Old 07-28-2020, 02:58 PM   #1
Allybye
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Default get more gain on tracks?

I thought I read some comments about this in a thread but I forget where and what was writ!

Advice please.
Trying to investigate noise on various mics. My preamps can only give +60dB gain so could do with higher signal levels and am trying to add gain to tracks within reaper. Recorded at 24 bit so it should be ok from that perspective. If it can be done that would be useful to avoid another stage of gain prior to AtoD (which I do not currently have to hand).

I see that the display can be "enhanced" by 30 odd dBs but actual amplitude increase is only 12 dB....30+ calibrated dB would be great!

Is that understanding of what can be done in Reaper correct and is there a way to increase signal level once recorded..or am I going to add hardware in the signal paths?
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Old 07-28-2020, 03:12 PM   #2
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Originally Posted by Allybye View Post
I see that the display can be "enhanced" by 30 odd dBs but actual amplitude increase is only 12 dB....30+ calibrated dB would be great!
Too much science. I don't get it.

You do know that you can normalize items to 0dB (or any lesser value you like)?

Right-click on item > Item processing > normalize items

That enough amplitude for ya? Or am I missing the point here?
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Old 07-28-2020, 11:08 PM   #3
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With 24 Bit A/D converters, usually the input level should not show more that some -12 dB max before any processing in Reaper.
After recording you can do "normalize" or increase the volume to have a nicer looking display. (This does not alter the audio, but of course pushes more gain to the plugins)

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Old 07-29-2020, 01:19 PM   #4
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Now it's me confused by the science!! or rather what your suggestion will do!
Apologies this post is a bit long!

Let me explain and question a bit more. It might be I am trying to use Reaper/FX plug ins to do measurements that it is not capable of!....but I hope not. i.e. Compare and analyse noise under real world conditions rather than by specifications! In the professional world it used to be done by a measurement rig with over 90dB gain. I only have a good preamp with 60dB gain, interface and software (currently Reaper!)

So,

Generally recording low level signals.
Several mics involved so am using a low level noise (ticking clock) to set channel gains to try to get equal (but still low level' outputs for each mic channel involved. (Irrelevant to the question but for completeness - Max preamp gain being 60 dB - applied for example to dynamic mics is really insufficient but I have no more gain in hardware to hand).

This results in low signals levels in Reaper. Applying up to 12dB track gain on each track (nominally) and +30dB or so on the display gives sufficient display size to try to make each mic track have the same signal level on it's Reaper track i.e. by adjusting individual track faders -backed off from +12dB gain for the higher output mics.

So far so good.

Now with all the above settings just recording again without the clock i.e. just ambient, mic and electronics noise still leave a very low 'noise signals' on each track and that low level is rather small to do any analysis of level and spectrum. Hence the wish to get more real gain on tracks rather than just bigger on display!

So thanks for your responses re normalising! That might be the answer but I am not sure, so the questions:

How does this work in Reaper?
Does it provide actual track gain or does it just do a similar job to increasing displayed track size?

If the former, does it work on peak levels (the clock ticks in this case or what?

and if those peaks, can that normalized gain then be identically applied from the clock ticking track to the same track on the recording without the clock?


To sum up if I were doing this in hardware:
if I needed 30 extra dB gain on one mic and 40 on another I could add in another stage of gain. That gain would be applied irrespective of the signals being recorded and measured. Turning up the monitoring level of the speaker would not affect the track level. Analogy in software: equivalent to reaper track gain and display size.


Phew! Does that help or confuse?

Mschnell the track peaks from the low level ticking clock get nowhere near -12dBFS, more like -45dBFS.
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Old 07-29-2020, 01:46 PM   #5
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What kind of data are you trying to analyze and what led you to experimenting with audio hardware and software? It sounds like you are searching for something outside the realm of audio.

