Thank you for bringing the subject. I happened to utilize dithering technic for low level signal detection (as well as I am highly experienced in audio signal restoration) so I'll try to simplify the explanation of dithering. 1) The reason you can't hear dithering noise is because usually a noise shaping is involved pushing the energy of the noise to the higher frequency where the ear is less sensitives. 2) Noise shaping also increases resolution of the low level signals. 3) signals that are at level lower than the least significant bit cannot be restored without dither. In order to be effective The peak level of the dither signal is usually higher than the LSB by 2-3 bits. 4) A lower than 16th bit signal is not only a signal that is at loudness level lower than -96dB it can be an harmonic with signal level lower than 1 bit, say low level violin harmonics played together with a contra bass. These harmonics are also lost (or heavily distorted) by the challenge to affect the LSB bit 5) when (only) a high frequency section of -90db level white noise is played, the LSB bit will change it's status from 1 to 0 randomly, we will get a noise at RMS level of 3dB with average of 0 volt DC' meaning and equal avenge number of "0"s and "1"s . 6) When playing a 1KHz signal @ -108db level with no dithering, the digital recording value will return a constant 0 voltage because the level is too low to affect the A/D LSB bit. 7) When adding the two signals (1kHz and White noise) together we will still get a DC average of 0 Volts, we will get a noise of about 3dB level, and the long time DC average will be 0VD. However, the digital statistics of the noise will have more "1"s during 1 KHz signal period is at the positive area and more "0"s when the 1Khz period is at the negative cycle. 8) This means that when adding an LPF for a 2KHz at the output of the DAC, we will get rid of most of the noise and can reconstruct the -108dB level 1Kz signal. 9) Luckily for us as humans, our brain can recognize the 1khz since we barely hear the noise shaped dither' so no need for adding an LPF in order to hear the 1Khz. 10) When FR is limited to 20Khz (at 44KHz sampling rate), we will be able to recognize only medium frequency which is fairly good result. 11) A High res sampling rate allow pushing the dither noise shaped signal energy to above the 20Khz range, ( beyond human hearing capability*) and allows for higher audio frequencies to be restored. 12) Is Dither needed for 24 bit Hi Res Signals? to my experience the answer is yes. it is not needed for the recording itself' but it is useful for the reproduction. not all systems are capable of accurately produce a 24bit quality of signals (due to poor DAC circuitry for example or other reasons), in other cases the recorded signal is passing through digital compression channels such as streaming or Bluetooth and Etc. 13) For my work I always use a 24bit 96Khz production with noise shaped dithering of 20 bits (4 bits for the dither). 14) * We can hear information above 20KHz: We can't identify tones above 20KHz (on our age ithe limit is lowered to above 12Khz) , this is what makes dither works: We can hear identify a "20Khz to 40Khz modulated white noise carrying a low level hi-mid frequency (it is not realty modulated, just a sum of the 2 signals) Hope the explanation is not to detailed or too complicated.
Shure, However the subject here relates to final product. 16bits, 44.1KHz done properly, can sound amazing. so is there any added value for a 24/96 product? @@FirstLast-is9xe
@@FirstLast-is9xe 32 bit float is useful inside a DAW or similar, since it's 24 bits of data, shifted by an amplifier/attenuator that's capable of several hundred dBs. This allows you to work with a signal whose reference level is off by a bit, without losing quality; for example, if I've recorded a line level source using a 24 bit ADC while forgetting that my microphone pre-amp is turned on, I can keep the full 24 bit samples, and attenuate back down to line level; I can also take a microphone recorded with the wrong pre-amp settings and amplify it, keeping as much of the original sample set as I can. It's also worth noting that your ADC is not just picking up signal; it's picking up thermal noise in the electronics and cabling, too. A UA-cam comment is not a good place to go into how Boltzmann noise affects an ADC's effective number of bits, but if you do the maths, you find that for professional line level signals, you've got around 20 to 22 bits of resolution before all you're picking up is noise, even with a top-quality 24 bit ADC (and similar on the output stage - the bottom 2 to 4 bits of your DAC are effectively obliterated by thermal noise). As a result, the "optimum" is to accept that your signal has around 20 bits of information in it, and to build your workflow around that. One option is 24 bit integers and being careful with your chains of (digital) attenuation and amplification in the studio, so that you've always got 20 bits of information flowing through the system. Another is to use 32 bit (or even 64 bit) floats and then have less concern about foolish chains of attenuation and amplification, since you're unlikely to damage your 20-odd bits of source information even if you do something completely crazy like attenuate by 130 dB followed by amplifying by 120 dB (which can be replaced by a 20 dB amplification).
