I am a professional classical violinist, but I also have a business doing live concert recordings of classical ensembles. On occasion, I end up recording a concert on which I also perform. When that happens, I can't just sit out in the venue during the dress rehearsal to work on levels. The best I can hope for is a few moments at the beginning of rehearsal where the conductor will have the orchestra play one of the louder sections of the concert while I frantically set levels. After rehearsal, I would then go home, listen back and make notes about what I needed to do differently for the concert the following night. Still, I always had that "did I leave the oven on" type of feeling on stage wondering if I got the gain staging right. After all, the dress rehearsal doesn't account for the audience clapping or the performers getting extra amped up for the concert. This is especially true of vocalists, who rarely sing in full voice during the dress rehearsal. The obvious solution, of course, would be to not record concerts on which I also perform, but then I would make half as much! Then, 32-bit float entered the picture. I replaced my Zoom F8 with the F8n Pro and have not had that feeling since. Turn it on, hit record and the levels are perfect every time. This also comes very much in handy when I am not performing. I no longer have any fear when accepting recording gigs on short notice. I can walk into a venue, spend all of my pre-concert time assessing the room and deciding on a stereo mic technique and not waste a single second worrying about gain staging. As for the sound quality, I have recorded, mixed and mastered nearly 100 live classical performances and I cannot hear even the slightest decrease in fidelity when using 32-bit float. If you hear a noise floor, it's your mic, simple as that. And Beethoven's Symphony no 9 with 100 voices is not loud enough to clip the inputs when mic'd properly. Just stop with all of the caveats. If you're not recording Falcon Heavy launches, then you are going to be fine. If you are skeptical about taking the leap, think of it this way. When I first started doing concert recordings I used a 8U rack with a 24-channel mixer bolted to the top and recorded everything on a Roland VSR-880 hard disk recorder. All of that functionality, and immensely more, is now packed into my little F8n Pro. Now, do you think I have any nostalgia for the good old days of pushing 100 pounds of equipment all around town? Hell no! And I won't miss worrying about gain staging when I only have one chance to get the recording right. I am an enthusiastic convert to 32-bit float and I will never go back.

I had no idea anyone would get so worked up over 32-bit FP. I record almost everything at 32-bit FP because I work in film, specifically as a sound designer, and the dynamic range is awesome. Same with dialogue. When working in 24-bit I would do something similar to what Sound Devices does with their 32-bit FP; meaning I would record to two channels, with one channel properly gain staged and another about -10dB below that. Using 32-bit FP saves me setup time, and editing time. My deliverables are still almost always in 24-bit 48kHz because that's the current industry standard. But my preference is recording in 96kHz 32-bit FP then downscaling. I've never heard a client complain, and frankly most of them don't even know enough about sound to complain. I will add that I've clipped in 32-bit FP a few times when making sound effects, and had to adjust how I'm setting up mics.

such a great video, thank you so much for sharing this with us!

I love it when musicians talk about computer science. As usual the marketing department chose to misuse a technical term because it sounded cool not because it was accurate at all for what it was describing and thus "32 bit float" somehow became an "audiofile" buzzword...even though this is all about sampling and compression. The real issue is everything is optimized for MP3 and Mp4 and all the 32bit float formats are proprietary.

Great explanation! It reminds me of the early days of digital stills photography (late 90s). As a co-chair of the Digital & Advances Imaging committee for the Professional Photographers of America we were introducing and teaching thousands of our members and non members about the new tech. Almost without exception, we heard “digital will never replace film”. As you said it was fear of the unknown new tech and workflow. Try to find a place to develop film now. As I transitioned into cinematography, the film and television industry went through the same “digital will never replace film” denials 10-12 years later. Some of it is protectionism and resistance to change as you said. Ten years from now only the die hard OGs will be clinging to their antiquated 24 bit recorders and looking for people to repair them.

Very cool and helpful video, thanks!

It's interesting that half a year after you posted this video, ZOOM released their essential series 32-bit recorders with NO gain controls. I jumped into 32-bit float (Tascam X8) because I'm a solo musician and record all my live sessions. I have enough to do without having to run a soundcheck and try to get optimum levels. 32-bit allows me to just hit record and not worry about if the recording is too loud or too soft. I was fine with 24-bit for years, but sometimes the levels weren't set well (I had to guess) and the recordings were way in the red, making them unusable. 32-bit gives me a usable recording as long as I don't overload the front end.

