Topic: Recording in 24-Bit vs. 16-Bit

[Dan33185]

This is probably a really dumb question, but I have been having issues with clipping and distortion with a few of my recordings recently, which I record at 16-Bit. I am wondering if recording at 24-Bit would reduce the risk of clipping and distortion, or would it not make a difference?

[rocksuitcase]

need more details:
1] internals or external mics?
2] You using a battery box or pre-amp if external?
3] From your profile sig, if you are using the Zoom by itself with internals, that is the clipping in the pre-amps, the bit rate will not change that. If you are using it with external mics, then the clipping could also be about sensitivity of said mics. but again, not bit rate related.

[Dan33185]

need more details:
1] internals or external mics?  (Internals with the Zoom ; XLR > SBD with the Tascam)
2] You using a battery box or pre-amp if external? (See above)
3] From your profile sig, if you are using the Zoom by itself with internals, that is the clipping in the pre-amps, the bit rate will not change that. If you are using it with external mics, then the clipping could also be about sensitivity of said mics. but again, not bit rate related.  (That's what I was afraid of, but thought it was worth asking...thanks!)

[Gutbucket]

As rocksuitcase mentions, choosing to record at a different bit-rate will not fix your problem.  It can be difficult to determine where overload distortion is occurring.  However in your case, since you are using two fully integrated stand-alone recorders, it can only be occurring either before the signal reaches your recorders, or inside of them.

Sometimes the sound through the PA (and potentially through the SBD) is distorted before it reaches your recording gear, yet due to hearing perception and loudness masking it might not be obvious during the performance, and only becomes more obviously apparent upon listening to the resulting recording. In such a situation you may be making a faithful high-fidelity recording of distortions that have already occurred, over which you have no control.  That might not be what's going on in this case, but is one possibility to be aware of.

Lets assume instead the signal reaching your recording rigs (through the air and/or through a SBD patch) was not overly distorted before it got to them.  In that case, the distortion must be occurring somewhere in your signal chains. And in the setups you describe, the entire signal chain other than the XLR cables from the SBD is located inside of the recorders.  There are several separate stages inside the recorders in which distortion is mostly likely to occur: The analog to digital converter stage, the analog input stage, and (on the Zoom at least) the internal microphones.

First thing to check is sufficient powering, the lack of which is a common cause of distortion. Are the batteries fresh and providing sufficient voltage and current?  If they are, you'll need to reduce signal level through whichever stage inside the recorder is distorting. Do so by setting input sensitivity lower (confirm low microphone sensitivity is selected instead of high, switch to line-in rather than mic-in if possible), and make sure your recording levels are set sufficiently low.  If not possible to reduce levels further via the settings of the recorders, you might use signal attenuators in the XLR connection between the SBD and Tascam. With the Zoom using its own internal microphones, it might be sufficient to move the recording position farther away to where the SPL is lower.  Yeah, the recording is likely to suffer from doing that, but is better than unacceptable distortion. Alternately you might switch to using external microphones capable of handling the SPL, the sensitivity of which are low enough so as to not overload the input stage of the Zoom. 

If none of that works or is doable, you'll need to switch your current recorder(s) out for something else capable of handling higher signal levels without distorting.

[rocksuitcase]

I like also what GB is telling you. THIS, in specific may be the case, especially SBD output distortion:

Sometimes the sound through the PA (and potentially through the SBD) is distorted before it reaches your recording gear, yet due to hearing perception and loudness masking it might not be obvious during the performance, and only becomes more obviously apparent upon listening to the resulting recording. In such a situation you may be making a faithful high-fidelity recording of distortions that have already occurred, over which you have no control.  That might not be what's going on in this case, but is one possibility to be aware of

.

[Gutbucket]

And to answer your actual question, rather than helping trouble shoot the cause of distortion, here is why recording in 24 bit rather than 16 won't help-

It is true that a 24 bit digital audio file is capable of storing a signal of wider dynamic range than a 16 bit file, so its reasonable to suspect it might help in this case.  But the overload point of a 24 bit file is exactly the same as a 16bit file. Both max out at 0dBFS. What changes is that the noise floor of a 24bit file at the quiet end of things is significantly lower than that of a 16-bit file. The point of clipping at the loud end is the same, but the level of noise (hiss) at the bottom is lower.

To take advantage of that for recording louder things, the signal chain ahead of the digital file being written to memory needs to be capable of handling the higher signal levels involved before overloading (that is the key in this case) while sending a lower signal level to the ADC which converts the analog signal to digital, thus keeping the the digital signal from exceeding 0dBFS. But that much applies to both 16 and 24 bit.  To take advantage of the greater dynamic range of the 24bit digital file format, the signal chain ALSO needs to have a low enough electrical noise-floor that the extra dynamic range provided by the 24bit file format at the quiet end of things doesn't just end up getting filled with a bunch of random noise.