Quote:
Originally Posted by Allybye View Post
How does this work in Reaper?
Does it provide actual track gain or does it just do a similar job to increasing displayed track size?
Reaper or any other DAW simply captures the data stream from the AD converters. There isn't any direct connection to the analog to digital converter circuits to enable gain-ranging, for example. Any manipulation of the (assumed) audio sample data by Reaper - after the fact or while recording - is manipulation of the data after the AD capture. Signal integrity and noise are directly attributed to the AD hardware in the audio interface.
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Old 07-29-2020, 01:58 PM   #6
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Originally Posted by Allybye View Post
So thanks for your responses re normalising! That might be the answer but I am not sure, so the questions:

How does this work in Reaper?
Does it provide actual track gain or does it just do a similar job to increasing displayed track size?
It is not the track itself, but the item(s) on the track that are normalized. This is not just a cosmetic change - the waveform on the item gets bigger because the item gets louder. The track fader stays where it is, but of course the level output increases.
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Originally Posted by Allybye View Post
If the former, does it work on peak levels (the clock ticks in this case or what?
Yes - the maximum peak is raised to 0dB. The signal is not compressed or otherwise changed in any way; the entire item is subject to the same gain.

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Originally Posted by Allybye View Post
and if those peaks, can that normalized gain then be identically applied from the clock ticking track to the same track on the recording without the clock?
Yes. In this instance, you would select both items and select....

Normalize items (common gain)

.... your clock tick peaks will now reach 0dB, but the peaks in the other recording will not - the same gain is applied to both items.

Take care in this instance not to select

Normalize items

as this would raise the peaks on both recordings to a maximum 0dB. (And the one without the ticking will sound NASTY.)

The above is a little over-simplified - it is possible to normalize to different values (with a little tweaking). Reaper's editing tends to be non-destructive where possible, and I think that applies to normalizing.... but all this is way beyond me. I can't even science.

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Phew! Does that help or confuse?
Two meny werds.

Last edited by Fex; 07-29-2020 at 02:14 PM.
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Old 07-30-2020, 02:10 AM   #7
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Thanks Serr, I appreciate what you are telling me.
What I am trying to do is certainly within the realms of audio. It is the noise and frequency performance of the front end equipment, and the differences between them that I am trying to measure and demonstrate. I am really just short of the proper hardware to do it! Not trying to add gain to the pres or AD circuits but rather compensate as far as possible for the lack of it in Reaper (or any other software for that matter) but not something extra that will cost me dosh!!

Fex thanks for those succinct and specific answers. Will have a play with becoming 'normal'!!
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Old 07-30-2020, 03:39 AM   #8
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Quote:
Originally Posted by Allybye View Post
I thought I read some comments about this in a thread but I forget where and what was writ!

Advice please.
Trying to investigate noise on various mics. My preamps can only give +60dB gain so could do with higher signal levels and am trying to add gain to tracks within reaper. Recorded at 24 bit so it should be ok from that perspective. If it can be done that would be useful to avoid another stage of gain prior to AtoD (which I do not currently have to hand).

I see that the display can be "enhanced" by 30 odd dBs but actual amplitude increase is only 12 dB....30+ calibrated dB would be great!

Is that understanding of what can be done in Reaper correct and is there a way to increase signal level once recorded..or am I going to add hardware in the signal paths?
I have digimax fs interfaces and they give >55db gain and that is plenty even for my shure sm7b. What are you short on gain on?
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Old 07-30-2020, 07:37 AM   #9
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Hi Coachz,
For most recording not usually short of gain but what I am recording is very low acoustic pressures such as the ticking clock at about 5m distance and without that clock the very low acoustic noise in the room.
The dynamic mics such as the Røde M1 give very little output with that acoustic input. The higher output condensers not really much of a problem.
The clock is really only being used as a low level reference!

To put a bit more context on this I am now more aware of a low frequency rumble in the room that I can, now I am aware of it, just hear by old fashioned ears!
It is exceedingly quiet and must be not much more than the threshold of hearing. This is in a very quiet country location and I do not know where it comes from it certainly is not from equipment in the guiding (they can be heard separately when switched on) and my best guess is rumble coming through the rock strata from a hydro generating station about a mile away! It's either that or vibrating electricity cables in the supply to the house! Not an issue but an interesting (to me!) aside!
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Old 07-30-2020, 07:43 AM   #10
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Quote:
Originally Posted by Allybye View Post
Hi Coachz,
For most recording not usually short of gain but what I am recording is very low acoustic pressures such as the ticking clock at about 5m distance and without that clock the very low acoustic noise in the room.
The dynamic mics such as the Røde M1 give very little output with that acoustic input. The higher output condensers not really much of a problem.
The clock is really only being used as a low level reference!

