The preamp has nothing to do with the number of bits, one is analog and the other is digital. In side the device is a 24bit ADC those 24 bits map directly to the 24bit part of the 32 bit float. That is a 24 bit int has exactly the same information as a 32 bit float so far as the static representation of an audio signal. The 32 bits float only becomes useful when processing the signal with effects or when mixing. The actual bit depth does not change but remains 24 bits of actual audio information. Most audio interfaces from a semipro on up have 24bit (int) ADC and that is mapped to a 32 bit float for use inside the computer. Ardour uses 32 bit floats throughout for all audio, even audio from 16 ADCs is converted to 32bit float. There really is no difference between 32 bit float and 24 bit int, they are a different representation of the same thing. That is, 24 bit int converted to 32 bit float converted to 24 bit int should be bit perfect. The use of 32bit float in this device allows keeping all 24 bits of information intact at different levels. That is 24 bits in from two or three ADCs at different levels can only be expressed with 32bit float while keeping 24 bits of information. Most ADCs are single and so can express the input directly with 24 bit int. In both cases the resolution is still 24 bits.
BTW imagining there will be a difference is not the same as there actually being one. The only way to find out is testing… from the math, I would expect that any measurable difference between this device and another would be in the analog components rather than the digital representation. This is not a put down of this device using 32bit floats. Using more than one ADC at different levels requires 32 bit float and that is what it would end up at in the computer anyway. I am just saying that it is still 24 bits of information. It is not a higher resolution or a lower noise floor (as Robin pointed out, the analog circuitry already limits the noise floor to less than 20bits of information) but rather an easier format for manipulating audio information.
I agree that 32-bit float is just 24-bit scaled up and down. The question here, I think, is whether the levels at which the mic input hits the preamps makes any difference to the sound quality. Again, it may not be noticeable but in others areas in the audio world it doesn’t stop people wanting the very best quality – think SRC and dither! Chris Johnson (airwindows) not only has what might be the world’s best dither (NJAD24 and 16 as part of StudioTan) but also a float dither both separately and baked into every one of his plugins. Could people tell in a blind test if 32-bit float dither is being used? Probably not, but it’s nice to know that using them is not negatively affecting the sound. And there’s the math to back it up. Same for SRC…Ardour’s Secret Rabbit Code seems great but it doesn’t stop programs like Brick, Smarc etc taking things to an extreme.
And, yes, testing is the only way. I would love to hear a recording of something acoustic, say a choral or symphonic concert, both in 24-bit and 32-bit float turned all the way down. As long as I don’t hear any difference over my monitors or hi-fi speakers I’ll gladly show up and use 32-bit float without even turning up the gain. If one sounds more sterile, then I’ll be sure to at least turn the knobs to unity gain. I’m not interested in null tests per se, but each to his own.
@ LEN yes I understand that of course this is all in the digital realm. but i can apply before different mic gains? if i can, then this will alter the sound, as mic preamps do usually not work 100%linear with every gain? and cant i saturate a mic preamp on purpose to get some sound colouring there? something i couldnt do to the extremes with 24 bit? there i was aiming at… but correct me when i m wrong, this is pretty new to me…
A device that has 32bit float outputs has a chain that looks like this:
Audio in->preamp with gain control->24bit int ADC->internal cpu->24bit->32bit float->to user’s computer. The gain control must be in the analog part of the circuit in order to make the maximum use of the bit depth without clipping. With two (or more) ADCs it would be possible to have the level control set two levels one you see and one that is hidden and exactly 20db down (or some other know value). The ADC at the higher level would be used for the output most of the time but if that adc goes over full scale the second one at 20 db down could be inserted by increasing it’s level by 20 db and switching to use that. The 32bit output would then faithfully represent the portion of the waveform that was over 0dbfs. However during that clip the bit depth would actually drop from 24 bits to 20-ish bits. It would still sound better than a digital clip. Also, it would allow recording at a higher level overall meaning a better use of the 24 bits available. However, it is questionable if regaining the last 4 bits is worth anything if the limits of the audio circuitry is only 19 to 20 bits anyway. There is not any way that using 32 bit float will help reveal preamp sound. The normal way to do that is to use a separate preamp that has the desired sound, set it’s level for that sound and then set the output level for correct ADC level in.
