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Gear / Technical Help => Recording Gear => Topic started by: dallman on October 01, 2019, 11:37:46 AM
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This thread is here for anyone interested or already recording in 32bit Float to share gtheir experiences thoughts, ideas, techniques, questions so we may all benefit and learn through experiences abouit 32bit Float recording:
Please resist the urge to hijack, there are plenty of threads for those pro, con or unhappy.
To keep it simple, I'll start over:
I thought with a few new decks having 32bit float recording and more coming, it might be nice to have a thread dedicated to recording with this process. There are plenty of threads debating the merits and features of the decks but many of us are interested in focusing on how these decks record in 32bit float and what the results are, so I thought this might be useful.
I used my MixPre6 II on Saturday to record Gov't Mule. I ran hotter than I ever had, and I did lower the levels about 2db after the first song which was in the red the entire time. My goal was to run hot, but not super in the red at all times hot. I ran 4 channels, Schoeps MK41's Hypercardioid, and Superlux S502 Cardioid. Here are my thoughts:
I started recording in 1971. It was really hard, really hard to watch levels run regularly in the red after 48 years of avoiding the red. I did it, but I figured how hot to run will be a gradual process.
The results though were excellent! There was nothing even close to distortion. I opened in Soundforge which is what I like to use nest. I transferred the poly file to a stereo file for the Schoeps and a stereo file for the Superlux. On both files I lowered the gain on the first song 2 db and then the entire file another 2 db. The files looked perfect and sounded great. I see obvious advantages, especially when the levels at the start of the show are much hotter than expected.
The learning curve might just be does anything matter? Is any gain too much? Do I care that I have to lower the gain in post if it sounds great? Will it at any point with high hot gain sound compressed? These seem to be the details that we will all be looking at, and there probably will be a few differing opinions just like in all of our recording techniques. However, there is much more freedom. After 48 years I do not make too many mistakes and I know my mics and most local rooms, but...this was really cool, I totally look forward to using the deck more. Tonight, I will record 6 channels at a DSO show with different mics and while I do not plan to run superhot, will push my levels into the red with confidence.
To add: The DSO show sounds great. Not necessarily different than if I recorded at 24bit, but the room and volume of the show was louder than usual for the venue and that did not matter to me, I didn't need to worry. Also in post I need not worry about levels being too low. Personally I do not like to normalize as it makes my ears get tired, so sometimes if needed I play with the levels of each song especially if I recorded conservatively and the process is time consuming. I no longer have that problem, or maybe I'll still be adjusting in the future but that might be more about the difference between say an acoustic or very quiet song versus the loudest of the sets songs. I'll try to post the visual from Soundforge when I have an opportunity.
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Only 2 shows so far, but I ran my levels conservative and boosted in post with no issues, which is something I did not like to do.
In 24 bit, I ran hot and relied on my limiters to capture what I felt was a more accurate dynamic range.
I liked the original MixPre series, but thought the limiters engaged too early. It was nice to run without limiters.
The new series does allow more adjustment to the limiters in 24 bit, but I really like not having to use them.
I feel my 2 recordings have a nice sound to them after boosting, and I think i will continue to run that way on my next shows.
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Only 2 shows so far, but I ran my levels conservative and boosted in post with no issues, which is something I did not like to do.
In 24 bit, I ran hot and relied on my limiters to capture what I felt was a more accurate dynamic range.
I liked the original MixPre series, but thought the limiters engaged too early. It was nice to run without limiters.
The new series does allow more adjustment to the limiters in 24 bit, but I really like not having to use them.
I feel my 2 recordings have a nice sound to them after boosting, and I think i will continue to run that way on my next shows.
It would be interesting to run 2 decks same mics one hot, one conservative and then boost or lower to the same levels and compare. According to what is being preached, they would be as identical as you'd get in that type of real world scenario, but that is the sort of thing which I hope comes to light as more samples and styles get discussed. I always ran the reverse of you, conservative with the boost of gain in post. Now I'm going hot and bringing it down (or I assume I'll bring it down as I'm going to try and track them tonight, but even with the files looking huge and foreboding, they sound great, no distortion or compression at all, nice dynamic range.
