Taperssection.com
Gear / Technical Help => Battery Boxes, Preamps, Mixers, ADCs, and Processors => Topic started by: jerryfreak on July 15, 2008, 01:14:15 PM
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just wondering about your thoughts on this.
My new rig doesnt have a pre, and it sounds great to my ears (MK4v>CMR>battery box>AD2K)
normalizing once in soundforge at about about +25 - +30 dB doesnt seem to hurt the sound at all.
While minimalist, this is actually my new 'open' rig. I'd love love love to find a super small, super low noise floor stealth AD to eschew the gain stage entirely while stealthing. Whose got time to check levels when you're watching your back? the only good small AD of fthe top of my head is the sbm-1, but its only 16bit, and wouldnt have the headroom to pull this off.
i was hoping to get some feedback on this setup from the more electrically inclined. as to any major disadvantages with regard to headroom, dynamics, etc.
it seems with an AD capable of a clean -117db, and a mic with a noise floor of -75 to -80, id never be limited.
check it out. AD2K was set to its highest input level (+14 dB, minimum attenuation):
http://www.sendspace.com/file/uzt783
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so you're saying that you need to raise the level by 20-30db after the fact to get it up to a listenable level?? What are you're meters reading on the ad2k while recording??
Edit: the samples do not sound overly noisy, but man, that is a really low signal to feed the recorder. and could get noisy as stated below.
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Lots of Features, Small Size, Cheap Price .... you can have any (2). But not all (3) at the same time. :)
normalizing once in soundforge at about about +25 - +30 dB doesnt seem to hurt the sound at all.
But normalizing not only increases the level of your sound, it also increases the noise by the same amount. But if your music material is loud enough to mask the noise, then you'd probably not notice it... except for pauses or soft passages.... then the noise floor will rear its ugly head.
And just to be specific, there are "line preamplifiers" and "mic preamplifiers". Line preamp work with higher line level signals. Mic preamps work with input signals in the millivolt range. Both are referred to as preamps... and even in the mic peamp category, you can have a simple electret mic preamp, or a standard mic preamp that can work with regular dynamic, condenser (with the appropriate 48V phantom power) or ribbon mics.
Not all "preamps" are the same.
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to answer the poster above, the original wav peaks between -30 and -25 dB
while normalizing does increase the noise obviously, you run into the noise of the capsules well before the noise of the A/D, so regardless of where the levels are at, it all sounds the same. with this setup, once normalized you can clearly hear whispers across the room that peaked at like -70 dB (note the recording above is a noisy bar room, near a fan too)
preamp 'flavors' aside, there is always a compromise when you amplify anything (distortion). IMO, the noise introduced by doing it digitally is the same (or less) than a separate physical gain stage.
What i am concerend about, is the digital theory of defining a waveform at levels that low.
in other words, are their less bits available to define the waveform when recording in the range of -30 to -110 dB as opposed to 0 to -80 dB? or is it essentially all the same as youre above the analog noise fllor of the preamp, which would act as a natural dither? I'm thinking about how poor 8 bit audio sounds when recorded low and then normalized. obviously the bit depth of 24bit is all about increased resolution at lower levels, i just down want to lose any subtle transients or dynamics of soft passages.
in tests, it sounds the same to my ear as using a pre (preamp flavors aside)
Lots of Features, Small Size, Cheap Price .... you can have any (2). But not all (3) at the same time. :)
normalizing once in soundforge at about about +25 - +30 dB doesnt seem to hurt the sound at all.
But normalizing not only increases the level of your sound, it also increases the noise by the same amount. But if your music material is loud enough to mask the noise, then you'd probably not notice it... except for pauses or soft passages.... then the noise floor will rear its ugly head.
And just to be specific, there are "line preamplifiers" and "mic preamplifiers". Line preamp work with higher line level signals. Mic preamps work with input signals in the millivolt range. Both are referred to as preamps... and even in the mic peamp category, you can have a simple electret mic preamp, or a standard mic preamp that can work with regular dynamic, condenser (with the appropriate 48V phantom power) or ribbon mics.
Not all "preamps" are the same.
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Lots of Features, Small Size, Cheap Price .... you can have any (2). But not all (3) at the same time. :)
sweet, i dont need features! Brew me up a small, cheap HQ AD, please! ;)
FYI, here are the specs on the gear:
mics:
Sensitivity:
ca. 0.6 times the value as measured with a standard CMC microphone amplifier. For example with the MK 4 cardioid, the sensitivity is about 8 mV/Pa, the maximum sound pressure is 130 dB SPL and the equivalent input noise is
29 dB CCIR** or 18 dB A-weighted*.
Maximum output voltage:
900 mV with 20 kOhm load impedance (ca. -1 dBV)
560 mV with 2 kOhm load impedance (ca. -5 dBV)
Output impedance: 15 Ohms at 1 kHz
A/D:
THD + N = -107 dBFS (0.00033%) at -1 dBFS
Dynamic range = 117 dB, A weighted
Input level for 0dBFS = +14 to +24 dBu - switch selectable
Input Impedance = not specified.
By my logic, with a close to 900mV output from the mic, the mic would reach its max spl of 134 dB somewhere between FSD and -10 dB - in other words, you would never have to set a level with this setup, and it offers the full dynamic range of the mic. To reiterate, yes there is a noise floor there, but it is the mic capsule itself. so for a given sound source volume/mic location, the noise on your recording will be the same regardless of if you are recording at 0db or -30 dB and post-normalizing.
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Normalizing 24bit from -30 is probably fine but you need to have your ears checked if you don't hear a difference on 16bit sources.
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yes, this wouldnt work with a 16-bit AD, and i would not attempt that
Normalizing 24bit from -30 is probably fine but you need to have your ears checked if you don't hear a difference on 16bit sources.
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I dont know...., add a preamp and then see if its "no difference".
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By my logic, with a close to 900mV output from the mic, the mic would reach its max spl of 134 dB somewhere between FSD and -10 dB - in other words, you would never have to set a level with this setup, and it offers the full dynamic range of the mic. To reiterate, yes there is a noise floor there, but it is the mic capsule itself. so for a given sound source volume/mic location, the noise on your recording will be the same regardless of if you are recording at 0db or -30 dB and post-normalizing.
I don't think there are any gear with 20K input impedance... so I would use the lower value of 560mV at 2K load. That's more realistic.
By my logic, with a close to 900mV output from the mic, the mic would reach its max spl of 134 dB somewhere between FSD and -10 dB
If I understand your sentence right, you're looking at this backwards and equating apples to oranges.
The max SPL of the mic is the loudest level it can be subjected to. (And note that 130dB is 10dB more than the threshold of hearing pain, 120dB. )
134dB SPL is not the output level of the mic. The mic will never "reach" this 130dB SPL.
SPL = sound pressure level. The max SPL the mic can withstand... NOT the mic can reach.
the noise on your recording will be the same regardless of if you are recording at 0db or -30 dB and post-normalizing.
No, it's not the same.
Normalizing will multiply by a factor of X both the signal and noise. Your overall volume level may be up, but your dynamic range is still the same if you normalize it.
Amplifying the signal while keeping noise down not only increases your volume level but also your dynamic range. This is not something "normalizing" can give you. If it did, all pro recording engineers out there will be normalizing instead of working hard to reduce noise.
Earlier you said, but its only 16bit, and wouldnt have the headroom to pull this off.
I think you meant to say "dynamic range."
As 16 bit will only have 96dB dynamic range, well below the 120dB dynamic range of human hearing.
Headroom has got nothing to do with 16 or 24 bits or dynamic range. Headroom is NOT THE SAME as Dynamic range.
"Headroom" is the maximum voltage level that your gear can work with without clipping, and is measured in dBu (note the "u", it's not just dB). Where 0dBu = 0.775Vrms = 2.45Vpp
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Lots of Features, Small Size, Cheap Price .... you can have any (2). But not all (3) at the same time. :)
sweet, i dont need features! Brew me up a small, cheap HQ AD, please! ;)
FYI, here are the specs on the gear:
mics:
Sensitivity:
ca. 0.6 times the value as measured with a standard CMC microphone amplifier. For example with the MK 4 cardioid, the sensitivity is about 8 mV/Pa, the maximum sound pressure is 130 dB SPL and the equivalent input noise is
29 dB CCIR** or 18 dB A-weighted*.