To put a bit more context on this I am now more aware of a low frequency rumble in the room that I can, now I am aware of it, just hear by old fashioned ears!
It is exceedingly quiet and must be not much more than the threshold of hearing. This is in a very quiet country location and I do not know where it comes from it certainly is not from equipment in the guiding (they can be heard separately when switched on) and my best guess is rumble coming through the rock strata from a hydro generating station about a mile away! It's either that or vibrating electricity cables in the supply to the house! Not an issue but an interesting (to me!) aside!
You might like a cloudlifter or something like it. Radial makes great stuff too.

https://www.youtube.com/watch?v=RYhxtLFL27Y
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Old 07-30-2020, 09:04 AM   #11
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Thanks for the suggestion.
Short reply that is not a road I wish to travel.
Firstly I do not want to spend any more money on something I will not use in the longer term (this is just a short term exercise). I would need several Cloudlifters or fetheads etc. and that would be what 200 to 300 £s o $s ? Cannot justify that.

But you have prompted me to thinking that I could reduce the number of parallel channels I record and route each channel twice through my preamp to give extra gain. Could be that or normalising is job done!
At least those ways no extra cost, known very good preamps, calibrated gain etc. (Preamps are bipolar discrete bespokes, not off the shelf ones, in case anybody suggests buying better!)

I have looked at the Cloudlifter and the like before. I am a born skeptic and really wonder, apart from extra gain, why anybody would purchase them. No specs seem to be freely available. Advertising claims that the sound is improved by fet stage or two with no xformer, capacitor or resistor in a 2 stage amp is anathema. Reviews do not explain how you have an amp without components in circuits and thus in the audio path. Many comments have been made without proper comparative testing. One even wrote that 70dB gain was required for a particular purpose but only 45 to 50 with a Cloudlifter ignoring the 20 to 25 dB it provides! In my brain 45 + 25 equals 70! So no more gain and noise from extra stages! Maybe insignificant if the existing preamp is very noisy.....ooops nearly wrote a crap preamp!

Last edited by Allybye; 07-30-2020 at 09:09 AM. Reason: a few typo corrections mainly....
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Old 07-30-2020, 09:27 AM   #12
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Sanity check time!

Getting good signal vs noise with any average mic (that means even a beer soaked '58 - yes, literally) with any average mic preamp (yeah, the Mackie 1202 is fine!) is not hard! The digital system we have with 24 bit AD converters to capture audio is nothing short of full happiness and light here in the golden age of audio.

The analog stage at the input is the part that hasn't changed since 1940. Garbage in, garbage out. Like I said, even a beat up '58 through a crusty old Mackie mic pre delivers VERY usable signal just out of the box. Sure, there's stuff to talk about here and more finesse than that would be welcome! But it works.

Where I'm going with this is...
The above sounds like something is truly broken!
Then speculation above suggests that capturing anything accurately is such a technical stretch and requires significant expense and specialized software beyond a standard DAW. And that's just not the case.

This REALLY isn't that hard!
You either have a genuine edge case. Like a very low output mic in front of a super quiet signal that truly needs a more expensive analog input. Or something is straight up broken like a mic cable.


Now, sometimes you need to capture an event with more than one mic! In theory, you should be able to put a mic in any position in a room and record exactly what you hear when listening from that spot. Our hearing scales with our perception much like vision does. You get cases where you can't capture an enormous dynamic range with zero noise from one point field recording style. Unless maybe you threw $$$ at it with all class A mic preamps and Neumann or better mics. (Someone was going to say "Neumann isn't the end all be all." That's for you. )
But you could easily put a mic right up on that clock ticking you're trying to record and then a mic or pair of mics in the room to record whatever background noise - sorry, "ambience" - you're after. Then produce that mix.


