your articles concerning the differences in DAWs. I’ve tested 3 different DAWs. This is what I tried:

1) Open the same group of 16bit audio files and pan them hard left/right (to feed the inputs of the Yamaha DSP Factory card).

2) Set the internal resolution of ALL programs at 32bit resolution (the highest in every case) This are my personal (admittedly subjective) observations: 1) BrandX sounds the HARSHEST. Almost brittle and very FLAT (no dimension) 2)BrandY sounds the CLEANEST and MOST HI-FIDELITY (for want of a better word) 3) Brand Z is somewhere in between the above two programs My question is this: (1) Why is there such a difference in quality? Aren’t all programs using high resolution internal processing? (2) What is the effect of something as “innocent” as panning have to do with expanding wordlengths and the like? Sorry for rambling, but I amreally puzzled. In any case, thank you for your selfless contribution in writing these articles.

Hello, Nick. It’s a good question. If you were testing for perfect clones, the first thing is to get a bitscope.  In lieu of a bitscope, a free plugin by Stillwell audio called Bitter does a good job. It appears to me that you’re trying to see if the program is altering the files when you think you are telling it not to (that is, pan left right, do not change gain, do not equalize, etc.). Why not test to see if the program produces a clone by the methods outlined in my “more bits please” article, and also *get a bitscope!*

First things first: Find out if the program altered the data? Then we go on from there…

I’ve found that something as simple as Panning may involve changing gain if the program doesn’t do things the way you think! And, as per my dither article you know that changing gain involves are calculation. For example, the Yamaha consoles have unity gain at the center pan position, and +3 dB at the sides (panned). But most other DAWs make it unity at the sides and -3 dB in the middle (more logical to me). To get a perfect clone of a stereo source through a Yamaha system, you have to put the pans to the middle and assign each channel to only one output (L or R). As well as leave the masters at 0 dB. Thus, you need a bitscope to protect yourself from console designer’s bugs, or maybe your own assumptions…

The ultimate proof is to do a null test on your data. If you pass it out and load it back in, null them out, and don’t get a complete null, then the data has been altered. Now, since you are using the same hardware to do your auditions and only comparing software, I’ll eliminate jitter as a possible cause of the differences you hear.

Hope this helps,

Bob

From: Derek Casanares

Subject: Re: DAW Suggestions

Hello Bob,

…How using plug-ins effect your final product, using a combination of recording direct to hd and tape – best of both worlds.

There are some very good plug-ins. The weak link is inexperienced engineers who do not realize the limitations. For example, you can push an analog compressor very hard without getting harsh artifacts, but inexpensive and poor plugins simply cannot do so. As sample rates get higher, the artifacts of strong digital compression become more tolerable, but really the key here is moderation. An experienced engineer who has been mixing with the best of analog equipment for, let’s say, 5 to 20 years, could get good results mixing with plug-ins because he uses his ears, doesn’t push the plugins past their limitations.

I have great sympathy for inexperienced engineers attempting to do so, because your early products will not have the dimensionality, warmth, depth, impact, and other qualities that can be obtained from a good analog mix. It takes a whole new set of talent and understanding of the benefits and limitations of a digital mixing system for each type of music. Use good plugins, avoid the ones which make your mix sound smaller. I’m not saying it can’t be done, I’m only saying it isn’t as easy as it looks.

Also, I’m trying to decide if it’s in my best interest to butan analog
board for my DAW and get on of the new digital mixers.

Read “More Bits Please.” Digital Mixing has come a long way, both in the box and with external mixers. The limitations are often the quality of the equalization and compression algorithms, but the basic mix engines are usually high enough resolution to get good mix results. The included reverbs in the inexpensive digital mixers are usually poor to medium quality. In short, supplement your digital mixing, whether in the DAW with plugins or with a digital mixer—-with good outboard analog and digital gear and know how to use it to its best advantage and I think you will not need to invest in an analog console.

In my book, second edition, I cover the issue of analog versus digital summing in more detail.

Bob Katz

Well, I really was thinking something along the lines of, exactly is the
theory behind it? I know that dc is 0 hertz, and I know that dc
offset is when the soundfile is not centered around the baseline of the
waveform.

You’re correct! The way I determine if the DC offset is severe enough to have to deal with is if you hear a click or pop when starting or stopping the material during a soft passage. If you don’t then I would say ignore it as the cure can be worse than the disease.

What is the prefferred method for adjusting DC offset?

Very rarely is DC offset a problem in the first place. So I think the most successful and transparent method is not to do anything! However, if you have a DC offset problem that’s large enough to cause an audible pop when you start and stop the material, then the most successful methods I’ve used have been to apply an extremely-high quality linear phase high pass filter. Why linear phase? Because to my ears, the phase shift of a severely-applied minimum phase (standard variety) high pass filter is audible. Same as a capacitor in series in an analog output circuit can be heard by keen ears with a good monitoring system.

In order of preference, the best unit I have had for this is the 13.8 Hz high pass filter in the Weiss EQ1-Mk2 (in linear phase mode). Assuming that there is no extra low frequency information in the material that we want to keep (and sometimes there is, even at 13.8 Hz). The second most successful has been the high pass filter in the Algorithmix Red and the only reason it has been less successful than the Weiss is (currently) the Alg. red starts at 20 Hz; though it can be made steep, it sometimes eats into important low frequency information we want to keep (though this is not usually a problem).  Lastly, in Sequoia, the new FFT filter does a good job at this as well.

Hope this helps

Bob

John MCkinney wrote:

Thanks for the info on your site. Would you be kind enough to tell me what type of file a mastering engineer actually burns prior to pressing, and what is a DDP?

Dear John:

Thank you, I’m glad you like our site.

The first step in producing a master is creating a file or set of master files that (around here) are initially at 96 kHz/24 bit, downsampled to 44.1 kHz/24 bit, then dithered to 44.1 kHz/16 bit. Usually in a WAV format, but arranged in an EDL in SADIE or Sequoia for cutting to a master. At that point we cut to a DDP file or to a CD-A (a CD Audio Disc, in regular language). If to a CD-A, then we error-test it and produce a PQ list and seal it up for the plant. If for a DDP, then we cut what is called the “DDP fileset” which is a set of special files that the plant can use to cut a glass master. There are several files in the fileset, one of them contains the audio and the others contain PQ code, CD Text, and other information. The DDP fileset is then either uploaded to the plant by FTP, or placed on a CDROM or DVDR and sent to the plant, which copies it to their hard disc system and then they can cut the glass master directly. We also include an MD5 file to prove that there have been no errors in transmission or copying.

Hope this helps,

Bob

From: Sal Vito

My comments are: Bob, I read your review of the Weiss POW-r dither unit… fascinating!

Thanks, Sal.

Do you know anyone who is implementing POW-r in 24-192? Ideally I would need a direct-x or VST plugin… If not, is there any program that would work on a pc that can do this conversion? A hardware box would also do, I guess…

I think that the advantages of POW-r pretty much become insignificant at larger wordlengths. A simple high pass dither is suitable for reducing greater wordlengths to 24 bit.

In addition, if you’re remaining at 192 kHz, who would want to reduce the wordlength? What application would you conceive of where you would want to reduce the wordlength and stay at 192k? Every application I could think of would first be to sample rate convert to 44.1 K, and finally reduce the wordlength (dither to 16 bits) for compact disc.

What kind of dither do you guys usually use to convert 24-192 to 16 bit 44.1? Just curious…

First things first… you have to sample rate convert. The SRC I use is the Weiss SFC2. Say the output is a 24 bit, 44.1 kHz file. At that point, I usually use the POW-R dither to bring the signal down to 44.1/16. Always dither to 16 at the last step, never in any intermediate step.

Hope this helps,

Bob

From: David H

I do not want to take advantage of your help, but… I’m also trying to get a better understanding of dither. The “Secrets of dither” was excellent, it really cleared up a lot for me. However, in trying to get a grasp on all these digital theory concepts, I try to summarize the concept so I keep it straight in my head. In a nutshell, how does dither increase apparent dynamic range. How does it make soft sounds more pleasing? Is there an analogy that can be made to analog tape machines?
Also, could you please provide simplified answers to the following questions: What are the effects of jitter on signal reliability and noise floor. How do you reduce jitter? What benefits are there to using a master word clock generator (such as the Aardvark Aardsync) to synchronize an entire digital studio? Thank you for all the time and effort you put forth in educating the masses! I promise these are the last questions I’ll bug you with. 🙂 I will of course keep reading and re-reading your informative site!

Dither: Increases dynamic range by adding a small amount of random noise to the long word source BEFORE truncating to a lower wordlength. The low level material rides on this noise and produces the same average level that any low level material would on an analog noise floor. For example, in a totally analog system, let’s say that tape hiss at a certain moment in time is 1 millivolt in level, positive going. And that the audio signal at that same moment in time is 2 millivolts, positive going. The sum will be 3 millivolts, positive going.

If you perform the same operation in digital, creating a random noise and add that to your signal BEFORE truncating, then the sum at that moment in time will still be 3 millivolts. And that can be encoded to the lower wordlength. At other moments in time, of course, some low level signals added to the noise at a given moment in time will be below the minimum level of the lower wordlength. BUT ON THE AVERAGE, the dithered result will be the same as it would have been in an analog system with the same noise floor.

In my book, Mastering Audio, I diagram this in detail and a picture is worth a thousand more words  :-).

Jitter, in short:
If you already have made a conversion (A/D) and have it stored digitally, then jitter in your system only affects the listening. The apparent noise floor is higher during signal, it can sound harsher, and less deep (spaciality). This is a function of monitoring in the D/A converter.

On playback and mixdown, the best way to reduce jitter is NOT TO REDUCE it, but to obtain a D/A converter which is immune to its effects! Such as a Prism converter, which is known to be extremely immune.

During recording, jitter can permanently affect the recorded signal, and the best way to keep it down is to use an A/D converter on Internal sync.

The benefits of word clock are to have a continuous, stable, timed signal feeding all the gear that can be slaved. However, the central “generator” should be your A/D converter on INTERNAL sync. For word clock distribution, let the A/D be the master, then have its AES or WC output feed the word clock distribution, and everything else slave to the WC distribution.

Again, I refer you to the book for a more detailed explanation with diagrams to help.

Best wishes,

Bob

From: Tim Ashworth

I have a simple question that I can’t find a concrete answer to and thought you may be able to shed some light. Should dither be applied when converting 24 bit lossless source (FLAC) to 16 bit lossy output (mp3)? Thanks if you can shed some light on this topic.
Dear Tim:

Normally, as long as wordlength is being reduced, DITHER MUST be applied.  But mp3 (as well as AAC) is a special case because the LAME and Fraunhofer encoders work in 32 bit float in their encode side, and accept up to 32 bit words without penalty and without need for early reduction. They then use all the information you send them (24 bits in your case) to produce the best mp3 that they know how to produce, and internally take care of getting the most resolution encoded into that final 16 bit mp3 word. In fact, although the mp3 word is a 16-bit file, when playing the mp3 it will decode to full 32-bit float. It’s still lossy, but it’s not as lossy as you might think!

In other words, an mp3 made from a 24 bit or 32 bit float source file has the potential to sound better than one made from a 16-bit; you may perceive more depth or better resolution, if you use the highest bitrate mp3 you can make. You may not even hear a difference, however, if you make only a 128 k mp3…

Don’t forget to convert the lossless FLAC to a 24 bit wav first! For more information on this subject, I suggest you read my book: iTunes Music.
Best wishes,

Bob

Galen writes:

“Dither and Sample rate conversion question- 

Reading Bob Katz’s book, mastering audio. 

I want to clarify- That I have heard from 2 independent sources, the Bob Katz book, and Dan Worrall, that one should use 24 bit dither when bouncing to 24 bits, because it is a wordlength reduction, coming from the daw’s 64 bit floating point. If I am incorrect I would like to know but I believe those of you saying dither only when converting to 16 bit are incorrect. If I am misunderstanding Dan Worrall and Bob Katz please correct me. 

I have a project which I was mixing, the file is 24/96. 

I bounced the project to one stereo file, also 24/96, and used goodhertz’s good dither as the last part of the chain on the output, to dither at 24 bits. 

Now I am mastering this song, and I am trying to clarify in what order I should ideally do SRC to 48khz and 24 bit dither, to get 24/48 master. 

I am reading in the book that Dither should be the last stage, even after SRC. 

In Izotope RX8 I can do SRC and also host audio units, so I was thinking I could do SRC in RX, then follow with a 24 bit dither, then export directly from RX. But I have mostly exported from Logic. 

Anyway, if there’s something else I should be doing ideally, let me know.”

Dear Galen

Hi, Galen. My book explains that wordlength grows with any multiplication operation. SRC uses a ton of multiplications, and the SRCs on the market are all floating point and their output wordlength is mostly 32 bit float with some 64 bit. No matter what source wordlength, e. g. 16 or 24 bit, you’ll get 32 float out of the SRC, or even a simple gain change in a floating point DAW. 

So, if possible, leave it at 32 bit float till you are done with ALL the operations in line, until the final wordlength reduction. 

How audible is a truncation to 24 bit? Let’s just say “audiophile-level audible”, which is quite subtle. Typically Some slight loss in depth or soundstage width. People say, if that’s the case why bother. No one’s gonna notice. 

But keep in mind that distortion is cumulative. A little loss here and there can add up, particularly if the end product is lossy coded. And being technically correct doesn’t take much brain power once you learn the simple rules. 