Nothing is “scaled” up or down, 24bit int and 32 bit float are the same information in a different format where each format will give the exact same output. Now it is true that once the data is reformatted to 32bit float it can be scaled up beyond limits imposed by 24bit int and that scaling down will not loose bits of information as the 24bit int does. But after conversion from one to the other there is no difference besides format until someone decides to modify the data. That is why most audio interfaces are raw 24bit int and most daws use 32bit float.
Of course it does… but that is not really the question at all. The reality is, in any recording case the preamp will be set in a similar position so that the peaks are less than 0dbfs and most of the audio is no more than about 20db down. Changing preamp level for colour is not in the cards in this case. Preamp colour means using some sort of extra level control after the preamp to ensure the signal level falls within the best recording range of the ADC. So the levels going into the ADC are pretty much constant for any circumstance. Dither is for exporting at 16 bit, not for signal processing a 32 bit float over and over as it goes through each effect in a chain. Dither is by definition noise that is added to the signal. If done right and only once it is useful, if done wrong and over and over the noise floor goes up and you loose bits of information. SRC is generally frowned on, it is only used where no other method will work (like those cheap USB microphones so many people want to use). SRC means something has failed and SRC is there to rescue an otherwise unusable situation.
What you hear over your monitors level wise, is not anything digital, but rather the analog level on the amplifier connected to the speakers.
I’m referring to the effect of having up to 3 different input levels (padded or otherwise) coming together to make up the 32-bit float file. If there are 3 different levels to accommodate a gain range extending beyond that of a good microphone, at least one of them will be hitting the preamp/convertor at a low level. I guess it doesn’t make too much of a difference in the real world though…
I disagree. If I record an orchestral concert at 96kHz/24-bit I always want to create FLAC and MP3s from that highest quality file and then down-sample (externally if necessary) to 44.1k in order to create a physical CD from the same markers. That’s standard practice and Wavelab includes a great custom montage feature just for that reason! Most people go to external SRC because the ones included in DAWs are generally worse but Ardour (Secret Rabbit), Wavelab (SoX) , Pyramix (Hepta Apodizing) etc are superb. There’s a reason people use Weiss Saracon on a regular basis – precisely to be able to master hi-res digital files at 96k or higher and then again at 44.1k for CDs using all the markers and edits they have already completed.
@ len thanks for clarifying, this all makes more sense to me now. so I d say for my usecases: recording music wont make a difference as i can set the gain as i wanted. when it is for set recording in a chaotic situation and a lot of channels 32bit floating point can make a huge difference.
https://people.xiph.org/~xiphmont/demo/neil-young.html https://xiph.org/video/vid2.shtml
96k equates to twice the space, twice the noise, twice the cpu, distorts on most systems that can actually reproduce anything above 20khz.
Anyway, yes SRC on export may be useful for 48k to 44k1 or back (48k is standard and 44k1 is CD) But we were talking about audio interfaces, in the case of audio interfaces, SRC is a fix for a broken situation.
Eek. You should probably tell this to Linn Records, a superb hi-fidelity classical label with perhaps the finest recording engineer alive to today, Phil Hobbs. They record at 192k for SACD and, I assume, down sample afterwards for the various formats including, on occasion, 44.1/16-bit harpsichord discs. SRC is far from a “fix for a broken situation”. They produce their own music systems and speakers too. I believe that in this conversation I personally have only mentioned SRC with regards down-sampling in software after capture.