Maximum output voltage:
900 mV with 20 kOhm load impedance (ca. -1 dBV)
560 mV with 2 kOhm load impedance (ca. -5 dBV)
Output impedance: 15 Ohms at 1 kHz
A/D:
THD + N = -107 dBFS (0.00033%) at -1 dBFS
Dynamic range = 117 dB, A weighted
Input level for 0dBFS = +14 to +24 dBu - switch selectable
Input Impedance = not specified.
By my logic, with a close to 900mV output from the mic, the mic would reach its max spl of 134 dB somewhere between FSD and -10 dB - in other words, you would never have to set a level with this setup, and it offers the full dynamic range of the mic. To reiterate, yes there is a noise floor there, but it is the mic capsule itself. so for a given sound source volume/mic location, the noise on your recording will be the same regardless of if you are recording at 0db or -30 dB and post-normalizing.
I wont try and teach you about audio ;) Or preamps because l I know you already know all that stuff. But one of the reason why your recordings are so good is because the self noise on the mics your using and your signal path is very low and your source was very loud thus your signal to noise was very good at the settings needed to obtain the level to capture the show. Adding a good preamp will mean your signal to noise actually improves but in some situations in some recordings that's not something you might notice. It all depends on the level of your source. But good preamps also add flavor to the signal something I know you are aware of. I think in the end there are times when you dont need one and there are times when you do. But the nature of the concert taper is one of never knowing for sure who is mixing the live sound and how loud it will be to a certainty. Thus having a preamp available at all times is a good thing provided its not taking away all the sweetness of your great mics.
Chris
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Thus having a preamp available at all times is a good thing provided its not taking away all the sweetness of your great mics.
So can I therefore conclude, Chris, that since I use preamps, I'm not a pussy? ;D ;D
Bumper sticker idea...Preamps are for REAL men! [Notice font color is pink.] ;D
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Thus having a preamp available at all times is a good thing provided its not taking away all the sweetness of your great mics.
So can I therefore conclude, Chris, that since I use preamps, I'm not a pussy? ;D ;D
Bumper sticker idea...Preamps are for REAL men! [Notice font color is pink.] ;D
Thats right man... Preamps are for real men.. Who like it "hot"
Thank you I will be here all week folks... Thanks for comming out! :P
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I don't think there are any gear with 20K input impedance... so I would use the lower value of 560mV at 2K load. That's more realistic.
thanks, that actually jives better with my observations
By my logic, with a close to 900mV output from the mic, the mic would reach its max spl of 134 dB somewhere between FSD and -10 dB
If I understand your sentence right, you're looking at this backwards and equating apples to oranges.
The max SPL of the mic is the loudest level it can be subjected to. (And note that 130dB is 10dB more than the threshold of hearing pain, 120dB. )
134dB SPL is not the output level of the mic. The mic will never "reach" this 130dB SPL.
SPL = sound pressure level. The max SPL the mic can withstand... NOT the mic can reach.
yes i understand that. My point was that the (lack of) gain strategy allows for the full range of the mic, and that the mic would suffer input distortion before the rest of the signal chain.
the noise on your recording will be the same regardless of if you are recording at 0db or -30 dB and post-normalizing.
No, it's not the same.
Normalizing will multiply by a factor of X both the signal and noise. Your overall volume level may be up, but your dynamic range is still the same if you normalize it.
Amplifying the signal while keeping noise down not only increases your volume level but also your dynamic range. This is not something "normalizing" can give you. If it did, all pro recording engineers out there will be normalizing instead of working hard to reduce noise.
i have to disagree with you on this one For this example, im going to refer to the self-noise spec of the mic as opposed to dynamic range, please correct me if im wrong in any assumptions here.) Ok here goes. The signal coming off the mic is what it is. The noise coming off the mic is what it is. These are in a fixed ratio (maybe not ratio, but fixed dBA difference lets say) This is why placement is so important, to get the highest level possible into your mic and keep the noise down. The noise level of the mic relative to the input signal is entirely independent of ANYTHING you put downstream. Lets assume my source has a maximum volume level of 100 dBA (typical not-too-loud concert environment) and the mic noise is 18 dBA . If i use a preamp and record with peaks at 0dB, the self noise of the mic is at -80 dB. If i dont use a preamp, and my peaks are at -30, the self noise of the mics is recorded at -110 dB. When normalizing, you amplify the following three signal (or noise) levels):
1. peaks get normalized from -30 to 0 dB
2. self noise of mic gets amplified from -110 to -80 dB (the same as if you had used a preamp for gain)
3. self noise of the A/D gets amplified from -117 dBA to -87 dBA (inaudible? relative to mic noise)
4. the 'quantization noise'? (digital noise floor, LSB, whatever) gets amplified from -144 dB to -114 dB and subbsequently dithered - absolutely inaudible
With this setup, no matter what you do, i beleive it is impossible to create additional audible noise that wouldnt be masked from the mics in the first place. in other words, regardless of what gain you use, if your having noise problems, its because the volume of your source wasnt high enough and the self noise of your mic is controlling. i have tried to hear the noise floor of he ad2k, i cannot distinguish it from the noise floor of the mics themselves.
Lets take an extreme case: peaking at -70 dB vs using 70 dB of gain in the field - all youre gonna hear is predominantly the noise of the mic, and thats a mic placement issue, entirely independent of signal chain - 'garbage in, garbage out.
Earlier you said, but its only 16bit, and wouldn't have the headroom to pull this off.
I think you meant to say "dynamic range."
As 16 bit will only have 96dB dynamic range, well below the 120dB dynamic range of human hearing.
Headroom has got nothing to do with 16 or 24 bits or dynamic range. Headroom is NOT THE SAME as Dynamic range.
"Headroom" is the maximum voltage level that your gear can work with without clipping, and is measured in dBu (note the "u", it's not just dB). Where 0dBu = 0.775Vrms = 2.45Vpp
thanks, yes i was speaking of dynamic range. i kno wim not the first one to brutally use those words interchangably :)
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I wont try and teach you about audio ;) Or preamps because l I know you already know all that stuff. But one of the reason why your recordings are so good is because the self noise on the mics your using and your signal path is very low and your source was very loud thus your signal to noise was very good at the settings needed to obtain the level to capture the show. Adding a good preamp will mean your signal to noise actually improves
again, this is where i disagree, sure a preamp would boost the signal, but it would also boost the self noise of the mic equally - and thats the limiting factor in this signal chain.
but in some situations in some recordings that's not something you might notice. It all depends on the level of your source. But good preamps also add flavor to the signal something I know you are aware of. I think in the end there are times when you dont need one and there are times when you do. But the nature of the concert taper is one of never knowing for sure who is mixing the live sound and how loud it will be to a certainty. Thus having a preamp available at all times is a good thing provided its not taking away all the sweetness of your great mics.
again, i think the extremely low noise floor of the AD2K offers me the full dynamic range of the mics , even at quiet levels, so unless i want a 'flavor', i theoretically would never need a preamp.
as i stated before, no gain is perfect. if you us a pre, there will be distortion (measurable, not necessarily audible). by doing a single normalization, there will be distortion in the form of rounding errors (also measurable, not necessarily audible). i propose that even considering the rounding errors, a simple digital algorithm of normalization would introduce less distortion than ANY analog audio component.
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LOL - my obviously inflammatory post title is attracting attention and generating the traffic on this thread that i had hoped for.
I still think ALL you guys are too cool for skool!
Thus having a preamp available at all times is a good thing provided its not taking away all the sweetness of your great mics.
So can I therefore conclude, Chris, that since I use preamps, I'm not a pussy? ;D ;D
Bumper sticker idea...Preamps are for REAL men! [Notice font color is pink.] ;D
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i dont really know much about the techno-babble, but doug oade once said, "the best pre-amp, is no pre-amp."