Lastly, I will say that going to properly dithered 24 bit as an intermediate step is not sonically costly at all, as the dither noise at -141 dBFS is inconsequential. So if you have to deliver, say, a 2444 master to some service, and also a 1644, I wouldn’t worry about using the properly dithered 2444 as an intermediate product, making the 1644 from the 2444, there’s no hill to die on there. 

If you ever do a null test between the 24 bit and a 16 bit result, 9 times out of 10 the null residual will only be dither noise, no musical information or distortion. What’s that all about? Just noise? But that 16 bit dither noise is usually enough to subtly change the depth or tonality by virtue of masking. So choose your dither shape wisely! It’s also the main reason why I avoid making mp3 or AAC refs from a 16 bit source when possible, and go back to the 24 or even the 32 float to create the mp3. Because the tonal shift, the depth losses are even more audible at the lossy stage. Always consider that there are going to be more stages and manipulations to the product AFTER it leaves your hands!

From:     Christopher Hicks

 I don’t know if this is interesting or helpful, but here it is anyway…

Now, take two dice of different colours, red and blue say. Roll them both, and write down the total. Now pick up the red one (leave the blue as it is), roll it and write down the total. Now roll the blue one (leaving the red as it is) and write down the total. Continue in this way, rolling the red and blue alternately, and you’ll end up with a sequence of numbers from 2 to 12. This sequence has a triangular probability graph, but each number is now related to the one immediately before it; if the blue is showing a five or six as we roll the red, thenthe total we get on this roll is likely to be high, and vice versa. If you plot a graph of the successive scores it will look a little smoother than a similar graph drawn from the “white” sequence of the previous paragraph. This is low-pass triangular dither, not much used in audio.

To get high-pass triangular dither is a bit more complicated still. Roll both dice, and write down the blue score (B) minus the red score (R) to get B-R (which may be negative). Now roll the red, and write down R-B. Now roll the blue, and write down B-R. Repeat this sequence, rolling the dice in turn, and subtracting the one you left on the table from the one you just rolled. The probability distribution is triangular again, with numbers around 0 being the most likely, and +/-5 being relatively unlikely. Successive numbers tend to be at opposite ends of the possible range because of the alternating subtraction, and a graph of the sequence of numbers comes out relatively jagged. This is triangular high-pass dither.

P.S. –  By the way, it is a common misconception that “white” and “Gaussian” are synonymous when talking about noise. This is totally untrue – “white” is a noise colour, and “Gaussian” is a distribution, and any given noise signal can be either white, or Gaussian, or both, or neither. You can get reasonably close to a white, Gaussian sequence by taking a large number of dice (20, say) rolling them all together and writing down their sum. Repeat ad infinitum.
 
-Chris Hicks

 
By the way, Chris, can I take you along with me on my next trip to Vegas? Enhancing the explanations of the complex stuff in easy-to-understand ways is what  my book is all about! And your explanation beats the pants off of everyone else. I would include a graph picture of gaussian versus triangular in order to increase the explanation, as in “based on this shape, it is more likely that the middle number ranges of the dice will come up. In other words, it’s just as unlikely to roll snake eyes (two 1s) as it is to roll boxcars (two 6s).”

– Regards,
Bob

Hi Bob,

It’s been a while, I hope you are doing well. Dare I say — I miss our loudness group’s regular calls!

I’m writing to ask for your advice. My colleagues at Vox and I are having a discussion about encoding. All of our podcast content is distributed at 192kbps stereo MP3. The question on the table is — when encoding 24-bit WAV to MP3, is dither necessary?

From my understanding, bit-depth only applies to PCM. Pro Tools (and some other DAWs) won’t even give you the option of selecting a bit-depth when exporting to MP3. So does that negate the need for dither?

I found one source that say that some older software or devices might actually convert the MP3 to 16-bit PCM on playback. In that case, dither is helpful. Otherwise, I’m thinking it’s not.

Might you have any advice to offer on this?

Best,

Rob

Hi, Rob:

No need to dither to mp3. The encoder is 32 bit float. It will try to fit as much of a 24 or 32 bit source into the mp3 encode as possible. Since it’s a coded system, the output of the 16 bit encoded word will look like 24 bits, with about 18 of them psychoacoustically useful, but use them all and treat them like 24 bits in the DAW. 

As far as older playback devices or codecs that apparntly convert mp3 or AAC to 16 bit on playback, your best safeguard is to inspect the output of any encoder with a free plugin from Stillwell audio called: Bitter. Try it… it’s very helpful. if it comes out as 16 bits, get rid of that codec. There is also some confusion since the codec packs the coded information into a 16 bit word, but its effective psychoacoustic resolution is around 18 bits or possibly more when it is reproduced, and Bitter will probably show 24 bits active on the output of the codec, which is normal and expected behavior.

How’s that sound!

Bob

Bob,

Thank you for this. You have soundly settled our debate 🙂

I will check out the Bitter plug. We have two MP3 encoders in use by our engineers — Fraunhoffer and LAME. I hope they both pass the test.

I and the team here appreciate your guidance on this!

Best,

Rob

From: Mark Luckett

Hi friends. I consider your site one of my most valuable resources. Thanks for your work and help. I have a question. After reading your articles on dithering and word lengths I am much more informed about the subject. I recently got a new masterlink and mixed a “practice” mix to it at 96/24 as you recommended. I wrote a 24 bit .aiff file to cdrand transferred the material to pc to edit in sound forge. Did my edits,got ready to save the file and convert to 44,1/16. Opened Waves L1 ultra to dither then suddenly realized that I didn’t know which order to do this in. Should I resample, then dither, THEN run bit converter? Or resample, bit convert, THEN run L1 maximize + dither, I’m really confused here and I want this to be the best that I can do here on my own. Dither then convert, or vice-versa? Please help. Thanks in advance.
Best Regards,
Mark

Hi, Mark. Dithering to 16 bit should always be the last step.
Resampling to a new sample rate should always be the penultimate step. Keep your sample rate and wordlength to the longest possible until the next to the last step, then resample, then dither!

Hope this helps,

Bob

From: John La Grou

HI Bob 
You are on Len Moskowitz’s WWW page, so I guess I’m writing this through the back door.. Your MIX article is terrific. I think it will open a lot of eyes. Did you say that “all” digital gain reductions require a certain amount of dither?

John La Grou

To be very exact, all digital gain reductions (and most other non-trivial DSP operations) require a certain amount of dither if the signal is to be truncated to a shorter wordlength (there is only one correct minimum amount of dither). If the signal is going to be transmitted or broadcast at its original wordlength, then redithering will not be necessary. For example, I start with a 20 bit recording, and I drop the gain by 4.5 dB. This becomes a long word (48 bits or more) within the DSP (for more information on this, see the dither article). Internally, in the processor, this long word should be dithered up to 24 bits. Then you receive it at 24 bits. If you can use or transmit these 24 bits, then so much the better. Otherwise, when you reduce the word to 16 for storage or transmission, you will have to dither.

Hi again Bob,

I am thinking about the dithering and was wondering what about mixing and dithering?  Nobody is talking about it, but we are using all this plugins in mixing with not a single dithering. What is your recommendation about dithering when mixing and using the plugins?

Lets say I am mixing and my session is 24bit 48khz if I use the compressor on my Master bus I should use dithering when exporting/bouncing the mix right? And how should I use the dither, to 24 bit or to 32 bit output?

If I use the eq on my bass I should apply dither after EQ on the bass track etc….

Another confusing thing is if I have 24 bit 48khz  file and I apply lets say SRC plugin with 32 bit float output resolution the file gets bigger and is not 24 bit file anymore but 32bit file.  This is all very confusing.

Thanks,

Nenad

Plugins and wordlength are almost all taken care of by the architecture of the DAW. The signal from the object is going to be in 32-bit format no matter what format the object is. So if you have a 24 bit object and you change gain—presto, that’s 32 bit in the dsp, and this gets passed from plugin to plugin at 32 bit float with no need to dither until you reach the outside world or the need to store it in 24 bit format. So if you are storing the output in the plugin chain for later processing, ideally, store that in 32 bit float. If your DAW prevents you from doing that due to weaknesses in it’s design, then dither to 24 before storing at 24. This could be accomplished, if the DAW does not provide dithering while capturing, by you inserting a dithering plugin last in the chain, set to 24 bits.

Floating point does not require dither because it can express from the smallest to the largest quantity. So if you can save the output of your chain at 32-bit float, then there is no need to dither (yet). When the wordlength is further reduced, you should dither. You can go direct from 32 to 16 in some cases or intermediately through 24 (dithered). Do not dither to 16 until after all the processing has been performed, In other words, sample rate convert first. I hope this is starting to make more sense.

Best wishes,

Bob

Hello Bob,

Just wanted to drop a line and let you know how much I’m enjoying both your book and the website.  I’ve only one question, what program can I use to read the CD booklet templates?  I’m running the latest MacBook Pro currently.

Most of our templates were produced for QuarkXPress with some in Illustrator. But people are increasingly using InDesign as their standard! Until we change our templates over, I recommend an indispensable extension to InDesign called Q2ID, which should allow InDesign to open a Quark document. Sorry for the inconvenience!

From: Dale Bryant

Is there anything to gain by converting my CD’s to DSD using Korg’s Audiogate software?

Dear Dale,

This depends on whether the reproduction equipment you use for DSD is as good as or better than the reproduction equipment for cd. Starting with CD (1644) material, I found that using a $20,000 system consisting of a cd player equipped with a proprietary FireWire output into a DAC which has many choices of upsaming including to dsd, there was a mysterious increase in depth with dsd compared to 192 kHz linear PCM. This could be unique to the brand that I tested, or something freaky about the dsd format. So it sounds different, even with the same source! After “upsampling” to DSD. So your mileage may vary. Use your ears and decide. It’s all in the architecture of the DAC as far as I’m concerned because if you start with 1644 you can’t create new information unless the “new” information is a false reality.

Then again, you may REVEAL more information with different reproduction methods and filters. Which is exactly what DSD is compared to PCM.  It is possible that the apparent increase in depth could be something psychoacoustic due to the out of band noise doing something with your tweeters. Is it biasing them into a more linear region? Or is it adding distortion in the audible which the ear interprets as more depth? More questions than answers. In the end you’re doing it to tickle the ears, and your ears are the judges. I just hope you are not fooling yourself. My standard is PCM, 2496 and above, and that is what I consider the source. I always hear a loss when I reduce it to the CD rate, and I work my butt off trying to minimize that loss during my mastering. But you can’t fit 24 pounds into a 16 pound bag. And PCM is the standard, so DSD is something you and I can play with and enjoy, but it’s not what typical consumers are going to buy.

Also, will recording with DSD and converting to 24/88.2 sound better than recording to 24/88.2 initially. I’m using the Lynx L22 card with a PC and considering a Korg mr-1000 or Tascam DVRA 1000 recorder.

Some people think so. How do you know it isn’t the euphonic character of the initial DSD conversion. To my experience, dsd (the low speed version) softens the sound slightly, a bit like magnetic tape. This produces a more pleasant sound with some types of music. But what if the converters you are using for 88.2 are more accurate but less flattering — in that case you might choose the more flattering route. If you’re getting into these kinds of subtleties, my approach would be to record at 192 kHz, which I find to be the most transparent and accurate, and by choice of preamps, mikes and outboard gear and recording technique, to get the sound I want. Given the limitations of current hardware in track count, make your own choice by careful listening and comparison. And realize that the type of music you are recording, your tastes and your mike techniques will dictate your choices. Anything over 44.1/24 bit will sound bett tha 1644, and slightly different as you go up. 48 kHz/24 bit really doesn’t sound that bad! So I’m not going to dictate your choices for you past 2448 without actually being there and able to audition a shootout with your recording techniques and material.

Personally I prefer to record and master using an accurate format (2496 in most cases around here) and use my outboard processing gear if I need a more euphonic presentation. It suits my philosophy.

Hope this helps

Bob

Is this a substitute for a good 2-track mixdown machine?

Addition: as we don`t have the big money to buy a Studer 2-track mastering recorder, would it be smart to dub to a multitrack machine (instead of hard disk) to> achieve “louder”, more retro/analogue sound?

Or dub the digital mix to analogue tape and back again?

and I replied…

Well, it has to be the right analog tape. If it’s wide-track, well-maintained multitrack analog, then the “cure” may be better than the disease. If it’s the good parts of analog you’re looking for you have to have an analog machine that makes it “fatter” without getting fuzzy.

But for the past 10 years or so, and in the article, I’ve been preaching that good analog tape has higher resolution than cheap digital. If you dub from your harddisk to analog tape, saturating it too much, you’ll get bad analog sound! If you dub to the analog and use its gentle compression characteristic to get a fatter, more full, warmer sound, it can be a win-situation. But it’s not win-win! Because you still haven’t replaced the mixdown DAT machine. Mixing down to the 16-bit DAT from your multi is like taking your hard-earned Corvette, and feeding it cheap, watered-down gas. Please don’t do that… Both attempts are compromises, in my opinion.

In my opinion, if you have the time and the energy to dub it to multitrack analog and then mix then you also have the ability to mix it down to a good 2-track machine. I’d rather you stayed on the digital multitrack without dubbing it, and then mixing to the 2-track analog. The losses are much less that way, in my opinion.

That way, you haven’t lost any additional resolution, haven’t done an extra D/A conversion, and you’ll still have an analog mixdown with the best qualities of analog. And then don’t dub that analog mix tape any further. Send it direct to the mastering house, where it will be transferred with loving care through the best A/D conversion in the world…

After 3 projects, the 2-track pays for itself. Try to work on “cures” instead of bandaids.