Its sort of a Goldilocks thing - comfortably fitting the dynamic range of the analog signal into the digital file container without exceeding its limits at both the top and bottom.  With 24 bits, the limit a the top is the same as with 16 bits. The limit at the bottom is lower.  That extra range at the bottom may not be useful.  It won't be useful if the electrical noise-floor of the signal chain through the recorder is already higher than the noise-floor of the 16 bit file format, which is not unusual. And it won't really be useful if the acoustic noise-floor of the recording environment is higher than that as well, which is pretty common.

[Dan33185]

Thanks for all the information...I know it's not a power issue, batteries are freshly charged every time I head out to a show and replaced every so often because as they age the charge obviously doesn't hold as long. I'm thinking I might just need to lower the recording level, I was experiencing hiss with some recordings because I had to amplify it so much in post that I was trying to minimize the need to do that by raising the recording level, but I think I over estimated how much I could. I'll play with some levels as I go to shows and hopefully figure out a happy medium. I just figured if a higher bitrate took some of the guess work out of it, that could be helpful, but that won't make a difference as I'm discovering. The plan is to upgrade my recorder and get externals at some point when finances allow, but just trying to make the best of what I have at the moment.

[Gutbucket]

Was adding a bit to the bottom of my previous post when you replied which somewhat addresses that.

In short, you do need a file format that provides sufficient dynamic range in which to fit the music.  But 16bits provides a pretty wide range.  The dynamic range of the music itself is defined by the acoustic noise-floor of the recording environment at the bottom, and the loudest sound being recorded at the top of that range. Its almost always smaller than that of the 16 bit file format. But you do need to fit it in there comfortably, so 24bits makes fitting it in easier while requiring less adjustment of levels, yet only if everything else in the chain is capable of handing that additional range as well.

The actual thresholds of hiss and overload in your recordings are probably not defined by the bit depth of the file format but rather by the circuitry which precedes it.. or the recording environment itself at the quiet end.  The noise-floor at the bottom could be the acoustic noise-floor of the recording environment, which unless dominated by background music, folks mulling around talking and making noise, is most often dominated in otherwise quiet sounding recording environments by acoustic noise that sounds like hiss when amplified.  Sometimes hard to tell what is dominating the noise-floor of the recording - the self-noise of the mics, electrical noise of the gain-stages, or the acoustic noise floor of the room itself.  They can all sound like hiss.

[robgronotte]

To get this out of the way, look at the wave form in any music editing program while listening. If it's only sounding distorted when the levels are peaking at 0, you had the levels set too high.  If you're hearing it other times, look at what the guys said above.

[Dan33185]

To get this out of the way, look at the wave form in any music editing program while listening. If it's only sounding distorted when the levels are peaking at 0, you had the levels set too high.  If you're hearing it other times, look at what the guys said above.

Good tip, I'll check that on the next show I run through Audition.

[rocksuitcase]

To get this out of the way, look at the wave form in any music editing program while listening. If it's only sounding distorted when the levels are peaking at 0, you had the levels set too high.  If you're hearing it other times, look at what the guys said above.

Good tip, I'll check that on the next show I run through Audition.

I was going to add this IS a good step to go through before you decide what is causing the distortion.

[Gutbucket]

Yeah that!

If so all you may need to do is set levels somewhat lower to avoid exceeding 0dBFS, just not so much lower that the quiet parts begin to get swamped in noise.

[DSatz]

Digital audio recording is easiest to understand correctly if you think of the samples as fractions rather than integers. They're binary fractions, but that's a technicality. The number .25 has the same magnitude as the number .250, but when you see the second as opposed to the first, you have _less uncertainty_ about what's going on in the "thousandths' place". That corresponds to a lower "noise floor" in the purely digital part of each recorder channel--though not its analog circuitry or A/D converter (where any audible noise in a recording usually comes from, if not from the venue or the microphones).

24-bit recording offers you no greater capacity to record higher numeric values (representing greater sound pressure levels) than 16-bit; ±1 is still the absolute outer limit that you don't want to crash into. But up to a point, if your signal path is generally sensible, 24-bit recording can let you reduce your record levels enough so that you don't run out of room at the top, and still not pay a price in increased audible noise. Not always--your mikes themselves and/or your outboard preamp, if you use one, can be overloaded before the signals get to the recorder; again, the ceiling isn't being raised but the floor is being lowered, one may reasonably hope.

The amount that lowering is never anything like what the additional 8 bits would imply (48 dB!), though, because no A/D converter or analog input circuitry has that much dynamic range. One well-known recorder that I measured some years ago gave only <2 dB improvement when going from 16 to 24 under the particular conditions I was measuring it. It wasn't a typical scenario, but it does illustrate the real dependency that exists on all the other factors in your actual scenario. If someone had used that recorder with those signal levels and gain settings except that they thought, "With 24 bits I can reduce my levels by (say) 12 dB to be absolutely safe from overload", their 24-bit recording would then have been ~10 dB noisier than the 16-bit one. Again that is based on an atypical scenario in terms of signal levels and gain settings, but not impossible by any means.

