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Readers' Letters: Why a digital signal is never accurate, and more...

A post by David Mellor
Thursday November 30, 2006
A selection of readers' letters recently submitted to Record-Producer.com
Readers' Letters: Why a digital signal is never accurate, and more...

In response to Why a digital signal is never accurate, Magnus Olofsson writes...

You claim that there's always added noise and distorsion when going digital with a 24 bit signal. That is completly untrue.

The added noise and distorsion is way way waaaay below the noise and distorsion of the analog part of the DA and AD converter. So all noise and distorsion from the digital conversion is buried in noise floor of the analog components.

So there's no loss of detail because of the digital conversion initself. It is the analog side that can't keep up.

The quantization noise and distorsion is simply irrelevant in a 24 bit converter.

RP response: One could go so far as to say that it's the engineer who can't keep up! We take your points, but many people are concerned about the quality of converters. If they were all perfect, then we could just use them and forget about them. One day...


In response to Can the E-mu Proteus X replace the hardware sound module?, Karl Moeller writes...

Where have all the MIDI MODULES gone?

Once a mainstay of the electronic music business, and of home and pro studios, the hardware MIDI sound module has become nearly extinct.

The latest issue of Sweetwater shows that Yamaha sells exactly TWO MIDI sound modules. Kurzweil sells two, I believe. E-Mu Systems, sampling pioneer, was bought out by Creative Labs, the PCI soundcard company, for their chip technology, and sells exactly ZERO sound modules, voting in favor of computer based streaming audio libraries which require - you guessed it - high end sound cards. Creative Labs also bought Ensoniq for their technology, and disappeared the entire product line.

Using a capable multitrack sequencer and a multi-output MIDI switch such as a MOTU Midi Time Piece, a rack of MIDI modules - whether synth or sample - based, is a powerful, easily affordable way to extend the audio capability of any studio.

But a recent eBay search for Kurzweil 1000 rack modules displayed none available. Only a few E-Mu rack units were available, and the prices are climbing. Take your own survey.

What has replaced the MIDI sound module in the music industry? Computer based streaming audio libraries. There are a dozen clamoring for our attention, few standards and little interoperability. On a sound to sound basis, the quality may be somewhat higher than a MIDI ROMpler can provide. However, the computer platforms the libraries run on must be very robust, with fast processors, 2-4GB memory minimum, and large, fast disks. And, unlike a dedicated hardware module, they run on deliberately - obsoleted operating systems which, at least in Microsoft's case, have traditionally been riddled with security holes. Product incompatibility, high prices, computer platform fragility and high prices don't make this a very appealing alternative to dedicated hardware MIDI sound modules.

One has to ask if this is better for the consumer, or the manufacturers, given the trend to O.S. obsolescense (what exactly was wrong with XP?) and application version churn, it appears to favor the manufacturer.

Karl Moeller

Tucson AZ, USA

mkarl2@qwest.net


In response to Seven pro microphones tested, with audio - THE RESULTS!, Larry@adkmic.com writes...

I wonder what your favorite voice-talent mic might be had you included a Hamburg or S-7 ADK in the test you performed(?)

Those mics are fine-tuned for male Vox; there are indeed optimal response curves for every instrument in an orchestra !!!

The SM Series, with it's lack of sonic information in the 10k to 20k range make them useful for emphasizing 'chest-tone' and for taming sibilance. There are other ways to acheive this and still have sonic information in the upper frequencies (Notably the curves of the U-67). Anyway - that's my point of view as an end-user of microphones for 40 years . . . It's all about Matching The Timbre of the Mic to the Timbre of the Source (by way of the Timbre of the Pre)



Respectfully,

Larry J. Villella, Founder, ADK Microphones

RP response: Hi Larry, if you have some demo files you could send over, we would love to publish them on the site.


In response to When to pan left, when to pan right, Jonas/studielyd.dk writes...

Well. An obvious example would be toms in a drum kit. Do you mix it like the drummer hears it or from the perspective of the audience. In my experience producers are doing both in todays records. I do a lot of live sound, so obviously I do it from the audiences perspective.


In response to When to pan left, when to pan right, Ferry Verhoeve writes...

Hi,

Wy wife is a pianist and years ago when we start recording together we had some discussions about placing and sound.

She, as the player wanted to hear the piano from her perspective, and I was used to listen "sideways" we choose for the panning her perspective and of course not fully left right but a slightly smaller "view " to keep the sound as realistic as possible.

In pop music the choice is as far as I look at it more free.


In response to When to pan left, when to pan right, Walt Rogers writes...

The decision on panning should be determined after giving thought as to how you want the final aural image to be presented to the eventual listener. If the realism of a concert performance is to be reproduced, not only should panning be considered from the final perception of the audience, but microphone proximity and location which will give that desired perception, as well.

