get more gain on tracks?

[Allybye]

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?

[Fex]

Originally Posted by Allybye

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?

[mschnell]

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)

-Michael

[Allybye]

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.

[serr]

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.

Originally Posted by Allybye

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.

[Fex]

Originally Posted by Allybye

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.

Originally Posted by Allybye

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.

Originally Posted by Allybye

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.

Originally Posted by Allybye

Phew! Does that help or confuse?

Two meny werds.

[Allybye]

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'!!

[Coachz]

Originally Posted by Allybye

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?

[Allybye]

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!

[Coachz]

Originally Posted by Allybye

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

[Allybye]

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!

[serr]

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.

[Fex]

Originally Posted by Allybye

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.