Thus you should always use the input to the recorder and the menu and level settings, if any, that are most appropriate for the signal levels you're dealing with, and always try to get your absolute peak sample values somewhere in the top, say, 6 dB or so of the recorder's range without hitting 0. It's just that with 24 bits as opposed to 16 (and with everything else optimal, please note) there should be no audible "noise penalty" if your highest peaks are only at, say, -8 dB rather than -2. So by setting somewhat conservative levels when you don't know what's going to happen (i.e. life), you're better prepared in the event of the unpredictable. Murphy's Law says, after all, that the unpredictable is the most predictable thing there is.

[Gutbucket]

The amount that lowering is never anything like what the additional 8 bits would imply (48 dB!), though, because no A/D converter or analog input circuitry has that much dynamic range. One well-known recorder that I measured some years ago gave only <2 dB improvement when going from 16 to 24 under the particular conditions I was measuring it. It wasn't a typical scenario, but it does illustrate the real dependency that exists on all the other factors in your actual scenario. If someone had used that recorder with those signal levels and gain settings except that they thought, "With 24 bits I can reduce my levels by (say) 12 dB to be absolutely safe from overload", their 24-bit recording would then have been ~10 dB noisier than the 16-bit one. Again that is based on an atypical scenario in terms of signal levels and gain settings, but not impossible by any means.

This makes me think about the 32-bit floating point recorders using multiple A/D converters along with proprietary schemes of switching dynamically between them.  Rather than folks thinking, "With 24 bits I can reduce my levels by (say) 12 dB to be absolutely safe from overload", with 32bit fp the ability to adjust recording level is essentially removed entirely. 

Seems to me these recorders incorporate two separate innovations which needn't necessarily be linked.. yet both are present in every product I've seen.  Ever since these recorders became available it's stuck me that the more fundamental innovation is not the use of the 32-bit floating point file format, but rather the scheme of switching between multiple A/D converters to provide a significantly greater "real world" dynamic range capability.  That same real world improvement should apply regardless if the recording format is 24bit fixed or 32bit fp. A true 24 bits worth of dynamic range already exceeds the specified input range of the recorder.  The Sound Devices MixPre II recorders with this feature claim a dynamic range of 142 dB, slightly less than what can be mathematically represented by a 24bit file format.  The Zoom F6 claims 131dB.

What am I missing here?  Why did manufacturers feel the need add the 32bit fp format when no 32bit fp recorder is capable of capturing a dynamic range greater than what can be handled by a 24bit recording format? 

More specifically, does this mean folks using these 32bit fp recorders may be able to record 24bit files which represent the same real-word dynamic range as a 32bit fp file made on the same recorder?  That assumes the multiple A/D scheme remains in use when recording to the 24bit format as well as the 32bit format, of course.

[aaronji]

Seems to me these recorders incorporate two separate innovations which needn't necessarily be linked.. yet both are present in every product I've seen. 

The original versions of the MixPre (such as the MixPre-6) have multiple ADCs and record a maximum of 24-bit. They even cite their patent in the specifications in the user manual. This is also the case for the higher-end Sound Devices recorders, such as the Scorpio, 888 and 883. This was discussed quite a bit previously; with those recorders, it is generally possible to set the gain such that the entire dynamic range of most microphones, which far exceeds the dynamic range of a performance, can be recorded.

[WiFiJeff]

Recent 32-bit experience, and why I don't intend to go back to 24 bit:

After a nearly 4 year hiatus since the onset of Covid, I went back last week to recording a weekly chamber music series that has just started up again.  I had not used any of my higher-end equipment since early 2020 and so knew I had to do some maintenance.  I replaced lots of batteries and power bricks that had died or lost reliability, and went with my Zoom F6 with triple-redundancy powering (and extras on the side).  My cables and stands were stored in a closet at the venue, so I brought a fresh set of XLR cables and some cable cleaning fluid.  I relearned how to run the F6 (a little nervous not to have the redundant second SDHC card that the F8n has, but with a loose extra card along anyway) and set it for 32-recording alone, no 24 bit parallel so I wouldn't have to worry about levels.  This immediately saved me from a heart attack when I found that the headphones stored at the venue were 1/4" plug but the F6 only has 1/8" output.  Yak!  But, hey, with 32 bit I could actually fly blind, and I had recorded enough piano solo in that hall to know how to setup without soundchecks, though I was not exactly happy about that (I arrived early enough that the lights in the room were not on, and I got through half my setup, from memory, in very very dim ambient  illumination).  I eventually found a 1/4" female to 1/8" male connector (and a second one turned up hours later), so could monitor the session.  I was recording two stereo pairs: two omni mics on a Jecklin Disk over the piano tail and a new set of wide cardioids (DIN, sorta) centered.  I monitored the Jecklin Disk set which was my usual main recording, the wide cardioids (Nevaton 59W-S) were experimental.  I bought them just before Covid and had never used them in the field.  The experiment was interesting, mainly I think because I did a bad thing.  In seeking to make sure a signal was coming from the 59W mics on the F6 meters, I unplugged and replugged the Nevaton mics _without turning off phantom power_.  Naughty naughty.  All four tracks were recorded as individual stems, not stereo linked.  When I got home and looked at the files in Wavelab, the omni tracks were perfect but the 59W had the right track at about the same raw levels as the omnis but the left almost invisible.  Quick loudness check in the wavelab "Analysis" menu showed the left track was almost exactly 20 dB lower (the 59W mics do not have pads, and tests of the mics on a 24 bit Tascam recorder showed the mics were pretty closely matched; the F6 settings for both channels were identical).  So I boosted the left track 20 dB, and ended up with a perfect stereo pair, no sign of any artifacts from system or background noise, which is exactly what 32-bit is supposed to do.  I suspect that fiddling with the connections during setup, with phantom on, caused a transient that "fooled" the F6 A/D to record the left channel lower.  Do the 32-bit A/D converters work in 20 dB jumps? Is this scenario possible or plausible?

Other things I forgot I needed: velcro cable ties, tape scissors for cutting tape to 1) replace velcro cable ties and 2) narrow the Shapeways printed Nevaton 59-S DIN mount so it is actually the right size to fit 59-S mics.

Both recordings came out great, in my opinion.

Jeff

[Gutbucket]

The original versions of the MixPre (such as the MixPre-6) have multiple ADCs and record a maximum of 24-bit. They even cite their patent in the specifications in the user manual. This is also the case for the higher-end Sound Devices recorders, such as the Scorpio, 888 and 883.

Thanks. I was unaware of previous SD products incorporating multiple ADCs limited to 24bit. That just makes me wonder even more about all this.

This was discussed quite a bit previously; with those recorders, it is generally possible to set the gain such that the entire dynamic range of most microphones, which far exceeds the dynamic range of a performance, can be recorded.

Yes, and the question remains..  If the real-world dynamic range capabilities of the recorder is smaller than that of either file format, its the recorder's dynamic range limitation that is the bottle neck, not the file format.  In such a case, what is the need or value of switching to the 32bit fp file format? 

I see claims of post processing, but that's typically done in a 32bit fp workspace regardless of the bitdepth of the source files.

[aaronji]

Thanks. I was unaware of previous SD products incorporating multiple ADCs limited to 24bit. That just makes me wonder even more about all this.

Not just previous. The Scorpio is their current flagship model.

Yes, and the question remains..  If 24bits exceeds the analog dynamic range of which the recorder itself is capable, the recorder's dynamic range limitations that are the bottle neck, not the file format.  And in that case, what is the need or value of switching to the 32bit fp file format?  The real-world dynamic range capabilities of the recorder fits within either file format.

I see claims of post processing, but that's typically done in a 32bit fp workspace regardless of the bitdepth of the source files.

The need/value is you don't have to figure out the range of gain settings that work best with your mics. I think the appeal of "string everything together and press record" reigns supreme for a lot of us, for obvious reasons.

When I got my first gen MixPre-6, I recorded a show where the opening band didn't provide a board feed, but the headliner did. Somehow, though, a low-level signal was coming through. I didn't even notice until I got home. It was peaking at -33 or -35 dB, but there was zero noise when normalized to -1 dB. After that, I played around with it a bit and realized that I could figure out the gain setting such that, given the mic sensitivity and SPL handling specs, the input and the mic would simultaneously overload. Normally, I add a bit of gain to that setting (peak SPL is extremely unlikely at most shows I attend) and turn on the MixPre's excellent analog limiters. To my knowledge, the limiters have never actually kicked in.

[Gutbucket]

That the Scorpio is their flagship recorder targeted at professional uses, and does not include 32bit fp could be a confirmation my suspicion. 

The need/value is you don't have to figure out the range of gain settings that work best with your mics. I think the appeal of "string everything together and press record" reigns supreme for a lot of us, for obvious reasons.

I totally get that. But you should be able to "string everything together and press record" using a 24bit recorder that features the same multiple ranging ADCs in exactly the same way as with a 32bit fp recorder because the actual dynamic range thruput of the 32bit fp recorder itself isn't any larger than the 24bit one.  The increase in dynamic range that was necessary to accommodate that kind of flexibility is being provided by the multiple ADCs. 