Good luck in your search for the perfect sound,

Bob

I’ve just finished reading the updated version of Bob Katz’ – Mastering Audio book. One thing that caught my attention was when Bob makes the following statement in the context of a section about achieving dynamic impact (punch): “Does this mean that a punchy master can’t be produced from a MIDI’ed rhythm section based on 808 kick, synthesized sampled bass and sampled handclaps? Frankly, that’s an uphill climb, it requires a team of engineers with ability and knowledge, and it wouldn’t hurt to have at least one focused lead or rhythm line played on a standard instrument.” I’m wondering what other people out there make of this? Is there really a problem with their sound that can only be overcome by a ‘team’ of engineers? Is there an explanatory theory behind this statement? I’m not making any judgements here – just curious – what do you think?

Thanks for writing. That opinion of mine is based on years of experience and lots and lots of submissions for mastering, and comparing what we get versus recordings made with a larger portion of real instruments. Remember that many samples you use have been peak limited or compressed or inferior technology was used. If you make your own samples, combine real with artificial, and have good ears and talent, you can beat the reaper. I’m just saying that the odds are stacked against you.

Of the recordings I’ve heard, only a small percentage of totally sampled material makes the grade for dynamics and impact. So the deck is stacked against you.

From:mblock

MBlock: Bob,
Would you please help me to settle a debate I’ve been having with a friend of mine? I maintain that technically it is wrong to digitize audio in the -10 to -20 (peak) range. You’re just throwing away bits.”

Bob Katz: Hi, Mark. In general, from the point of view of the A/D converter, more is better, as long as you are not overloading any intermediate analog stage in line in order to do it. And as long as you have an accurate overload indicator. (See my article on levels: Part I)

OtherGuy: “Your interest in making a track as loud as possible would seem to sacrifice dynamic range.”

BK: Making an initial recording peak near 0 dBFS is not the same as making the final mixed production as loud as possible. Making the track peak near 0 actually maximizes the dynamic range of the recording with respect to the signal to (dither) ratio of the recorded multitrack medium. Afterward, in post production, you can alter (lower) the mixed or post produced level of that previously recorded track to any point you wish, so, once again, the dynamic range of the final production is also not sacrificed. Unless I totally misunderstood the poster’s motives, he is wrong.

MB: “No, just the opposite. The dynamic range is the difference (in dB) between the noise floor and the maximum signal a system can tolerate without audible distortion.
(The term “audible distortion” requires a definition in the analog world, but with a digital signal it’s clear cut.) Full 16 bit dynamic range requires peaks near 0.”

BK: Correct, as I just said. But “Requires” is a dangerous word. For example, if I were recording a live performance direct to 2-track with a world-class properly dithered 16-bit A/D converter, and one peak in the entire hour hit 0 dBFS, but the second movement was all at -20 dBFS (the original range of the players), it would probably just be fine. The sound would be world-class. As long as you did not have to raise the gain of the second movement in post production or do any post production at all. The point being that in direct to 2-track of live material, if the original dynamic range is good-sounding, then you have made a proper recording. But if you have to do any post production on it, potentially needing to raise or lower some tracks, then it pays to record the 2 track (or in this case, more likely, multitrack) 24 bit with good (“24 bit”) A/Ds, so that you have more room above the quantization noise floor when getting into post, where you might have to raise or lower gains. More on this in a moment.

OG: “It seems that your rationale forces you to constantly lower the music level in the mix anyway.”

BK: That’s fine, not a problem. By recording “hot”, you have made a good, clean original recording which you can then later lower in level inthe mix or post stage. That’s called “having your cake and eating it, too.” It’s a good idea: Maximizing the record level of A/Ds, especially the cheap, shoddy ones, is a very good thing. Another way of thinking of it is you are increasing the signal to garbage ratio of the original recording, where the garbage is at the bottom of the A/D converter. More on this, as we get into the “24 bit rules”….

MB: “My aim is simply to bring audio into the Avid at maximum resolution, and that means digitizing as hot as possible without distortion.”

BK: Correct. But as ADCs have gotten better it is no longer necessary to digitize to full scale. Regardless, as I said, you are preserving dynamic range if you are simply recording the unaltered microphone output to the ADC at near full scale.

OG: “You are equating bit depth with volume. Are you sure that is really true? By definition, this would imply that a 24-bit system can achieve louder levels and a wider dynamic range. most commercial music is highly compressed to start with and really doesn’t cover this full dynamic spread. As a result, if you have 20dB of actual (mixed) dynamic range with “loudness” that falls in the bottom, middle
or top of the available range, it really doesn’t end up sounding different – or so it would appear to me.”

BK: You are more and more correct for the more modern ADCs recording at 24 bits. Set your average level to -20 dBFS and let the peaks fall where they may. However, the older and more inferior ADCs may benefit from “maximizing”.
MB:My understanding is that digital audio sounds best as you approach maximum level; it sounds pretty nasty down in the bottom of the range.
BK: Shall we clarify that to say: An Analog to digital conversion sounds best as you approach max. level….. (within reason).  On the D/A side and mixing side, it’s a bit more complex, and it is possible to remain clean without needing to hit full scale. Otherwise you could never mix productions with reasonably wide dynamic range that are designed to be reproduced well (gotta sweat the soft stuff….).

MB: “I try to record with pre-mixing in mind. I have the playback levels all set at zero on the console (10dB down from the top of the fader.) I set my record levels so that I have a pretty good mix coming back with all faders in this ruler-straight line. If a guitar peaking at full scale digital is too loud, I can either pull down the record level or the monitor level.

BK: As long as you are talking about the A/D conversion (tracking) situation in 24-bit then I have to agree, you do not have to “maximize” your record side as long as your average forte levels measured with an averaging meter are working at or higher than about -20 dBFS.

OG: “Now… There is no difference between recording the guitar at -12 or pulling it down 12dB in the final mix.”

BK: There certainly is, from the point of view of the sound quality of the original recorded track, in a 16-bit recording, which I think your friend is referring to. As we moved to 24-bit, I would correct that to say, “there is no perceptual difference between….” meaning that there is enough signal to noise ratio in good 24-bit ADCs to allow more latitude in record level.

OG: “And, you are not really losing any resolution. If you are recording at 24bits, even if you record at -40dB you still have 24bits of resolution in the data you are recording.

BK: Depends on what you do with that data and how you gain stage it in the final. If that -40 dB represents a pianissimo passage that is going to remain that way and not be turned up, then you haven’t lost any resolution. If instead that -40 dB represents the peak level you are putting on the 24 bit recorder, then you have lost almost 8 bits of resolution! So I agree with him if his forte passages are at -20 dBFS approximately, average-reading meter.

OG: In my definition of “resolution”, resolution and noise are directly related. The closer the recording is to the quantization distortion noise) level (the lower the recorded level), the lower the resolution of the recording, and the fewer effective bits it has. If I record to a 24 bit recorder peaking at 48 dB below peak level, I have essentially made a 16 bit recording, from the point of view of resolution, and signal to garbage ratio. Resolution has nothing to do with record level. Resolution and noise floor both change values when talking 16bit vs 24bit, but they don’t really have much to do with each other.

BK: My definition of resolution has EVERYTHING to do with record level. You can have a high resolution tape recorder but if the highest peak of the entire program is -24 dBFS, then you have probably lost 4 bits of resolution which you will never get back. If you have to raise it later in mastering it may not sound as good as if you peaked your max. peak to full scale. The difference will be subtle with modern ADCs, perhaps inaudible, but theoretically I would try to avoid making a recording whose highest peak doesn’t reach at least, say, -10 dBFS peak. This is just a conservative application of the principles…. not too low, not too high  :-).

OG: “When you record at lower levels, you still have the same resolution, the only thing that changes is the noise floor. With 16bits the noise floor is always 96dB down from full digital level. If you record at -10 your noise floor is 86dB below the signal. Still very good. Since audio reference levels are at -18 or whatever it is for digital audio on video for the final mix, then it doesn’t matter if you record at the lower level and mix with the master fader up, or record at the higher level on each track and pull the master fader down. The final noise floor is the same. Quantizing error is a function of the converters used, and does not figure in to the recording levels or resolution. So, it doesn’t matter, and I would just get everyone to agree on one way to do it so that everything is easier and you don’t have to switch back & forth between methods.”

BK: I disagree with his stating that resolution is the same when the signal is closer to the noise. The closer the signal to the noise, the less resolution that low level signal will have, at least in terms of signal to noise ratio. Yes, the noise floor is the same IF in the end case you do not raise the gain. If the original soft signal remains reproduced soft in the final, then you have not worsened the perceived noise of the final product.
Now, if the original A/D and recorder was 24 bit, there is considerable leeway in how far you pushed the original track, absolutely, but it is still true if the original track was recorded rather low, and for esthetic reasons you must raise the level of the mix fader—-then the level of the original quantization noise and distortion of of the original A/D conversion gets raised above the  noise floor in the final mixdown. This is true whether you have a floating point or fixed point mixer.

Hope this helps,
Bob

From: Anuroop

Hello Sir, I am Anuroop from India and I study at SRFTI, Kolkata, pursuing Audiography at the institute.  My question/doubt to you is as follows:  Yesterday I was mixing a hindi song which was originally recorded in the 1970s. Although I had re-recorded all the instruments and vocals as well, a part of our curriculum; I wanted to mix it just as the original one sounds, keeping in mind the dynamic range as well.  After the mix, I measured the loudness with an LeQmeter, which showed the mean at 84dB.  It was a 5.1 mix and the machine was calibrated with nuendo (-12 dB 1k tone) – 0 VU – 93 dB SPL / C Weighting.  As I compared my mix to others, I thought of increasing the level a little bit, but I didn’t do it, as it was killing my dynamic range (after all my ear would also compress the sound after a certain level).  The peaks in my mix went to 95 dB and the lows to 70dB.  Is this a right way to mix?  Is there anything more I should concentrate on while mixing a track?

Thanks and Regards,

Anuroop Kukreja Satyajit Ray Film and TV Institute, Kolkata

Hi Anuroop,

As you probably know, mixing a track is a very complex job that involves a lot more than low levels and high levels and SPL.  No one but your ears and your experience and your monitors and your room can tell you whether the dynamic range of the material you are mixing is the right dynamic range.  An SPL meter is not a good way of judging low level passages.  Some things will sound good at 70 dB and some will not.  The ears are the only way to judge.

The best way to see if you have “too much” or “too little” dynamic range includes:

1) listen at different monitor gains (quiet and loud, in ranges which are appropriate for the audiences who are going to listen to the recording) and see if the important elements of the music are still in good balance and can be heard

2) listen in a quiet room and to listen in a slightly noisy room to see if it translates

3) Lots of experience dealing with many different kinds of music and how they sound to the average listener

4) Check with an “average” listener to see if he/she is comfortable with the mix
And lots more!
Hope this helps,
Bob

Bob,

Hope you are doing well. I have your Mastering Audio book that I reference quite often and have question that I hope will help me in reconfiguring some new gear.

In terms of bit depth and dither it is always best to dither at the very final stage. However, there are times where dither may come into play before that. I believe you mentioned that it is always best to dither when reducing bit depth or risk truncation distortion.

Audio is passes through the mix engine of the DAW at 32 bit float and then leaves the system at 24 bit at the converters. I have read where some people consider adding dither across the monitor outs. If that is the case would it matter that the dither used might not be the same dither type in the final master? In other words would it make a difference say in imaging, depth, etc. or anything else? Would adding dither here be worth considering, even if not the same type in mastering?

Hi Barry

In the case of monitoring you’d be using 24 bit dither. The audible impact of 24 bit dither is so low as not to change the tonality, sound or depth of the music, it just prevents low level distortion. So don’t think of that 24 bit dither as impacting your workflow in any way. In fact, you could cumulate several stages of 24 bit dither (if necessary) without any affect on the sound. Regardless, dither at the 32 bit float to 24 bit fixed boundary will be very subtle. Some people claim they can hear an improvement, others do not. 

My second question that goes along with this deals with digital outputs of the interface. Also I have heard people say I don’t want to give up an analog output on my interface and my monitor controller has “better” converters. They will send the audio to the monitor controller via digital outputs, being TOSLINK, SPDIF, or AES. Well isn’t the spec on those protocols only 24 bit? So wouldn’t that be subject to truncation distortion as well? Would this be a spot that dither might be used? I would assume it is still applicable with digital transfer as well, correct?

Any time you reduce wordlength, you should dither. Whenever I feed an SPDIF or AES/EBU output, I try to dither to 24 bits when it’s possible. Truncation is truncation, wherever it occurs, and SPDIF and AES/EBU are included. 

Hope this helps,

Bob

From: Thomas Johansson

Hello, Mr Katz!

I’m wondering if maybe you could answer a question for me…
When you’re eqing an album, or doing any eq at all for that matter :).. How much in your opinion do you cut/boost a band on one song before making compromises?

I made a little demo cd master today for a friend, which consisted of two songs.. sounding a bit like Limp Bizkit but more industrial metal type. On one of the songs I had to cut about 0.9db at 208Hz to get the tone I was after. Then on the other song I had to cut 3.3db at the same freq. Is this to much eq in your opinion. I think it sounds ok, but should I have made a compromise and not eq the first song which might have resulted in maybe just a 2db cut in the other song. What would you have done?

There is no rule. If your loudspeakers are very accurate they will tell you if you are having a problem. Usually, because 2 track mastering is a compromise, one frequency affects another very easily, so if you are cutting as much as 3.3 dB at 208 Hz for some instrument or resonance you hear, you are likely severely affecting the balance or sound of other instruments. But again, there is no rule.

Is it “legal” for a master to sound slightly different from track to track or does it have to be exact?