I don't use 32-bit float because people are now selling perfectly good 24-bit equipment for bargain prices. 16/24-bit was fine for decades for professional workflows. It's certainly good enough for my silly hobby projects for the next few years.

Hats off, man. Grest subject to cover. I agree, the experience I had with the F6 was a real gamechanger, albeit for field recording and location sound but now I want to try it for music purposes thanks to you! Subscribed.

Great vid. Just bought a Zoom H6essential, and this one of the few pro-32bit Float essays out there. I embrace the future.

The argument I have made for years is my consoles in the studio have about 6 DBU (+26dbu) more headroom than the best 24/32 integer converters (which run at +20dbu) on the market so if I want to slam the mixbus or push a mic preamp I am going to overload the converter, the only option was a 32 bit floating point converter like we see today. The dual gain stage of the converter gives us the headroom needed. Sadly there are not any large 24 or 32 i/o interfaces for console users like myself. But I have been using a Mixpre for about a year now as my final mixdown rig. So When we do our final mixes the output of the console is captured to a separate rig running a Macbook Pro and a mixpre. But we really need a Thunderbolt interface with DB25s and around 32 i/o to really use in commercial studios!!

Great video!

These vids are quite good, I’m keen to get into 32 bit mode.

You can hear the distortion on the back end of transients for 32-bit float. Specifically zero in on the snare and hat sounds. They aren’t as crisp. This is a known issue with the dual ADC that was only recently patched by Sound Devices. I have not seen any sort of fix from Zoom yet.

Hi! I am just learning about these mobile recorders and wonder how you would attempt recording a band, say drums first, then bass, etc.. Do you hear playback when overdubbing? What about different takes? ... Probably trial newbie questions. Your videos are 👍

Thx for the explanation. I mainly use the SD Mixpre6ii for voice. The levels are set in pre with the limiters activated in 24/48kHz. Saved a lot of processing time in post and a lot of HD-space. 32-bits came in handy when the sound of thunder and rain was needed and when I was filming on a construction site. Those situations were unique and it was handy to be better save than sorry. At that moment a time saver for shure.

Great video, I shoot flying events, or just hang out at a local airport where anything from large Boeings to a Piper Cub can be out and about. The 32 bot float on a Mixpre-6 Mk.2 is fantastic!

Great video. Once again, as with all processes in recording audio...it's "Horses for Courses".

I'd like to know more about your dual F6 12 channel recorder setup and what you think about the results.

I’ve always worked in 24 bit and I’ve never had a problem when I’m in the studio alone. BUT I often record people that have no to little experience speaking or singing into a microphone and I’ve had to do many takes over because someone thought when they suddenly scream they have to get closer to the mic to accentuate that -- and whoops ruined… I assume I should try working in 32 bit (considering to buy the SSL 12) to enable a 32 bit workflow…

If you export your 24 bit 32 bit float recording example to Soundcloud in 24 bit wav format i think we probably will hear a difference. But You tube audio quality MP4 is often about 192 kbps MP3. What do you upload?

Your killing me with that groove. You have demonstrated what everyone always says in these videos; what really matters is the performance' lol Great video man thanks!

I have Behringer XR-18 in the details I read it says 40-Bit floating-point DSP features “unlimited” dynamic range with no internal overload and near-zero overall latency , is that correct if anyone knows! Thanks

With the zoom, which I believe does not have a gain control, how do you deal with the noise floor of the preamps themselves? I would think that on a much louder source, turning down the input gain would net you less noise induced by the analog part of the preamps themselves into the 32bit digital recording. Even with two separate preamp circuits, that is two different gain levels, and a compromise on the analog side, I would think? Especially if you are using a microphone like an RE-27 that requires a lot of gain to get a usable level out of it, about 60db of gain on an 1173 with a singer right on the grill, but less if you are using it on a floor tom with a heavy handed drummer, or a kick drum, etc.

Can you explain the normalization advantage? In traditional DAWs you just click the normalize button and that’s it so I’m wondering how it gets easier. I understand how it gives you more flexibility, but not how it makes it easier to normalize or faster.

When I record my acoustic guitar with the Zoom H4 essential (about 12" from the neck/body joint) and bring the Zoom file into Reaper, the signal level is so low I have to boost that channel to its max (+24db) to get peaks anywhere near 0db. Is this normal, Am I missing something?

what do you mean by audio normalizing? Thx

Great video. I'm going to say the problem with 32-bit is that we lack a device that can handle say - 12 channels. Presonus new interface has 8 channels at 32, which is great - but then I have a bottle neck with all my outboard ADAT gear. I felt if Presonus made two interfaces that you could mate for a total of 16 inputs at 32 bit float.