This isn't that hard! Something weird is messing with you.
The discussion sounds like someone trying to control the speed of their car by measuring how far to press the gas pedal in mm.
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Old 07-30-2020, 09:54 AM   #13
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I could reduce the number of parallel channels I record and route each channel twice through my preamp to give extra gain.
What? I mean.... what?

If scientific experimentation is the order of the day, maybe try to build a better mousetrap. Let's not waste any more time trying to reinvent the wheel.
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Old 07-30-2020, 10:20 AM   #14
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There are data acquisition devices made for other fields than audio with more "dynamic range". Sample size might be the better term. Look for a forum closer to the application in question. Trying to use audio AD converters and running a signal through AD stages multiple times is only going to add noise. The very first pass is really the only one that captured anything. Further passes will only add noise and erase original data and precision.

Look for medical or other scientific fields for analog data acquisition with higher sample sizes. And then the analysis software made for that field. Clearly audio equipment is not up to the task! Sounds like it was a creative experiment to try all the same!

Not sure how else to respond!

For the speculation that capturing audio with ANY precision and with more signal than noise is nearly impossible to be remotely true, we would have to assume all audio interface devices and all DAW software to be equally crude and inaccurate. And, you know, there are all these recordings out in the world and people using this stuff day to day. They're not making everything up. I haven't been hallucinating listening to music all these years. (Hallucinating while listening is a different subject, thank you very much!) Just... no!
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Old 07-30-2020, 12:01 PM   #15
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The newest Zoom Recorder captures more then 24 Bits in Floating point format.

This seems appropriate for such applications.

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Old 07-30-2020, 01:10 PM   #16
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I read what you say and realise you are trying to help and keep me on the right track, as it were!

But please do not think I am trying to make a good music recording and that I do not know that lots of equipment out there in the worldrecords good music.
That is not my aim for this exercise.

It is more akin to a scientific investigation/demonstration and I just want more gain! I thought it might be possible in Reaper and I still have to try out the suggestions regarding that.

I would humbly suggest you do not limit the scope of what you think audio equipment can be made to do. I do not need high accuracy precision a couple of dBs is fine not a fraction of 1% it is after all being done for audio purposes albeit mine! I'll try and link to my results if I get there if I think it will be of value.

I will not be going down the scientific equipment route for good reasons. I have been an engineer and have been involved in Instrumentation tasks (not meaning here musical instruments but measurement instruments) for a number of years and know the capability and the cost. As I have posted earlier I could not justify my (pension) income to be spent there plus it is not necessary.

I'll just carry on with what I know and what kind posters, maybe frustrated by my questions and explanations, have nevertheless given much help for me.

My current devices of Class A preamp stages and Reaper via 24bit interface will have to do!

As Fex wrote "Let's not waste any more time trying to reinvent the wheel." I'll do the lateral thinking, but let's end the thread, at least for now. Cheers all for input to date.

See you on other threads!
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Old 07-30-2020, 04:57 PM   #17
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You can use multiple analog mic pre-amps. You can put them in series but really you should only need one preamp before the interface (each preamp will add interconnect noise which gets amplified). Most people do not do this, we rely on the preamp inside the interface.

Your interface, regardless of how many bits, has a preamp in it. And then it goes to A/D.

Depending on the mic you can have:
mic --> powered preamp --> computer audio interface (which includes preamp and A/D)


A preamp does not need to be expensive.
You can build a preamp using a small 8-pin LM386 chip and a few external components, the circuit has been reliable and stable and very low noise since the 80's, and it runs on 12VDC (but higher voltage for the power like 18VDC gives better signal to noise). And the gain can be very very high, simply depending on your component choice, so high that you can purposely make it go unstable (into self-feedback oscillation). There are diy plans published everywhere for this.

The entire circuit costs under $5. So if you look to buy a preamp (especially with overinflated "hifi audio" prices) consider that really it only costs $5 to make.