For me it's all about headroom. This is important for the recording and production phases. I was given a raw 24bit live recording where the levels had been incorrectly set for the first few minutes and I needed to bring the levels up to match the rest of the recording ... I feared the worst whilst remembering trying such things in the old world of analogue recording where the previously inaudible noise floor would intrude unpleasantly. I therefore was blown away when I simply selected the low level section and brought it up to the rest of the recording using Audacity. To my ears it was audibly perfect with no noise floor issues at all! However, an end product distributed in a 16bit format (once all the production tweaks have been done) is fine for my ears.
THIS^^^ There are also many reasons why you might lose bits while trying to reproduce the audio. For example, my Motu M4 will just throw away bits if it doesn't get the packet in time, and if you don't add latency, the sound will lose audible information. some cheaper DACs even intentionally lower the bit depth to lower the volume. 😳
I’m sure people are tired of hearing from me A EE degreed person who understands digital signal processing and audio as well as higher than audio frequency audio processing and equipment that I am involved with. The 24 bits does nothing but lower the already noise floor even lower or correspondingly gives you many more higher register bits for a more high dynamic range headroom. 24 bits is usually useful only internal to audio processing systems like DAWs and mixing console type systems because as you add any EQ reverb chorus Flanger delay, d-Essing special effects (to remove sibilants from vocals) it’s nice to have all that headroom as all those numbers get crunched together with those calculations getting on but on the final rendered output once you master it to 16 bits to make a CD there should be really no quality loss at all. I get so, so tired of the 24 bit vs 16 bit debate everybody has. Tsk, tsk, tsk. 🤨
I would just like to re-iterate, but I love this UA-cam channel master audio class it is or something because it gives you so many different viewpoints about what audio files and music can be. It lets engineers, audiophiles and just average music listeners come together to Appreciate music, and I guess we can debate and dawdle about the details, but it is just such a fun field to be a part of. I love this channel!
I'm in University for Comp Eng in my 5th semester. From what I understand the bit depth determines the quantization levels for each given sample. Now I didn't learn about quantization in systems class yet, but I can imagine that it is useful to capture either more fine grained, or a higher dynamic range with no loss of detail. I came here cause I saw an add for some headphones that were boasting their 24 bit capability, and it just felt off to me and like an unnecessary gimmick. Also they were hinting towards having a higher spectrum which also didn't make sense to me.
I'm completely new to recording and I have a 16-bit dynamic mic with specs 44.1-48KHz and a sample rate of 192, but my mic audio is so muted compared to my other inputs. I can level my inputs with my mic audio to make them more balanced, but then I have a quiet video. How can I boost the audio by 4-5x without risking static?
Nice video. I like your distinction between ‘enthusiast’ and ‘audiophile’. I’ve come to realise I’m the former as I don’t have the money to become the latter. I like being happy enjoying the music with my tech that is good enough. 16-bit 44.1kHz for me 😊
Why do we have to have a war on the the word "audiophile" ??? I know major recording engineers (real names) and they don't mind being referred to as "audiophile" because it sends a message that they are interested in doing quality word whereas other in their trade are just in it for turnover and money and that brought us the compression wars otherwise known as loudness wars. There is some really bad recorded stuff out there or good stuff that was then wrecked in the mastering process. Trust me, I am speaking from experience. I don't mind being called and "audiophile" and I have good ears and claim no more. Don't make war on words, make war on bad recordings and rubbish.
@@D800Lover Agree, I don`t even know where there is this urge to attack the people in the first place. Why not just leave them alone? Sure, you can tell them that they get ripped off, but why make fun of them all the time?
@nrezmerski - And are you a music lover? What great sin to call yourself an "audiophile" because you like music to sound like real music in your own home? I say, keep it up!!! And audiophiles often spend within their budgets and still call themselves audiophiles.
