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I understand now. I should have made the connection on my own.
Typically iZotope64-bit SRC with MBIT+. I have also used R8 Brain pro and it is indeed excellent.
The E-MU USB 0202 comes close:
http://www.libinst.com/EMU%200202%20USB.htm
I'd guess that Ethan was playing it safe; I'd expect 80 to 90 dB down in general to be enough but you might want to put different numbers on the different contributors.
The Crown PowerBase 2 power amp I bought almost 20 years ago claims, "Hum and noise better than 100 dB below 310 watts" which is the output power rating. For line-level gear it's even easier to achieve. My aging Mackie 1202 mixer also claims noise at -100 dB with distortion at 0.0007 percent.
I'm sure that's not correct, but I'll be glad to look at any evidence you have.
--Ethan
Yes, exactly. I used to say "artifacts 80 dB down" which I'm sure is low enough to be transparent. But a few months ago while preparing my AES presentation I decided to err on the safe side in my "standard blurb" to be sure even the most golden-eared listener won't be able to hear it.
--Ethan
That's quite deceptive of you.
Crown also quotes for the PowerBase 2:
"Total Harmonic Distortion (THD): Less than 0.05% at full bandwidth power from 20 Hz to 1 kHz increasing linearly to 0.1% at 20 kHz."
"Intermodulation Distortion (IMD): (60 Hz and 7 kHz 4:1) Less than 0.05% from less than 158 milliwatts to full bandwidth power."
This means your amplifier has the sum of all noise and disortion around -60dB (the spec does clearly say 310W @ 0.1% THD+N), and does not even come close to passing your own transparency goals, whichever one you like to pick and choose.
That's also rather deceptive of you. Mackie also quote for the 1202-VLZ Pro:
"Main Mix Noise = 90dB Signal to Noise Ratio, ref +4dBu."
"Main Mix Out THD @ 1kHz = 0.005%"
You picked an absolute noise figure in dBu with the knobs turned down! And a THD for a single mic pre insert. The sum of all noise and distortion for your mixer is at least -86dB and probably around -80dB. It does not come close to meeting your latest goals of transparency either.
My evidence uses accepted calculations, treating speaker cable as an effective voltage divider. For a typical speaker of impedance 3-30 ohms, and a typical amplifier of output impedance 0.05 ohms, then:
10ft of standard 18AWG lamp cord causes an amplitude variation of 0.32dB, and 10ft of 16AWG lamp cord causes an amplitude variation of 0.20dB.
So, your speakers range from what to what in impedence? Are they misdesigned then? Is there some reason you're using #18, too? How about you try your experiment with #12 AWG?
Then again, exactly how is this any different than any other cable with the same effective series resistance? Care to compare apples to apples, now?
As to the SNR vs. distortion issues, you need to separately talk about noise floor vs. distortion. Noise floor can be examined without signal present, with no masking due to signal. Distortion, on the other hand, happens during periods of masking by signal, and as such, the spectrum of the distortion is of utmost importance. Still, 60dB of spectral dynamic range in the instantaneous sense is not quite sufficient for the most pathological signal possible, but is quite sufficient for most normal low-pass signals.
So you can't say anything definitive there without the signal and error spectra.
Yes - the reason I use 18AWG is because that is standard lamp cord as mentioned in the video, and the reason I use 16AWG is because that is heavier gauge lamp cord as recommended for short runs such as this by Mr. Winer himself on other occasions.
I never compared different speaker cables, so I'm not sure how you can criticise a comparison that was never made.
I used speaker impedance results from Canada's NRC labs as the basis for my representation, they can be found on soundstage.com and such variations exist for many popular and respected speakers such as from B&W.
It is Mr. Winer who is making definitive statements about the sum of all distortion and noise and audibility, not me.
I'm so sorry, I stand corrected. I didn't realize that distortion products from a 20 KHz signal are audible or important.