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Here is a picture of the wave as recorded Neumann SKM 185 Hypercardioid to MixPre6II Set 1 Dark Star Orchestra First Ave Mpls MN Sept 30 2019. The second shot is after lowering 4.5db. I hope to track and upload this and Gov't Mule from Sept 28 tonight or tomorrow.
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It would be interesting to see/hear your results if you purposefully set your levels way too low and then have to boost them in post.
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It would be cool to hear some samples from people who own 32-bit float recorders.
I am especially interested in hearing things with quick changes in dynamic extremes, to hear how seamless the hand-off between ADCs is (or isn't).
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got my second mixpre 6 II (1st was defective) and tested it out just recording the stereo. for parts I cranked the gain all the way hitting +15 or so over and some I had it at -30 or so. as expected the +15 was heavily distorted and the -30 was barely audible. Both sounded perfect when getting them to about -1 or so.
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It would be interesting to see/hear your results if you purposefully set your levels way too low and then have to boost them in post.
ive done that a lot with mics right into ad2k peaking below -30
sounds the same for our application, really. no audible noise
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Here are two recordings (Gov't Mule and Dark Star Orchestra) that started life as 32Bit Float/ 48kHz recordings:
http://bt.etree.org/details.php?torrentId=607192 (http://bt.etree.org/details.php?torrentId=607192)
https://archive.org/details/dso2019-09-30.from32bitfloat_Neumann (https://archive.org/details/dso2019-09-30.from32bitfloat_Neumann)
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Here's a short sample of a marching band show I recorded yesterday. 32-bit float recording has made this SO much easier to no longer worry about levels.
This is the original 32-bit float / 48 kHz WAV. Nothing has been done to it except a level adjustment of about -10 dB in RX.
https://we.tl/t-vGQHM3iOET (https://we.tl/t-vGQHM3iOET) (expires in 7 days)
Details:
High school football stadium
Rig at the very top of bleachers at 50
Pit instruments (mallets, synths) are amplified through portable PA stacks on field, near each 25 yd line. Everything else is acoustic.
Chain:
Line Audio CM3 in SRS Hybrid Mount (110 deg; 30 cm)
Movo WST50 windscreens
Zoom F6 at 32-bit float / 48 kHz
EDIT: Fixed upload.
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Something wrong. Link already expired
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Something wrong. Link already expired
Just uploaded again with another service.
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Heh, well that's peaky stuff! Here I had to add considerable amounts of HF boost for it to sound "right" but that's probably down to my 70 year old seriously shot ears. BTW it played just fine in "VU Player" - I wasn't sure whether the 32 bits might cause a problem. (And right now I'm updating from version 2.49 to version 4.8!)
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My guess shared file was unprocessed 32 bit source, it needs to be adjusted and compressed. Same goes to 24 bit recordings captured with low levels to avoid clipping.
Here is my very basic attempt to adjust file in Reaper. Source file gain was boosted +20, and Limiter was applied to Master Track. As a result - loud, but same time distortion free output.
https://i.imgur.com/jorv9zz.jpg
https://www.dropbox.com/s/3nxd6g3k4xij5qp/191026_002excerpt-processed.mp3?dl=0
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My guess shared file was unprocessed 32 bit source, it needs to be adjusted and compressed. Same goes to 24 bit recordings captured with low levels to avoid clipping.
Here is my very basic attempt to adjust file in Reaper. Source file gain was boosted +20, and Limiter was applied to Master Track. As a result - loud, but same time distortion free output.
https://i.imgur.com/jorv9zz.jpg
https://www.dropbox.com/s/3nxd6g3k4xij5qp/191026_002excerpt-processed.mp3?dl=0
Surely you mean +2, and not +20 dB? The peak value of the excerpt I posted is between -3 and -2 (don't remember exactly, and not at that PC).