Why doesn't SoundDevices more expensive 8 series recorders offer 32 bit float as an option?

I am confused, just casually saying to normalize the audio like its nothing. Isn't normalization similar to compression? Its bringing up the quiet parts and bringing down peaks? Thats going to change the sound? If it didnt change the sound then it would just be gain and not called normalization.

The use case I can imagine for 32bit is for when you need to record something that you don't have a chance to try out first to set your levels. If you're recording a jet flying by or a building blowing up, or a space shuttle launching, that might not be repeatable and you won't have a chance to check your levels.

As a 16 year location Sound Dept. veteran (and hack home studio enthusiast), I can tell you that it is definitely the under paid post department that is to blame for the lack of utilizing 32-bit float. I get blow back for recommending the post department download the Sound Devices "Wave Agent" software, so that they can edit/recall meta-data or edit the mono/poly wave data...let alone, asking them to get into the weeds with editing 32-bit float. Another issue with 32-bit float is "what else is in the signal path?". Your lavaliere's element (or diaphragm) can distort (unless you use the new DPA's) or your transmitter can over-modulate (maybe not if you have the latest A20s by Sound Devices). You'd have to REALLY plan out your signal chain to fully utilize 32-bit float....and at the end of the day, you're not getting paid any more money to rent this kit out...nor is there any guarantee that post will use it. It's not like the camera world, where RAW recording and LUTS were accepted with open arms. Sound Department is literally One-Half of the product, but is viewed as a minor annoyance. It is going to be a LONG time before 32-bit float catches on. In the studio world, there still are no multi-channel audio interfaces with 32-bit float. Steinberg has "32-bit converters" on their interfaces, but they are not full on 32-bit float. It's like "we have a cure for cancer!" but the world responds with "meh....". Frustrating for sure.

WHAT? You have an ADC. In the analog case you have a noise floor like the guy said. If you aren’t concert with the volume and play low you get noise. When you come over to the digital side its better with 32 bit float, that is saving the audio on on the SD card. But on the analog side gain is always the case! If you have a auto normaliser function on the zoom, good for you. Still volume is important.

Its crazy that there is any resistance to 32 bit. The benefits are so obvious

Equipment shut have opition to listen to it any formate 16 to 32 bit

I found your post about the 32-bit recording recently so, sorry for my delayed comment. You made quite good content. In short, I’ve 5 points in my mind. (1) Firstly, it s speculative to say that the equalisation and normalisation are so fast and take "10 seconds" or so. For drums recording, obviously yes, but when you're recording environmental sounds, for example, you cannot avoid extremely high peaks randomly recorded. Then the post-production takes significantly more than 10 sec and the final quality would be questionable. (2) You said nothing about the file size which is at least 30% larger compared to the other encodings because of the high amount of data. Actually this is a significant consideration, especially when you're creating sound library - on every 3 Gb you are loosing 1Gb compared to 24-bit (consider a library or project of 1TB or more with all draft recordings and intermediate sessions). (3) The quality of the sound practically remains the same, so 32-bit recording doesn't bring any benefit here. (4) You said nothing about the wave form and sound pressure coming from your input devices and how all imperfections are recorded in 32-bit, and how devices with dynamic range much lower then the recording capacity are performing and recorded. (5) This point is a bit provocative, but anyway, why the clipping is always a bad thing? There workflows, for example, where the clipping is used as markers... 32-bit recording has its benefits as 24-bit and 16-bit as well... Doesn't matter what technology you're using - the answer is not black-and-white for sure. May be the key dilemma is „think before“ or „think after“ your recordings, or „do not think at all“. Like in the real life :)

i think 32bit float is great. the only concern i have is what some people pointed out; pushing work to post can disrupt existing workflows. i work in radio broadcast as an audio engineer, and the recent introduction of izotope RX and supertone clear made for some pretty grueling experiences where reporters asked me to fix their shitty reverbant (and sometimes clipped) field recordings, because they know we can do that, to an extend. if we had 32bit float some issues we have now wouldn't be there, but i feel like reporters would care even less about standards that make sense beyond the old tape noise floor logic. having the tools to fix things in post is nice but doing it right from the start is often very easy, especially if the workflows for it are long established. if i end up freelancing as a videographer again, i would probably consider using 32bit float, because i'm on my own and i save time by fixing it in post. at my job, there is no time to fix it in post because it's going on air in 5 minutes. different workflow :D