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I dont know...., add a preamp and then see if its "no difference".
i did, i didnt hear one.
i was looking for a technical discussion on the digital theory of turning your music into bits in the lower range. in other words, in a 24-bit realm would an ADC digitize a waveform peaking at -30 db with the same resolution as one peaking at 0? by my understanding, a 24-bit recorder will have the same resolution as a 16-bit recorder when peaking at -48dB. That said, noise of the adc aside (which is lower than the mic anyway), i think i answered my own question.
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please, please, somebody make me that sticker for my ad2k
i dont really know much about the techno-babble, but doug oade once said, "the best pre-amp, is no pre-amp."
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i dont really know much about the techno-babble, but doug oade once said, "the best pre-amp, is no pre-amp."
i was going to say this exactly (quoting Doug...who ironically makes a living making/moding preamps. maybe he's just gutting our all in one boxes to the "no preamp MOD" )
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interestingly enough....the KMD Neumanns come to mind.
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I started using an outboard preamp at a time when very few open concert tapers did so, 1990. What I found then, and still content today, is that an outboard preamp can harness very loud sources better than any recorder where the mics are plugged directly into it.
So my personal experiences with the benefits of preamps does not deal with very quite and/or extremely dynamic sources where noise levels are a big consideration, as the OP seems to be concerned with.
Most all past portable DAT recorders, and most of today's wave/DSD file recorders, seem to operate best when they are fed a clean line level signal that falls within the normal line level ranges (-20 consumer/+10 professional). So, by placing an outboard preamp in the path to deliver a line level within the optimal range the recorders are designed to handle to my ears usually sounds better.
All the Sound Devices users I know personally, while willing to use the unit with mics straight in when a stripped down rig is required, they all swear by outboard preamps. I have heard comps between V2/V3 versus no preamp, Sonosax versus no preamp, Neve, PSP3, VMS, MP-2, etc.etc. and rarely have I been unable to pick out the difference between one of those preamps versus using no preamp at all.
I will not attempt to discuss the benefits of one brand against the others, since part of the art of taping is matching and tuning a rig based on how different mics and preamps sync together. Some are matches made in heaven, and others are downright ugly.
Bottom line, the original poster's joy in not using a preamp for HIS particular setup is great for HIM. If HE likes it for HIS recordings, that that is all that should matter to HIM. I think it's a waste of time to attempt to draw others into supporting a rationalization based on technical arguments.
To repeat Doug Oade's best and most famous quote, "trust your ears".
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I started using an outboard preamp at a time when very few open concert tapers did so, 1990. What I found then, and still content today, is that an outboard preamp can harness very loud sources better than any recorder where the mics are plugged directly into it.
So my personal experiences with the benefits of preamps does not deal with very quite and/or extremely dynamic sources where noise levels are a big consideration, as the OP seems to be concerned with.
Most all past portable DAT recorders, and most of today's wave/DSD file recorders, seem to operate best when they are fed a clean line level signal that falls within the normal line level ranges (-20 consumer/+10 professional). So, by placing an outboard preamp in the path to deliver a line level within the optimal range the recorders are designed to handle to my ears usually sounds better.
All the Sound Devices users I know personally, while willing to use the unit with mics straight in when a stripped down rig is required, they all swear by outboard preamps. I have heard comps between V2/V3 versus no preamp, Sonosax versus no preamp, Neve, PSP3, VMS, MP-2, etc.etc. and rarely have I been unable to pick out the difference between one of those preamps versus using no preamp at all.
I will not attempt to discuss the benefits of one brand against the others, since part of the art of taping is matching and tuning a rig based on how different mics and preamps sync together. Some are matches made in heaven, and others are downright ugly.
Bottom line, the original poster's joy in not using a preamp for HIS particular setup is great for HIM. If HE likes it for HIS recordings, that that is all that should matter to HIM. I think it's a waste of time to attempt to draw others into supporting a rationalization based on technical arguments.
To repeat Doug Oade's best and most famous quote, "trust your ears".
Very well said ;)
+T
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thanks for the input, brad.
im not trying to justify or talk anyone out of their preamps, just trying to have a good tech discussion on why this may or may not be acceptable.
as for your example, i would say that a preamp is absolutely essential for recording to 16-bit recorders as you describe
the theory here is that the super low noise of the ad2k and the 24 bit theoretical noise floor allows you to skip the pre. while some may consider this 'skimping', if i can save the cost of a pre and the weight and power in the field and not hear a difference , thats enough for me
i just got off the phone with John Siau (designer of the AD2K, who graciously took time to support a legacy product). what he explained to me is that noise is cumulative and is slightly higher than the level of the loudest noise. something along the lines of youve got two levels of noise, one (i think he said 18db) lower than the other. the combined effect of this noise willl be 1 dB higher than the loudest one. so that said, the post-normalized noise floor of the ad2k is NOT totally masked by the noise of the mics. But realistically, to me, and increase of 1db of low level noise is essentially inaudible.
Hes looking at the schematics of that unit to see if we could mod the unit to more closely match the output of the cmr mics.
i think if i could get it to peak at -12 vs. -25-35 dB that would be a win-win
I started using an outboard preamp at a time when very few open concert tapers did so, 1990. What I found then, and still content today, is that an outboard preamp can harness very loud sources better than any recorder where the mics are plugged directly into it.
So my personal experiences with the benefits of preamps does not deal with very quite and/or extremely dynamic sources where noise levels are a big consideration, as the OP seems to be concerned with.
Most all past portable DAT recorders, and most of today's wave/DSD file recorders, seem to operate best when they are fed a clean line level signal that falls within the normal line level ranges (-20 consumer/+10 professional). So, by placing an outboard preamp in the path to deliver a line level within the optimal range the recorders are designed to handle to my ears usually sounds better.
All the Sound Devices users I know personally, while willing to use the unit with mics straight in when a stripped down rig is required, they all swear by outboard preamps. I have heard comps between V2/V3 versus no preamp, Sonosax versus no preamp, Neve, PSP3, VMS, MP-2, etc.etc. and rarely have I been unable to pick out the difference between one of those preamps versus using no preamp at all.
I will not attempt to discuss the benefits of one brand against the others, since part of the art of taping is matching and tuning a rig based on how different mics and preamps sync together. Some are matches made in heaven, and others are downright ugly.
Bottom line, the original poster's joy in not using a preamp for HIS particular setup is great for HIM. If HE likes it for HIS recordings, that that is all that should matter to HIM. I think it's a waste of time to attempt to draw others into supporting a rationalization based on technical arguments.
To repeat Doug Oade's best and most famous quote, "trust your ears".
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well...I guess if it ain't broke....., FUCK WITH IT MORE !
my new motto.
:)
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Thanks MS!
ok, that said, im currently using it as a 0dBFS =+14 dBV device, would i assume EIN = 117 dB at that setting. it appears the nature of the mod we want to do will decrease the attenuation, making it a 0dBFS=-0 dBV or 0dBFS=-10 dBV device. Perhaps this is why they say in the manual "Sensitivities greater than +14 dBu have not been included since they would compromise the
dynamic range of the AD2402-96."
so basically if we go from +14dBV to 0dBV, we are going to take the dynamic range from 117-14 = 103 dBA (or alternatively 93 dBA at 0 dBFS = -10dBV) .
as for dither im talking about in the course of the final product in 16 bit, i wouldnt dither a 24bit file just to do it. the dynamic range of a 24-bit waveform is self dithering on all real-world equipment
This is largely correct, except for a couple of points, one minor and one major.
First, the major point: a 117dB dynamic range on an A/D converter does NOT equate to equivalent input noise of -117dBA. That's the difference between 0dBFS on the converter and its noise floor. If the converter is say 0dBFS = +8dBV (a -10dBV nominal device, perhaps), then the noise floor is at -109dBA.
That means if you don't amplify the mic signal, then the converter noise will dominate. More accurately, as you note in your last post, the noise will sum and the result will be about 3dB of extra noise in this case.
