shame

recording engineer for 30 years??? duuuude. i wonder what your home studio looks like @_@ can i get your professional opinion about something? ..i came here because i was wondering if it would be worth it to get a Zoom H2, which records up to 24bit/96kHz. my goal is to get as superbly low-noise vocal recordings as technologically possible, using a Rhode NT1 on a Yamaha MG10XU mixer, via XLR Mogami Gold mic cord, with USB output to a laptop, all plugged into a surge suppressor that has an RFI filter. is the Zoom necessary, and can the Zoom record from USB anyway? also, if so, is 24bit/96kHz a bit too high for the output of a NT1? #20questions

@@prodigalus i would totally bypass your mixer, buy a class A preamp go condenser mic into that preamp the go out of the preamp to input of any interface but make sure that interface channel has zero internal preamp because you only want your class A preamp in your audio path then for monitoring either open your daw load your stereo mixed song stem and assign your mic to a new track on your daw and you can change your buffer samples to 64 so that your not getting audible latency unless your preamp has a real-time zero latency monitoring like a focusrite isa one which is perfect because on that you can also connect a compressor to your focusrite to do a bit of compression on the way in which is a really good idea anyway after all this you have to make sure your mic setup is in the best spot in your room, if you don't have a treated room your always going to get weird artifacts from the room this is incredibly important, the idea is to get the cleanest source on your way in to your daw, and if you don't have a treated room? Set up your mic where you can surround it with blankets and stay away from room corners or you will get an ugly residule bass artifact, do this and you are golden you will have a super duper clean vocal

Also note that floating point cannot be properly dithered, and 32 bit float is a 24 bit file with an 8 bit multiplier... www.thewelltemperedcomputer.com%2FLib%2Ffloatingdither.pdf

That's because 99% of so-called "mastering engineers" are frauds and audiophools who reject knowledge and embrace gossip and magical thinking. All they have is stories, and their stories are always BS. As for 24 bits, you'll never see that IRL with a signal recorded from an analog signal because Johnson noise (a.k.a. thermal noise) puts a hard limit on dynamic range there. In theory you could use high voltage electronics and/or cryogenic cooling to improve that a few dB, but nobody seems to want to go to that expense. If you recorded completely inside the digital domain, you could set your noise floor, I suppose. Excepting that, the noise from the analog sections will dictate the noise floor regardless of how many extra bits are added. It goes to show that not measuring things like the true SNR of a track can lead to lots of wrong assumptions. Good to see a well-informed post!

You might want to add, that in tracking not every 32 bit recording is necessarily floating point. 32 bit PCM still does clip if you hit 0dBFS. Also, unless the recording hardware itself produces a floating point input, that includes signals over 0dBFS properly, you can still clip in tracking.

Thanks. Great to hear. Glad to be useful.

Awesome, so great to hear! Thanks for being here.

So glad you found it too Jeffrey! I hope you’ll join us for some more videos :-) -Justin

That is true, if you were actually able to hear a noise floor that 96dB below your peak level. Since you can’t, trained listeners aren’t able to distinguish between dithered and undithered 16 bit audio in proper blind listening tests. -Justin

This is the way

And you can also just think of harmonies as polyrhythms

100%. Very insightful! I have a whole article on some of these ideas that you might enjoy: sonicscoop.com/beyond-the-basics-harmonic-motion-and-the-root-of-all-music/ -Justin

@@SonicScoop Thanks! I'll check it out!

@@SonicScoop I loved the article, however there appears to be some missing links to some of the images

Alternatively, humans were designed...

No, you!

Digital is NOT a stepped wave. It's recorded as a set of data points but once they have passed through the reconstruction filter, they are not only a smooth wave but they are the SAME smooth wave as went in. Exactly the same. That graininess you hear on 8bit recordings is noise. Also analogue isn't a perfectly smooth rendition of the original sound wave. Everything in nature is quantised at some level be that physical, molecular, atomic etc. An analogue tape does not have an infinite number of combinations of magnetic particles. A vinyl record cannot be cut to perfectly model a soundwave. Vinyl is a soft material. Very small peaks would get scraped off by the needle. Large variations in sound cause the needle to jump and not track the groove. And it turns out the limits of analogue recordings are measurably worse than digital at adequate sample rates.

Truth! The analog part of the converters is the limitation, not the digital part.

24bit? or did you mean 16?

@@prodigalus 24bit. I still prefer that to 16bit. On playback it makes a difference. Also the cost of using 24bit to 16bit is next to nothing

Yes dither your 32 bit floating point audio when converting to 16 or 24 bit because of rounding errors. It's all about the math precision.