32 does sound a bit more dynamic (open)

The question of whether it has anything to do with Led Zeppelin depends on how you spell it... Fabric? or Geographical Region? :)

It is different

🙂👍

A good comparison would be a camera's focus (or f-stop/ISO/shutter for that matter). Imagine shooting a 30 minute interview with the camera out of focus. Imagine being able to pull that focus in post. That's what 32 bit float can theoretically offer. Most professionals manually pull focus in the same way audio engineers gain stage, There's a specific person that solely operates and pulls focus and they get paid LOTS. Most resistance to utilizing auto focus was purely due to it's poor and inaccurate implementation, usually resulting in undesired, adverse, effects. People need to experience 32 bit float in order to trust it, in the same way you would need to experience and play around with good quality, modern, auto focus. Auto focus works well for people who don't have the in-depth knowledge or patience to dial in exactly what they want, 32 bit float satisfies that exact notion. Someone who's been working in audio for ages would have zero need or use for 32 bit float because they know exactly how to capture what they want, the same exact way a photographer uses manual mode because they've learned exactly how and why to configure the settings to accommodate the desired outcome. Technology like 32 bit float audio recording is such a massive step, that it's hard to even justify the amount of freedom it could restore to an artist. We've grown so accustomed to only being able to capture a hollow representation of audio and that it's been proven time and time again that any step further gives us diminishing returns. So despite the heavy resistance in standard use cases, 32 bit float can and should thrive within it's own, separate, classification of audio capture. The same way we have stereoscopy, or 360 action cam videos. At the end of the day it's just simply a different way of representing captured information, and therein lies the root of the problem; since it's a different way of representing audio, the thought process revolving around it should also be reconsidered.

Why won’t it catch on? Does it sound worse? No Does it sound better? No Will any of us ever record anything with more than a 90db range? Probably not. Save time by not setting gain? Combine your process of normalizing with 40 tracks, many with multiple takes, over a 12 song record, and I bet getting it right at the front end is quicker. Still, how am I supposed to send a decent headphone mix and monitor what I’m recording without proper gain ? You almost get one of the most important parts about the actual sound at the end. Using an API (or any analog pre) will require you to adjust the gain, and how you adjust that gain has a significant impact on what you get out of that pre. Studios aren’t going to give all their preferred pre amps even if the above mentioned problems weren’t a thing. This set up is nice for throw and go, but it’s kinda solving a non existent problem.

Files are massive.

It all boils down to: digital gain vs analog gain.

Auto Align 1 is great on drum tracks. Auto Align 2 is even better. kzfaq.info/get/bejne/gZl0gqaF3cXVg2w.html Of course wav 16, 24, 32 float or 64 float bit ...it is an upgrade but these things like Auto Align give a lot of depth and sound to the drums that is far beyond the improvement going from 24 to 32 bit float. Phase and alignment is important/essential for drums and for a mix.

The brave thing would be to make recording devices with no gain control whatsoever. The gain control doesn't set the "zero point" but the "one point": you're deciding how much dynamic range is between the number zero, and some other number that is not zero. Also, your introduction of the term "dithering" here is inappropriate, it has nothing to do with the range selection/combination process. Also, having more than one ADC is not strictly necessary, and for all intents and purposes 24-bit integer PCM is also fine for gainless line level ADCs... and in practice probably also fine for gainless mic preamp-ADCs, but there are extreme outlier microphones for which this is not the case.

24 bit integer precision makes zero sense for listening to music. 24 bits are only needed for writing music because you can record your signals low and then boost them without boosting the noise floor caused by imprecision errors. You need to record your source at a very low volume, like barely hearable, much lower than whispering, for 24 bits to start having issues - but why would you even do that? 24 bit precision is 256 times more precise than the default 16 bits and a finished track will be indistinguishable between two formats. 32 bit float precision is also quite questionable because floating point numbers can be very imprecise and result in errors. To me the format always seemed like an internal way for very old DAWs to work around the hard limitation of very old audio interfaces that could only do 16 bits at a cost of extra CPU power.

320 bits 32bit

DAWs use 32 bit internally anyway.

The best analog stage is barely 20 bits (122 dB) signal to noise ratio, the claim of 1528 dB is complete BS. 32 bit linear is theoretically 194.4 dB.

2024 shut have all type up to the people if won't to listen to it