When I got my first gen MixPre-6, I recorded a show where the opening band didn't provide a board feed, but the headliner did. Somehow, though, a low-level signal was coming through. I didn't even notice until I got home. It was peaking at -33 or -35 dB, but there was zero noise when normalized to -1 dB. After that, I played around with it a bit and realized that I could figure out the gain setting such that, given the mic sensitivity and SPL handling specs, the input and the mic would simultaneously overload. Normally, I add a bit of gain to that setting (peak SPL is extremely unlikely at most shows I attend) and turn on the MixPre's excellent analog limiters. To my knowledge, the limiters have never actually kicked in.

Good gain staging.  When setup so that the mic and input reach overload simultaneously I wonder if it was the microphones or recorder that dominated the electrical noise floor of the recording chain.. not that you would ever actually encounter that in your recordings

I note that the EIN of all 32bit float recorders seems to be somewhere around -130dBu.  Their max input level varies a bit more.

[Gutbucket]

Recent 32-bit experience, and why I don't intend to go back to 24 bit:

Quick loudness check in the wavelab "Analysis" menu showed the left track was almost exactly 20 dB lower (the 59W mics do not have pads, and tests of the mics on a 24 bit Tascam recorder showed the mics were pretty closely matched; the F6 settings for both channels were identical).  [..snip..]  I suspect that fiddling with the connections during setup, with phantom on, caused a transient that "fooled" the F6 A/D to record the left channel lower.  Do the 32-bit A/D converters work in 20 dB jumps? Is this scenario possible or plausible?

That was my immediate suspicion.  The 20dB difference could represent a "step" of the auto ranging display I suppose, or my first thought upon reading your description of events was that 20dB represents the available headroom present before the impulse exceeded the dynamic limits of the analog input stage and was clipped at that point.

So I boosted the left track 20 dB, and ended up with a perfect stereo pair, no sign of any artifacts from system or background noise, which is exactly what 32-bit is supposed to do.

 
Okay, but a 24 bit file created from the same multi-ranging ADC should work the same way, because the mathematical range available in a 24bit fixed point file also exceeds the total dynamic range capability of the recorder.   In either case, that unplugging transient would exceed the recorder's real world headroom limit before exceeding the limits of either file format.

I don't see any difference in this scenario if you had recorded in 24 bit rather than 32.  I do see a difference if you weren't employing the recorder's multiple ADC feature.

Thanks for the discussion guys.  Not trying to be a contrarian here.  I just want to better understand this and have never heard a really convincing argument for 32bit fp recording on its own terms. In contrast to that I totally get the advantage of a multiple ADC design.  In most folks minds those two things are one and the same, but they needn't be. Perhaps its because that's how these recorders are marketed.  I kind of suspect its a marketing thing.  32bit is a bigger number and so is interpreted as being better.  Too hard for a competitor to explain why its really not, as long as the total dynamic range capability of recorder itself remains the true bottleneck.

[live2496]

24-bit recording will give you a lower noise floor. So the advantage of that is that you can record at a lower level and allow yourself some more headroom.

[Gutbucket]

^ Only in some cases..

24-bit recording can provide a lower noise floor, yet will be able do so only if the noise floor which is a result of the combination of everything else in the signal chain is lower than the least significant couple bits of the 16bit file space.  If that is the case, the maximum potential gain in headroom that can be achieved without incurring an increase in noise-floor will be the difference between those things. 

If the noise floor of all that other stuff is only a bit or two lower, that's all the additional headroom you get.  If the noise floor of all that other stuff is higher than the least significant bit of the 16bit file space, you can gain no additional headroom and just end up recording four additional bits of random noise.. that is, if you were to switch from 16 to 24 bits and leave the gains where they were.  If you reduce gain to increase headroom, you end up raising the noise floor of the recording by that amount, and produce a recording with more than 8 additional bits of noise at the bottom.

If the intent of the brief post above was to sum up the discussion (at least up to the point where we diverged to discussion of multi-ADC designs and 32bit fp), it's difficult to improve on the summary quoted below form an earlier post in the thread-

[..snip] 24-bit recording offers you no greater capacity to record higher numeric values (representing greater sound pressure levels) than 16-bit; ±1 is still the absolute outer limit that you don't want to crash into. But up to a point, if your signal path is generally sensible, 24-bit recording can let you reduce your record levels enough so that you don't run out of room at the top, and still not pay a price in increased audible noise. Not always--your mikes themselves and/or your outboard preamp, if you use one, can be overloaded before the signals get to the recorder; again, the ceiling isn't being raised but the floor is being lowered, one may reasonably hope.

The amount that lowering is never anything like what the additional 8 bits would imply (48 dB!), though, because no A/D converter or analog input circuitry has that much dynamic range. One well-known recorder that I measured some years ago gave only <2 dB improvement when going from 16 to 24 under the particular conditions I was measuring it. It wasn't a typical scenario, but it does illustrate the real dependency that exists on all the other factors in your actual scenario. If someone had used that recorder with those signal levels and gain settings except that they thought, "With 24 bits I can reduce my levels by (say) 12 dB to be absolutely safe from overload", their 24-bit recording would then have been ~10 dB noisier than the 16-bit one. Again that is based on an atypical scenario in terms of signal levels and gain settings, but not impossible by any means.