That’s a great question. Intuitively, you would think it would work that way. I did too! But that’s not exactly it. Just like with sample rate, you have infinite variability, but within a limited predetermined range. Think of it this way: with a 44.1 K sampling rate you can reproduce a sine wave at 1000 Hz or 1001 Hz or anywhere in between those two value. There aren’t steps in between those frequency options that are missing. The limitation is simply that you cannot reproduce frequencies above 22k. It is similar with bit depth. You can reproduce -1dbfs or -2dbfs or anywhere in between those two values. The limitation is that you cannot reproduce amplitude levels below -96dB. This seems counter intuitive at first. But recognize that in order to produce a sine wave at 1khz at -1dbfs, you need that sine wave to pass the zero crossing, infinitely below -, tens of thousands of times per second. And that is where the resolution loss occurs: At he bottom of the dynamic range, registering as noise. (And fairly ugly sounding noise too, before it is dithered and made random, like white noise.) That is my understanding of the theory of it. But even if that were wrong, the reality of how it functions in practice can be confirmed by your own tests. Go ahead and load up an 8 bit audio file and dither it. You’d expect it to sound weird and distorted and “crushed” in some way. But it isn’t. It doesn’t sound like 8 bit video game music or something, which is what people are usually expecting. It’s literally just noisy. This shocked me the first time I properly tried it too. Hope that helps make sense of it! -Justin

It should be, sure. It's still a best practice to always get the full resolution files, and if they are available, you may as well request and work from them, but it's REALLY not going to make or break the project when they are not available.

@@SonicScoop Thanks for the response, it's really appreciated. I've started the mix for her already using the 16bit files. There seems to be more hiss than usual, but I'm not sure if that's how they've been tracked or if it's to do with the noise floor being higher as you've mentioned? I've had to use Waves x noise on a couple of them. I think she was using a novice producer/home studio for tracking.

If the original files are 24 bit, then get the 24 bit files because bit reduction should be the final step.

@@380stroker This.

I get the confusion there. I had it too at first. But what you're really hearing with old school Atari and Nintendo is primitive 8 bit synthesis. Not 8 bit audio recordings. Totally different ballgame! 8 bit digital recordings, when properly dithered, just sound like regular recordings, but with a fairly high noise floor. Try to run this experiment for yourself! Convert a file down to 8 bit (and dither it). Assuming you keep the sample rate the same, it probably won't sound nearly as bad as you expect it to sound. It'll just sound fairly noisy. If you went down to 1 sample per second, you wouldn't really hear anything as that would only be cabable of playing a sound of 0.5 Hz well below the range of human hearing. But if you used a sample rate of say 8k, you'd only hear frequencies up to around 4k. This was actually a sample rate used in some early digital telephone systems I believe. Works OK enough for speech. (By the analog standards of the day at least.) But it's way too dark for music. That would sound even worse than AM radio, which already sounds pretty bad! Hope that helps.

The 8-bit sound you got from an old Nintendo console is because they didn't use a proper antialising filter and no dithering. So if they used modern stuff to record that 8 bit sound it would actually sound pretty good.

@MF Nickster Thank you Nickster. It confirms what I suspected. What you are saying is the binary coding has been used to determine amplitude only. The more bits you have the louder the amplitude you can take (like the difference between using metal tape and ferric tape in a cassette deck of old) ie you can have greater dynamic range between noise and clipping. However you have a fixed minimum dynamic change as defined by one bit. This is your quantitation limit. Like I said before, real music contains allot of information, some of this may involve voltage fluctuation smaller then defined by a bit. This is the minimum resolution. signal that fall between inside this range will be rounded up ( or down ). The bottom line is when you do the D to A, your wave form will be missing that info - re harmonic content and replaced with different harmonic content consistent with statistical musical error correction used to "Join the dots". As I said before the missing content will hold information detailing the richness of the instrument(s), in a stereo recording positional placement, ambiance and a three dimensional presence. You can increase the time resolution by increasing the sample rate but not amplitude. This would explain why so many people think DSD recording sound so much more real where your sample rate is the most important variable.

@MF Nickster That video is brilliant. Thanks.