It gets worse if you're using a +4dBu A/D, it might have 0dBFS at +16dBV (+18dBu), that means your converter noise now lives at -101dBA, so you need much more gain to offset that. If we use the figures in your post, that's +27dB (although I think it's more like +19dB; noise power sums, not noise voltage).
Further, I would suspect that could be a bit dangerous for people who don't have converters with 117dBA dynamic range, who see this conclusion and think "Hey I don't need a preamp either!" With a quiet mic on a quiet source, they could suffer from audible converter noise in that case. So the conclusion on whether or not to use a preamp is going to depend on the rest of the taper's gear.
But will any of that ever be noticed at a loud show? Probably not. That's because the acoustic noise level could be high enough that none of these electronic noise sources matter.
Now, the minor point is about dither and quantization distortion. First, you can't dither after conversion (or truncation), dither must be added to the signal before conversion. If you dither a signal that already has quantization distortion, then you just get noise + distortion. When quantization distortion exists, it is at a MUCH higher level than the nominal dynamic range of the converter system.
Having said that, all existing 24 bit converters are sufficiently noisy to self-dither any signal at any level (unless we are talking about a digitally-generated signal that does not have dither; perhaps a synth). So you need only worry about converter noise, not quantization distortion.
If you are talking about quantization distortion upon truncation to 16 bit, well then that couldn't exist if you dithered the normalized 24 bit signal first, but as you have already noted, the noise floor of the recording would be high enough to self-dither.
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well...I guess if it ain't broke....., FUCK WITH IT MORE !
my new motto.
:)
qft
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qft
ha! i finally looked up that acronym. for a lng time now,i thought it meant 'quit f$#^kin talking!'.
kinda changes the context of message board discussions!
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quoted for TRUTH!
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Wow, even my brain is gonna explode from reading all of this. Greetings to all I haven't seen in a long time.
Going to give this one a cursory glance, don't have time to respond to every detail of the very fine discussion here.... but....
First, I'd be inclined to believe that there will simply be inherent differences in coloration attributable purely to different impedances between the output of a preamp at line level vs. the pure mic level output.
Second, as someone already said here, (sorry, too lazy to quote, so I'm gonna paraphrase) using a preamp will in most cases improve signal to noise ratio at the A/D. The quieter the material, the worse the signal to noise ratio. Noise being not just self noise from the mic and analog stage in the A/D, but also quantization noise from the A/D process.
Third thought, the best way to prevent any further noise from being added to the signal and still amplify the signal is to use bit shifting (e.g. some software... I think samplitude has a 2x volume function). Any sample value multiplied by exactly 2 will not generate a mantissa, and therefore, will not create additional noise. Amplification through bit shifting would be the only way to preserve the effect of dither and noise shaping once applied. Of course, in using this method, how close you can get to 0dBFS would depend on how close your loudest peak is to -6.02dBFS.
Fourth, I usually consider it simply like this. You really only have about 25dB extra headroom with the AD2K over the best 16 bit A/D. So -48dBFS is only in theory the same as 16bit at 0dBFS in terms of S/N ratio. In practice, it's probably closer to -24dBFS being about the same. Granted, when comparing differences down to this level, you'd also have to consider the differences in color of *all* frequences over the dynamic range for any given analog stage. I believe most specs center their S/N ratios, etc... around a 1kHz tone. As such, every analog component's self noise will have it's own unique FFT. The degree to which the noise signature with a preamp or without a preamp is complementary to the live source is purely up to the listener. In other words, the differences in self noise will impart a coloration to low level sounds. It is possible that cumulative (i.e. greater than the loudest noise, as John Siau puts it) noise dithers the sound better to your ear one way over another for your recording, much in the same way that white noise dither would sound different than "pinker" noise.
Bottom line, practically, I'd say, if you were willing to carry a Benchmark AD2402-96 to a show, you should be willing to carry a Grace Lunatec V3 to the show instead, and kill two birds with one stone.
I still love that V2->AD2402-96 combo, and for anyone curious, for 2 track live recordings (as infrequently as I do them these days), I still use the DAARWIN-24 Sony picturebook with the VXPocket, Win2K, and that Pre/AD combo. The combo continues to do exactly what it was designed to do 8 years ago, no reason to mess with a good thing. A solid state disk sure would be nice though. ;-) For the multitrack Hot Tuna shows, well, that's a whole other ball o' wax. Gotta love the Alesis HD24-XR for that.
(no, I don't read this board regularly at all.... too busy coding stock trading formulas these days. ;-)
jerryfreak roped me in on this thread. Somewhere along the line I also created two ID's here as well. And so it goes...)
Cheers, and keep up the quality discussion. Mmmmmmmmmmmmmore bits please.
Dan Heend
High Fidelity Microsystems
HFM Studios
www.hottunatunes.com
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http://www.valvotronics.com/httpdocs/d19.htm for the big boyz
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thanks for chiming in dan!
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I also liked doug's old saying.. "what we need is a straight wire with gain!"
i dont really know much about the techno-babble, but doug oade once said, "the best pre-amp, is no pre-amp."
i was going to say this exactly (quoting Doug...who ironically makes a living making/moding preamps. maybe he's just gutting our all in one boxes to the "no preamp MOD" )
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Wow, even my brain is gonna explode from reading all of this.
Mine too. Thanks everyone for the great posts. I've learned a lot ( and that's why I'm here in he first place ;))
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Nothing to add, except thanks for the great discussion on a topic I often wonder about.. and still do.
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I'll 2nd that.
+Ts
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nothing brings out the pros like some good old-fashioned name calling!
at this point, im still waiting to hear back from john siau on what mods we can do to 'open up' the input of the ad2k to lower levels. were talking about bypassing a resistor in the input section, or alternatively, a switch to a second resistor that would change the attenutation range of the front-panel knob. With no trim available, it would still need to be set conservatively, but if we can get the peaks up in the -20 to -12 range that would be better i think
to recap:
-to my ears, i cant hear additional audible noise when doing this
-we agree that there is adequate digital resolution to allow this, and a single normalization in soundforge has no negative effects
-ampification of the noise floor of the AD2K is the limiting factor, and appears to be minor, if audible at all.
tell ya what i'll do - I'll record both a high and VERY low level test recording with and without a pre this weekend, and we'll see if you guys can blind AB them
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i dont really know much about the techno-babble, but doug oade once said, "the best pre-amp, is no pre-amp."
Hell, yeah. And while you're at it, get rid of those pesky mics and A/D distorters as well. I just bought one of these:
http://cgi.ebay.com/Windup-Premium-Gramophone-Kit-by-Gakken-for-SP-EP-LP_W0QQitemZ140248973205QQihZ004QQcategoryZ1442QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
That's real analog sound for brave hearts.
Jeff
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that would go great with gutbucket's avatar
i dont really know much about the techno-babble, but doug oade once said, "the best pre-amp, is no pre-amp."
Hell, yeah. And while you're at it, get rid of those pesky mics and A/D distorters as well. I just bought one of these:
http://cgi.ebay.com/Windup-Premium-Gramophone-Kit-by-Gakken-for-SP-EP-LP_W0QQitemZ140248973205QQihZ004QQcategoryZ1442QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
That's real analog sound for brave hearts.
Jeff
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just wondering about your thoughts on this.
My new rig doesnt have a pre, and it sounds great to my ears (MK4v>CMR>battery box>AD2K)
normalizing once in soundforge at about about +25 - +30 dB doesnt seem to hurt the sound at all.
While minimalist, this is actually my new 'open' rig. I'd love love love to find a super small, super low noise floor stealth AD to eschew the gain stage entirely while stealthing. Whose got time to check levels when you're watching your back? the only good small AD of fthe top of my head is the sbm-1, but its only 16bit, and wouldnt have the headroom to pull this off.
i was hoping to get some feedback on this setup from the more electrically inclined. as to any major disadvantages with regard to headroom, dynamics, etc.
it seems with an AD capable of a clean -117db, and a mic with a noise floor of -75 to -80, id never be limited.
check it out. AD2K was set to its highest input level (+14 dB, minimum attenuation):
http://www.sendspace.com/file/uzt783
One reason why there is a trim control on mixers is to adjust the gain so that you get the best signal to noise ratio. It's all about proper gain staging. Some preamps have a variable impedance. Matching the impedence allows for a better signal transfer between the mic and the destination circuit.
