restorer-john
Grand Contributor
In early digital era, for example, you can see that this DAT machine has marker on -20dBFS when looking at the meter.
Oh, that Sony Pro DAT is real purty...
In early digital era, for example, you can see that this DAT machine has marker on -20dBFS when looking at the meter.
You have to be careful with Mic preamp gain marking in digital audio interfaces, since there is no standard defining what those "gain" marks mean (or in other words, what's the 0dBFS level in dBu or dBV).Even for the low sensitivity dynamic mics? I suppose proper interfaces with 50dB+ worth of gain are going to work well with something like a Shure SM7B, but something like a Focusrite Solo that maxes out in the high 40's if I recall, may leave something to be desired. Though still good enough nonetheless.
I see 0dBFS = 20dBu on (pro) Yamaha preamps.Nowadays +24dBu = 0dBFS
That's why my original post saidI see 0dBFS = 20dBu on (pro) Yamaha preamps.
And I have a Focusrite Liquid 4Pre where 0dBFS=+22dBu
But that's purely abstract, since it's just based on what figure they use to label each gain position.
These info are listed on product specs, so not abstract at all. Users need to offset that value to achieve the desired headroom.Nowadays +24dBu = 0dBFS is common used, albeit some lower-end or bus-powered equipment often fall into somewhere between +10 to +20dBu.
True.These info are listed on product specs, so not abstract at all.
Those I listed are not exactly "Lower-end"my original post
You have yo be careful with Mic preamp gain marking in digital audio interfaces, since there is no standard defining what those "gain" marks mean (or in other words, what's the 0dBFS level in dBu or dBV).
As an example, I have a Motu 828mkIII (FW), which seems to reach 0dBFS with a -58dBu signal for "53" gain mark, while a Yamaha AD8HR would allow up to -33dBu for the same gain mark.
Thats a massive 25dB difference.
My original post mentioned tape recorders (analog and digital), so the subsequent example of 10-20dBu and 24dBu are also not preamps. So it also makes perfect sense for line-ins to have slightly higher headroom. Anyway, if the term "lower-end" offended you, I am sorry about that. That post was not intended to reply your previous posts.Those I listed are not exactly "Lower-end"
I Kinda get this. Still doesn't fully gets that this would mean fully fixed gain analog stage, maybe part of the answer here with a previous post: @jerryfreakA good page on 32 bit float
https://www.sounddevices.com/32-bit-float-files-explained/
Sound devices control automatically the gain of preamp for avoid clip. This gain is encoded in the 32 bit float.
A fun video and good demonstration that show the possibility atat 1:16
I don't think it works that way. It is said to use two ADCs. One for high and one for lower level signals. Then they are used to create a clean 32 bit result. The combining is the tricky part. And no I don't mean 32 bit or even the full 24 bit range.I Kinda get this. Still doesn't fully gets that this would mean fully fixed gain analog stage, maybe part of the answer here with a previous post: @jerryfreak
"The dual ADC is used when recording in 32 bit float and when recording in 24 bit. The Trim allows you to set the level that will be recorded to the 24 bit file. This makes it so that if the signal clips in 24 bit mode, it is due to the file clipping from the limitations of the 24 bit integer format, not from clipping at the converter."
Could we assume from this, maybe it's my understanding that is not fully arrived, but the first adc in the chain would be used to set the gain but it would control an analog preamp, some sort of resistor network, but only the second adc would be the only actual one responsible of of the digital conversion of the recording? Or does that really mean that the gain stage is fully fixed and would accomodate enough headroom for any input signal and that it is all fully digital from there?
Thanks for the explanation, Interresting thread.I don't think it works that way. It is said to use two ADCs. One for high and one for lower level signals. Then they are used to create a clean 32 bit result. The combining is the tricky part. And no I don't mean 32 bit or even the full 24 bit range.
As an alternative imagine this. I split my mic feed to two inputs. One I run at a normal gain, maybe 40 db thinking it probably won't clip. The other I run at 0 gain or maybe only 10 db gain. And occasionally that 40 db gain side does clip some. I can take the low gain version for just that segment that clipped, amp it up digitally to the proper level and replace the distorted clipped section. It would have a raised noise floor briefly, but would beat being clipped.
Guessing Zoom are targeting that crowd.
Of all the people I've known in broadcasting, I can't think of one who would overlook those terrible A/D results.
It doesn't.And this matters exactly why?
It did not.My original post mentioned tape recorders (analog and digital), so the subsequent example of 10-20dBu and 24dBu are also not preamps. So it also makes perfect sense for line-ins to have slightly higher headroom. Anyway, if the term "lower-end" offended you, I am sorry about that. That post was not intended to reply your previous posts.
I don't think it works that way. It is said to use two ADCs. One for high and one for lower level signals. Then they are used to create a clean 32 bit result. The combining is the tricky part. And no I don't mean 32 bit or even the full 24 bit range.
As an alternative imagine this. I split my mic feed to two inputs. One I run at a normal gain, maybe 40 db thinking it probably won't clip. The other I run at 0 gain or maybe only 10 db gain. And occasionally that 40 db gain side does clip some. I can take the low gain version for just that segment that clipped, amp it up digitally to the proper level and replace the distorted clipped section. It would have a raised noise floor briefly, but would beat being clipped.
If there is a downside I'd say safety and compatibility.Is there a down side (isn't there always?) to 32 bit float and the and the ability to recover in post? Perhaps a stupid comparison (and possibly outside the groups expertise), but is this similar to what RAW files allow in photography?
that mic will still need gain, but at the end of the day the >110dB dynamic range of modern ADCs still exceed the dynamic range of real-world recording situations, limited by self-noise of the mic, and background noise (as well as max SPL)
Sensitivity in the mic won't help. Eventually all mics have a self noise defined by the individual air molecules hitting the diaphragm. Larger diaphragms have a lower self noise floor as they average out the impacts more. Similarly the analog stages are limited by the various thermal noise sources they contain. Lots of care can be taken, but there are fundamental limits that can't be got past no matter what.For sure agree.
Yah I was wondering what people do to remove noise. Is the solution simply using sensitive mics? Also I recall Okto Research talking about when they moved into their Stereo version of their DAC. They said they combined the 8 channels of their Pro version and that leads to better performance.
Are designs like that possible with ADCs or perhaps post processing of you natively havr mote channels to work with, but havent had a hardware design where this "combining of channels" done by the company?
Backgroud hiss combined with noise is what I feel id mostly be concerned with becoming better tackled from the ADC side. Do you have any tips on how that can be achieved barring expectations of "summation of channels" like Okto did? And of course simply buying more sensitive mics?