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Heh, well that's peaky stuff! Here I had to add considerable amounts of HF boost for it to sound "right" but that's probably down to my 70 year old seriously shot ears. BTW it played just fine in "VU Player" - I wasn't sure whether the 32 bits might cause a problem. (And right now I'm updating from version 2.49 to version 4.8!)
The highs come through pretty strongly for me on my HD650s, particularly the snares and the vibes. It does sound like the band is a considerable distance away, because they are. But my setup was in the place you are supposed to listen to these shows. They are trained to play to the press box, which I was right next to.
In the past, I have used my little Naiant omnis for these shows which have a treble bump - that compensates for the HF loss at distance. I suppose I could use my DPA 4061s which would be even better with their short boost grids, but I never want to risk losing them as I'm racing to set up on the stands. I also would need to get much better windscreens for them - sometimes there is serious wind at these shows, particularly up high.
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I add track gain (volume) +20 db.
Screenshot shows slightly different setup. I normalize item and add +18db gain to track. And use Limiter on final master track. Sort of extreme test, but you can do the same with 24 bit source if it was not clipped during recording.
32 bit allow to record accidental overloaded sounds, and bring clipped data back in post. Some Sound Devices tests explain that 32 bit also less noisy when you try bring back ultra quiet data, but i am afraid no one will hear differnce in real life, because that difference located lower than microphone self noise and ambient noise. In most real life situations if you adjust preamps record track without clipping 32 vs 24 will be the same.
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My guess shared file was unprocessed 32 bit source, it needs to be adjusted and compressed. Same goes to 24 bit recordings captured with low levels to avoid clipping.
Here is my very basic attempt to adjust file in Reaper. Source file gain was boosted +20, and Limiter was applied to Master Track. As a result - loud, but same time distortion free output.
https://i.imgur.com/jorv9zz.jpg
https://www.dropbox.com/s/3nxd6g3k4xij5qp/191026_002excerpt-processed.mp3?dl=0
Surely you mean +2, and not +20 dB? The peak value of the excerpt I posted is between -3 and -2 (don't remember exactly, and not at that PC).
yea left channel is -2.99 peak and right is -2.04
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32 bit float will not change any of the old arguments regarding limiting. Some take the view that any change in the overall dynamic range of an acoustic (usually classical) recording is heresy, and those that take that view will be delighted by the very lifelike dynamic range these new recorders will offer. Others take a more pragmatic view, that most people's listening spaces have noise levels such that low level signals will disappear in the room, or that high level signals will cause the neighbours to complain.
My own concern in days gone by was that I often came home with recordings which might have one or two peaks which were considerably higher than all the rest of the recorded material - sometimes more by accident than design. If the recording was circulated uncompressed, it would sound pretty low level almost all the way through. But if I did use a limiter, it would be a very fast one which only affected the smallest number of samples possible, and which would be undetectable by the listeners. It would leave 99.99$% of the audio unchanged.
An example of the kind of "accidental" peak I'm thinking of would be where the last chord of a mighty organ work would build up in a cathedral, the level being compounded by reverberations and pressure waves bouncing around the building, although the level originating from the instrument wasn't varying. Or when a timpany player got a bit more enthusiastic than the composer had in mind. That kind of thing.
One thing is clear - these new devices offer a lot of choice over levels in post production and that enables recordists to make their own choices according to their own tastes. Or the clients'. That's good.
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32 bit float will not change any of the old arguments regarding limiting. Some take the view that any change in the overall dynamic range of an acoustic (usually classical) recording is heresy, and those that take that view will be delighted by the very lifelike dynamic range these new recorders will offer. Others take a more pragmatic view, that most people's listening spaces have noise levels such that low level signals will disappear in the room, or that high level signals will cause the neighbours to complain.