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Hell, yeah. And while you're at it, get rid of those pesky mics and A/D distorters as well. I just bought one of these:
http://cgi.ebay.com/Windup-Premium-Gramophone-Kit-by-Gakken-for-SP-EP-LP_W0QQitemZ140248973205QQihZ004QQcategoryZ1442QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
That's real analog sound for brave hearts.
Jeff
Oh yeah. No pesky electronics at all. I love how it records on CD's.. by scratching in an analog groove! Just wind up the clock spring and point the horn. You gonna thread the base for a lightstand mount?
that would go great with gutbucket's avatar
Yup. I run two of 'em in a giant kangol cap.
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Heres what John Siau had to say. Note his comments are for the CMC6, the CMR have a very slightly higher noise spec, so it seems with the mod hes talking about, the noise level of the AD would be comparable to that of the mic, and more importantly inaudible in all real world field recording environments
mk4+cmc6, sensitivity = 13 mV/Pa, max SPL = 132 dB, EIN = 24 dB CCIR/ 15 dB A weighted
mk4+cmr, sensitivity = 8 mV/Pa, max SPL = 130 dB, EIN = 29 dB CCIR/ 18 dB A weighted
John's comments:
Jamie,
Your analysis is basically on track but has some errors that are
throwing the final numbers off a bit.
I am including my microphone data spread sheet which already has the
MK4V+CMC6 combination. This shows a self noise of 14 dB SPL and given
the sensitivity of the microphone this means that the noise at the
microphone output is -115.5 dBu. The AD2402 has a signal to noise ratio
of 117 dB and a maximum sensitivity of 0 dBFS at +14 dBu. This means
that the equivalent input noise of the AD2402 is +14 dBu-117dB = 103
dBu. The self noise of your microphone (measured at its output) is
actually 12.3 dB lower than the equivalent input noise of the AD2402.
In a perfectly quiet room the AD2402 would add about 12 dB of noise to
your recording. Obviously the noise level in a live venue exceeds 14 dB
SPL even when nobody is in the room. If the noise in the room measured
14+12.3=26.3 dB SPL the AD2402 would add 3 dB of noise to your
recording. At a room noise level of 32.3 dB SPL the AD2402 will only
add 1 dB of noise. Your system is working because you are recording in
a noisy environment. In a quiet studio you would benefit from a
microphone preamplifier.
Here are some easy rules-of-thumb for adding noise sources:
1) Two equal amplitude noise sources added will degrade the SNR by 3 dB.
2) If a new noise source is added that is 6 dB lower than the existing
noise, the resulting noise will increase by 1 dB.
3) If the noise sources differ by more than 6 dB, the quieter source can
be ignored.
4) Use RMS calculations for summing noise sources when you need exact
numbers.
The above assume random noise and no correlation between the two noise
sources. This is usually the case with microphones and electronics, but
be careful about room noise because it is often not random when the room
is empty. On the other hand, the random rustle of a crowd can approach
white noise under certain conditions.
Another interesting and useful rule of thumb:
The human ear can detect a tone that is 30 dB lower than the ambient
noise level. I frequently demonstrate this is the lab. What this means
is that a TPDF dithered 16-bit signal with an SNR of 93 dB can audibly
reproduce a signal that is recorded at -123 dB FS.
FFT analyzers can easily resolve signals that are 50 dB or more below
the noise floor. Our ears have about the same resolving ability as a 4k
to 8k point FFT - this is amazing. In the lab we have the luxury of
using 64k FFT analyzers so that we can analyze distortion signals that
are too small to be heard. Every time the number of points is doubled,
the resolution of an FFT analysis increases by 3 dB (sound familiar? -
see rule of thumb #1 above).
Here is a modification that will boost the input sensitivity of the
AD2402 by 8.2 dB (making the self noise of the AD2402 only 4.1 dB higher
than the microphone self noise - please note that the signal to noise
ratio of the AD2402 will still be 117 dB). With this modification the
AD2402 will only add about 1 dB of noise to your microphones in the
quietest studio.
Modification to increase sensitivity of AD2402 by 8.2 dB:
Each of the input attenuators has a 5.62K Ohm and a 649 Ohm resistor.
There are 4 of each. If you swap the position of the 649 and 5.62k
resistors, the +14 dBu switch position will become a +5.7 dBu switch
position (meaning that 0dBFS = 5.7 dBu). All other switch positions
will remain unchanged.<snip>
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as a point of reference: Dr. Booty (LOL) says a library is 30 dB
http://home.new.rr.com/trumpetb/audio/dBexamp.html (http://home.new.rr.com/trumpetb/audio/dBexamp.html)
I find his analysis excellent. There's your answer right there, more succinctly than I have said it.
sounds like a go.
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Got a chance to listen tonight. I must say the recording quality sounds really good. :)
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thanks. that was in a corner of a small bar room, far from optimum conditions. just goes to show how forgiving the schoeps are.
Got a chance to listen tonight. I must say the recording quality sounds really good. :)
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Well I can't take it quite as technical as some of the posters on this thread, but my rule of thumb has always been as close to 0 db without going over.
I am always looking to attain the best result with the least post recording processing. Before I went to 722 and finally 744T, I ran 140's/50's > v-2 > ad2k+ > HHB PDR1000.
It took some experimenting to finally get the AD2K+ dialed in. In general, the best performance out of the ad2k comes when you run it on the warm side. You want to see plenty of activity in your level meters peaking at yellow and keeping away from red. The yellow light means something to the effect of full digital signal without saturation ( which would be red)
I apologize for the lack of technical speak, but I love the AD2K...I'd run it in front of my 744T in a second , but I've been trying to scale down the sze of my already over full gear-slut bag >:D
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historically ive always run my ad2k righ tup to -1 as well, but now that i compare it, i cant hear a difference when peaking at much lower levels.
if youre trying not to do any post at all, of course higher levels will always sound better
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When I moved to 24 bit when I went to the 7 series recorder, I kinda had to tighten up my post-recording game out of necessity. I have been playing around with my buddy's masters of Davis Gilmour he stealthed at Radio City. He was running full body mk4's >744 from the lower level balcony (DFC, by the way) and ran the levels low and sfae, to prevent messing with it and getting busted.
I used Soundforge 7.0, and pumped the levels in the volume control to like -2 @ 24 bit then dithered to 16 and these disc sound amazing...zero hiss, nice high's and lows.
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i wonder what mike grace would think of this thread?
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i would never call grace a pussy, thats for sure!
im sure he would agree with john siau's excellent technical assessment, that in the case of concert taping with an AD wit a very low noise floor, it really does not have any drawbacks.
just wait till you guys see the rig i have in the works ;). having a hard time keeping it to myself
i wonder what mike grace would think of this thread?
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i would never call grace a pussy, thats for sure!
im sure he would agree with john siau's excellent technical assessment, that in the case of concert taping with an AD wit a very low noise floor, it really does not have any drawbacks.
just wait till you guys see the rig i have in the works ;). having a hard time keeping it to myself
i wonder what mike grace would think of this thread?
Um speaking of PREAMPS :) Your parts came in today YOUR new preamp will be shipped out next week ;)
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Oh, burn ;D
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that preamp is not part of the new rig.
thanks for the update, chris!
i would never call grace a pussy, thats for sure!
im sure he would agree with john siau's excellent technical assessment, that in the case of concert taping with an AD wit a very low noise floor, it really does not have any drawbacks.
just wait till you guys see the rig i have in the works ;). having a hard time keeping it to myself
i wonder what mike grace would think of this thread?
Um speaking of PREAMPS :) Your parts came in today YOUR new preamp will be shipped out next week ;)
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that preamp is not part of the new rig.
thanks for the update, chris!
i would never call grace a pussy, thats for sure!
im sure he would agree with john siau's excellent technical assessment, that in the case of concert taping with an AD wit a very low noise floor, it really does not have any drawbacks.
just wait till you guys see the rig i have in the works ;). having a hard time keeping it to myself
i wonder what mike grace would think of this thread?
