Discussion in 'Pro Audio Equipment' started by Mundox, Nov 6, 2003.
OK.This is a little weird and I should know this, but I really am not sure.
i'm no expert, so i welcome corrections if i spreading misinformation here, but all of the theory that i have read indicates that 24 bits gives you increased dynamic range... however whether you are using/need all that dynamic range is another issue entirely.
if the track has less than 96dB of dynamic range then having the extra 8 bits of data will not get you anything you didn't have with 16 bits.
since i cannot listen to a guitar playing and tell you how many dBs of dynamic range it covers (and i don't want to spend the time to scientifically measure it) how does this knowledge translate practically? i track with 24 bits so that i can keep my digital meters well below 0 and avoid any digital overs. after that, and after i apply compression etc. it may be, and probably is, the case that i don't have more than 96dB of dynamic range.
In the controlled environment of a mixdown i can make sure that if i am dubbing to a 16bit source, as you are, that i am not getting digital overs (in a way that could not be predicted during tracking) but still utilizing as much of the 16bits as i can.
long answer short, for rock/pop music, no... you are probably not really losing anything by dumping down to 16 bits if you are maximizing your levels. classical or jazz might be a different story as they may be more in need of that extra dynamic range.
Truncation is the result. In other words, if you go from 24 to 16 without dither, you lose the bottom 8 bits.
if you have less than 96dB of signal to noise dynamic range (AKA 16bits of dynamic range) the lowest 8 bits would actually contain nothing... correct? so while you may be 'truncating' you would be cutting off... nothing. so nothing would be lost except 8 bits of nothing.
i ask for my own understanding as 'truncating', while that may be the technical term, sounds inherently insidious and bad.
i guess everybody is on the same page with this, including me .Since I use heavy compression on my tracks I don't really utilize those lower 8 bits.(except for cymbals which go to the analog outs)
Its the highest 8 bits that contain nothing in the recording you are using as a reference.
Just imagine looking at the LEDs on a circa 1980's tape deck. As the signal gets louder, the LEDs grow from green to yellow to red at the top. What you lose in truncating is a significant portion of the green. You don't lose the red.
Now what happens is the quantization effects of this lead to a lot of ugly noise. So you suffer even more if you truncate a quieter signal.
What's really interesting is that dithering is, in a sense, adding a small amount of broadband noise to hide the quantization.
i think perhaps we are mixing up our analogies somewhat (or just visualizing it differently... which happens a lot, i've found, when imaging sound) but i do understand what you are talking about mitz. by the "lowest" 8 bits i was referring to the quietest 8 bits... which would indeed be contained in the green leds on the meter (the green bits?).
but what i was really wondering was if the quietest 8 bits were in fact silence what damage truncation would actually be doing.
in doing some deeper looking it seems that, as usual, the differences in results you get from truncating and dithering vary depending on the material.
in more dynamic detailed material it seems the noise from quantization errors could be problematic without dithering(as mitz said).
so, as usual, the answer seems to be... i dunno... do you hear a difference?
the more i learn the more it all seems to come back to that... but learning is such fun.
Now you've learned a valueable lesson. More people could answer their own questions and learn more by taking the time to apply that lesson.
The result of 24 to 16 without noise shaping or dither is called truncation distortion. How much audible damage is done by it is related to the type music and how the headroom was used going through the 24-bit conversion to begin with. It is a common practice that is done all the time. Use those tracks for the less important and/or less dynamic sources and you should be fine. Chances are that you or anybody else won't ever notice it in the end mix or that your monitors are good enough to hear it. There are far more critical things to worry about.
I myself, prefer analog audio. To my ears, it sounds the best. I am not saying it is the most accurate, just that I prefer its sound.. I don’t think anything we have come up with at this point is a “mirror perfect” representation of real world audio.
That being said, I work in digital for many reasons. It is more affordable, for me and my clients. Maintenance is an issue for me. Property tax each year and space to house a big 2” machine, a ½ track and a large format console can add a lot of numbers to the bottom line. The ability to manipulate the audio is in demand. Clients look for it. Random access is a real time saver. The ability to completely recall a mix with a keystroke is phenomenal. And it sounds fine. Not as good as a 2” and a large format console, but good enough.