BTW, there's no intentional deception. My printed original manual does not contain your quote above. I'll gladly scan the page and email it to you if you don't believe me. In fact, THD is not mentioned anywhere, only IMD and noise as I listed them.
I know what a s/n of 70 dB sounds like from when I owned a large pro recording studio and we had pro quality analog tape recorders. The tape hiss was always obvious. Versus my Crown and Mackie gear. Perhaps A weighting is a factor. Mackie claims unweighted for the EIN spec, but doesn't say for the s/n spec I quoted. When I get a chance I'll try to measure it myself and report here, though I don't know if the limited test gear I have is up to the task. But I'm absolutely certain my Mackie mixer is much quieter than only 10 dB less hiss than an analog tape recorder. If it weren't, it would have been sent back the same day I bought it.
A 100 dB s/n ratio is not difficult to obtain with modern gear. Even when you have to amplify very low level signals, an NE5534 dual op-amp costing 40 cents is more than up to the task. And that's the real point. The idea that you have to pay handsomely to achieve audio transparency is a myth propagated mostly by the people who sell expensive gear.
--Ethan
This is about right. I happened to use this nice little mixer for a bit for line level summing and its noise floor was in this range when there was no signal, idling with levels set. Of course, this all depends on levels, the numbers of faders in use, how quiet the source (Alesis HD24) etc.
I can't tell you what the pres are like noise-wise as the ambient sound of any place other than a very quiet studio or excellent concert hall is going to be above this anyway. (I rarely see below perhaps -65 when recording on location.)
No kidding. I wish more people understood this when they claim they can hear dither and jitter and the benefit of 24-bit recording ad nauseum. Stuff at -100 dB is 30 dB or more below the music. Nobody hears that, they only think they do.
--Ethan
I don't know what the practical real-world limit of audibility is.
However, I do know that it is very easy to hear further into the music when it is recorded in a truly quiet space with little reverberation. Artifacts are certainly going to more audible in these circumstances.
OTOH, white noise like "presence" at minus 65-70dB on a location recording is not that easy to hear unless you turn the volume up a good amount. While it doesn't appear that far down in level it really is.
No one has ever asked me to apply noise reduction to get rid of this sound on a final product. Then again, maybe they hear it and accept it as the sound of a live recording - even though they fuss over lots of other things.
j_j what are your thoughts on the limits of audibility in practical terms in this context?
Not important with pure tones, Ethan, but poor linearity at 20kHz is associated with the production of audioband intermodulation products with music having significant top-octave content. Your Crown would appear to have limited open-loop bandwidth.
I have this Mackie, so I'll measure its S/N performance at the weekend with the Audio Precision. I measure S/N ratios with the input shorted but the level control at its maximum, which gives you a true idea of the circuit's self-noise.
Not with the products I test. An unweighted audio-bandwidth S/N ratio of 100dB is relatively rare unless you turn down all level controls to their minimum setting or choose a very high output level as your reference level. Or apply an A-weighting filter but neglect to mention that fact. :-)
Incidentally, Ethan, regarding your mentioning the analog noise floor of real recordings, you can't discuss the effect of a S/N ratio without also specifying the spectrum of the noise. For example, as I mentioned earlier in this thread, typical room tone has a red spectrum, ie, it is weighted heavily toward the low frequencies. It therefore offers no masking in the region where the ear is most sensitive, ie, 2-4kHz.
John Atkinson
Editor, Stereophile
What about the other frequencies? How is -66dB lower than -100dB?
Do you stand corrected that the Crown does not achieve any of your transparency goals, let alone come anywhere close? Do you also stand corrected that the Mackie does not achieve your latest transparency goals either? And that nor does even a few feet of lamp cord as speaker cable like you also were sure about?
I can well accept that you are totally incompetent and ignorant rather than deceptive, my apologies. You did not list any IMD numbers by the way, and that you thought a manufacturer's SNR sufficed as the sum of all noise and distortion says it all!