My own concern in days gone by was that I often came home with recordings which might have one or two peaks which were considerably higher than all the rest of the recorded material - sometimes more by accident than design. If the recording was circulated uncompressed, it would sound pretty low level almost all the way through. But if I did use a limiter, it would be a very fast one which only affected the smallest number of samples possible, and which would be undetectable by the listeners. It would leave 99.99$% of the audio unchanged.
An example of the kind of "accidental" peak I'm thinking of would be where the last chord of a mighty organ work would build up in a cathedral, the level being compounded by reverberations and pressure waves bouncing around the building, although the level originating from the instrument wasn't varying. Or when a timpany player got a bit more enthusiastic than the composer had in mind. That kind of thing.
One thing is clear - these new devices offer a lot of choice over levels in post production and that enables recordists to make their own choices according to their own tastes. Or the clients'. That's good.
I don't have a problem with limiters in classical / acoustic recording, so long as they don't draw attention to themselves, and affects the bare minimum amount of the recording, exactly as you describe.
I have never been successful with this in my personal experience though, through a combination of my lack of skill and the limitations of the limiters in recording devices I have owned.
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Oh, I wouldn't dream of limiting at the recording stage - and of course with 32 bit float you don't need to. I'm referring more to post-production - and to avoid putting a load of stuff here that is marginal to the topic, here's a link to something I posted way back which illustrates my personal approach and method (and the now unavailable VST I used to use).
https://www.gearslutz.com/board/showpost.php?p=5291625&postcount=32
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Oh, I wouldn't dream of limiting at the recording stage - and of course with 32 bit float you don't need to. I'm referring more to post-production - and to avoid putting a load of stuff here that is marginal to the topic, here's a link to something I posted way back which illustrates my personal approach and method (and the now unavailable VST I used to use).
https://www.gearslutz.com/board/showpost.php?p=5291625&postcount=32
I have often used limiting in post with classical recording; only when necessary of course.
Your procedure for testing VST limiters is very clever.
The "Soft Limiter" in Audacity is actually quite decent and simple to use. I have often applied it to random percussive peaks that are far outside the RMS level of the concert, so that I can normalize several dB higher.
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Observation: Zoom F6 vs SD MixPre6 II
Zoom claims that you cannot set gain on 32Bit Float, Sound Devices claims you can set gain on their MixPre6 (and other 32Bit Float models).
I am not sure if I can explain this and I am not claiming this is scientific, but my observation is that in reality they both do exactly the same thing in this regard. (I am not talking about sound, preamps or anything in that regard, just the use of gain for the MixPre6 II and Faders on the Zoom F6 on the wave form. Here is what I have observed:
On my MixPre6 II, I can set gain (pre-recorded level of sound), and there are a few ways to do this. The bottom line though is that it "feels" like there is a set sized file imprinted on my SD card that I can enlarge or shrink in my post opened file, and it stays proportionally the same. That in essence means that the ratio of loudest to softest sound is what is being set by the MixPre, and I am enlarging or reducing that proportionally. If too big, it will distort, but then I can just reduce and it will be fine. How high or low I set my gain seems to have no (or little) effect on the final file. (we'll learn more as the samples become more prevalent, but this is what I observe)
On the Zoom F6, their claim is that Gain is preset or predetermined and that you have fader (post recorded level of sound) control. Like the SD MixPre6 above, the file seems to have a predetermined ratio of loudest to softest sound, and the saved file on the SD card is determined by how loud or soft you set the faders. So gain is pre-set according to Zoom, but if I have my faders really low, I see a very small file when I look at it in post (I use SoundForge), and if I set my Faders high, I end up with a large (loud) file that gets printed on the SD card and observed when I open it in SoundForge. Again the ratio appears set. If it is too hot and distorted, I lower it and the wave file is fine and if it is too low, I add gain and the file proportionally gets louder.