I often record grand piano, and few instruments can be closely miked without getting undesired action noise. Often, I use my personal rule of "sevens". For a 9' Hamburg Steinway piano, I place two microphones seven feet apart, seven feet in the air (as if the inside of the full-stick lid was directing the sound to the microphone), and seven feet away from the center of the piano (figuring into this dimension the curvature of the side of the piano so that the microphone on the narrow end of the piano is closer to the edge of the piano than the one on the keyboard end). The microphones are far enough away to lose most action noise in relation to the musical sound, this arrangement seems to channel the best combination of sounds to the level of the microphones, and I can stand in the center of the microphones and hear a fairly accurate approximation of what the recorder will hear (including a small amount of room reflections). Further panning adjustments can be fine tuned by using monitoring headphones and removing them to get an A-B comparison of what I hear without them and what the recorder will hear. By this I can make final adjustments to match the actual sound.

Placing microphones under or inside the piano may give a grandiose sound, but it is not what the audience hears.

This works for me. The bottom line is to do what it takes to get the sound you want (whether one wants natural or not).


In response to Three microphone preamps tested. (One of them is pretty bad!), Karl writes...

Hmmm, Here are my findings:

Test 1 , decent balance of noise to signal, but the voice sounds hollow.

Test 2 revealed the lowest noise floor (background noise is almost gone), but the voice didn't sound that great (lacking on the high end).

Test 3 revealed the highest noise floor, but the voice sounds the most present and in your face

Test 4 Has a healthy balance of signal to noise, but sounds harsh on the high end and lacking in the mids)

It sounds to me the speaker may have slightly changed microphone positions during each recording. If not, these test really reveal the effects of a preamp on a microphone.

None of these preamps wowed me and really, I could work with any of the 4, but I would need additional signal processing with each one too.

My conclusion, if the differences are thousands of bucks, it wouldn't be worth it.


In response to Why do old people listen to old music?, Keith Kurtiss McIntosh writes...

To me, it's simple. People of all ages listen to the classic era artists( I'm more of a hard-rockin' guy)because it's BETTER music. The artists of the late 60's and 70's: Montrose, Aerosmith, Styx, Sweet, Deep Purple, Uriah Heep, Led Zeppelin, my fave Blue Oyster Cult, and many others of the day.

Most of what you hear these days simply doesn't have the substance of the classic era bands. Listen to the new Scorpions CD "Humanity:Hour One". This is a brilliantly produced and performed collection of thoughtful, intelligent hard rock. The classic era artists just did it better.


In response to Why aren't the record labels actively looking for YOU?, Prometheus writes...

"But that's the way the music industry works. Unless you can come up with a way to change the industry, you had better learn to live with it."

And they try to make us feel guilty about downloading mp3's... What should be on trial here is not the poor artist struggling for recognition, but an slothful, besotted industry that works on dodgy deals, back handers and corruption and yet has the audacity to place merit at the bottom of its list of requirements...

Personally, I feel they might have the curteousy to at least send an email back to the artist who has put a lot of time, money and effort into his project, just to say, "Sorry, not interested." if nothing else.

There's nothing worse than sending a press pack out and hearing nothing back so you don't even know whether or not it has even arrived at the hallowed halls of these pathetic creatures who play God with the life's work of the artists in question...


In response to How to tell if your preamp is bad, Calaveras writes...

Your logic makes no sense.

a preamp at full gain usually has more thermal noise appearing at the ouputs than one at 20db less gain. Of course at lower gain you will also often be routing the signal through a bit more resistive material, depending on the topology of the preamp and gain control of course.

Furthmore, most manufacturers specs that I have seem on prosumer/MI gear like Mackie Focusrite etc, spec theri gear at whatever their gear is comfortable with to get a good figure. It is not uncommon to see mixers specced for one channel at unity gain input to master out, all other faders down. This gives a flattering picture of the noise SNR but is not realistic. As far as how important preamps are. I would place them before mics, but after mic techmique. Some of my recent preamp acquisitions have rendered previously unusable mics, very usable. Also, a really pro preamp will have neat stuff like variable HPF, variable impedance, and will be just plain euphonic.

RP response: A preamp at 70 dB gain (for instance) should in theory produce 20 dB more noise than a preamp set to 50 dB gain. The signal-to-noise ratio will be the same. In practice however, preamps often produce a better noise specification at high gain, which is why manufacturers often quote the EIN (Equivalent Input Noise) spec at full gain, which unfortunately does not reflect typical use. If a preamp has an unduly high noise output at high gain, this is indicative of a fault, or poor design. audiomasterclass.com recommends that the selection of microphone will normally take precedence over selection of preamp. We do of course recognize individual preferences.