Are you changing the goalposts now? You clearly say "the sum of all noise and distortions" should be lower than -100dB, and freqency response should be flat to 0.1dB. You are on record as saying most gear achieves this, even cheap stuff. You gave two examples of your own equipment that you thought achieved it, but you were wrong weren't you?
I am still waiting for you to show me all this cheap gear that achieves your measurement goals. $100 CD Player and $500 Amp you say on another thread here - which models please? Should be easy enough to answer if most gear does it and easily too like you claim...
An excellent point.
I am curious to learn what the measurements are for the Mackie.
Well, first, you have to separate the issue of noise in quiet and noise during signal.
Sensitivity to noise in quiet is well described by Fletcher's zero loudness curve, or if you prefer, Stevens' curve, which vary in the way they handled the ear canal resonance.
Now, noise while music is playing. This is a much harder subject to discuss. Simply put, you have to know the signal and the error spectrum (i.e. noise plus everything that isn't the intended signal).
Inside of a critical band/ERB (see one of my tutorials on hearing at www.aes.org/sections/pnw/ppt.htm for what ERB's or critical bands are, basically they are the "in band" part of the ear's built-in mechanical filterbank) you can get about 30dB in simultaneous SNR, with the most sensitive signal. With noise, it's more like 3dB to 5.5 dB, depending on the distribution and type of noise. (No, even in narrow bands, not all noise is the same.) So if you can keep a 30dB SNR always, after analysis by a cochlear filterbank, at all times, you're cool. BUT this is, if I may put it politely, damnibly hard to do, and this is not anything even remotely like what electronics or loudspeakers naturally do. If you look from low frequencies, where most energy usually resides, up to high frequencies, 70-80dB instantaneous SNR is sufficient, and 60dB is for MOST signals, but not all. (I'm presuming good recordings with dynamic range and some spectral slope. For "modern recordings", well, we're back to 30dB.) If we're talking about very strong high-frequency energy and noise at low frequencies, which is again quite unusual, you might get to 90dB with a contrived signal and contrived noise, in a very quiet room with excellent equipment.
So, I realize this isn't the most useful answer, but it's the actual answer, which, as usual in the real world, is "complicated".
So, I'd say that a noise floor that isn't audible during playback, PLUS 90dB for any noise/distortion that isn't below the threshold of audibility given the gain profile of the equipment, is sufficient.
Electronics may do this pretty well. Speakers, hahahahahahahahahahahahaha... You're joking, right?
Did you watch my video? Have you tried experiments yourself like the one I did? I just updated the download files page on my site with the noise file so people can do their own tests.
--Ethan
All good points John. Yes, please measure your Mackie 1202 and let me know what you find.
--Ethan
Thanks, j_j. Sometimes a good answer is "it depends."
Ethan, I played with your "noise injected" files quite some time ago and it is a great experience.
I found that I got better at hearing noise at lower levels with practice. Like many things audio, you can more easily hear what you learn to hear.
Okay, got my Mackie 1202VLZ on the test bench this morning.
All measurements were taken with one channel soloed and with its input shorted, and are referenced to 1V RMS at the main output.XLRs, which is SOP for preamps. The Mackie clips around 18V, so you can add 25dB to every figure if you want the ratio referenced instead to the maximum possible output voltage.
As any noise measurement depends on the meter bandwidth, I have quoted 2 bandwidths as well as the A-weighted figure.
There are 3 gain stages on Channels 1-4 - input level trim, channel fader, output fader - which are used for both Mike and Line input jacks, so I have presented several sets of measurements. I didn't measure the S/N ratio with all 3 controls set to their minimum; while this will give the lowest noise, it also gives the lowest signal level, so is not very helpful. I used three conditions with the Mike input:
Best case: Input trim minimum, channel and output faders at 0dB.
Typical case: Input trim at maximum, channel and output faders at 0dB.
Worst case: Input trim at minimum, channel and output faders maximum.