So if I set gain on the MixPre loud and in the red, I get a huge file that may be distorted, but reducing gain (in post) brings me a wave perfect file. If I record too low, increasing gain (again in post), gives me a nice noise free clean file.
So if I set the faders on the Zoom F6 loud and in the red, I get a huge file that may be distorted, but reducing gain (in post) brings me a wave perfect file. If I record with my faders too low, increasing the gain (again in post), gives me a nice noise free clean file.
I hope this is clear, in essence they do the exact same thing. Now the ratio's between loudest and softest could be different (that is a different set of tests I suppose) but my point is that with these two 32Bit Float decks, setting the gain on the MixPre6 II or the faders on the Zoom F6 seems to do the same thing and yield the same results.
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Intersting, thanks for the report.
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When recording 32bit, I can set Gain using the menu controls on the MixPre-6 II. I set it so that my level displays show me ‘normal’ looking audio volumes to monitor that I’m getting audio. The knobs are Faders and don’t need to be used.
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When recording 32bit, I can set Gain using the menu controls on the MixPre-6 II. I set it so that my level displays show me ‘normal’ looking audio volumes to monitor that I’m getting audio. The knobs are Faders and don’t need to be used.
You can set the knobs on the MixPre6, 3 or 10 II or older original MixPre 3,6 and 10 version so that the front knobs control gain. That is how I prefer it, rather than go into the menu. This has been well documented, custom setting, advanced setting for all else except gain set as basic setting. While it may be less important at 32bit, the more I use 3 2bit the more I seem to retain the habits I have always used, but with far less worry particularly at the beginning of a live set for an audience recording.
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Thanks for the tip!
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Some thoughts:
I done one more quick test with 32 bit sample files from SD website https://www.sounddevices.com/sample-32-bit-float-and-24-bit-fixed-wav-files/
1. I lower item volume to -150db (lowest possible volume in Reaper app)
2. I render one file as 24 bit and another as 32 bit.
3. I open both rendered files in Reaper and Normalize them back.
After Normalize rendered 32 bit file looked 100% like original
After Normalize rendered 24 bit file was crazy distorted.
Next i repeat test with source item volume lowered to -130db and -100db, but even so 24 bit file show very visible distortion after normalize.
Next i repeat test again and again and only with source item volume lowered somewhere to -60 -70db (at these levels waveform start to became rather visible), rendered 24 bit file starts looks more less like original after normalize. But i can still notice some tiny digital noise pattern artifacts if look very close on spectre image. Where is that truly safe limit in 24 bit file? It is fixed limit or smooth undefined limit? I don't know...
I see that 32 Bit Float allow to create really huge volume adjustments not only during recording but also during any other workflow steps and allow re-save truly lossless files 100% without any digital limits at all possible volume levels. So it seems like perfect archiving and editing format.
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For apps like DaVinci Resolve that can read 32 bit files but don't fully support 32 bit processing yet, the most universal compromise way is to normalize 32 bit files in Reaper, Soundforge, Izotope RX, and export with preserved Timecode in same 32 bit float depth.
In this case you can work with file in Resolve and same time you keep your actual 32 bit source data formally untouched, so it may be useful for other apps and processing plugins that operate in 32 bit float.
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You may notice in older Curtis Judd's videos he discover that the normalized sound character is slightly different at different gain amplification levels.
My guess it is not because A/D converters, but because analogue circuits inside microphone body itself.
In 32 bit at any gain level we always capture all possible dynamic range of microphone and so we can always digitally normalize it without any loss. So when we provide more gain to microphone we actually provide more voltage to its circuit and so internal electronic components inside microphone reacts somehow to this voltage and adds some specific character to the sound.
When we provide less gain/less voltage to its circuit, the internal electronic components inside microphone reacts somehow in different way to this voltage and adds some other character to the sound (less bass, less warm in that video example)
So probably with 32 bit recorder we can shape the sound of microphone by experimenting with different gain settings and find which gain will produce better sound on specific microphone model for specific needs.