Thus you should always use the input to the recorder and the menu and level settings, if any, that are most appropriate for the signal levels you're dealing with, and always try to get your absolute peak sample values somewhere in the top, say, 6 dB or so of the recorder's range without hitting 0. It's just that with 24 bits as opposed to 16 (and with everything else optimal, please note) there should be no audible "noise penalty" if your highest peaks are only at, say, -8 dB rather than -2. So by setting somewhat conservative levels when you don't know what's going to happen (i.e. life), you're better prepared in the event of the unpredictable. Murphy's Law says, after all, that the unpredictable is the most predictable thing there is.

[Gutbucket]

The details about all of this were discussed early in the thread if you care to go back and read through it.

[Gutbucket]

David, I'm hoping you might comment on the question I posted earlier which remains unresolved:

What am I missing / misunderstanding?  Why did manufacturers feel the need add the 32bit fp format when no 32bit fp recorder is capable of capturing a dynamic range greater than what can already be handled by the 24bit recording format?

[Rairun]

If you have multiple ADCs and you're using a fixed point format, wouldn't you have to assign the clipping point of the least sensitive one to 0 dBFS, and then the more sensitive ones to lower ranges of the digital format? So a Sound Devices MixPre II (dynamic range of 142 dB) would have to commit to recording very quiet sounds (converted by the more sensitive ADC, providing less analogue noise) at the lower end of its 142 dB dynamic range, so that it if it recorded an explosion immediately after, we would never get digital clipping? The problem is that the resulting file, if the sounds recorded were quiet enough, would sometimes need to be boosted by something like 100 dB in post, as we wouldn't be allowed to set the gain!

It seems to me that the advantage of using a 32 bit floating point format is that 0 dBFS is not clipping, and the digital noise floor is much lower anyhow, so you don't ever have to record the quieter sounds anywhere near it.

[DSatz]

Gutbucket, you asked for my reply to this:

> What am I missing here?  Why did manufacturers feel the need add the 32bit fp format when no 32bit fp recorder is capable of capturing a dynamic range greater than what can be handled by a 24bit recording format?

I'd say that it's business economics (in a market segment different from the one that most of us are in) mixed with behavioral economics. It's straightforward business economics in the market for $20,000 recorders for film sound, where 32-bit recording offers advantages like those we discussed w/r/t digital microphones a month or so ago. It's also a good fit when the production sound crew hands its work off for post-production and mixing on computer workstations. The data conversion step that it saves is small but still greater than zero. It's no mystery why 32-bit recording is used in that work environment.

When and if 32-bit recording offers real advantages for the average person doing live music recording, however, it isn't because of 32-bitness as such, but because of everything else that WOULD go with it in an optimal implementation. It can just as well be implemented in ways that offer no advantage to anyone, as an empty selling point. If a $300 recorder really had similar capabilities to a $20,000 recorder, word of that fact would seep out sooner or later, I'd imagine. Back in the 1980s plenty of CDs were made from PCM-F1 recordings; despite certain problems, most of what was needed was there. I don't see any similar situation happening with low-cost 32-bit audio recorders, though.

"Behavioral economics" controls the world we actually live in. It's the study of the difference between (a) the way people hypothetically MIGHT make choices if they had all the relevant information about the value of something, could interpret it properly, and had no conflicting psychological baggage versus (b) the way people DO make choices, given fragmentary information of uncertain quality and a whole slew of biases that they're in denial about. Marketers and other business executives study this field closely; social media platforms make many billions of dollars from putting it to use; as I write this, some people on this board are very likely wearing clothes with outward-facing insignia and/or brand names that they paid to wear, rather than being paid to wear them. Supply and demand still exist, but corporations also create demand--often by letting customers feel that their individual identity and personality are being expressed by representing the brand.

The difference between (a) and (b) above is famously much bigger in consumer audio than in any other technical hobby or field. Few amateur astronomers would spend five times as much for telescopes based on the supposed properties of their exterior paint; therefore such telescopes aren't generally offered to them. Few ham radio operators would imagine that signals sent through a transceiver carrying the insignia of a long-dead brand will reach farther than one with identical measured performance but no such brand; therefore such transceivers aren't generally offered to them. In consumer audio, however, we're totally swamped with such things. Even pro audio has bent in that direction over the time I've been in it. It's profitable to cater to the credulous; significant opportunities are lost if a manufacturer fails to do so.

--best regards

[Gutbucket]

^
Thanks for that.  Good to know I'm not nuts.  At least with regard to thinking that through.