Um speaking of PREAMPS :) Your parts came in today YOUR new preamp will be shipped out next week ;)
ZING ;D
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that preamp is not part of the new rig.
thanks for the update, chris!
i would never call grace a pussy, thats for sure!
im sure he would agree with john siau's excellent technical assessment, that in the case of concert taping with an AD wit a very low noise floor, it really does not have any drawbacks.
just wait till you guys see the rig i have in the works ;). having a hard time keeping it to myself
i wonder what mike grace would think of this thread?
Um speaking of PREAMPS :) Your parts came in today YOUR new preamp will be shipped out next week ;)
Not a problem. Thank you for your purchase. I hope you enjoy your new preamp. I am sure your mics will work very well with it.
Chris
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looking forward to it
4V>cmr>ca-ugly>mod-sbm
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I think this all boils down to this: How do we store sound waves in state as accurately as possible to the original sound waves and be able to play these back in the original form? Which is impossible at present as it is not possible to record all possible entities (sound w in the original environment at the exact point in time and space and in even close representation (a constant stream of differences) - maybe quantum physics will bridge this gap :)
So it is all up to best guesses and reproductions. MP3 bad; Analog can offset some; tubes/transformers can offset some; pure interpretation impossible at present.
What I am saying is reducing the amount of processing/coloring/interpretation in the middle is great (i.e. no preamp) as long as you know what your reducing. But maybe in some cases pre-amp coloration offsets the lack of our ability to accurately reproduce the original sound image into a more comfortable representation of the original.
Long night - ok I need to go to sleep now :)
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Are preamps for pu$$ies?
If so get my maxi pad ready. I'm about to start. ;)
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Are preamps for pu$$ies?
If so get my maxi pad ready. I'm about to start. ;)
:lol: :lol:
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Sorry for jumping into this thread late, but I was discussing this very subject with my brother just a few weeks ago.
Since reading this thread I did some casual empirical testing. When plugging a dynamic mic directly into the (balanced) line input of a sound card and normalizing the level, the background noise/hiss is pretty gnarly. Running the mic through a preamp at unity gain is just as bad. Adding just a small amount of gain from a mic preamp, say 20 - 30 dB, makes the recording much quieter. Anything above 30 dB of preamp gain brings up the noise floor in the analog domain (which is to be expected) and doesn't buy you anything you can't get through normalizing. I don't understand the theory behind the 20 - 30 dB sweet spot. Maybe it is the difference between the dynamic range at 24 bits (144 dB) and the noise floor of my sound card (144 - 30 = 114 dB). My sound card is an EMU 1212m. After some preliminary testing I made some recordings with a Sennheiser MD421 at a distance of 12". With only 20 - 30 dB of preamp gain the levels were really low before normalizing, but after normalizing the noise was no worse than if I had brought up the gain of the preamp.
One advantage of this approach is that you have tons of headroom. Clearly it works best at 24 bits.
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The published specs for the sound card give +6 dBV as the maximum input and 120 dB A-weighted for the SNR. So that puts the sound card's noise floor at -114 dBV?
I don't see how you arrive at 26 dB. If the preamp's input noise is -120 dBV, 26 dB of gain brings it up to -94 dBV. Adding 6 dB of gain would bring the preamp's noise up to -114, the sound card's noise floor.
The 114 dB I originally gave as the dynamic range of my sound card was but a lucky guess.
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looking at your card's specs for balanced input it states the maximum level is +20dBu (+18dBV), so if you had the card set to balanced, that should put the noise floor at -102dBA. I don't know if it's possible to set the card for balanced and -10dBV nominal
+4 and -10 are the only options. My preamp has an unbalanced output.
Let's say the preamp's noise floor is -120 dBV and the sound card's noise floor is -114 dBV. That gives us a 6 dB difference, plus the 6 dB you add for summing, for 12 dB of amplification.
I honestly don't know the S/N of my preamp, but it is based on this chip:
http://www.national.com/ds/LM/LME49720.pdf
Scroll way down to the balanced input mic amp schematic. I am using just the two opamps on the left. It just occurred to me that I could give it a balanced output if I didn't ground the output of the bottom opamp.
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by my experience, internal computer soundcards are rarely capable of exceeding 90-100dB signal/noise due to the large amount of rf noise typically found in a computer case, i'm guessing you have nowhere near the dynamic range they quote with that setup. a quick peek at the emu specs for that card show that they do quote 120 dB dynamic range, but i find it hard to beleive they actually got that unless on a test bench and not inside a computer chassis.
even the ad2k, which is designed all around a philosophy of ultra-low noise and distortion, still has a slightly higher noise floor than my mics. Keep in mind that unit is solely a DA and costs over 10x what the emu card does, youre asking a lot of that emu card. i dont think preamps can be eliminated in ALL cases.
looking at your card's specs for balanced input it states the maximum level is +20dBu (+18dBV), so if you had the card set to balanced, that should put the noise floor at -102dBA. I don't know if it's possible to set the card for balanced and -10dBV nominal
+4 and -10 are the only options. My preamp has an unbalanced output.
Let's say the preamp's noise floor is -120 dBV and the sound card's noise floor is -114 dBV. That gives us a 6 dB difference, plus the 6 dB you add for summing, for 12 dB of amplification.
I honestly don't know the S/N of my preamp, but it is based on this chip:
http://www.national.com/ds/LM/LME49720.pdf
Scroll way down to the balanced input mic amp schematic. I am using just the two opamps on the left. It just occurred to me that I could give it a balanced output if I didn't ground the output of the bottom opamp.
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I am basing my gain calculations on human speech, which is typically 94 dB SPL at a distance of one inch (63 dB SPL at three feet). Conveniently this is equal to one Pascal. The output of a typical dynamic mic is -54 dBV at one Pascal. At two inches human speech is -60 dBV, at four inches it's -66 dBV, at eight inches -72 dBV. If my mic preamp is set to 30 dB of gain, it is delivering -42 dBV to the sound card, not -10 or +8 dBV. That's why I said you have gobs of headroom.
if you are just running the two amps on the left, you don't have a balanced circuit at all
Balanced input circuit or output circuit? With just the two on the left I'm giving the mic a balanced input, aren't I? Would the addition of the third opamp justify the attendant noise and distortion of a second stage? I'm OK with running an unbalanced output if it will result in lower noise, THD, etc.
But even at -126dBA, that's a good reason to use more preamp gain.
I'm doing listening tests as I build this out. Using only the two opamps on the left, I've got a 1.5K resistor in place of the 200 ohm resistor at R1. I haven't done an actual measurement yet but I suspect the gain is somewhat above 20 dB.
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I would take a balanced input over the tiny difference in distortion any day of the week.
A balanced mic input, of course, or you'll pick up the nearest AM station. Did you mean balanced output? BTW this circuit replaces one with a THAT 1510 chip but has a nicer square wave response.
The usual approach is to use a pot (with a resistor in series to limit maximum gain) or rotary switch in place of R1, so you have gain control.
I can do that, but I'm deciding whether I want a small amount of fixed gain, say 20 dB, or a variable amount, say 20 - 40 dB, and normalize. The gist of this thread is to use a small amount of preamp gain, giving a large amount of headroom, and bring up the levels digitally. IOW what is the smallest amount of preamp gain I can get away with and still get good results?
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How would this work: use a LME49740 which has four opamps on one chip, giving me two amps on the "right" side instead of one, and make a balanced output out of them? Then make R1 a pot variable from 20 to 40 dB gain? Or unity to 40 dB? If more gain is needed, normalize.
This has been very instructive, thanks.
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use an impedance balanced output (30 ohm or so resistor from pin 3 to ground)
You're saying make R7 33 ohms.