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Some thoughts:
I done one more quick test with 32 bit sample files from SD website https://www.sounddevices.com/sample-32-bit-float-and-24-bit-fixed-wav-files/
1. I lower item volume to -150db (lowest possible volume in Reaper app)
2. I render one file as 24 bit and another as 32 bit.
3. I open both rendered files in Reaper and Normalize them back.
i would expect that as -150dB is below the resolution of 24-bit. same as if you rendered a file normalized to -100dB to 16 bit
Next i repeat test again and again and only with source item volume lowered somewhere to -60 -70db (at these levels waveform start to became rather visible), rendered 24 bit file starts looks more less like original after normalize. But i can still notice some tiny digital noise pattern artifacts if look very close on spectre image. Where is that truly safe limit in 24 bit file? It is fixed limit or smooth undefined limit? I don't know...
depends on the noise floor of your source recording. most mics have a self noise around 20 dB. in a studio setting its hard to get more than 80 dB of dynamic range. in a live concert its hard to get close to 50 dB over the crowd noise
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There’s nothing like experience to judge that as the ambient noise and the mic noise will vary. I’d also appreciate some guidance on this from those who are recording indoors but not in a studio setting.
But if you don’t have the experience, this might just help. The new MixAssist plugin for MixPre:
https://store.sounddevices.com/product/mixassist-plugin/
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Would this - very small and inexpensive - box work for recording loud music then? If I understand the logic of 32 bit float, surely it would??
https://www.newsshooter.com/2019/09/14/tentacle-sync%E2%80%8B-introduces-track-e-32-bit-floating-point-audio-recorder/
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yes but its mono
it is discussed here
http://taperssection.com/index.php?topic=191874.0
any recorder can record "loud music", provided you select a microphone with the proper sensitivity.
this is why people prefer DPA 4061s over the 4060s. Both can handle similar high SPL, but the 4061 has an output thats more appropriate for our application
as for the "logic of 32 bit float", you dont need 32 bits (or even 24-bit for that matter) to make a great recording... in fact there are tons of great recordings on 16bit digital and many more from analog gear that offered equivalent 70-80 dB dynamic range that 12-14 bits would allow.
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yes but its mono
it is discussed here
http://taperssection.com/index.php?topic=191874.0
any recorder can record "loud music", provided you select a microphone with the proper sensitivity.
this is why people prefer DPA 4061s over the 4060s. Both can handle similar high SPL, but the 4061 has an output thats more appropriate for our application
as for the "logic of 32 bit float", you dont need 32 bits (or even 24-bit for that matter) to make a great recording... in fact there are tons of great recordings on 16bit digital and many more from analog gear that offered equivalent 70-80 dB dynamic range that 12-14 bits would allow.
We’ll no, not all recorders that don’t have technology like this can record loud music. If I plug my mics into a voice recorder it won’t be able to handle it.
I know you don’t need 32 bit float to make a good recording. Funny enough I’ve made plenty of good recordings without it. That’s not the point of the discussion.
The deal breaker for me would be Mono, but I’d imagine it’s only a matter of time until the 32 Bit Float option is available via a very, very small handhand recorder like the Roland.
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We’ll no, not all recorders that don’t have technology like this can record loud music. If I plug my mics into a voice recorder it won’t be able to handle it.
nor can this, necessarily. They havent released any specs on max input level yet. i wouldnt expect it to be much out of range of the normal handheld, say 0 to +4 dBV or so, which very hot mics or a SBD could still overload
I know you don’t need 32 bit float to make a good recording. Funny enough I’ve made plenty of good recordings without it. That’s not the point of the discussion.
it kind of is, as it seems a lot of people misconstrue 32bit as being able to get around the practical limitations of the analog stages that precede it.
This other thread with a technical discussion of 32-bit float is worth a look, the input level limitations are discussed extensively
https://taperssection.com/index.php?topic=192100.0