Thanks’ Nickster. I think I have finally got my head around this. The key point here is when we talk about Dbs. This isn’t an absolute measurement as say a meter, kilometre, second. It is a logarithmic ratio requiring two numbers. In the context of sound, it is essentially: Db=10 x log{Base10} abs((Amplitude{max}/Amplitude{min})) This is the standard definition of a Db. When we look at an analogue signal, we can calculate dynamic range in Dbs by using to quietest bit above the noise floor and the loudest bit. When you are looking to record sound, your medium needs to be able to accommodate this variation with out loosing detail in the noise floor or clipping. This was true when we used to record music on cassette decks where the record level had to be adjusted to maximize the headroom available on the tape. This would also be true of the digital medium where the dynamic range available will be defined by the number of bits available. At this point what you say bit numbers have is mostly true. There seems to be a rule quoted by various documents saying each bit is approx. 6.02 Db. This number various with number of bits being employed in your process =10 x Log{Base10}(2 ^n x (1-2 ^(1-n)) ^2) Where n is the number of bits. I built this table based on this formula: No. of Bits No. of steps DBs per step Dbs per Bit 1 2 0.0000000 0.0000 2 4 2.3856063 4.7712 3 8 2.1127451 5.6340 4 16 1.4701141 5.8805 5 32 0.9321011 5.9654 6 64 0.5622939 5.9978 7 128 0.3287193 6.0109 8 256 0.1880110 6.0164 9 512 0.1057977 6.0187 10 1,024 0.0587866 6.0198 11 2,048 0.0323351 6.0202 12 4,096 0.0176380 6.0204 13 8,192 0.0095540 6.0205 14 16,384 0.0051445 6.0206 15 32,768 0.0027560 6.0206 16 65,536 0.0014699 6.0206 17 131,072 0.0007809 6.0206 18 262,144 0.0004134 6.0206 19 524,288 0.0002182 6.0206 20 1,048,576 0.0001148 6.0206 21 2,097,152 0.0000603 6.0206 22 4,194,304 0.0000316 6.0206 23 8,388,608 0.0000165 6.0206 24 16,777,216 0.0000086 6.0206 25 33,554,432 0.0000045 6.0206 26 67,108,864 0.0000023 6.0206 27 134,217,728 0.0000012 6.0206 28 268,435,456 0.0000006 6.0206 29 536,870,912 0.0000003 6.0206 30 1,073,741,824 0.0000002 6.0206 31 2,147,483,648 0.0000001 6.0206 32 4,294,967,296 0.0000000 6.0206 Clearly when we talk about bit resolution, we need to be careful of what we are saying. Yes number of bits does set the dynamic range available but the fact you are talking about bits also limits the amplitude resolution you can code. When I asked about the quantisation limitation you kindly pointed out that excellent video describing dithering. After reading other documents it is clear that this is a useful statistical “sleight of hand” where you overwhelm the anharmonic distortion (caused by the “stepping” from one time sample to the next), with a harmonic distortion. We as humans are very, very, sensitive to anhormonic distortion but will tolerate fairly high levels of harmonic “natural” distortion. This is why we are happy listening to a valve amp with 2 or 3 % harmonic distortion and hate transistor amps with far less anharmonic levels. There is another type of error that is simply born out of the fact you are sampling. Sample rates can be increased but bit resolution can’t be. It is effectively fixed to anything above 6.02Dbs. Now remembering that a Db is a ratio and not a fix quantity, this means you sound engineers need to maximize you record level such that it accommodates the actual dynamic range of the music fully. So if we look at a particular time slice and look at how the signal has changed from the previous time slice, the difference needs to be greater than Dbs per step. Log{Base 10} (Abs(Voltage level {current sample} - Voltage level {previous sample}) / Voltage level {previous sample})) >= Dbs per step for your bit level. eg for 16 bit it needs to be better then 0.0014699 Dbs. This was really the point I was trying to make. Look forward to you take on this..

@MF Nickster Thanks Nickster. I have to say I am enjoying this debate. Remember I coming from a position of almost complete ignorance re sound recording practices. I come from a scientific background And later with data analysis in retail (last 20+ years !). I have been interested in hifi since the late 1970's. I agree with you re there should be a sensible point where the level of resolution becomes nonsensical. My view was I suppose centered around live recordings like orchestras. I have heard a very expensive £50,000 violin played by the No.2 violist from the LSO and was struct by all the harmonics I was hearing. All these are very fine details. If you had a whole orchestra then this becomes even more complicated. And then there is stereo content phase/group delays and the rest... I did read some where that we as humans were sensitive to phase angles of 30 degrees or more and that this suggested that whilst we couldn't hear above 20K we are still sensitive to information contained there. I have yet to find an official position on this though. I got my first CD player Philips CD303 in 1984 and was struck by the clarity and bass coherence. In 1995 I listened to a LP again on a LINN Lp12 and was struck by the depth and feeling ambiance and space and naturalness. Despite the detail and dynamics of the CD something was missing as well. This really where I am coming from. I have started listening to High Def audio using my Christmas present - Cambridge audio Azure 851N. I have listened to CD 16bit/44.1 , 24bit/96K 24bit/192K and DSD64. To my ears CD was the worst where I felt the DSD64 sounded the more natural , 24bit/192K wasn't far behind. Obviously 50% of the comparison where done without being certain that they were the same masters sources. My feeling is we still don't know enough of how we hear and all the subtleties that real analogue we use/hear in building our sound model in are minds.