In response to Q: What is the best way to work in a studio if you are the musician, producer and engineer, all by yourself?, Calaveras writes...

Working alone has its benefits. Besides modesty, there are also cost considerations. In both cases, the best bet is to either get an engineer, or yourself, to set up 3 or 4 signal chains. Lets say one for vocals, one for guitar, one for bass one for drums. Have them all patched an ready to go at any time. Just flip on the power, wait a few minutes for warmup and go!

This is how my studio is set up. I even have the computer set to power on automatically on powerup of my studio gear. All Ihave to do is hit one switch, type in a password and click on the icon for Sonar 7.

Also in a "real studio" even working by yourself, it is a great asset to have a seperate room. One word guitar amps. Ok thats two words, but it is so great to be able to record an amp in a seperate room and really hear just what the mic hears.


In response to When to pan left, when to pan right, Stephen Olcott writes...

The placement of the piano is more important then the length of the strings. The panorama of the audio canvas is 8:00 to 4:00. Center stage, Stage Left, Stage Right, and the increments in between. You then must take into account whether the instrument is Stage Front or Stage Rear of the Conductor for placement in the audio field. So after you establish where the piano is, ie: 10:30 to 2:30 you then use gain to place it by level fore or aft Stage Front (louder) or Rear(softer) in relation to all the other instruments, String section from 12:00 to 4:00, Brass 9:00 to 11:00 etc. Watch a performance on PBS or the BBC and block out the sections and you'll get the picture. You slice the sound canvas like a pie, 9:00 to 3;00 or 8:00 to 4:00. Individual effects but one verb for the whole room. I prefer to let them hear what the performer hears, mostly left right, but tight so it's more like a solo vocalist with an orchestra. Good luck.

RP response: The 'clock' theory is good, taking into account the actual width occupied by the piano in the sound field. We would not advise using effects on an orchestral recording however, unless there was some special purpose in mind. Digital reverb may be used as a matter of expediency, but the natural reverberation of the auditorium is always preferable.


In response to 1000 dB - how loud is that?, Jake writes...

Rubbish! a 1000w amp and speaker can easily put out 100db, even low wattage horns on cars can put out more dB than you claim an F1 car could generate.

RP response: It's a thousand dB, not a hundred! A thousand is like a hundred, only ten times bigger. Peace :-)


In response to When to pan left, when to pan right, Simon writes...

If a piano is position on the left hand side of the stage it would seem logical that the one channel should just be panned more left than the other channel, otherwise the instrument would appear to be, horizontally, much larger than actually it is. Also room must be left for you other instruments in the stereo field.

RP response: Very true. We will be looking into the apparent size of instruments in the stereo sound field in a future article in Audio Masterclass


In response to When to pan left, when to pan right, Chrisc_o writes...

Good point.

I liken your piano example to my panning "style" when approaching drums: I prefer panning from the drummers perspective rather than the listener's perspective.


In response to Do you find this photo scary? You should., Gerard writes...

There's a difference between marketing microphones and using microphones. If you believe everything you see in marketing collateral, you'd be a poor engineer indeed! Ears are the judge, and the 4061s are omnis, so dismissing the technique based on a photo alone is a bit nearsighted.

Mic placement for piano recording is one of the most complex tasks I know. It depends on so many variables it can make your head spin. Mic type, pattern and technique, mono or stereo, piano make and model, lid up or down, room size and reverberation characteristic, live, broadcast, or studio, solo, duet, ensemble, orchestral, jazz...! Not to mention micing for effect. Keith Jarrett uses DPA 4011's stuffed inside to get a particular focused sound. I prefer stereo 4015's - wide cardioids - to get a little more room sound, but still maintain reasonable isolation from other instruments.



But in the end...the ears have it.

RP response: Microphone technique is very subjective and open to opinion. Our subjective opinion is that the technique in the photo will, among other things, achieve the sound of putting your head inside the piano. There are indeed times that you might want this. But it's good to be aware that this is not at all a natural sound.


In response to 'Reverse latency' - is there such a thing? One Audio Masterclass visitor seems to have a problem..., Danny Binders writes...

Hi there, just to let you know you're not alone on this. I use Logic with an Audigy and get exactly the same thing. I tried the original Realtek soundcard and was still the same. I tried another recording program and yep, was still doing it. So i'm ready to throw the whole set-up straight out the window now. Just wondered if you have solved it or can shed any light on this. Heres hoping,

Danny.

A post by David Mellor
Thursday November 30, 2006 ARCHIVE
David Mellor has been creating music and recording in professional and home studios for more than 30 years. This website is all about learning how to improve and have more fun with music and recording. If you enjoy creating music and recording it, then you're definitely in the right place :-)
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