Also, there are plenty of people who say that 96k or higher is essential for capturing complex reverb tails (and plus I find it is much better to SRC from a higher samplerate down to 44.1). I tend to be sceptical of ultra hi-res download options and often default to 44.1/16 for audio-only productions but a couple of people have convinced me recently that it is worth capturing at a higher sample rate at least (given HD space is cheap as chips these days). And, honestly, in conversations with Phil Hobbs while I was working in Glasgow, I trusted everything he said about audio and watching him coax a good performance out of an ill-prepared performer was magic in of itself. I digress…
The link len posted is my favorite when it comes to the sample rate discussion. Although i have to say there are moments 96kHz or beyond might make sense: some classical music recordists say it adds a silky “coloration” to the sound , that is favorable (so lets say high sample rate as a “pleasing effect”) [I have never witnessed that with my own ears.] or when you do fieldrecordings (bats) and want to analyse ultrasonics in baudline , or when you want to apply weird downward pitch shifts from crash cymbals for sound design. , etc…
for “general” production I don´t see any sense to it.
There is marketing and then there is physics. Marketing is about getting the highest dollar for a product. Physics is reality. Linn Records is selling a product for maximum profit. They are selling to the same people who are willing to pay extra for high sample rate reproduction equipment and gold speaker wire. They may feel they have to work in the fashion they do so that they can compete with other companies advertising high resolution audio. This does not mean anybody, and I do mean anybody, can tell the difference a higher sample rate than 48k makes. Very few people who have made past their teens can hear sounds as high as 18k, let alone 20k. Any difference heard is the separate mix done for the higher sample rate product… as in you’ve been hood winked.
Feel free to record and sell audio in any format you wish but do understand there is no audible difference aside from the greater distortion higher sample rate audio produces on high frequency playback equipment.
I’m genuinely enjoying this discussion. As long as you are comfortable let’s continue. In the case of Linn, they are recording at 192k with pristine microphones, preamps, beautiful spaces and the like. It’s not like they are recording at 44.1 and upscaling to 192 to pull a fast one. I don’t think marketing plays a huge part in this, honestly. They were one of the first to offer it so it wasn’t as if they were needing to compete. Physics says that 44.1/16 is plenty for dynamic range and frequencies that the best human ear can respond to but it does not address the “rate” part. 96k captures complex tails better because there are more sample points than at 44.1k. Has anyone studied at what rate the human ear can discriminate sample points? In down-sampling this has a better effect than simply recording straight at 44.1k. There are plenty of folks in the business with “golden ears” who can reliably tell the difference a good deal of the time. I’m not one of them unfortunately but something “feels” different when I switch between SACD and regular CD mode on my Onkyo player. As someone wise recently told me, music isn’t just about frequency…it is also about time.
We had that discussion a lot here and in other places, so I m not sure how far this still makes sens… in the case of Linn, I dont know, they surely have great sounding recordings and let me be provocative: I think they would even sound better when recorded at 44kHz/24bit and dithered beautifully to 16bit.
" I’m not one of them unfortunately but something “feels” different when I switch between SACD and regular CD mode on my Onkyo player. As someone wise recently told me, music isn’t just about frequency…it is also about time."
I get that totally. And whatever sounds better subjectively IS better for your personal case. Noneoftheless there are various issues about verifying if it is “objectively” better, on the recording side, and on the playback side: Where the SACD and the CD mixed and mastered exactly the same? maybe the internal SRC of your onkyo player is better than the one they applied in the studio? would the SACD still sound better if the CD would be recorded and mixed exactly the same way but at44kHz with no SRC? if the SACD is just mastered 0,1 db louder, it might appear to sound better. no audio system can reproduce 24bit, so what is happening at that stage in your hardware? does the dac in your system has the exact same circuitry? and there are a lot of more questions. …
Check also the nyquist-shannon theorem the digitisation process relies on.
but then: imo whatever sounds and feels good is fine on the music part, because it is about the fun of it. It s just not right to use myths and marketing to justify a supremacy of a recording chain no one can scientifically proof it is better (and that for the time being is proofed to be worse). The reason for that I think is also there is just so much money in marketing for products that is misleading and no one will spend millions in ads on the nyquist-shannon theorem, too bad
Yes, I’ve actually had that bookmarked in my browser and read it a good number of times – it makes very convincing reading! I want to believe it is true, I really do, and came close to releasing everything I do purely at 44.1/16 and I still might In terms of “beautiful dither” what do you use? Good ol’ triangular, shaped, “non-dither” a la airwindows?