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I know it's an incomplete schematic. You shouldn't make inferences about the THAT 1510 preamp from the LME49720 data sheet schematic you are looking at. The 1510 is all built and has phantom power protection, RF bypass caps, an output resistor, a split power supply (two 9-volt batteries) and an unbalanced output by design. See figure 4 of the THAT 1510 data sheet for an idea. It does not provide phantom power by design -- that has to be provided by an external supply. It works great and has served me well for a couple of years now. The new preamp will have all of those things added to it which you don't see in the incomplete schematic on the data sheet. Last night I built the output stage adding two more opamps in a balanced configuration. It is basically a carbon copy of the left half except there is a 10K resistor where R1 is. So far it tests great, but needs to have the caps and diodes added to it.
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ms, is there a tiny circuit than can take an unbalanced signal from the cmr and simply convert it to balanced?
It is basically a carbon copy of the left half except there is a 10K resistor where R1 is. So far it tests great, but needs to have the caps and diodes added to it.
That will still depend on a differential amp in your converter. In other words, if you plug that into an unbalanced input, the entire signal chain will be unbalanced.
Copying the left buffer amps again will just add gain (6dB with 10K, I believe) and slightly degrade CMRR.
What you probably want to do is build the schemo as shown in the datasheet, but feed its output to pin 2 out AND the negative input of an opamp in inverting configuration, with that opamp's output to pin 3.
Good basic opamp page:
http://www.bcae1.com/opamp.htm
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CMR? You lost me there ???
i was referring to the unbalanced schoeps cmr amplifier body, which is where this thread all began
http://www.schoeps.de/E-2004/cmr.html
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similar schems here kindly drawn up by mshilarious...
http://taperssection.com/index.php/topic,102355.0.html (http://taperssection.com/index.php/topic,102355.0.html)
i gathered all the parts to make it, but then I got a good deal on some PM-4's, so it never happened. This version doesn't use transformers though, so I still might cobble it together at some point as I have a hunch it would be more transparent.
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yeah, that was me, but i dont need to use phantom, just wanted to know if there was an adapter circuit that could be built and used close to the mics to allow for longer cable runs
Oh yeah! There was another thread on the mic board where I talked about that. In fact I think I built such a device for somebody, but I forgot who it was, and I don't recall if they reported back on how it worked.
I think I posted this somewhere too:
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I think normalising is much more pu$$y.
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you mean 3 conductor cable, right?
so, bear with me, this is explaining the cabling running backwards counter to teh aactual signal flow:
from device with balanced input (ad2k/v3/etc), run normal 3 conductors up to the mic, then at the mic, terminate the (-) balanced leg from the cable to ground thru a resistor matching the output impedance of the mic.
thus the mic itself is not balanced, but the cable leading to the mic will be balanced, and reject most all of the inducted noise. again, i will see some rejection, but probably not the full cmrr due to matching.
that said,the mic is spec'd at 15 ohms at 1 khz. im sure its an approximation of sorts should i get a batch of 15 ohm resistors and try the various values with a long mic cable and do some sort of simple test to see how much noise results?
yeah, that was me, but i dont need to use phantom, just wanted to know if there was an adapter circuit that could be built and used close to the mics to allow for longer cable runs
Yes, just use 2-conductor cable, and tie the unused lead across a 30 ohm resistor (or whatever the output impedance of the mic is) to the cable shield on the mic side of the cable.
You will need a differential (balanced input) amp at the other side of the cable to take advantage of that though.
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the 100 ohm pot sound slike the way to do it.
should i use like 100+feet of mic cable and how should i 'look' at the noise, i dont have a scope
you mean 3 conductor cable, right?
2 conductor plus shield, your standard basic mic cable.
so, bear with me, this is explaining the cabling running backwards counter to teh aactual signal flow:
from device with balanced input (ad2k/v3/etc), run normal 3 conductors up to the mic, then at the mic, terminate the (-) balanced leg from the cable to ground thru a resistor matching the output impedance of the mic.
Yes.
that said,the mic is spec'd at 15 ohms at 1 khz. im sure its an approximation of sorts should i get a batch of 15 ohm resistors and try the various values with a long mic cable and do some sort of simple test to see how much noise results?
Put a 100 ohm pot in the circuit and turn it until the noise is minimal. Then disconnect it and measure its resistance.
The other way to do it is to measure the actual output impedance. There are a few ways to do that. I have a variable input impedance preamp, so I just measure the signal difference at two different input impedance settings, and calculate the impedance from there.
Or just use 15 ohms . . . I always buy 1% tolerance resistors, so I don't think trying out a bunch of different 15 ohm resistors will help too much. It's more of a issue of whether you buy 18 ohm or 15 ohm or whatever.
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- an impedance-balanced line. Figure out the output impedance of your device, and match it with a resistor from pin 3 to ground.
What becomes of R7? Pin 3 is the non-inverting input of the final opamp. You've already got a 10K resistor (R7) between it and ground.
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match it with a resistor from pin 3 to ground.
Pin 3 is the non-inverting input of the final opamp.
The impedance balancing resistor does not connect to either input of the opamp.
Do you mean pin 3 of an XLR connector or pin 3 of the opamp chip?
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Arriving late in this thread ... seems to me that there are a few things some people might want to think about.
One is that you need to make a fair comparison. Yes, theoretically every piece of equipment contributes some noise and distortion. But when you're asking whether or not to use a preamp, the preamp isn't the only source of noise and distortion to be considered! The recorder's input circuit--which you've been using in the absence of a preamp--also has noise and distortion. If you leave that out of the comparison, you'll be off on a completely wrong tangent.
Practically speaking, the mike inputs on most mass-produced portable recorders are mediocre because there is no economic reason for them to be better; people buy them anyway. A 30-cent increase in parts cost either removes a dollar from the profit picture of the manufacturer, distributor and/or retailer, or forces them to raise the selling price a dollar, making them vulnerable to price competition from other manufacturers' products. And the present-day audio equipment market is driven much more by price than ever before.
Equipment built to professional standards of quality, reliability and adaptability is usually quite a bit more expensive. It seems as if we all hope for a miracle each time we buy a (relatively) bargain-priced piece of mass-produced equipment. There may be something like a gambler's psychology at work. The moment a low-cost alternative can really replace a high-cost item of equipment, you see the professionals adopting the low-cost gear.
I'm not saying "more expensive is better;" I'm only saying that in some things, "better is more expensive." And since most people who buy recording equipment don't do any live recording at all, or do so only in their own homes and/or with consumer-type microphones, the biggest place where the mass-produced recorders almost never meet professional standards is their mike inputs. We can get more specific, but I just wanted to lay down a carpet of what I consider the reality to be.
--best regards
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dsatz - on point as always
one flip side to your discussion of 'pressing the limit of input noise' is the concept of my original post
schoeps>ad2k (pro mic>pro converter) reaches its limits in a quiet studio, as the input noise of the converter is *slightly* greater than that of the mic. Is very but This setup however is perfectly applicable to our environment as the noise floor of both the mics and converter are far below even the quietest background noise of a live venue. so in this case, it works!
oddly enough, i stubmled across this setup by accident, while doing a test of a bunch of schoeps capsule/preamp setups micing monitors in my home, i found that the cmr>ad2k was pretty much as good as it gets, so it became my field rig. (for now)
I've got some new things up my sleeve that are along the same lines as this setup, technically addressing the fundamental 'limitations' if you call them that, the goals are two fold:
-ideally reduce the input level of the converter to 6 dB or more below the mic noise
-increase the input sensitivity of the converter to get a signal thats in the 0 to -20 dB range as opposed to -24 to -36 range, to reduce or eliminate normalization. but it all goes back to #1, converter noise vs mic noise. if you can get the converter noise 6dB below the mic noise, it is really irrelevant what level you record at, as you have essentially allowed for the whole range of mic input levels to be accurately recorded.
a trick is to have a device that can allow for low level inputs but maintain enough headroom to take a signal off a pre when you want to use it.
theres a couple other features my 'ideal box' would have, its really in its infancy, it is a modded piece of gear, but not an ad2k per se.
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how not cheap are we talking? also what would be the rough size and power requirements of a simple assembled fixed or trimpot-variable gain? are wer talking larger battery box or ad20/mic2496 size?