Best case was 78.2dB (<10Hz->500kHz), 88.3dB (22Hz-22kHz), 92.1dB (A-wtd)
Typical case was 55.6dB (<10Hz->500kHz), 71.0dB (22Hz-22kHz), 73.8dB (A-wtd)
Worst case was 27.4dB (<10Hz->500kHz), 39.7dB (22Hz-22kHz), 42.3dB (A-wtd)
So the little Mackie is pretty good regarding noise for such an inexpensive product, though using the input trim above the 12:00 position rapidly degrades the noise performance. Recommended for hot mikes and/or close-miking, therefore.
I use mine for location monitoring and foldback, so have never felt the need to measure it before.
John Atkinson
Editor, Stereophile
Excellent John, thanks for doing that. I have only one comment:
Mackie mixers offer a lot of extra gain, so I always leave mine with all level controls at the center detent, doing gain changes through the mic preamp trim only. I think a more reasonable "typical case" is with all channel volume controls centered, the microphone preamp set for 30 dB gain (because pros use condenser mics), and the master output volume set between 1/4 and halfway up. If you don't want to bother measuring again with those settings I understand. I don't have a real analyzer, but I could easily record the output of my Mackie with those knob settings into my Delta 66 sound card, and see what happens.
Also, are your figures A-weighted?
--Ethan
Thanks, John. Very interesting.
I have yet to meet a "pro" yet that doesn't regularly use dynamics, especially the SM57, its siblings and cousins.
Additionally the Mackie is hardly a high end unit. It is more likely to be used with dynamics as a result.
Ribbon mics have exploded in popularity as well, although the Mackie would be an unlikely choice as pre for a ribbon - not quiet enough and not enough clean gain.
For the line inputs 5-12, the gain with both channel fader and output fader set to 0dB is just 1.6dB. While Channels 1-4 can be used for high-level signals, inputs 5-12 offer the lowest noise for line-level use. Ref. 1V with the input shorted and both faders at 0dB, I measured 72.2dB (<10Hz->500kHz), 80.1dB (22Hz-22kHz), and 82.9dB (A-weighted).
You're basically attenuating the channel by 10dB in this situation, and therefore reducing the noise of the first stage by the same 10dB. The resultant 20dB of total gain is 10-12dB too low for the DPA capacitor mikes I use in a typical live jazz club recording situation. However, with the Mackie set to these conditions, I measured 81.1dB (<10Hz->500kHz), 93.9dB (22Hz-22kHz), and 96.6dB (A-weighted), all figures ref. 1V output as before.
The first two figures in each triad are unweighted, the third A-weighted.
John Atkinson
Editor, Stereophile
The Mackie's mike pre is okay with my (low impedance) SM57 for close-miking but is a little noisy for distant use.
John Atkinson
Editor, Stereophile
Do the mic pres remain in the circuit for 1-4 even when used for line level? Is this why they are noisier?
It is interesting how much the noise varies depending on configuration and gain settings.
Marc Aubort(elite recordings)uses a mackie mixer..and he is the man..the creme of the crop.
no noise complaints either.
I've rebuilt the odd 24 channel board. Holy crap, is that a big job. I've rebuilt them to audiophile standards. Man, you've no idea how incredible that sounds. You end up having to actually mix live bands down by at least a few db. The dynamics are HUGE. I mean --off the scale HUGE. (compared to any 'normal' board).
remus, is there a particular reason you are acting like an ass, or is being an ass your natural state?? Either way, you are an ass. .
I don't have a schematic but it does appear that the front-end gain stage remains in-circuit for line signals on these inputs.
As I said, you have three gain stages in series, each with a level control. And, of course, this is with one channel soloed. When you add a second channel to the mix, and every subsequent time you double the number of channels, in theory you will reduce the S/N ratio by 3dB when all the channels are set identically.
John Atkinson
Editor, Stereophile
You got that right, Bubba.
It's never the gear that makes great recordings (or great music).
Particularly good performance, but I have not yet tried this one: http://www.esi-audio.com/products/julia/
D'oh! Note to self: Learn to read. Sorry John.