To clarify, you would consider the whole higher samplerate / complex tails phenomenon nonsense too? I may be overthinking things but I’ve tried to use the Xiph article in discussions previously and it has been shot down as not telling the whole story. They say that phase and aliasing are the most important factors when listening to music as we do it both in the frequency and in the time domains. 44.1, they claim, is not good enough to avoid phasing issues and why perhaps people still enjoy vinyl and tape because it is more relaxed.
I appreciate the time you have taken to discuss this and what you say does make a lot of sense. I hope you agree that it’s healthy to question the various viewpoints and what Xiph has going for it, obviously, is that it isn’t trying to market anything
@anon60445789 I usually use the shaped noise dither within ardour after some reading I did. Not that I remember any of the arguments exactly but it sounds good to my ears and I m happy with it! I m a fan of some of airwindows plugins and would like to test how to dither with his method… how would i do that? apply his dither plugin while exporting, without applying the internal dither option in ardour on export?
The phasing issues: what are they? I have not heard that before…
And yes imo the analog <–> digital discussion is a whole other thing better not to get into in this thread. but within the digital realm, things can be pretty binary so to speak
Exactly this. On the master bus I use NJAD24 and NJAD16 as part of his StudioTan plugin but his new monitoring plugin will also allow by selecting either Out24 or Out16. Either way, disable any DAW dither and you will be good and be sure to leave master fader at unity so that dither (or “non-dither” in this case) is the very last calculation before export. If you watch Chris Johnson’s videos on NJAD, StudioTan etc you’ll get a great idea about the math behind it (Benford realness calculations) and his claim that even NJAD16 will give a CD a sense of infinite resolution.
Actually for a design like this it is more (excuse poor ASCII diagram)
/|----- preamp gain 1 -----> ADC1 --|\
audio in -<-|----- preamp gain 2 -----> ADC2 --|-> DSP to selectively blend between three inputs
\|----- preamp gain 3 -----> ADC3 --|/
The thing to watch out for is that the noise floor can change with signal level as the DSP fades between the three (in the case of Sound Devices; four in the case of Stagetec) different preamp gain/ADC combinations. You have to be aware of that fine print when looking at the specs, and understand what the measurements mean. For example, Stagetec claims that their input stage has a 158dB SNR (with 32 bit integer output, 143dB when converted to 24 bit output). I don’t believe that a 158dB SNR is physically possible, I suspect that if you looked at the noise floor and signal at any one signal level you would see something more like 120dB SNR, but because of the way the output is derived from multiple input stages with different gain settings the 120dB of SNR is moved around to different places in the available code points of the output word.
I have not heard one of those gain ranging converters myself, reportedly it sounds OK when done well. It obviously gets pretty expensive since you are effectively taking three or four normal input channels and then adding on additional DSP. I’m also not sure how the input clamping is handled, the stages with higher gain that get selected for low level signals must get overloaded when the signal level goes higher, so something has to take that into account in the circuit design so that when the signal goes below the overload point those high gain stages recover back to full performance very quickly.
It does sound very appealing to just plug in a signal source and not have to worry about gain staging. Maybe calimerox can report back in a couple of months and let us know how it works out in practice.
Well that doesn’t look right at all. Something ate the extra spaces I put in to try to make the ascii bars line up. Hopefully you can figure out what I was trying to do, else I can try to paint up a png file and see if discourse will let me add that in a forum post.
In terms of “beautiful dither” what do you use? Good ol’ triangular, shaped, “non-dither” a la airwindows?