@MF Nickster Actually I totally agree with you. The problem is not in the technical way we do things nor our technical /scientific way we encode data. Data is king here, information rules. The cutting edge of physics seems to be saying the very essence of reality is information (see kzfaq.info/get/bejne/jt6Gn6pyq76bpnU.html this is good!). The problem is how we actually hear ie biological process involved in how we assemble auditory information into a coherent sound in our brains. This is why I keep harking back the degree of resolution and having more then we currently think we need. This is I believe where the issues are arising, not just in the digital world but also in analogue. Back in the early 1980s, there were allot of medium to high end Japanese amplifiers (from different manufactures) that boosted ridiculously low distortion figures of 0.000005% and yet they sounded awful ! The reason was they used used high degrees of negative feedback in esoteric named circuits. This killed phaseual information and smeared the information in the time domain. Turns out we are very sensitive to this type of distortion. When we hear, we hear with more then I ears. We use low frequency vibration in our bodies, we re aware of very low level ambiance - so much so that they limit the time people work in anechoic because near total silence can have a psychological effect. (www.theguardian.com/lifeandstyle/2012/may/18/experience-quietest-place-on-earth). You say you don't have a scientific background (yet!) What are you studying ?

I don’t make records, but am very into listening to them. I’ve come to you via DAC processing per Chord QUTEST and wanting to understand what bit depth and sample rate mean. Your presentation helped a lot. Thank you!

Nope! Sorry :-) The kind of hum your are talking about on a guitar is likely 60 cycle hum, especially with single coil pickups. Different animal altogether. Try humbuckers! (or double stacked single coils.... which are effectively humbuckers that sound like single coils.) Hope that helps, Justin

I'm having trouble finding this study. Can you link to it? Was it double blind? Under anything approximating normal listening conditions? Based on the 94.1% discrimination rate, I'm going to guess the answer to that is a resounding "no" :-) Likely scenario is that either it wasn't double blind, or there was some type of trickery being done that does not remotely mimic normal listening conditions. Probably both! I recall a paper submitted to AES where they took the fade out tail of an extremely quiet moment in a piece of music, jacked up the level by like 70dB and then had people listen for differences in the noise floor. Yes, THAT is possible, obviously! X-D But the part this leaves out is that if you played back the rest of the recording under these conditions it would probably blow out your speakers and your eardums :-) Is this that "study"? If so, it fails to demonstrate what it seeks to demonstrate, and confirms what we already know: That the bit depth only makes a difference if the noise floor it adds is loud enough to be heard. That just isn't the case with normal program material mastered properly at 16 bit. Adding more bits above that demonstrably makes no difference at all. This is not really controversial. I wish it were different too, but it doesn't appear to be. I hope that helps make sense of it! -Justin

48khz at 16 bit. Anything more than that is diminishing returns and no one would ever notice it (No one has ever said: “Wow, this video sounds like 16 bit”). I think the only exception to go higher is if you work on professional music videos, theatrical movies, etc that will be played on 5.1 and 7.1 speakers/headphones

The SACD is not from an identical source. SACD were often remasters. If you take the same master and put it on CD, SACD, or Hi Res formats, it is impossible to tell the difference. There are about 2 studies out of hundreds where someone could tell the difference but the subjects were "trained" i.e., biased. Physics says that it is impossible to tell, not opinions.

SACD and regular CD are being mastered differently. That's why SACD sounds better. They do this on purpose.

Maybe sample rate went from 48 to 44.1, all I know it was frustrating

There isn't much point in producing a master over 16/48 but inside a DAW or when using any other DSP, using higher rates make complete sense

@@RaveyDavey Most DAWs are able to do internal handing of up to 32 bit, but you still need plugins that support it. It's highly discusable tho if it makes any sense to use the much higher processing power needed to do so. But there is absolutely no point in home studio's to track (input) in higher than 44.1 or 48k/16bit I bet you noone can hear the difference if you go 24 bit or 96/192k. You are just moving 2'nd, 3'rd etc. distortion around. you might get slightly better results dependent on the material your tracking by going higher sample rate, and you might make it worse. Our ears don't have the ability to detect the dynamic range provided by higher than 16 bit resolution. Theoretically 44khz/12bit is enough to cover the human ears dynamic range.

Well, it also depends on how good your monitors are. But yes... with good speakers and a good audio interface, you will be able to hear the difference, assuming of course, that you're also a good listener(literally... as in if you're not and don't pay attention, you won't. But it will also let you get more feeling when playing on your guitar). The question, in reality, is really: What kind of music do you play?