Whenever I eventually end up purchasing one of these multi-ADC design recorders I suppose I'll need to do some testing to make sure I could actually use that particular recorder to record in 24bit mode using the same recording modality as 32bit fp operation, thus avoiding the increased storage space requirements and the need for format conversion back to fixed point.

[Gutbucket]

The problem is that the resulting file, if the sounds recorded were quiet enough, would sometimes need to be boosted by something like 100 dB in post, as we wouldn't be allowed to set the gain!

That describes 32bit float recording!  Problem or feature?

It seems to me that the advantage of using a 32 bit floating point format is that 0 dBFS is not clipping, and the digital noise floor is much lower anyhow, so you don't ever have to record the quieter sounds anywhere near it.

 

Not an actual advantage because the bottleneck is not the 24 bit fixed-point file format, but the clipping and noise floor thresholds of the signal path through the recorder upstream of file format.  In this situation, both 24bit fixed and 32bit floating point formats are similarly capable.

[admkrk]

... and the need for format conversion back to fixed point.

If I am not mistaken, most DAWs work in 32 bit float, so there is no conversion in that stage and the conversion to a lower bit depth is going to happen either way when rendering. Peter's set I posted earlier today spiked at +3.? dB. Listening to it from Reaper, it did not seem to clip, but did over power the headphones. Lowering the gain bellow 0 dB got rid of the overpowering and still provided a full wave form.

One of the mics was 20 dB off, and I compensated for that, but it had other issues as well. It is interesting that someone else had the same offset regardless the cause for it.

[Rairun]

Not an actual advantage because the bottleneck is not the 24 bit fixed-point file format, but the clipping and noise floor thresholds of the signal path through the recorder upstream of file format.  In this situation, both 24bit fixed and 32bit floating point formats are similarly capable.

Is noise floor not cumulative, though? Meaning that when you add 100 dB to a recording, you're boosting both the noise floor of the signal path AND the noise floor inherent to the 24 bit fixed-point format?

I've just recorded 10 seconds of audio with my laptop's webcam mic, and silenced the audio on audacity. Once the audio is silenced, you can amplify it by however much you want, and the result is still going to be silence (internally, Audacity works with 32 bit float). Then I did the following experiment:

(1) I exported the audio to signed 16 bit and 24 bit PCM format.

(2) I opened the 16 bit file and boosted it by 50 dB. The noise level became very significant!

(3) I opened the 24 bit file and boosted it by 50 dB. It still sounded completely silent. I boosted it by 50 dB again, and then the noise became very significant!

Now, yes, if the noise floor of the signal chain is boosted by the same amount, you're right  that it will be louder than the noise I heard in (3). But surely the total noise level will be a combination of both noise floors, and not just the highest one? This is usually pretty irrelevant because we users do reasonable gain staging, so the signal chain's noise floor is always much, much louder than the 24 bit noise floor. But if your recorder always needs to record the quietest sounds at something silly like -140 dB (to make sure a sudden, deafening noise will not clip), then suddenly both noise floors will be pretty similar, and adding one to the other will result in a noticeably louder noise once boosted! 

[wforwumbo]

Just to chime in, the reason DAWs use 32 bit (some are even moving to 64-bit capabilities) has less to do with dynamic range and more to do with overall numerical precision. This is noticeable when we start manipulating audio - speaking in general hand waves, a basic bell/peaking filter will perform better if the bit depth is larger.

From a dynamic range perspective, it can certainly help with overflows when gain staging, but that's secondary to overall sonic performance of manipulating the audio. Keep in mind, the vast majority of DAW users are distinct from us tapers in that they're producing full records.

[Gutbucket]

Now, yes, if the noise floor of the signal chain is boosted by the same amount, you're right  that it will be louder than the noise I heard in (3). But surely the total noise level will be a combination of both noise floors, and not just the highest one? This is usually pretty irrelevant because we users do reasonable gain staging, so the signal chain's noise floor is always much, much louder than the 24 bit noise floor. But if your recorder always needs to record the quietest sounds at something silly like -140 dB (to make sure a sudden, deafening noise will not clip), then suddenly both noise floors will be pretty similar, and adding one to the other will result in a noticeably louder noise once boosted!

Yes the noise floor is cumulative. If both noise-floors are close enough in level, the resulting noise floor will increase. But the cumulative noise floor of the signal chain is likely to be higher than the dither in the least-significant-bits of a 24bit file by more than a little.  And both will be considerably lower than the acoustic noise floor of any of the environments in which we are recording, including church mouse quiet, purpose build classical halls.  The real world result is the same - the acoustic noise floor of the venue almost always dominates the noise floor of our live music recordings, followed by the self-noise of the microphones.