Thomas. It totally depends on the music. I just did an album whose philosophy was such that every track had to sound different. I did another album a while back with the same philosophy. However, despite that, there was a unity in that the tracks worked with each other, despite their differences, and that was part of the skill of the mastering engineer.

I have further suggestions on this in my article Secrets of the Mastering Engineer.

Best wishes,

Bob

From: Enrique Ocasio

My comments are:

Great website. Your articles are very informative and direct to the point.
One question: What is your advice in EQing a final mix taking in account the growing-population of X-Tra Bass, T-Bass, etc… home stereos and car stereos?

Thanks
Enrique Ocasio

Dear Enrique:

Thanks very much for your comments!

You should try to make your final mix as neutral as possible, as good as possible on a wide range of systems. I’ve listened to a few of these 3-box systems with the T-Bass and they tend to deal with bass in the 70 to 100 Hz range, not extend it down below 60, as they’re just too small to reproduce those fundamentals, so I would not worry about them. Just EQ your mix so it sounds good on an accurate system and it seems to translate to the T-Bass systems almost 10 times out of 10.

Cars are another issue. Many of them have resonances at 70 Hz and below and these cause a bad impression of solid bass notes. Ever since the advent of car stereos with hyped-up woofers, many recordings have been compromised with too light sub bass. Cars have got to get their systems better. I do like the Bose high end car systems, they seem not to be hyped and have accurate low bass. But our Harmon Kardon premium car systems always seem to be artificially boomy.

Be very careful about compromising a recording to sound less boomy in a car or it will sound thin, even on the T-Bass enhanced 3-piece systems, which really do not have any fundamentals as I said.

The better the mix you make sounds on a great system, the more systems it will translate to. And, hopefully, we will in the mastering, help you to NAIL IT!

Hope this helps,

Bob

Un saludo a todos, permitanme que escriba en español, pues me sera mas facil que algunos me entendais. Mi cuestion es que trabajo como tecnico de sonido en una emisora estatal de television en España, donde como es habitual cada vez hay mas equipos digitales, practicamente todo ya es digital. El problema que veo, es que los equipos entre si estan conectados casi siempre con conexiones analogicas, cosa que no creo sea lo mas correcto, incluso nunca se conecta la conexion word clock. Todo esto me parece una falta de conocimiento que no es bueno para la señal. ¿que me podeis contar sobre esto?? ¿se degrada mucho la señal de audio cada vez que es convertida? ¿es acumulativa la degradacion en cada conversion A/D o D/A? ¿se puede medir esta degradacion?

Muchas gracias por vuestra ayuda y felicidades a Bob por su magnifico libro.

I’ll answer in English and Spanish! The questioner works in television in Spain and notes that many pieces of audio gear are digital but interconnected via their analog connections. I agree that this is is not a very good practice, and that the more conversions you can avoid, the better. Yes, this can be measured. And it does degrade the sound, subtly at each conversion.

In a large television installation, it is almost obligatory for each piece of gear to be synchronized to the master wordclock. You can get away with AES/EBU sync for a bit of time in gear that is serially connected, but with wordclock controlling the sync of at least the first and last piece of gear in a digital processing chain, you are more likely to have a stable signal for the broadcast or final destination.

For example, if the last processer is connected via AES/EBU and locked to the incoming AES signal, it will probably produce more jitter on its output than if it were locked via wordclock. This is especially problematic if you are using the analog output of this last processor. And could cause problems if this high jitter AES signal feeds another digital processor in line.

Hola! Gracias por la pregunta. Estoy de acuerdo de que conectar una cadena digital en serie usando analog puede degradar el sonido. Se puede medir la degradacion aunque puede ser subtil, la distorcion se acumula. Asumiendo que el equipo esta compatible, es mejor conectarlo de manera digital.

Wordclock es mejor por lo menos controlar el sync de la primera y la ultima pieza. Y si es posible, para todo. Para minimizar el “jitter”. Por ejemplo, si el ultimo procesador se conecta con AES/EBU, puede tener un nivel de jitter acumulado de la cadena, y se minimiza usando wordclock. Sobre todo si se usa la salida analog de la ultima caja.

Espero que esto le ayude.

Re: Overload light flashing on the Finalizer 96k…should I be worried?

Bob,

Greetings! James Trammell here, and I have another question to ask you. Are you ready for the bad weather? Oops, that’s not the question.

I have a Finalizer (96k version). I use an Apogee PSX-100 for A-D conversion and the Finalizer is strictly a D-D tool. Recently I’ve been using it to convert things below 80 Hz to mono. With all other function blocks bypassed, I’ve set Insert 1 (I1) to the Spectral Stereo Image function. My crossover points are 80Hz and 8.00 KHz, with the 0-80 Hz band set to -100 (the minimum) and the other bands untouched. Things get fairly loud at times, and while I’m careful not to give myself overs on the PSX, digital signals entering the Finalizer cause the Overload light to flash. Not constantly, but enough to make me worried. The overloads occur in the I1 section where I am converting the low end of the spectrum to mono.

I’m sorry for being so long winded with the above, but I had to give background for my question: When the overload light flashes, what is happening mathematically in the Finalizer? Is it throwing away the calculation for that sample and just letting the original data through?

Dear James: There are probably two reasons why you are getting overloads. One is that combining two stereo channels to mono may produce a combined level that is greater than the individual channel levels if the two channels are very much in phase. The solution to that is to drop the overall level until the overload goes away.

The other reason is that Digital Filtering requires some headroom. It sounds funny, but even when you’re reducing gain at a certain frequency you need more headroom, because the nature of the filtering process can easily cause overloads, especially with sharp filtering. Filters ring near the cutoff frequencies, and can produce more level on the output than the input. In other words, overloading is quite possible even when the filter is doing attenuation, not gain. If you do not hear the overloads and you are confident of your monitoring, then you can effectively ignore them, but internally in the Finalizer, I can guarantee that some overloading is occurring, and it may be audible, depending on the severity of the source. By the way, it is not always dependent on the source level per se, but on the transient nature of the source material near the crossover frequency, so your source metering going into the Finalizer will not necessarily reveal any trouble. It’s a good thing the Finalizer has internal metering to alert you that its internal processing is going into overload. To be 100 percent safe, attenuate the signal prior to the filtering section until there are no more measured overloads. This is highly conservative, since the ear often does not recognize overloads of short duration. We don’t like to take those kinds of chances in our work, especially since one or two of those hits may well be audible, while the rest are not. Usually we attenuate and then make up gain at the end, if necessary.

Otherwise, continue to watch the overload light, and every time it flashes, listen VERY CAREFULLY.

Bob

Hi Bob,

Do you think it’s better to use HP and LP filters during tracking?  Will that help out the A/D converters get a better representation of the signal?
Dear Sam:

In my opinion, never apply a filter unless you are certain it’s going to produce better sonics. And only with proper monitoring can you determine that. When in doubt leave the filtering out UNTIL you are able to determine with GOOD monitoring if the filtering is needed. And if you follow the logic that filters will “help” out the converters, then why didn’t the manufacturer put them into the converters by default  :-(.  The right place in the chain to use a filter is the last stage that you can without affecting other things. For example, if you can wait until mixdown to apply a filter that might be needed, it gives you more flexibility, especially since your decision cannot be undone.

Best wishes,

Bob

Dear Bob,

Would you be so kind to give me info (or links) about 32-bit audio format. I’m interesting in what’s the difference between 32-bit audio format and 32-bit floating point audio format, used in most audio processing software for PC. Why are the mix engines of soft audio sequencers like Cubase VST and Cakewalk based on 32-bit floating format? I’ve heard this format operates audio with level exceeding 0 dBFS. Is it possible? What is dynamic range of such system?

I’ve spent a lot of time finding answers in WWW, but still unsuccessful.

Thank you in advance,
Fyodorov Alexander, sound engineer, Russia

Dear Fyodorov:

I am not a mathematician, but I will explain in the simplest words what I know to be true. If you need a more mathematical explanation, you’ll have to crack a textbook!

Fixed point format is the language of the “outside world”. That’s because in the real world, full scale is full scale—it represents the highest analog value that can be encoded. 24 bit fixed point is the language of the outside world, and its encodable dynamic range is 144 dB. This is the highest resolution allowed in the current AES/EBU transmission standard.

Some designers choose to use floating point chips in their internal calculations. This is very popular with native applications like Cubase because the computer CPUs that they are using like to talk floating point. I hear that the Power PC chip can work in either fixed or floating point, but for some esoteric reason, designers like to use its floating point capabilities. Probably because you can take an existing library of floating point code, and compile it for the Power PC very easily if you stay in floating point.

Once a number has been encoded into floating point, yes, it is true that the numbers can now represent overflow (above full scale) without overload, as well as smaller values than the 24th (LSB) of a fixed point number. So you end up with more internal dynamic range than 24 bit fixed point. This allows easy calculations for the “mathematically impaired”. You don’t have to worry about overload when increasing gain, boosting a filter, summing channels, etc. Many authorities also claim that this improves the internal dynamic range of the calculations (particularly filtering and compression algorithms) inside the processor. My distortion measurements comparing some devices using floating point calculations against others doing fixed point show that with some kinds of filtering work, the floating point processors show less distortion. However, other designers working in fixed point produce just as low or lower distortion. Depends on the designer and his/her DSP talent.

For example, in the most expensive and advanced processors, modern designers using fixed point processing have progressed to internal calculations using “double precision” (48 bit in most cases), which doubles the internal dynamic range, and many authorities feel this performance produces better sonic results than 32 bit floating point. This is at the cost of cycles and power, but with more chips and more speed, it’s not a big cost deal at this time.

But the whole “race” changes again when the floating point designers start using 40 bit floating point. At that point, using equal types of algorithms, the two types of calculations likely produce equal sonic results, without quibbling. When working with double precision, it is very easy for a designer to design 24 dB (or more) internal headroom without losing meaningful dynamic range, so when working with double precision, fixed point becomes as powerful (some say more powerful) than floating point.

However, a certain really talented designer working “only” in 32-bit floating point produces excellent, low distortion results. My take on the matter is that designers concede that it takes a lot more effort (design talent) to prevent floating point work from giving you trouble than fixed point, perhaps because of rounding errors from calculation to calculation. But one programming mistake, or a few cost-saving shortcuts, can ruin ether fixed or floating point work, especially if shortcuts are taken at the most critical time, when the final output number is converted to fixed point 24 bit at the end. If those numbers are not converted (and properly dithered to 24 bits) at that time, then the sound of the entire system can be compromised.

Bottom line: Don’t be confused by the specs or the numbers or the claims. Distortion measurements may give us some clue as to why some systems sound better than others, but even distortion measurements don’t tell the whole thing, because they don’t always reveal all the shortcuts that a designer is taking under all circumstances. All other things being equal (and they never are!), it doesn’t matter whether you’re working in fixed or floating point. Only the results count. And there are real sonic differences between platforms. “Cheap digital” still costs—it does not sound as good as cheap analog.

Best wishes,

Bob Katz

From: Leo Cottakis

Dear Bob,
I have been reading your articles at Digital Domain for some time now and I would like to tell you that I consider them an invaluable source of information for all things audio and an incredibly useful resource for the audio engineer. I particularly appreciate the way you discuss – or should I say, rightfully expose – the various manufacturer’s limitations and remove the “gold-dust-in-your-eyes” factor from the picture. All too often audio equipment manufacturers make us believe that their creations are the best thing under the sun only to discover that, under scrutiny their equipment should have probably not been released in the first place. I commend you on a very good effort and would love to see those articles coming.
I would value your detailed opinion on the following: I am in the process of putting together a project studio which will handle mostly classical work but also a combination of electronic/acoustic non pop&rock multitrack work. Quality and purity of tone is paramount and to that end I am proposing to put together a system based around a premium quality analogue desk, a 24 track HD recorder (RADAR II), some high quality outboard, good mics and mic pre’s and monitoring by Genelec and ATC.
The RADAR II is a 24/48 recorder.

The new Radar from IZ corporation claims to support higher sample rates. You should look into that.

I have evaluated other digital recorders and found the Radar to be somewhat closer to analogue 2”/30ips tape – at least in the bottom end – despite its 48khz limitation. During a typical multitrack session (analogue desk/digital recorder) a lot of ADA conversion is taking place between desk and recorder. Knowing that successive ADA conversions can degrade the signal could this eventually become a problem?

We’re currently in a hybrid world—Leo, and will be there for a long time. I feel that with the current state of the art in conversion (particularly at high sample rates/wordlengths) and with current “cheap and bad sounding DSP processing”, the losses due to degradation with a couple of extra conversions and an analog mix with a GREAT analog console are less than the losses with a bad digital console and DSP. This situation has changed with Moore’s law and now (2012), I would go with the digital mix over the analog mix. USING SOME GOOD ANALOG OUTBOARD gear for processing. Plenty of people still believe in the analog mix, for good reason, but I think it’s becoming less necessary if you have the best digital gear, recognize processor and plugin limitations, and supplement your outboard with good analog gear.

Is the previous statement true in this particular case? Radar converters are quite good especially if clocked properly. Could it be said that an analogue signal exiting a DAC (24/48) would somehow benefit from entering an all analogue chain prior to being converted back to digital for mixing down? In other words, would that signal be “enriched” by the addition of 2nd/3rd order harmonics potentially generated by an all-analogue mixing console which also has a better frequency  response (10Hz-50Khz)?

It’s now 2012 and my previous answer to your question leaned on the analog gear as being less harsh than the inferior digital processing. But as I said, this is no longer the rule. Now your chances of doing better with analog or digital depend far more on your skills. You still need a lot more skill to avoid digital problems, so if you feel less technically inclined, mix in analog. That’s my recommendation in 2012.
 