I've got some new things up my sleeve that are along the same lines as this setup, technically addressing the fundamental 'limitations' if you call them that, the goals are two fold:
-ideally reduce the input level of the converter to 6 dB or more below the mic noise
-increase the input sensitivity of the converter to get a signal thats in the 0 to -20 dB range as opposed to -24 to -36 range, to reduce or eliminate normalization. but it all goes back to #1, converter noise vs mic noise. if you can get the converter noise 6dB below the mic noise, it is really irrelevant what level you record at, as you have essentially allowed for the whole range of mic input levels to be accurately recorded.
An easy solution is a good transformer. A converter usually has a 10K input; the CMR is 15 ohm output, right? So something like the Jensen JT-13K7A does +14dB of almost noiseless gain. It's not cheap though!
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An easy solution is a good transformer. A converter usually has a 10K input; the CMR is 15 ohm output, right? So something like the Jensen JT-13K7A does +14dB of almost noiseless gain. It's not cheap though!
Transformers are often a useful addition to the front end of an amplifier, but you have to be careful about getting too much gain out of your transformer. It really depends on where the noise in your signal path is coming from. Your mic can be modeled as a signal voltage source in series with a noise voltage source and a resistor whose value is the output resistance of your mic. The noise voltage source is to model the acoustic noise due to random motion of air molecules that impinge on the face of the mic's diaphragm plus the thermal noise that is produced by the output resistance of the mic (equal to the sqrt of 4kTBR, where k is Boltzmann's constant, T is the absolute temperature in degrees Kelvin, B is the Bandwidth of the signal path and R is the output resistance of the mic). We'll call this the microphone noise.
You also have a couple of noise sources right at the input to your amplifier. One can be modeled as a series noise voltage source and the other as a shunt current noise source. The resulting noise at the output of the amplifier is given by:
Vnampout = (Inamp * Rmic * N^2 + Vnmic * N + Vnamp) * A
where:
Vnampout is the noise at the output of the amp
Inamp is the current noise of the noise current source at the input of the amp
Vnmic is the noise voltage of the mic
Rmic is the output resistance of the mic
N is the turns ratio of the transformer
Vnamp is the noise voltage of the voltage noise source at the input to the amp
A is the gain of the amp
The signal voltage at the output of the amp will be
Vampout = Vmic * N * A
So, the S/N is given by:
S/N = Vampout/Vnampout = Vamp/(Inamp*Rmic*N + Vnmic + Vnamp/N)
Typically, the largest contribution to the noise at the output of the amp is Vnmic, but notice how Inamp*Rmic is multiplied by N. You have to be careful not to use too large of a turns ratio, N, or your input current noise of the first amplifier stage after the transformer will begin to dominate the noise budget.
So, if you have mics with lots of self noise, then you can use lots of "free" gain from a transformer with a high turns ratio, but if your mics are pretty low noise, then you have to be careful not to use too much transformer gain or the input noise current of your amp will start to become the dominant noise source. The goal of any preamp should be to get the signal level boosted to a useful voltage level without significantly degrading the S/N ratio.
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> [ ... ] I thought the idea was to operate at a fixed level where the mic noise slightly exceeds the ADC input noise?
OK, I think it's time to call "shenanigans." Why bother to point out again and again that noise can't be adequately described by a single numeric value? People nod their heads sagely but then they say, "Well, I have to use SOME number,"--so even though the basis of the numbers may 10 to 12 dB apart, they just turn right around and act as if it's OK to compare them and to make engineering decisions on that basis. Sorry, it doesn't fly.
The noise spectrum of a typical condenser microphone is quite far from flat. At low frequencies it's dominated by 1/f noise from the capsule--the source impedance driving the FET or tube input stage of the amplifier. The spectrum of this noise has an inherent downward slope. At some higher frequency the noise of the amplifier's input stage gradually takes over--generally with a flatter noise spectrum, though this varies.
See the attached graph, for example, in which a good professional-quality preamp was driven by a very high-quality transformerless, low-impedance, balanced condenser microphone. A measurement test head was used in place of the capsule; it offers the same source impedance as a capsule of the microphone's own type would have (ca. 35 pF), but doesn't pick up sound from the room. The gain of the preamp was set at a typical level for the kind of recording I mostly do. Phantom powering was turned on, of course.
The rise at low frequencies is a realistic representation of what's generally going on beneath most of my recordings. The noise at the lowest frequencies is ~25 dB (!) above the noise at high frequencies where the ear is most sensitive.
To quantify that noise, you have to take into account the ear's drastically different sensitivity to different frequencies at different sound pressure levels--and that throws the whole thing into a cocked hat, because that relationship depends greatly on the playback levels. There's just no way to condense all that into a single, meaningful numeric value--and we haven't even touched on whether the noise maintains a steady amplitude over time or whether it's full of moment-to-moment variations and impulses ("shot noise").
So yeah, we'd all like the equivalent input noise of our preamps to be well below that of the microphone. But at what level does that occur if the input noise of the preamp/converter has a flat spectrum while the microphone's noise spectrum is tilting downward? Frankly, it's not very hard to build a preamp that will be quieter than even the best condenser microphones at low or mid frequencies. Measure around 4 kHz, though (where the ear is most sensitive at low sound pressure levels), and it's a rather different story.
Let's make the discussion as simple as possible, but please--no simpler ...
--best regards
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mshilarious, apparently you feel personally affronted by what I wrote. I had no such intention at all; I was actually trying to be adorable by quoting something from "South Park," which no one expects somebody my age to know. But since I quoted part of your message, I can perfectly well see why you got that impression, and I apologize for that.
All I'm trying to say is that a few more specifics need to be taken into account before you can know whether or not a given input arrangement will have lower input noise than your microphones (a) at one particular frequency or (b) across the audible band. In practice it is very hard to predict this sort of thing from specifications and theory; you usually need a real, working circuit as a starting point.
--best regards
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> [ ... ] I thought the idea was to operate at a fixed level where the mic noise slightly exceeds the ADC input noise?
OK, I think it's time to call "shenanigans." Why bother to point out again and again that noise can't be adequately described by a single numeric value?
That's a very unfair statement to make when you've arrived on a thread two months after I began trying to explain this as best I am able.
For example, on July 16 I wrote:
S/N ratios are either stated as unweighted or A-weighted. A-weighting is quite a bit broader than the 1kHz tone; more like 500Hz to 10kHz, with the peak at 2.5kHz.
Microphone self-noise is a combination of capsule noise (1/f) and white noise, usually with capsule noise dominating the A-weighted range. Once you get into preamps and converters, the noise spectrum is much more uniformly white in the A-weighted range.
And just a few days ago:
If you are constructing an amp for a single application, you could safely select a fixed gain. Otherwise, I would add at least a couple of switching options.
jerryfreak is looking at interfacing two very specific pieces of gear, and with that in mind, yes, a gain figure can be derived. And that was also described by the designer of one of the bits back on July 16.
Jon dont take it personal Dsatz really does know his stuff. I can personally attest to that :) He has corrected me on more then on occasion. I for one welcome his expertise to the community!
Chris
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What you probably want to do is build the schemo as shown in the datasheet, but feed its output to pin 2 out AND the negative input of an opamp in inverting configuration, with that opamp's output to pin 3.
mshilarious -
I built the circuit as you suggested above and it works. Thank you for all your help. One last question: do I need anything between the output of the third opamp (on the right) and the new, fourth inverting output opamp, such as a resistor or capacitor?
Here is a link to the data sheet for the four-banger version of this chip:
http://www.national.com/ds/LM/LME49740.pdf
Thanks in advance.
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Actually I added three resistors: one between the output of the third opamp and the inverting input of the fourth opamp, a feedback resistor between the output of the fourth opamp and its inverting input, and one from the non-inverting input of the fourth opamp to ground, all 10K.
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just wanted to unearth this topic now that i have more expereince with this rig under my belt.
and yes im quite happy with the results
check out red rocks and gorge if youre curious
http://bt.etree.org/?searches=jerryfreak&cat=0