I have Cubase VST 5.1 and I run it at 24 bit 44.1, through two ADAT AI-3’s for 16 analog and 2 s/pdif insanoutz. I monitor and mix at 16 bits through a Fostex CR220 CDr burner via s/pdif. This CDr has a dithering feature built into it and it is connected to a Nakamichi 410 all discreet stereo pre to a pair of Haffler 3000’s. I can also monitor at 24 bits through the AI-3’s and a Mackie SR24 vlz into the Nakamichi ...
Now I may be fooling myself but I think the digital output of the DAW (s/pdif at 16 bit) sounds waaaay better that listening to 24 bits through the Wackie! I think it’s the Wackie that gooeing up the works. But the point is, even 16 bit can sound very good, if it is done right. In some situations it can even sound better that 24 bits done poorly (through a Wackie or similar) .
All that said, I think keeping the sample rate at 44.1 throughout the production process is the best way to avoid sample rate truncation.
Kurt, weren´t you meaning sample rate conversion?
One of the things with Pro Tools TDM systems bounce to disk that gets nasty when going from 48k/24 to 44k/24 and then finnaly dithering to 44k/16 ( many people misconvert going diredctly from 48k/24 to 44k/16) is that the process itself changes your master levels.
I reported this at the DUC in 2002 and a few other guys around the globe described the very same symptons.
Even if you define a ceiling, let us say -0.1dB, you shall end up with some red lights at the final product.
This proves that frequency conversion, at least with the mighty PT, is not 100% accurate.
Now I also work at 44k/24, eliminating one bounce, and avoiding artifacts from a frequency conversion.
Sorry for being quite off topic. :
The great thing is digidesign pretty much got rid of that problem when HD came out with BTD.
Let's simple it up:
An expansion on gaff's statement: The closer you get to the 24 bit maximum, the less truncation will bother you. If you are starting with a super hot kick track, it will be much less noticible than if you were starting with a classical violin.
The quietest parts of a digital recording (like the very end of a long reverb tail) use bits 1-8. The middle of the line signal fall somewhere between 9-16. The loud stuff is up there at 17-24 bits.
When you truncate to from 24 to 16 bits, you are effectively throwing away bits 1-8.
So the violin, which should fall, for the most part, between 9-16 bits, will become 1-8 bits of the new 16 bit audio recording. Right down there with the silence. But we know its not silent, right?
That is the effect of bit-depth truncation.
Alecio and Kurt:
Sample rate conversion will exibit a loss of data on any DAW. I'll explain it by example:
At 48k, you record 48000 samples per second. At the very beginning of that second, you're at sample 1. 1/48000th of a second later, you're at sample 2, 2/48000ths is sample 3 and so on.
Now if you convert to 44.1k, sample 1 is handled, no problem. 1/48000th of a second later, the original 48k recording is presenting sample 2. However your 44.1k file has not reached sample 2. So how does the data get incorporated into the destination of 44.1k?
By a process called interpolation.
Sample 1 at 44.1k is identical to sample 1 at 48k. Sample 2 at 44.1k takes samples 2 and 3 at 48k and uses an algorithm to determine what the sound is at 1/44100th of a second, which is somewhere in between 2/48000 and 3/48000 of a second. In other words, sample 2 of the destination format (44.1k) is interpolated from samples which come before it and after it in time, in the source format (48k).
Interpolation is a necessary evil, we can't convert sample rates without it. The better interpolation algorithms take a look at several source samples before and after the destination sample to determine what the sample will be.
How does this affect us real world? Well just imagine that in your source file (48k) you have a transient. Chances are it happens somewhere in time that is not accounted for in your destination format (44.1k) How does it account for it? By moving that transient either forward or backward in time, and smearing it over two or more samples.
Of course, this is separate and distince from bit depth truncation, but the subject came up
Nope, no mirrors here, but recording analog is sampling at a sample rate of infinity, much more accurate than digital will ever be. 192k, 384k? Bah.
When you sample a sound at discrete time slots, you lose everything in between those time slots.
I ODed on theory in undergrad
oops my bad, hit the wrong button!
Nice thread guys,
If you work at 96khz and then dither down to 44.1khz, You are essentially just making things sound worse than if you had just started working in 44.1khz, am I right? I've noticed a lot of plugins sound better at 96, but I've never a/b a source through one on 44, and through one dithered from 96 - 44 before.