I have read your article on back to analogue and that resulted in me asking the question: Is properly aligned 2”/30ips with Dolby SR better sounding than good 24/48 digital processed through a very good analogue desk mixed onto Sadie and dithered to 16/44.1 using a good algorithm like Pow-R? Also, is that true of a second hand 2” MTR, however well maintained? I know these last two questions are very broad but can you nevertheless offer some feedback?

I am considering interfacing all my digital equipment via an AES patchbay using either an RJ45 unit or a Ghielmetti AES patchbay which is specially designed to handle high-rate AES digital audio. I am also considering using a Z-Sys router which will itself interface to the patchbay and into which signals from other points of the bay will be fed for routing splitting etc. Any potential problems here? Can you offer any feedback on Ghielmetti AES patchbays? Also, is there any real difference between good 110 Ohm AES cable and Belden Mediatwist?

For short runs, either 110 ohm AES or Mediatwist is fine. Mediatwist is cheap but good if you can terminate it properly and keep it from bending and twisting. I recommend a router over any patchbay for 110 ohm stuff. Skip the RJ-45…it’s not worth the trouble. Plug everything into the router and have the router feed an external XLR rack for interfacing ancillary digital gear. Avoid too many connectors in the signal path—since the XLR as you know is not true 110 ohm. Especially at high sample rates, it’s murder.

Are you working with high bit rate formats like DSD, SACD? Can you offer any comments, experiences, etc?

I’m working at high sample rate PCM but generally I’m not a DSD person, and it’s arguable whether that is higher resolution or just nice-sounding. I think that mid-rate DSD sounds a bit sweeter because it softens the transients. High-rate DSD seems to be about the same sonically as 192 kHz PCM to my ears. And the PCM is much easier to deal with.

I’m looking forward to your reply and thank you in advance for your time
Leo Cottakis

Hope this helps,

Bob

Dear Bob,

I write because I noticed you apply a slightly tilted room curve to your mixing room. But why not make a flat response as reference and then mix your recordings from that? To a “straw man” like my that would seem more logical because a tilted curve will introduce a spectrum bias which I as a consumer would need to implement in my system as well. Whereas if a mix made from a flat response means that if my speakers (whether equalized or not) are pretty much flat I will get the approximately same sound impression as in your studio. Of course, a complete replica is not likely unless I also aim for the same RT level as in your studio.

Anyway, it was just something that got me curious and finally I just had to ask for the reason for a tilted curve in a mixing studio, which for the uneducated man (like myself) seems odd.
I certinaly mean no disrespect, since your knowledge in this field is manifold greater than mine. So it with the utmost respect that I ask :-).

Best regards,

Mikkel G. Hangaard

Dear Mikkel:

Thanks for your letter.

What you have brought up is the classic “chicken and the egg” situation. It is true that our recordings have to complement our reproducers. if we equalize our reproduction system to flat (however that is measured), it would cause us to produce mixes and recordings which would be duller sounding than our current mixes! But historically, that has not been the case. We have a large recorded legacy of recordings that sound perfectly good on playback systems that measure some high frequency rolloff. All the major accepted reproduction systems since the beginning of time measure rolled off at the high end to some degree. So basically you are asking us to try to produce new recordings that would not be compatible
with the older reproduction systems.

It’s academic: If today, suddenly, all playback systems were made flat, then all or most recordings which already have been made would sound too bright. It’s a long, historical precedent and consistency is more important than absolute conformance with “flat”.

Lastly, there is a whole question of how “flat” is to be determined. Loudspeakers which have a wide, flat dispersion characteristic sound very different than loudspeakers which are flat on axis but rolled off off axis. How do you determine what is flat in that case? Then getting into questions of reflections from the side walls in the room and how they are treated. Lastly, the question of the measurement method and the measurement window, should it be anechoic or should it include some reflections in the room?  There are so many different ways of determining the window that unless you know exactly how a measurement was taken, you cannot effectively judge a published loudspeaker measurement.

All these variables produce varied amounts of “measured” rolloff. In other words, even the definition of how to define “flat” is put into question. So even if we were starting from scratch with new recordings and new reproducers, there would still be a fundamental disagreement as to how to measure and determine what “flat” really means and we would still get nowhere!

So in the end, the egg came first, and the chicken followed, and we just have to keep on keeping on!
Hope this helps,

Bob

From: “john conner”

Subject: Re: The real advantages of limiting and compression?

Hey, Bob. John Conner here from Capstone Studios (also Nashville). Enjoyed the wise words. Hope all is well.

Thanks. John is referring to my mini-post “How To Be Honest With Yourself”, I posted on rec.audio.pro.

Here’s what I wrote…

From: Bob Katz

Newsgroups: rec.audio.pro
Subject: The real advantages of limiting and compression?

On the Sonic Solutions maillist, a participant asked a question and I answered it in the most cogent, succint manner I possibly could. I think my answer is about the shortest, complete answer on this topic that I’ve been able to do, and so I thought I’d share it with rec.audio.pro. Here’s the conversation.

I’m sorry to be the lowbrow of the group, but when it comes to pop, rock, country, rap etc., I like the sound of proper compression and limiting. When done right it can add to the power and depth of a big electric sound.

I’m not ranting here… just subtly asking you to think and check some things which I’ve done a hell of a lot of working on…and a hell of a lot of listening….

Here goes…

Ahhhh, compression versus limiting.

COMPRESSION:

Yes, Compression can certainly affect the apparent “power” of music, and I have no objections to the use of compression to affect material’s “sound”—that’s all to taste. Just (in my not so humble opinion), please do yourself and your clients and the future of the audio world a favor by carefully comparing the compressed versus the bypassed version at equal apparent loudness.

Adjust the makeup gain of the compressor so that in bypass, the apparent loudness is the same. Now listen again… is there really a subjective improvement? At least 5 times out of 10 you will be very sobered to realize that what you had thought was an improvement in “power” or “punch” was really just a loudness difference, and at least 2-3 times out of 10 you will also be sobered to realize that what you thought was an improvement was actually a degradation, at least to some aspects of a sound. I don’t often find even the best multiband compression to be totally a win-win situation, and yes, I use it, but I’m much too honest with myself to think it’s always a win-win situation.

LIMITING:

2 to 3 dB of digital limiting (carefully designed as a sample-accurate, razor-sharp limiter, not a “fast compressor”) does nothing but raise the apparent loudness while also subtlely or seriously screwing up the transient response, depth, imaging, or tonal characteristics of the music.

Please, do the world a favor. Next time you want to limit to impress yourself, try this test:

Try 1 dB of limiting, thus raising the gain by 1 dB.

Now, A/B compare the limited versus the unlimited version. First impression: Oh boy! The limited version sounds more impressive, more powerful, doesn’t it? Second impression: Well, here’s a very sobering thought. While comparing the limited versus the unlimited version, move your volume control up when you bypass the limiter and back down when limiting is in, by that same exact 1 dB. Hmmmm…. what did the limiter do, exactly? Did it really make your material sound more powerful, or did it just make it sound hotter (louder)? And in the process of making it louder, what did it do to the sound? Better, or is it really worse?

When that fact sinks in, then it might be possible to extrapolate to: if the client is there, and you want to impress him with what you’ve done, instead of patching in a limiter, why not just turn the volume up and say, “OK, here’s the mastered version. Gee, doesn’t that sound better?” It certainly would be no more dishonest than just patching in the limiter without giving him the attendant lecture about the possible negatives of what it may do to the sound. When was the last time you did mention to your client what limiting actually does? And whether it accomplishes anything, even in a 6 CD carousel player. For every 6 CDs you pick to go with your new CD, I can pick 6 others that are either louder or softer.

What the hell’s the point? It’s lose-lose, big time. And carousel players are a big PITA…. sell them a compressor to use at the bar…

Of course, more than 2 or 3 dB of limiting, and it starts to sound a bit like very severe compression, not just a “loudness increaser”, but that’s not my point here…

I am extremely careful when I master my records. I often use a *lot* of processing. But before I declare the record finished, I do a very sobering thing: I listen once again to the raw, unprocessed, original material, at a matched apparent monitor level (sometimes they’re 4 to 6 dB apart!); I carefully evaluate what I’ve done, and if what I’ve done doesn’t sound *better* than the original material (not just *louder*), then I start once again from scratch, until the master is actually better…. or I just leave it alone! And you know what…last night it happened again. I was really impressed by what seemed to be coming out, but when I compared my “better” result at matched loudness, I found I really hadn’t made it better, just louder…. At least I hadn’t made it (much) worse, so I just called it a day. But don’t pat yourself on the back quite so hard until you’ve made that comparison. You may discover… the original “impressive” result turns out to be just a simple volume difference!

Many thanks for listening,

Bob

From: Danny Leake

So am I right? Is the epidemic of inappropriate hypercompression almost totally traceable to incorrect assumptions about the end user experience and a flawed listening procedure engaged in by artists, producers, and record label executives?

Part of it. The rest can be attributed to the following adding up together:

1) Louder sounds better, even .2 dB, so a little push in each step seems to be the thing to do. Everyone wants to sound better, so they turn it up.
2) Many (if not most) listeners are pretty insensitive to crest factor and the impact of transients, so they don’t notice the degradation. If the RMS is loud enough, they don’t notice the loss of peak clarity.
3) Combine that with the invention of digital audio and severe digital processing, which permitted average levels to be almost at the peak level.
4) Combine that with the widespread practice of peak normalization

I had an artist call me about an album I had mixed for him. We had
kicked it back at least once because we thought some cuts were a tad
overcompressed to gain level. After we were happy with everything I got another call. He was wondering why his CD was a tad lighter than a CD I had done for another artist. I told him aside from the fact that his record was a well recorded Jazz
record and the other track was an Urban influenced, machine driven
track; to raise the level of his CD would mean “crushing” the dynamics again
and compromising the sound of his music. In other words, “Just turn up
the amp and call it a day!”
 
We had agreed after the mix that level was not going to be an issue
for his CD, that we wanted the dynamics that he had worked so hard for to
come through…however, when the old CD changer came into play, he
panicked and forgot what we were trying to do.
 
I had to remind him. What if he hadn’t called me? Some artists will
but most Record Executives don’t.

Danny Leake
Urban Guerrilla Engineers
Chicago
 
I’m so sorry to hear that, Danny, happens quite often. You’re lucky you were able to remind your client of your “pledge” or it would have been another screwed-up record. We have to get out of this mess.

Bob

Marco wrote:
    Mi gran duda era saber cómo puedo corregir los picos de inter-sampling cuando se convierte a un formato AAC?
      Muchas gracias!
Translation: “My big question is how can I fix the inter-sampling clpis when I convert to AAC Format?”

Hola, Marco. Encantado “verte” otra vez!

Mi libro trata de hacer los conceptos claros pero voy a tratar de explicar aquí!.
No es posible “corregir” picos de inter-sampling. La única manera de “corregirlos” es de bajar el nivel hasta que los picos de intersampling (que siempre existen) no estén sobre 0 dBFS.
Esto dicho, hay métodos de limitación en PCM que pueden controlar los niveles de los picos de intersampling. Por ejemplo, un limitador certificado a ser sensible a los intersamples puede ayudar a controlar los picos de intersampling durante la masterización. Usado conservadoramente! Si oyes un artifacto de la limitación debes usar menos limitación. Este limitador funciona así haciendo intersampling si mismo, internamente.

PERO!!!!!! ———–  Despues de la conversion a AAC, ciertamente el nivel va a subir un poco, y los picos de intersampling que estaban abajo de 0 dBFS pueden re-llegar sobre 0 dBFS. Tienes que bajar el nivel antes de la conversion a AAC un poco para evitar estos picos. Quizás en el control de “output” del limitador. Normalmente no necesitarás más de un dB de ganancia (perdida), porque el limitador ya ha tratado de controlar (no eliminar, pero controlar) el nivel de los picos intersamples. Después de la conversion a AAC, AAC añade a los niveles a causa de su filtración interna. 
Al fin, si escuchas el AAC y hay pocos picos sobre 0 dBFS y no oyes un problema, y no mejora el sonido si bajas el nivel, puedes aceptar estos picos. Porque Apple no es la “policía de picos”. Ellos van a aceptar lo que tu envias, aun si es distorsionado.
Espero que esto te ayude,
Bob
Translation: “Hello Marcos, nice to “See” you again!
My book tries to aproach those concepts the easiest way possible, but I’ll try to explain again.
Its not possible to “correct” inter-sampling peaks. The only way to “correct” this is by lowering the level until the peaks (that always exist) are under 0 dBFS.

With that said, there are limiting methods in PCM that are able to control the levels of those inter-sampling peaks. For example, a certified limiter can be sensible to the intersamples and its able to control those peaks during mastering. Be conservative when using them! If you are able to hear artifacts of the limiting, you are using too much. This limiter does oversampling internally so it should handle the peaks that are between the samples that cause DACs and other processes to overload.

BUT!!!!———– After converting to AAC, it is certain that the level will go up a little, and the peaks that were under could get to 0 dBFS again. You have to lower the level before converting to AAC to avoid these peaks. Maybe lower the output on the limiter. Normally you won’t need more than 1 db gain (lower), because the limiter already tried to control (not eliminate, but control) the level of the peaks. After the conversion, they are not “New” peaks, they were always there, but because the AAC conversion will raise the level, they are now over 0 dBFS.
Finally, if you listen to the AAC and there are a few peaks over 0 dBFS but you are not able to hear any problems, and lowering the level won’t “fix” it, you can accept those clips. Apple is not the “Peak” police, they will accept your product even if its distorted.
Hope this helps,
Bob.”