The virtues of a multiple ADC design and 32bit recording needn't necessarily be linked, and I can see no benefit for myself or for other tapers in recording to 32bit, while I do see additional burdens.  That the files cannot be used as easily prior to conversion is something of an inconvenience, but the big thing is that the file size is 33% larger than 24bit. I record at 48kHz rather than 96kHz because it requires 50% less storage space and provides little to no loss of fidelity to my ear.   As for the processing argument, once imported to editing software the bit depth used in the calculation space is automatically increased to 32bit [edit- or more as wforwumbo mentions] and oversampling can be used were its beneficial, which some plugins do automatically.

I do realize that if I were using a multi-ADC recorder to record 24bit files using the same methodology as 32bit I'd need to normalize afterward, same as with 32bit, which works against the "used easily prior to conversion" aspect, but that's a burden I am willing to accept in exchange for the practical benefits of increased headroom and the ability to not concern myself with level setting at unfamiliar concerts.  All future recordings requiring 1/3 more storage space is not a burden I want to accept.

Based on all that it seems to me multi-ADC recorders writing 24bit files would be the most appropriate tools for live concert recording, as long as the manufacturers provide the capability of using their recorders in that way.

If anyone comes across a link to bench tests of multi-ADC recorders capable of saving to both or either 32bit and 24bit file formats, which specifically compare the results of both output formats, please post here or PM me.  Will need to confirm that the recorder uses its ADC in multi-ADC mode when recording a 24bit file.  Such a bench test comparison could answer this question definitively, at least for the recorder under test.  I do recall the SoundDevices representative posting here a few years ago when multi-ADC/32bit-recording was added to the Mixpres, confirming that they apply dither to 16bit output but not to 24bit.  Maybe that's because they determined the noise-floor of the signal chain up to that point is always higher, making the addition of dither unnecessary.  Testing for aliasing should reveal that.

[DSatz]

I hope this doesn't come as a blow to anyone's ego, but the data format used in 32-bit audio recording was defined by the IEEE back in the 1980s for other purposes, back before it was common for digitized audio to be processed on computers. There were no readily available hardware audio interfaces for that purpose, certainly not at the consumer level, and personal computers were scarcely powerful enough to be worth using for processing audio anyway. Who wants to wait minutes each time just to hear the effect of a simple editing decision? Plus hard disk space back then was limited and very expensive; in the mid-80s a hard drive with enough space to hold the contents of one audio CD would have cost something like $2,000 in today's dollars.

But over time the IEEE floating-point formats (plural--the one used in audio is one of a set) became an ISO standard as well, and Intel and other CPU manufacturers supported it in the form of native instructions of a peculiar sort: As you ran a program, if it contained those instructions, their execution would be farmed out to a physically separate "numeric coprocessor" that your main CPU could talk to IF you had paid the extra bucks for one. The Intel 80286 CPU in your IBM PC AT would talk to an (optional) 80287 numeric co-processor; an i387 co-processor was available for the i386. The story with the 486 is murkier and beside the (fixed or floating) point. If, however, like most customers other than science labs, you didn't splurge for the co-processor and wanted to run software that relied on having one, those instructions could be emulated in library routines. Those were of course much, much slower than dedicated hardware. On the other hand the internal coordination of the CPU with the co-processor also carried a certain amount of overhead. So ever since the Pentium hit the fan, numeric coprocessing capability has been integrated into all CPUs that you would ever find in a PC of any kind.

If a DAW operator is mixing many stems, trying out different EQ, reverb, levels and panning settings on each (some of which may be changing dynamically under software control), they don't want to have to wait minutes or hours for each proposed adjustment to "render" before they can hear what it will sound like. But they also want to leave the original stems unchanged so that they can revert any changes they've made, or revisit a decision later on. The more operations can be done in real time, the more flexible the software becomes, so on a dedicated workstation this can demand very large amounts of processing with multiple CPUs running in parallel. It makes anyone who remembers the original IBM PC with its 4.77 MHz 8088 (not even an 8086!) want to weep.

[aaronji]

^^ The ADCs in these recorders are all 24-bit, so it might actually be easier to record in 24-bit mode from the computational perspective. I think all current SD recorders are spec'd to 32-bit ADC precision, even though their top line models (Scorpio/888/833) don't even have a 32-bit floating point recording option. If you're interested in the math, read SD's patent. It mostly focuses on how they combine the outputs from the multiple ADCs. Pretty interesting, but not exactly beach reading...

[noahbickart]

Few amateur astronomers would spend five times as much for telescopes based on the supposed properties of their exterior paint; therefore such telescopes aren't generally offered to them. Few ham radio operators would imagine that signals sent through a transceiver carrying the insignia of a long-dead brand will reach farther than one with identical measured performance but no such brand; therefore such transceivers aren't generally offered to them. In consumer audio, however, we're totally swamped with such things. Even pro audio has bent in that direction over the time I've been in it. It's profitable to cater to the credulous; significant opportunities are lost if a manufacturer fails to do so.

^^^^^^^^^^^^
This is among the most important truths of audio.