Many scientists doing tests or field studies will build specialized preamps for their gear. Or they purchase expensive calibrated ones, called instrument amplifiers.

However with that said, one thing which bugs me about Reaper is that the UI slider to boost an item's gain in the property window, only slides up to +12dB, even when the "Normalize" button will boost to even +30dB, which is well-beyond where the UI slider can tune (the slider gets pegged and any attempt to move it will go back to +12 dB at most).
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Old 07-30-2020, 10:29 PM   #18
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Yep. It would be good to simply have an edit field to manually input am arbitrary dB value.
Please do file a Feature request.

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Old 08-01-2020, 01:12 AM   #19
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Waves L1
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Old 08-01-2020, 03:10 AM   #20
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f that might help, you could use some plugin in the Input Effect chain to boost the level while recording.

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Old 08-01-2020, 04:51 AM   #21
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Thanks all for the continued help and input.
I think I have a method now.

Will explain later.
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Old 08-01-2020, 07:43 AM   #22
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If the signal to noise ratio is OK as far as the mics and preamp you don't need any Cloudlifter type mic boosters. Raising soudfile after it is captured, to as far as you need it to go, will be fine. Putting a maximizer or brick wall limiter (like the mentioned L1) on it will enable you to raise the level as much as you want and if you set the ceiling at -1 it won't go above -1. It will hit against it harder as you raise it in the plugin, and go from just knocking down the transients to smashing the whole thing by 20 db as it raises it to -1. The general use of a brick wall limiter is to have it catch the transients and maybe a little more, and that can enable you to get 5 or 10 more db out of a track that was already normalized to -1.

There are some good free ones. https://loudmax.blogspot.com is one.

The low rumble you describe could be eating up a lot of the meter. If you haven't already, eq it out with a high pass filter and/or low shelf so nothing below the frequency of what you are recording comes through. That might give you another 10 db right off the bat. Might even be part of the reason why you weren't able to get a large enough waveform originally.

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Old 08-01-2020, 07:57 AM   #23
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Originally Posted by vdubreeze View Post
The low rumble you describe could be eating up a lot of the meter. If you haven't already, eq it out with a high pass filter and/or low shelf so nothing below the frequency of what you are recording comes through.
I could be completely wrong about this, but I inferred that recording the low rumble was the whole point of the exercise. If a ticking clock is what is required, one could just download a free sample.
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Raising soundfile after it is captured, to as far as you need it to go, will be fine.
I'm still wondering why Allybye doesn't just normalize the audio, because, well, that's what normalizing is for, and normalizing is how it's done.
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Old 08-01-2020, 02:11 PM   #24
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Well Fed is right regarding the rumble.....nearly.
Many are trying to help suggesting how to do a better recording, closer mic positioning, eq, better mic, preamp etc. but that is not the point of the exercise! Please refer to post #1 .

Please forget all about what we do can do or should do to get a 'better' recording or mix!

The object of the exercise is to inestigate on a qualitative and quantitative basis, noise and sensitivity comparisons between microphones. Nothing more, nothing less, based on real world info rather than specs, advice or sometimes dubious reviews.

All I started in the thread, in general, was is it possible to get more gain within Reaper?

Just as an aside to try and give some context, I mentioned something of the noise characteritics I heard at very low level that I had not really noticed before. That rumble is as Fex writes part of the noise that I am recording!

As far as normalising is concerned you presume, Fex, that I am not going to use that method. I have taken your advice on board. I plan to try it out when I have time but that will not be for at least a couple of days at the moment.
L1 sound an excellent FX process but as said before I will not be buying anything else at the moment owing to lack of budget. Maybe later if I need to but not just now so those suggestion, however good (and thank you for them) are on hold!

Last edited by Allybye; 08-01-2020 at 02:13 PM. Reason: first post ref.
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Old 08-01-2020, 02:24 PM   #25
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Quote:
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All I started in the thread, in general, was is it possible to get more gain within Reaper?
Simply, no.

Any gain on the original analog signal that will equally boost the noise floor can only happen pre AD conversion. Ideally from a first and only preamp stage to avoid more variables.