Paul Kelly wrote:

Hi Bob,

Can you tell me what software reads ISRC codes from a CD?  I’m trying to verify some of the codes and can’t seem to find anything (including Toast 9 and Jam 6.0.3) that will read the existing codes from a CD.

Thanks for your help!

Best,
Paul
Hi, Paul. There’s a trick in the terminal for reading ISRC codes on a Mac.
Launch the terminal in applications. You can add it to the dock if you want.
To read isrc codes, type: drutil subchannel
For CD text, in terminal just type: drutil cdtext
This will list out the text on the cd. for other usefull things using the drutil, type: man drutil
This will list out a bunch of other things you can do with your drive.

Let me know if it works for you.

Take care,

Bob, who thanks Michael Fossenkemper on one of my boards for promulgating that information!

From: Will Gibson

Subject: Re: isrc codes

My comments are: I’m trying to find a list/explanation of isrc codes on the net. can you help.

Dear Will, I don’t know where else to look on the net, but my mail list archives from the Sonic Solutions mail list reveal these facts, written by several of the members of the mail list

1. ISRCs – a unique code is indeed generated for each track on the CD.
This allows use of automated logging systems to be used at radio stations to track copyright ownership/royalties. The record label provides the codes to be entered for each track.

2. UPC/EAN code – is also called Mode 2 data and is a barcode that
contains info about the product. This does not have to be used with ISRCs. And is not obligatory on the CD since it’s a shrink-wrapped product!

3. Getting the codes – in the US a record label will write/fax the ISRC org in the USA to obtain a unique header for the ISRC code that designates their label.

If you are only needing one or two UPC codes, the UPC/EAN codes can be provided by numerous vendors/duplicators/replicators who will resell codes “onsie-twosie”. However, if you need a group of codes, you can buy a set from here. Or, this link will resell UPC codes in small groups. I don’t know about UPC codes but I suspect they are generated by the label for their own inventory purposes. Someone will undoubtedly correct me if I’m wrong.

ISRC codes come from the label releasing the title. Each tune (or
variation of a tune) is supposed to have a unique ISRC code.

Twelve digits only – no more, no less, don’t include the dashes (watch it if you’re using Sonic 5.3.x or 5.4, there’s a broken warning window which USED to tell you if you’d missed a digit).

In the ISRC code : US-MC1-98-12345 the first two digits are the country code, the next three digits are the company code, meaning the company which originally produced the song, the next two digits year the song was recorded, and the last five are recording codes designated to the version of the song itself.

Wayne Newton’s version of “Danke Schen” will have a different ISRC code from the Andrews Sisters’ version of the same song, each recording of Jimi’s “Foxey Lady” will have different ISRC codes, that kind of thing.

The UPC/EAN is the distributor’s bar code SKU number seen on the CD and finished artwork. It can be entered either by the mastering house, or at manufacturing itself. Usually the latter… at least that’s how we do it.

Greg discovered an apparent contradiction in my article on jitter: Why did I say that DATs don’t seem to pick up jitter, but AES/EBU CDR’s seem to pick up incoming jitter.

Date: Wed, 17 Jul 1996
Tp: Greg Simmons… Jodie Sharp [email protected]

Dear Greg:

Thanks for your comments. I’ll see if I can clarify…please consider the article on jitter a work in progress. Your question did hit the nail on the head on some missing information in the article, and I thank you. In my copious (????) free time I’ll try to clarify the article.

Dear Bob,

Your essays on dither and jitter are excellent, authoritive and very informative – I’ve learnt heaps from reading them. However, in your essay on jitter, you say some things that appear contradictory and have me slightly confused.

Firstly, you say that “…Playback from a DAT recorder usually sounds better than the recording, because there is less jitter. Remember, a DAT machine on playback puts out numbers from an internal RAM buffer memory,locked to its internal crystal clock.” Shortly afterward, you say: “…a compact disc made from a DAT master usually sounds better than the DAT…because a CD usually plays back more stably than a DAT machine.”

Under the heading ‘Can Compact Discs Contain Jitter’, you say “…An AES/EBU (standalone) CD recorder produces inferior-sounding CDs compared to a SCSI-based (computer) CD recorder. This is understandable when you realize that a SCSI-based recorder uses a crystal oscillator master clock.” The text continues by discussing the differences between the PLL system used by a standalone AES/EBU recorder and the crystal oscillator system used by a SCSI recorder.

The paragraph closes with “…No matter how effective the recorder’s PLL at removing incoming jitter, it can never be as effective as a well-designed crystal clock.”

Do you see the source of my confusion? When discussing DAT and dubbing in general, the essay suggests that jitter on the incoming data is irrelevant during playback (so long as the jitter is not so high as to cause actual errors) because after being recorded, the data is re-clocked by the playback machine’s internal crystal oscillatorlock. And yet, the essay also suggests that jitter on the incoming data (as in the case with the two different types of CD recorders) does affect the final sound.

Excellent point. I will have to include this answer in the next revision of the article:

First of all, the remnant playback jitter (“intrinsic jitter”) of a DAT machine is significantly higher than the remnant jitter of a good CD transport. Bob Harley measured 1.2 nanosecond RMS jitter on the output of a Panasonic 3500! The remnant jitter on the output of a great CD transport is on the order of 10 to 100 picoseconds. A nanosecond is 1000 picoseconds, so you can see there is an order of magnitude difference.

Power supply design, grounding, can affect the quality of these cloks, and audiophile CD transport designers pay special attention to the power supply. A poor power supply can affect the remant jitter both by contaminating the crystal clock, and the AES/EBU trasmitter in the digital output stage. Until someone examines the internal mechanisms of both reproduction systems with very sophisticated measurement equipment, we can only hypothesize. But for now, it is enough to say that the measured intrinsic jitter of a DAT reproducer is greater than 100 times the jitter of a good CD transport. We all know that shouldn’t be happening… all digital reproducers should measure perfectly–right? Good thing we are able to measure those differences, or the golden ears would all be in a pickle trying to demonstrate why DAT playback just doesn’t sound as good as CD playback.

Now for your question, you recall that my tests with DAT recorders seem to confirm that jitter coming in is irrelevant to the final playback. This does seem to confirm the proper action of the phase locked loops and FIFO’s in the DAT machine. But why doesn’t it seem to hold true with CDRs? Well, the deeper we dig, the more we learn. On the surface, the science holds true, but…

One of my hypotheses is that the residual jitter level of 1200 picoseconds in a DAT reproducer has a very large masking effect. It is highly unlikely that any small remaining differences or variances (say plus or minus 100 picoseconds) due to variant jitter of sources could ever be heard or reliably measured. Especially after reduction through the recorder’s input PLL, and especially after the separate blocks on tape are reassembled and then retimed through the reproducer’s output FIFO. So the question with DAT machines becomes a moot point, as far as I’m concerned.

Another hypothesis is that the data block structure of the CDR is different from that of the DAT, and may have an effect on outgoing jitter. In both types of playback, however, data is extracted in a “jittery” manner, and always smoothed by FIFO, so the different data block structure would have to have an indirect influence on the output clock.

Another possibility is error correction, and again, only through an indirect influence, common impedance coupling through the power supply. Perhaps the CD player’s design is more susceptible to that than the DAT. It seems the problems I was alluding to are only relevant to a very low jitter medium (such as CD). In a low-jitter CD player, we can examine and test for “microcosmic” influences on the stability of the player’s crystal clock and see if they are caused by “microcosmic” differences on the CD disc. It took David Smith and Sony Corporation months and months to devise some sophisticated audio tests in order to conclude that the golden ears were right!

This brave work was undertaken by design engineers working for the very company that had designed the “perfect” FIFO system for CD players which is supposed to eliminate all outgoing jitter (or at least reduce it to the residual of a crystal oscillator). It will take months to years before scientists with sophisticated measuring instruments find and eliminate the subtle internal mechanisms in a CD player that are somehow permitting jitter differences to be heard through a supposedly “perfect” system. So far, no equipment designer has succeeded in producing a jitterless playback system (everyone says it’s possible)–although great improvements have been made (listen to some of the newer audiophile digital reproduction systems and to a couple of the finest professional D/A converters).

We’re trying to find flies on an elephant, here. Unfortunately, on the audio side of this mixed metaphor, the human ear can hear the flies very well. A good scientist mustn’t assume anything, nor take anything for granted, and must recognize that all conclusions are based on some underlying hypothesis or axiom.

Surely a CD player is just like a DAT machine on playback, and uses it’s internal crystal oscillator to clock out the data, therefore reducing the problem to the jitter inherent in the internal crystal oscillator clock and eliminating any jitter caused by the disc recording process, whether the data was recieved via AES/EBU or SCSI. Or is there something I don’t understand?

I’ve heard terms such as pit jitter and land jitter, are these what you’re referring to?

Pit and land jitter on the CD may or may not be the cause of the differences we are hearing. Some other mechanism on the CD (size of pits, not necessarily the spacing of pits) may be causing the servo mechanism in the player to be more jittery. It is definitely not data errors. Research has shown that these CDs which we claim to sound different have identical data. But part of the problem may be due to error correction, with the error correction system causing problems, again by power supply coupling. Very far-fetched argument, yet to be proved. Same with the servo mechanism leaking into the power supply for the output crystal…engineers have found a 25 cent power-supply bypass capacitor in the digital section to do wonders on the audio quality, so this is pointing to the reasons.

And to complicate the matter, the analyzers which look at pit and land jitter on CDRs generally do not look at its frequency distribution. For example, 10 picoseconds of peak to peak jitter with a central peak at 3 Khz is likely more audible than 500 picoseconds of random (uncorrelated) RMS jitter.

It takes far more sophisticated equipment to make the second measurement. I’ve had a plant analyzer show the reverse result, RMS jitter was higher in the CD that played back with apparently lower jitter. That is, if pit and land jitter on the CDR is even the root cause of the sonic differences we are hearing. When we hear a CD that has a wider soundstage, greater apparent low level resolution, and other audible differences, we assume that is caused by jitter differences on the CDR itself. But this is only a very unscientific hypothesis. And no standards as yet have been developed that correlate measured jitter against listening differences.

However, advancements are being made, and a specialized test system that looks at the analog outputs of CD players (and D/A converters) has been developed.Paul Miller’s company employs Julian Dunn’s specialized test signal for this purpose.

Also remember that what I said in my article remains true: that you can copy from a CD that supposedly sounds “degraded” through a SCSI interface back to another CDR or to a hard drive, then cut another SCSI CDR, and the end result can sound better than the original if the new writer is better than the original writer! Jitter is NEVER transferred with the data to a new medium, if a clock is not involved. And SCSI does not involve a clock. Jitter is strictly an interface phenomenon, whenever a clock is involved.

Finally, I’m sorry to send you such a long winded message but I genuinely would like to understand these things. I suspect I may be interpreting your essay incorrectly. Please don’t take this message as a criticism of your essay – I have nowhere near enough knowledge or experience in this field to criticise your essay!! I’m just confused…

No problem… Jitter is a complex subject, no one knows all the answers. I ask more questions than I have answers. Someday I will try to rewrite my essay to incorporate all of these considerations. I hope this letter helped!

BK

On Tue, 30 Apr 1996, Bob Katz wrote:
I have observed that I can copy a CD into the DAW, and make a new CD that sounds better than the original.

Steve Potter wrote:
Bob, how would you define ‘sounds better’?

And I reply:

In summary, let’s compare two presentations of the same material, which is otherwise identical data. The one which is “warmer, wider, deeper”. (Pick one, two or three) is the one with less jitter. It makes sense logically, and it has been borne out by measurements.

How can a listening test be objective? How can we separate “different” from “better”?

This is a very tough question to answer. For example, regarding listening tests for jitter, no one that I know has reached the point of being able to say: “100 picoseconds of white noise jitter mixed with 25 picoseconds of signal-correlated jitter sounds better than 100 picoseconds of signal-correlated jitter mixed with 25 picoseconds of white noise jitter”.

No, we have not reached the degree of sophistication where we can judge “better” in that fashion.

However, a number of experienced listeners have been able to use the judgemental term “better” in ways that correlate quite well with the measurable physical phenomena that are under investigation.

For example:

First: one definition of “better” is the source which sounds closer to the analog master tape, or the live source if it is available.

Second: Those of us who have been chasing the jitter phenomenon have begun to educate our ears and recognize the sonic differences that different types and degrees of jitter cause. Please note that in every case I use the same D/A converter to monitor the different sources under test at repeatable monitoring gains.

If you wish to begin entering down this jittery road, I suggest you start listening to the easiest form of jitter to recognize, one which every user of Workstations can hear every day, while monitoring through a high-resolution playback system. (A high-resolution playback system consists of a good 24-bit D/A converter, wide range monitors in an acoustically treated room, power amplifier with wide dynamic range, and a quiet listening environment).

The easiest form of jitter to recognize is signal-correlated jitter. Signal-correlated jitter adds a high-frequency (intermodulation) edge to musical sounds when monitored on a susceptible D/A Converter. Obviously, a theoretically perfect D/A would not be susceptible to jitter. We must not blame the message directly…just that the message is contaminated with jitter.

Every day I load into the workstation connectors while listening through the desk, I hear (and am bothered by) the most primitive form of proof that correlated jitter is audible: The sound is different during the load-in than during the immediate playback from the hard disk! It also sounds demonstrably worse during the load in than during the playback.