Hey Missilanious!! Westchester representin' - I used to live in Mamaroneck - salutations.
mitz, very nice comments about the transient being slightly shifted.
That is why I decided a few years ago to do everything at 44k/24. I don´t think PT HD had the BTD fixed, however, I will check that.
I'd say 96 to 44.1 is a LOT safer than 48 to 44.1 - The ratio of the former is greater than 2. Practically anything captured on 96 that is lost in the conversion to 44.1 would be lost anyway in the A/Ds when recording directly to 44.1.
48 to 44.1, on the other hand, can introduce a whole set of problems. For starters, digital recordings ALWAYS need a rolloff that tapers down to silence by half the sampling rate. 24kHz should be silence at 48kHz sampling rate, for example. This is to prevent the appearance of aliased ghost notes from occuring when the A/Ds pick up signals greater than the media can handle.
Recording at 48kHz introduces one rolloff. Converting it to 44.1 adds ANOTHER rolloff to lop off anything above 22.05kHz. These rolloffs overlap and can technically screw the treble a bit.
I suppose a really cruddy analogy would be 48->44.1 conversion being akin to dividing three apples between two people, whereas 96->44.1 conversion is like dividing nine apples between two people. In the former situation, one person is more likely to feel cheated than in the latter situation.
but 88.2 to 44.1 is the best, considering its a 2 to 1 ratio with no decimels in the ratio make the conversion math easier, than again 44.1 is the best if your going to finish at 44.1 cause theres no sample rate conversion, but than again your plugins aren't going to be running at higher resultutions either, which is very evident that at highr sample rates plugins like reverbs sounds better than if you would use them at a lower sample rate, so I would say why work at 96k if you can work at 88.2k unless your projects going to end up on dvd's. And to my comment on the HD BTD fixed, I'm not talking about sample rate conversions cause thats going to sound different after the conversion I'm talking about the summing at the end of the 2-bus in protools. When using a mix system which uses an older TDM mix engine as opossed to the redisigned HD mix engine (TDM1 and TDM2 mix engines respectively) the problem many engineers were complaning about (not really a problem if you think about it) is that after a bounce to disk (if the summing was done in PT,and not just bouncing down at two track) it would sound different from what you heard out of your monitors when it was being played strait out of protools (not drastic) even when the bounce was of same sample rate and bit depth. When HD came out i don't see this problem anymore. And I did a test. I recorded a session at 44.1 24bit using my HD rig. Without any plugins I bounced that down without any changes to the sample or bit depth. That was the bounce off of HD. I took the same session to my schools mix plus rig via firewire. Loaded the session, again with no plugs or sample rate or bit depth conversions, and bounced that down. Both bounces were done using using tweakhead, 24 bit, aiff. I reimported both of those bounces and flipped phase of one, what I heard was some ambiance, upper mids, top end, a little bit of low end and a little bit of low mids, which says that is the difference of the two cause anything that is similar gets cancelled out. Alecio wouldn't see that because if I'm correct he uses an 02R to mix, so if I'm correct he'll mix in the 02R, which gets summed at the 02R's mix bus, and then he rerecords the 2 track back into protools or an outboard recorder.
This has no bearing on the practical discussion going on here but i thought that this statement is a little misleading. Analog tape (analog information) is just as limited by bandwidth restrictions as digital information is. If an analog recorder can record up to 45kHz (a good analog recorder at that) then that's what it can do... no more, no less. The same frequency can be captured digitally using a 96kHz sampling rate (with some to spare). That's what it can do, no more, no less. Call it discrete, call it continuous... that bandwidth restriction is the limit of possible information that can be captured.
In addition, the 'accuracy' of analog recording is limited by the amount of noise... as any dynamic changes that happen which are smaller than the size of the noise floor are indistinguishable from the noise. This limits the possible resolution of analog.
THAT being said, the band i am in just recorded bass and drum track to reel to reel because it sounds friggin fantastic (to us). That was the sound we wanted.
but sample-rate is different in the digital domain.
mitzelpik is saying that the sample-rate of analog gear is infinity - sure a digital sample rate of 96khz (half of the rate - possible freq range) will reach the same freq. as a good tape rig, but it will still be sampling less frequently than tape.
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