Sure, you can turn up the full contents of what you captured and converted to digital. But if you are after looking further down into the decimal dust and noise floor of the original signal, you need to capture that initially. Turning up a digital signal doesn't bring back anything deeper into the noise floor from the ether. The original samples captured are absolute. Even adding additional analog preamp stages could very well get into so much additional noise from signal chain issues that any additional original detail is lost.

If a standard mic preamp doesn't give you enough gain to measure down far enough into the noise floor for you, I think the only realistic solution would be to build your own preamp.
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Old 08-02-2020, 03:20 AM   #26
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" If a standard mic preamp doesn't give you enough gain to measure down far enough into the noise floor for you, I think the only realistic solution would be to build your own preamp. "

That is exactly what I have! ��

From post #11,
" ...... no extra cost, known very good preamps, calibrated gain etc. (Preamps are bipolar discrete bespokes, not off the shelf ones, in case anybody suggests buying better!)"

Ok you are not suggesting Serr, that I buy better but make better. It perhaps as not clear in that post that what you currently suggest is what I have done. My preamps are very high quality, built by me, very low noise and distortion figures, great frequency response covering more than the required spectrum etc. They are very close to the theoretical minimum noise of a resistor.
The only issue for my present task is that they were built to cover the normal gain required for recording and I have no wish to go inside them modifying them for a short term task!

At risk of repeating myself again (but I will do....��) all I am trying to get is a bit more gain at no cost to make analysis easier.
Thanks to posters I think I have a solution of two! ��

Last edited by Allybye; 08-02-2020 at 03:36 AM.
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Old 08-02-2020, 03:32 AM   #27
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Quote:
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I think the only realistic solution would be to build your own preamp.
I suppose this will not help without in fact building your own Mic, as the hardware - even just wires - will introduce too much noise.

Cooling down the mic plus the first preamp to some -200 °C might help.

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Old 08-02-2020, 03:46 AM   #28
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Ooooops, emoticons haven't shown up well in the forum!
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Old 08-02-2020, 06:01 AM   #29
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Ooooops, emoticons haven't shown up well in the forum!
Select from the "Smilies" list linked at the bottom of the page (under Posting Rules), or memorize some of these.

https://forum.cockos.com/misc.php?do=showsmilies
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Old 08-02-2020, 08:19 AM   #30
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Quote:
Originally Posted by Allybye View Post
" If a standard mic preamp doesn't give you enough gain to measure down far enough into the noise floor for you, I think the only realistic solution would be to build your own preamp. "

That is exactly what I have! ��

From post #11,
" ...... no extra cost, known very good preamps, calibrated gain etc. (Preamps are bipolar discrete bespokes, not off the shelf ones, in case anybody suggests buying better!)"

Ok you are not suggesting Serr, that I buy better but make better. It perhaps as not clear in that post that what you currently suggest is what I have done. My preamps are very high quality, built by me, very low noise and distortion figures, great frequency response covering more than the required spectrum etc. They are very close to the theoretical minimum noise of a resistor.
The only issue for my present task is that they were built to cover the normal gain required for recording and I have no wish to go inside them modifying them for a short term task!

At risk of repeating myself again (but I will do....��) all I am trying to get is a bit more gain at no cost to make analysis easier.
Thanks to posters I think I have a solution of two! ��
I wasn't meaning to suggest "better" vs "worse" in the traditional sense. I would suggest that that doesn't even apply here. If the issue is you need more analog stage gain, then you need a preamp circuit that delivers that. You stated that you needed to examine the signal in the noise floor and thus a standard mic preamp cannot deliver that range. A "better" mic preamp with the same gain range would not be better here if what you need is the increased signal amplification. It's all related of course. A higher gain preamp requires lower noise components and design. What I mean is "better" between two standard gain range mic preamp designs would be meaningless here.

Again, if you are playing with very small signals literally in the start of the noise floor, kludging preamp stages together is most likely to generate noise that eclipses what you are trying to measure.


I sense there might still be some miscommunication here. I explained the concept as well as I'm able right now. Anyway, I hope that helps!
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