This is attributable to the intermodulation of Sonic’s master clock by signal-correlated jitter during the loadin. During playback, since the source is no longer feeding digital audio, even though the DAW is still locked to the external clock, the external clock is much more stable from the DAW’s point of view. Its PLL is no longer modulated by the digital audio that is combined with the external clock. This well-known phenomenon, known as signal-correlated jitter, has been documented by researcher Malcolm Hawksford. See AES preprint titled “Is the AES/EBU S/PDIF Interface Flawed” AES Journal 1995. This form of jitter is quite audible.

After an engineer learns to identify the sound of signal-correlated jitter, then you can move on to recognizing the more subtle forms of jitter and finally, can be more prepared to subjectively judge whether one source sounds better than another.

Here are some audible symptoms of jitter that allow us to determine that one source sounds “better” than another with a reasonable degree of scientific backing:

It is well known that jitter degrades stereo image, separation, depth, ambience, dynamic range.

Therefore, when during a listening comparison, comparing source A versus source B (and both have already been proved to be identical bitwise):

The source which exhibits greater stereo ambience and depth is the “better” one.

The source which exhibits more apparent dynamic range is the “better” one.

The source which is less edgy on the high end (most obvious sonic signature of signal correlated jitter) is the “better” one.

Does this help you?

Seems like this could be very subjective. I could almost certainly agree on ‘sounds different.’ To be fair, I have not been involved in the kind of research you have done in this area but I still feel there is a lot of subjectivity involved.

I recently attended the NAB convention and watched some demos of video line doublers and quadruplers. While in some ways they ‘improved’ the projected image, I could not flatly say that they made every aspect of the picture ‘better.’ It was decidedly ‘different’ but among the group I was with we couldn’t all agree on what aspects we liked and didn’t like about the ‘improved’ picture.

That’s for sure. Well, line doublers actually alter the data which is sent to the monitor, so, unlike with audio jitter reduction units, the data is changed, and you get into very valid subjective questions. The question of “better” is definitely a slippery subject. I don’t pretend to have all the answers, but to me, the audio copy which sounds most like the original points the direction of the degradation. Then, I can relate an experience with two different jitter reduction units, both of which produced excellent-sounding outputs, but both sound very different from one another. One has a slightly “flabby” bass, the other a tighter bass. At least both of them sound “better” than the jittery copy as monitored without the jitter reduction. So, when comparing two different D/A converters or two different jitter reduction units, even more subjective judgment enters into the picture. I agree with Steve Potter that “Better” is a very complex subject! Here’s a followup on the maillist from Peter Cook of the CBC:

Date: Wed, 1 May 1996
From: Peter G. Cook
Subject: Re: DEFINITIONS OF “better” versus “worse” sound

A fine mini essay Bob. Perhaps you could add this on your web pages.

At 08:21 1/5/96, Bob Katz wrote:

Therefore, when during a listening comparison, comparing source A versus source B (and both have already been proved to be identical bitwise):

The source which exihibits greater stereo ambience and depth is the better one.

The source which exhibits more apparent dynamic range is the better one.

The source which is less edgy on the high end (most obvious sonic signature of signal correlated jitter) is the better one.

The better one, and it is better, is also easier to listen to. . . less fatiguing. I would also add to this that the low end just “feels” bigger and more solid. This is perhaps a psychoacoustic affect more than a measurable one. It may be that the combination of a less edgy high end and greater depth and width makes the bass seem better.

All of this makes sense if thought of in terms of timing (that is what we’re talking about isn’t it ;-]). With minimal jitter nothing is smeared, a note and all its harmonics line up, the sound is more liquid (a term probably from the “audiophile” crowd but one which accurately describes the sound none the less), and images within the soundstage are clearly defined.

-Peter

From: Joav Shdema

Hi Bob, Some discussions on the audio mailing list left me wondering should I make some changes to my audio cabling – regarding the digital audio transfer. I have searched your site, and others, but could not find a satisfying answer to my quest. Today I am, as most others , using 75ohm cables for S/PDIF and 110ohm cables for AES at 16/44.1 What are the recommended cables for 24/96 digital audio, are they different from the former and in what way (width, length, material, connectors, whatsoever), is there any new standard or practice I am not aware of?

As a producer/engineer who slowly make the transpose to higher bit and sampling rate – doing it right from the beginning is very important to me. Best, Joav

Joav Shdema
Desert Island Productions

Dear Joav:

As we move into 24/96, the integrity of the interfaces becomes extremely critical, the quality of the cabling, the length of the cabling, and impedances, voltages and connectors. There are only two high quality routes to go with, depending on your ingenuity.

The RCA connector is bad. It is not true 75 ohm. The voltage on S/PDIF is specified as 1/2 volt p-p, which does not survive long. Better to raise to 1 volt p-p. The XLR connector is bad—it is not true 110 ohm. Reduce the number of either connector in your system, and the performance improves, particularly at 96 kHz. I have seen some marginal connections that lock perfectly at 44.1, marginally at 88.2, and not at all at 96 K.

A router like the Z-Sys or the NVision is highly recommended. It will support all rates. By using a router, your connections get shorter and cleaner. The Z Sys can be outfitted with either BNC’s (AES -3 ID) or XLRs. With BNCs, you use low loss RG-59 U, BNC’s everywhere possible and convert all of your interfaces to 75 ohms. If possible, increase the output voltage of all your S/PDIF sources to either 1 volt p-p or 2.5 volts p-p, so they will survive cable routing longer. That’s the route I have taken.

Or with XLRs. In that case, I recommend Belden media-twist 110 ohm cable. It has the best bandwidth, reduction, and performance at 110 ohms. You can find further details on this topic at the mastering webboard website.

Hope this helps,

Bob

From: Kim

Ok I have been told that i can use a regular patchbay for my digtal equipment. I question the valitity of this request.(I’m talking about AES/EBU, Spdif). I would like to know if you have any thoughts on this.

I guess theoreticly it should work but what I’m seeing out in the world are a lot of active digital patchbays. So I guess what I like to know is this, have any of you built and used just regular patch bays for digital audio interconnects. Now I’m looking at just using a adc patch bay with a punchblock on the backside.

Will I have any signal disscrepencies ie ohm problems, the possiblity of blowing dacs ect…

Thanks kim

Dear Kim:

You won’t blow any DACs, but every impedance bump in the chain adds a degradation which will make your signal connections less reliable, and certainly add jitter, which adds distortion to the monitoring.

The punch block, even the XLR connectors (which are not truly 110 ohms), every point where one wire is connected to another…is a serious problem…these are all “impedance bumps”. The jacks are also not true 110 ohm. You may find your wire length only good to a few feet before you get crackles, pops, or no signal at all.

I advise against using standard “analog” patchbays for digital, unless you convert to 75 ohm technology and use professional (video level) patch bays, or a certified impedance matched active patchbay. The same is true of XLR “patchbays” for AES/EBU. Every XLR plug in the middle degrades the integrity of the connection. XLRs were not designed to be digital connectors. This was perhaps the biggest mistake made by the AES. It helped us to get into digital audio in the beginning, but it’s making lots of trouble now. I support the AES-3ID standard, which puts digital audio on 75 ohm coax with BNC connectors. Another valid technology is properly twisted pair CAT-5 cable, with RJ-45 connectors, running 10 base-T or 100-base T Ethernet. These can also be made with patchbays and cable runs that pass digital audio with high integrity.

I hope this helps!

Bob Katz

From: Grant Lyman

My comments are:
Bob, thanks for your informative web site and your great comments on the Mastering and Sonic newsgroups. I am hopeful I could get your opinion on a few questions I have. First, have you ever heard Toslink or Glass to sound as good as AES/EBU or SPDIF. My initial tests seem to indicate Toslink and even Glass tend to close the sound stage and thin out the sound. I have heard you mention in the past that the terminations on Toslink and Glass are critical for proper performance. Have you tested what you consider to be properly terminated Toslink or Glass against AES/EBU or SPDIF what was your conclusion.

I thank you, and value your opinion

Grant Lyman
Santa Barbara, CA

Dear Grant:

This is ultimately a jitter question, you know. My answer is that the apparent sonic differences between interface technologies such as Toslink, glass, and copper are IRRELEVANT when doing transfers or when passing signal from one processor to another. You can forget about that question with COMPLETE CONFIDENCE–since all of the technologies are capable of passing perfectly good data, within their specified cable lengths. Remember: the clock is not transferred along with the data. Only the data is transferred to the processor’s circuits.

The apparent sonic differences between interface technologies come into play in only ONE place… and that is at the input to the converters (A/D and D/A).

If the D/A is susceptible to jitter on its digital inputs (as most are), then you will hear differences between toslink (plastic fiber), glass fiber, and copper (hard wire). Some D/As reject jitter better than others, and that will determine the extent you can hear these differences. REMEMBER: This is only important to that particular listening session (the D/A only) and not to any other circumstance.

In the case of an A/D, if placed on internal sync, then its jitter (and subsequent distortion) is totally determined by its internal clock ciruits. But if you have to lock an A/D converter with external “AES” sync, the interface technology chosen may affect the stability of the A/D. Locking an A/D with wordclock produces far less jitter because there is no audio on the wordclock line, it is a pure clock. Wordclock is the second-best way to lock an A/D short of using internal sync. In the case of AES/EBU, the audio and clock are on the same line, and the audio (and other data) can cause interference during the critical clock extraction process. The different technologies (toslink, glass, copper) have different bandwidths, and reduced bandwidth (as with plastic fiber) can cause greater interface jitter.

In any case, it is preferable to put the A/D on internal sync for the lowest distortion. 2012: One possible exception. If the low frequency jitter (below the crossover point of the PLL) is better in the source wordclock generator than the internal jitter of the internal clock, then external sync can be better than internal. See papers at the Grimm website for this development.

Only by placing the master clock of the entire system within the D/A converter and feeding all devices as slaves to that clock can we eliminate these “ephemeral” differences. That way the D/A is immune to clock-induced problems on its AES or SPDIF inputs.

How can we reconcile this issue of requiring the master clock be inside the DAC, yet the A/D has to be on internal sync for lowest distortion? You can only have one master clock in a system. The answer is to design an INTEGRATED A/D and D/A system where the master clock is on a buss, feeding all the critical internal jitter-sensitive elements with a low-jitter buss-interface.

That’s what I’ve done recently in my system and I can attest that the “ephemeral” sonic differences have disappeared. (sigh of relief). At this time, I believe that integrated A/D/A systems with this technology can currently only be obtained from two vendors, TC Electronics (System 6000) and Prism. A consumer company called “Muse” has also adapted this technology on the consumer side, so there is hope. But it is sad and ironic that the audio industry has been so slow to adopt this technology, when the problem and solution have been known for years.

Hope this helps,

Bob

From: Everett Comstock

My comments are:
I hope you don’t mind, but I am writing in hopes that you could help to answer a question for me. I recently finished reading Robert Harley’s Complete Guide to High-End Audio, and I was wondering if there are jitter errors that take place within computer interfaces such as SCSI and IDE. I have read your article (which is excellent by the way!), and you seem to touch on the subject of computer based DSP cards, but what about the transfer within the computer? Can IDE or SCSI interfaces introduce jitter into the data stream, and if so, is it enough to affect the quality of the signal? Any info regarding this subject would be greatly appreciated.

Thank you,

Everett Comstock

Hi, Everett. Thanks for your comments. IDE and SCSI interfaces are unclocked interfaces which pass data asynchronously. As such, “jitter” is meaningless, because there is no clock! Data is passed completely irregularly over these interfaces and jitter at these interfaces is enormous. Sometimes it comes in bursts at 2 to 8 times speed, then there are periods of silence where there is nothing. It’s a “pass me data on demand” type of interface.

Then, the data goes to a new storage device, and that’s that. The data stored on the new storage device (the hard disc) has no jitter. That’s right. Jitter is only a question when it is introduced during the playback when a clock comes into play again. And in the case of SCSI, the hard disc system doesn’t operate in a clocked manner related to the digital audio rate at all, so even on playback, you get the same “burst” situation as in the first paragraph.

So, how does this affect your audio? Not at all. The date remains identical. It has been stripped of its clock and has no memory of how many “clocks” it has passed through during the hundreds of copies previous.

During the listening you may hear a difference between one or another hard disc-based system which is playing back identical data, you may appear to get differences when copying between such systems, but careful analysis of what is occurring will reveal that these audible differences are what I call “ephemeral”, that is, manifested by the particular clocking that is occurring, and the stability of the clock. Each time you play back the data, that is when you may examine how it sounds. Even the situation I relate in my article about how different CDs sound different, while it is quite special, is also ephemeral, but “semi-permanent”. What do I mean by that? Well, the CD player is a special case of a rotating medium where the rotational speed of the medium is intended to be related directly to the ultimate clock that is driving the final data. Thus, irregularities in the medium (the recorded CD or CDR) may affect the servo circuits in the player, which may then affect the power supplies driving the output buffer clock, which may then affect the sound during that particular playback.

Let’s say that “poor-sounding” CDR was made on an inferior writer, or one writing at a high speed. We seem to have some evidence that these differences are audible. So, what do you make of that? During the playback, at that time, if you transfer the output of that “poor-sounding” CDR back into your hard disc system and then cut once again at slower speed speed on a better-quality writer, guess what! The end result is a better-sounding CDR. You’ve restored the quality that you thought you’d lost. That’s what I mean by “ephemeral-semi-permanent”… The copy may even sound better than the “original”.

Your precious data is safe, and ultimately you only have to worry about the ultimate playback. In fact, if you have a great D/A converter, one which is relatively immune from incoming jitter, you can play back that apparently “inferior” CDR and no even hear a difference. And you’ll never know what hard disc(s) it passed through during the process.