Yeah, that seems closer to what I hear.
I'm not very anal with this stuff. I've been using the same setup for about 15 years. I turn it on, turn it up, and it sounds fine. Noise is never a problem. My room is pretty quiet as home studios go, and preamp noise is never the limiting factor. It's always the ambient room noise. I imagine it's the same when you record classical concerts in a hall, and live jazz in a club. Unless there are microphones placed very far away, and the players are soft, the room ambience will dominate. Further, when using a condenser microphone, there's already a preamp inside the mic itself! So that preamp establishes the initial s/n rather than the mixer's preamp.
--Ethan
Elk, meet Ethan. My Mackie has never once seen a dynamic or ribbon microphone.
--Ethan
I rarely have anything to do with dynamics (mostly classical recording) but you have got to play with some ribbons some time! They can be magical on strings and guitar amps - your instruments.
(Your recordings are very nice (and fun!) but don't you think of yourself as more of a serious hobbyist? Are you making a living/paid for recording?)
You're not the first to tell me that. One of my friends has some high-end ribbons, and we plan to fool around with them here some day. In the mean time I'm totally satisfied with my pair of audiotechnica 4033s, and my one DPA 4090, and my pair of AKG C-451s.
I used to record (and teach) audio professionally. I built my first "real" studio in the early-mid 1970s, then built a fully pro studio in 1977 which I ran until 1983 when I got sick of it. If you care:
http://www.ethanwiner.com/music.html
But you are correct that these days I spend much more time arguing with people about recording than actually doing it.
--Ethan
But you are correct that these days I spend much more time arguing with people about recording than actually doing it.
Some people argue about audio more than actually listening to it. (Not badmouthing anyone in this forum in particular, just a general view)
DPA 4090-- extremely noisy as is the 4060/4061.
maybe id use them for close micing or something..but never anything acoustic.
Cool!
I started using ribbons when they were out of favor and was tickled with what they can do. I bet that a ribbon would be gorgeous on your 'cello (one of the most gorgeous and expressive instruments there is).
I didn't know this. Neat!
You never used a dynamic or ribbon mic in your studio?
When I had my pro studio we had lots and lots of dynamic microphones. Offhand I remember having E-V RE20s, Sennheiser MD-421s, AKG D-110 (or was it 120?) and surely others. But my go-to mics were always the LDC types, and the dynamics just filled in for not having as many U87s as I wish we could afford.
--Ethan
So your phrase "pros use condenser mics" just meant that you are currently not using anything but condensers.
This is a pretty limited universe of "pros."
You and all the rest of us.
LOL, but yes you are correct. I do tend to live in my own little ego-centric universe.
--Ethan
As do we all.
Filtering everything through our own world view is hard to avoid.
I just thought of another relevant point:
Dynamic microphones are often used in recording studios, but mostly for close-mic'ing drums and guitar amps. So preamp noise is usually not a problem with such high SPL levels.
--Ethan
Wondering if you have one or a few favorite ribbon mics?
And vocals, and sax, and brass and . . .
Ribbons I like: The AEA R84, Royer R-121, Beyerdynamic 160 (double ribbon hypercardiod!). Don't forget the Apex 205 (amazing for $99 and responds well to mods).
Lots of other good ribbons, too.
Thanks, those are certainly familiar. I've been interested in the microphone end of things for many years since they're somewhat analogous to speakers and mic preamp design is an interesting challenge.
The transducers at the beginning of the recording chain and at the end of the playback chain have the largest impact on the sound we listen to in our homes.
The electronics and wires in between do relatively little in the overall scheme.
I have a number of Ribbon mics , the Beyer m130/m160 (the ribbon equivalent of a 57 ..work on anything) , Coles 4040s(my favorite ribbons, beautiful) and the AEA r84.
Id happily work with nothing but the coles.
I should have mentioned Coles ribbons.
I need to pick up an m130 to use with one of my m160's as a mid-side pair.
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