Bob

From: Paul R

My comments are:
Thank you for the best article on jitter I’ve ever seen anywhere.
Bob Katz’s comprehensive treatment in enlightening.
He loses me on a few points, though, and I’d love it if someone could clarify.

First, I believe he states that Jitter is introduced in the conversion process, but is eliminated in the digital storage (hard disk, etc.). But then he speaks of jittery CDs. How is a CD different from other storage media? Why is jitter recorded on a CD?

Hello, Paul… Thanks. your comments are cogent. Apologies for the “work in progress”. If we knew all the answers, we’d be geniuses! I will say that a large group of mastering engineers and critical listeners agree that CDs cut in different ways tend to sound different. The CD differs from other storage media in many ways, but the critical point is that the timing of the output clock and the speed of the spinning disc are related. The output of the CD player is a clocked interface, and the data are clocked off the CD disc in a “linear” fashion, one block of data after another. A buffer is used, which theoretically cleans up the timing to make it regular again. And for the most part, it does.

A lot of this is theory… no one has proved it as fact. And there may be more than one mechanism causing jitter taking place.

To obtain jitter in the low picosecond region requires extremely accurate timing. Any leakage current (interference) between the servo mechanism controlling the speed of the spinning disc and the crystal oscillator controlling the output of the buffer may unstabilize the crystal oscillator enough to add jitter to the clock signal. This does not change the data, by the way. If the servo is working harder to deal with a disc that has irregularly spaced pits or pits that are not clean, perhaps leakage from the servo power affects the crystal oscillator. It doesn’t take much interference to alter a clock by a tiny amount.

This jitter is “ephemeral”, though, because you can copy this data (irrelevant to the clock), and then play it back again from a more steady medium… and make it sound “good” again. This is not a permanent problem.

What makes the CD different from a hard disc, is the HD uses an asynchronous interface (SCSI or IDE). The disc is always spinning at the same high speed and the heads land on the spot you need when the data is requested. The data coming out is completely unclocked (it comes out in bursts) and has to pass through the SCSI barrier into a buffer located in a different chassis than the hard disc (the computer)… thus, there is great distance between the varying currents of the spinning disc motor and the oscillator driving the output of the buffer in the computer chassis. Since the computer chassis power supply only has to power the output oscillator, the result can be much more stable. Depends on how good the designer did his/her homework. Same for a CD Player… there are audiophile CD players where great attention has been made to power supply design, and these players exhibit much less jitter and better sound.

It is also possible to build a CD player based on a SCSI mechanism… possibly such a player would be more stable in playback than a standard CD player. You would have a computer in its own “cleaner” environment buffering the data. The Alesis Masterlink is such a player, and in another “chapter” of my work in progress I will have something to say about its audible performance.

I’d like to tackle a 200 page booklet to put all the pieces together someday, but haven’t the time. I think in our FAQ there are some explanatory letters which help to cover the rough spots.

He also states that a 99th generation copy of CD is apparently identical to the original. But then talks about the degradation of making CDRs at 4x speed vs. 2x speed. Please help me reconcile this.

The data is identical… It’s important to separate the message (the data) from the messenger (the clock).

It’s all in the playback of the last disc in the chain, Paul! The “old” clock is NEVER transferred on each copy, only the data. No matter what speed you write at, there is a new writing master clock in the CD recorder that determines the spacing of the pits on the newly written CD.

But each time you copy, that clock is not transferred through the SCSI barrier of the next CD Recorder. I will have to write about this in more detail and diagram it for my readers, hopefully soon…

And each playback is a new… if the clock of the final playback is irregular, you will have jitter on the final playback of the last generation.

But you can clean that up yet again and start the whole cycle all over again.

I’m hoping the answers to these questions are within my grasp.

I think they will be, if I can just get the hang of explaining it properly!

Take care,

Bob

From: Craig Allen

Thank you for your tremendous articles. I’ve read several — mostly above my head, but good for stretching me and reminding me of the value of true audio engineers.

In my budding home studio, I am currently using Tango24 converters, recording to a Mixtreme audio card via a Tascam IFTAD (lightpipe>TDIF converter) and mixing digitally via a Tascam TM-D1000 mixer (to Samplitude).

Several days ago I had a strange experience while recording a drummer via 7 mics onto 5 tracks, while playing back half a dozen or so tracks. On a particular take (where he played very well!), after recording — the playback of the new tracks was off. The tracks started together, but gradually and increasingly the newly recorded drums lunged ahead. Sure enough, by the end of the song, the graphic even revealed the drum track was shorter (faster) than the previously recorded tracks.

Thank you very much. I really value anything you can offer here.

Craig

Hello, Craig. Thanks for your comments.

If all the tracks are already in the same workstation, I doubt it’s a clocking problem. Sounds like something defective in the workstation, hard drives or something. It’s also possible you’re exceeding the capabilities of your CPU, and that’s its way of telling you to upgrade 🙁

I don’t think this is a clocking/jitter problem if all the tracks are coming from the same A/D converter at once.

Currently, I am running the Mixtreme as the master. The Tango converters are slaving via optical outs that get converted to TDIF by the IFTAD converter. The mixer is synced via it’s TDIF port. To my knowledge, prior to this week there has been no word sync/jitter issues — at least not in audible terms of clicks or pops, or in timing difficulties.

That’s why I think it’s something else.

Someone posting on the Tascam user’s site said you need to use BNC Word clock cables/connectors — that SPDIF could only be trusted maybe 5 percent of the time. Your article recommends using the AD converter as the master and other engineer’s article I read mentioned avoiding lightpipe.

This is for jitter issues, but for clocking issues, SPDIF is just as good as BNC word clock. True, however, that if you cannot use the A/D as I recommend as the master, then BNC word clock is the best second choice, from the point of view of jitter.

Can you explain and offer me guidance?

1) What benefit does BNC cabling/connectors give over TDIF cables, optical cables, and SPDIF cables?
Short of having the A/D as the master, BNC wordclock is the simplest clock, unencumbered by being multiplexed with data. In the other systems (except TDIF) the clock is multiplexed with the data, causing jitter problems.

2) Guitar Center sells 75 ohm Apogee BNC word clock cables for $50.
The local all purpose cheapo electronics store sells 75 ohm standard video connect BNC cables for $5. What is the difference? Do I need the high end Apogees?

In a word, No. Except there are some 75 ohm cables (made by Canare, for example) that are particularly well made and flexible and will not break if flexed. I’ve seen some poor cable in my time from El-Cheapo Electronics, foil shields that break if the cable is flexed. That is the only issue.

3) If I do go to BNC wordclock, the Tango and the TDIF have BNC connectors while the TMD1000 mixer has RCA wordclock connectors, and the Mixtreme only has TDIF and SPDIF. Can you recommend a best routing scenario for me?

The card should be your master, if it contains the A/Ds and D/As. Everything else should follow.

4) If the problem I had was timing (rather than sound quality), was my problem probably more likely hard disk or computer issues than clocking issues?
For what it’s worth, the Tango, the Mixtreme, and the mixer all alert me if there are clocking problems, and they’ve never indicated a problem with my current setup.

Exactly. I believe this to be a hard disc or CPU issue.

Hope this helps,
Best wishes,
Bob

Aaron Wrote:

i hope you are well and everything is fine on your end!

i just plan to talk about your K-14 theory and have an important question. The Loudness dB SPL is measured in stereo or must the 82 dB SPL setup by playing the pink noise a mono channel. Actually i have tested both but setting up the loudness with a mono channel made a better result if i remember correct. I have done this in an earlier studio of a friend and the K-14 worked out as a good and important rule mixing the music. Also the outboard / loudness attenuator is very important. 

thank you for this outstanding work!

PS: If you never did hear from Igor again, one client from Russia told me he is alive but he was deadly ill and he tried to get back to his audio pation but his health did not allow to go into it again. He did everything on his own, soldering, PCB making and probably this isn´t something that supports his health. He still recovers but seems to be cured. I just want to let you know that because when i heard that i was happy because i have never heard about him again after visiting him. 

Best Regards

Aaron

Hi, Aaron. Good question!

In Mastering Audio Third edition, I revised the K-System to use LUFS loudness, still on a relative scale with 0 LU being -12, -14 or -20 LUFS. So since loudness on an LUFS meter is measured integrating all channels to a single meter, the new K-meter should be a single LU meter where you adjust 0 LU to one of those values. 

Since the distance of the loudspeakers and their transient response have a big effect on the perceived loudness, it’s possible that in your room, your loudspeakers, the calibration point for 0 dB on the monitor control may be different than in mine. 

In my room with the loudspeakers about 9 feet away, the K-System works well with 0 dB on the LUFS meter corresponding with -14 LUFS. Narrow band pink noise at -20 dBFS in one channel reads 83 dB SPL at the 0 dB setting. Typical K-14 to “K-16” recordings are monitored around -9 dB on the monitor control in that room. The closer the loudspeakers are to you and the brighter they are the more you will find you have to derate the calibration so 0 dB on the monitor gain may represent, for example, 80 dB per channel (one channel playing at a time) with the pink noise signal.

Hope this helps,

Bob

Sindre Saebo wrote:
Hi! I just read your book “Mastering Audio” with great interest, and plan to implement monitoring with the K-system in my project studio. Just wondering: How does the K-system relate to the new standard ITU-R BS.1770? I understand both are based on average metering, do they behave very differently in use? 

Hi, Sindre. Good question!  I hope you enjoyed my book.

In use the current K-System meter is LESS ACCURATE as a loudness meter than ITU-R BS.1770. But no one has yet implemented an exact ITU version of the K-System. Depending on the amount of bass in the music, how much compression you use and other factors you will see more or less correlation to actual loudness in the standard K-System meter, as much as 2 or 3 dB difference. The size of the loudspeakers you use may affect the “absolute loudness” part of the K-System , but ITU will help on relative loudness if you are comparing one program you master to another. If you find a company you like who produces a K-System meter, please encourage them to add an ITU compatibility mode.  Regardless, the K-System meter is still far more useful than a simple peak meter.

I recommend ITU meters now that have a variable 0 LU, such as the TC Radar meter and the Grimm LevelView. It’s a new world and we may no longer have need for the K-System, for many reasons. I’ll have a lot more to say about that in the near future.

I hope this helps,

Bob

Hi Bob,

l have emailed you in the past and appreciate your thoughts. Got your book. I’m not technically inclined as such, but I enjoyed reading it and l seem to learn something from It each time l pick the book up to have a read.

This may sound like a silly question. But l’ll ask anyway. When lm mixing I have my meters set for k-20 rms (RNDigital). Now, should the levels be hitting max of 0db (nothing above) or is it ok to hit the orange and red zone for say explosion / impact parts? But it shouldn’t be hitting that all the time. Is that the correct way of using this?
Hope I’m making sense?

Cheers,

Chris

Dear Chris:

With a K-20, you have plenty of headroom, especially if you are not using any buss processors. In addition, part of the design is that 0 dB is considered to be “forte” and up to +4 dB is considered to be “fortissimo”, so your fortissimos (including VERY big climaxes and explosions) can go into the red. Above +4 and you are probably not listening loud enough 🙂

Best wishes,

Bob

Pamela Frangou wrote:

I’m recently delving deep into the K system and studying the honor roll. I have 2 questions:

1. After calibrating my system according to the k system and listening to the CD’s on the page, would Tunes be a good listening platform and should it be set to the loudest volume on the app and playing back through my monitor system with the gain at the written values for each album, hope I’m making myself clear.

2. After testing the SPL on the meter while playing back the music it can hit like 90-100 SPL, would this be correct as I do find it loud since i’m not used to mixing at these levels.

Looking forward to your reply.

Best regards,

Pamela Frangou

Hi, Pamela.

I think you’re on the way to a good understanding of the calibrated monitor and it will help you in all your work. Keeping in mind that loudspeaker distance from the listener and size of room affect the PERCEIVED loudness at the same measured SPL. 

So, for example, a -9 in my room might have to be a -10 or -11 in your room if your speakers are closer to you, even at the same SPL calibration. But after that the relative preferences should hold pretty well. 

It’s the same reason you are probably finding it loud. If your speakers are closer than mine, for the same measured SPL they will seem louder, so you may have to change your preference, but after that all of the relative preferences will probably reflect my relative preferences (reasonably!) and you’ll be able to make judgments about overcompression and amount of compression in any recording without necessarily looking at your meters. I hope this helps,

iTunes is not a good platform for this because its volume control is not marked in decibels. Nor is the Mac’s system volume. You need to have a more professional monitor level control marked in dB. 

Bob

Name: Stephane
Subject: Monitor calibration
Message: Hello Bob, I want to setup K14 system in my studio. I use to mix pop music and I am working in stereo with Genelec 8050A as frontfield and Genelec 1037C as midfield, in a “not so bad” LEDE control room. (90 percent of the time with 8050A). I see in the EBU documentation about R128 implementation that they recommend a listening level of 82dBA SPL per loudspeaker (using -18dBFS RMS). I understand that this listening level is for working at -23LUFS and not with K14. My question in about the weithing of my SPL meter. When I calibrate for K14 (0 dB on the meter yields 83dBC SPL per channel, slow speed), I need to boost the Genelec 1037C 2 or 3 dB and read 86dBC SPL to have an homogeneous sound pressure sensation when switching from one monitor to another. Metering with A weigthing (despite the SPL reading is diferent) give a more homogeneous response to my ear between the to monitor sets. As EBU recommend C weghting for 5.1 and A weighting for stereo I would like you to confirm that the right way to setup K14 is metering my monitors with C weighting. Thank you for all your fantastical educational work!!!