@nrezmerski - The reason people laugh at other people is bad behaviour. My mother thankfully taught me right. I am not rich and I can describe myself as an audiophile without somebody making videos at my expense? That's pompous and clickbait and deposits in the bank. So I decided, that deserves some pushback. Most audiophiles are not rich dudes.
This is anecdotal at best so take from it what you will. Years ago, we're talking Pro Tools7 era, I was recording a group and they were going to get the tracks mastered elsewhere. We were about finished when they finally mentioned that their mastering guy wanted 24/48. I said, "Sorry guys, I recorded 16/44.1" This is followed by a little bit of grumbling from the mastering engineer. A few weeks later I get a call from said engineer saying, "This is the best 16/44.1 recording I've ever heard." To me that just came across as silly. I'm sure you've heard better. I'm recording in a project studio in a log cabin with very minimal gear. My only goals were the right mics in the right place and capture the best performance. That same band came back to me for their next album as well. Don't get me wrong here, I wouldn't argue that 16/44.1 is objectively better than 24/48 but to me the difference is minimal. Prior to that time period I was working with ADAT XT20 machines. I subscribe to the concept that if you record garbage at any bit depth and sample rate you'll still have garbage. So, as I said, take from that what you will. I'm not saying that what I did was right/wrong or better/worse, only at the end of the day everyone was satisfied.
If you record multitrack in 16 bits you can mix to 24 and get 24-bit resolution because most or all of the tracks will be lower than 0 dB on the fader. Also reverb tails.
@AudioMasterclass Pardon me if the following question seems a bit naive, I know what sounds good but I don't really worry about the technical side of things that much. So, you're saying that I can multi-track in 16/44.1 but bounce out 24/48 stereo file assuming I'm not clipping the the whole thing out? (which, of course, I would not) What benefit, if any, would that provide? Just a little bit of extra wiggle room for the mastering engineer?
The benefit of 16-bit during production is that the computer doesn't work so hard, therefore you can have more tracks / more processes. That's the theory anyway but I doubt if many producers do this.
@@gabrielgodwin9953 When you apply effects there are math operations applied to samples, many involving multiplications (division is just multiplication by the reciprocal). These tend to product values with fractions of a whole sample step. If the project is in 24 bits or higher, these fractions don't need to be rounded off (rounding produces distortion). When the 16-bit mix is made proper dither will be applied. As mentioned, if you take a 16-bit track and halve the volume, you lose a bit if you stay in 16 bits, but in 24 bits you keep the full resolution. Or consider reducing to 75% volume. You can't reduce 65536 possible sample values to 49152 values without merging some pairs into the same output value. Distortion.
24 bit vs. 16 bit is dynamic range 144db vs. 96 db. Noise is from YOUR recording system, from the ambient noise of the room through the microphones through the mic preamps and then the rest of the mixing desk, including the AC system of your room or building. Cassette decks needed Dolby B because of the tape hiss inherent in the mylar tape medium itself which was noise at about -50db from the peak of 0db. With Dolby B, C, S; A for the pro recorders and on. You could get to a noise floor of -60db to -80db, maybe. In my DAW I can see the noise floor of MY recording system in my meters. With my blower motors off in my HVAC system and AC line filtering I can get, usually -70db to -80db down from 0db with my mics open in the recording chain. I can see it. Even with 16bit (-96db of dynamic range) I have still not used up all of the dynamic range and my noise floor is higher than -96db of the 16 bit system. I have not reached the bottom of the theoretical. Detail is a matter of sample rate so if I go above 44.1khz I can record more and I like 24 bit/96khz for all my masters. I have tried 16 bit/96khz and cannot hear the difference, but use 24 bit anyway. I read of some studios in NYC using 16 bit/96khz for their masters. I can't hear the improvement of 192khz or 384 khz using 32 bit float. It is just more work to get back to the 16/44.1 of CDs. In my DAW I have pulled some "professional recordings" from CDs and looked at them at the beginning of a track. They usually leave about 2 seconds of blank space before the music starts. I have seen in some commercial recordings a -40db to -50db noise floor before the music starts. They could have used a noise gate, but did not. This tells me their room is noisy from open mics through their recording chain.
Yeah, that's what I am saying in a comment above. And I am hobby drummer with a serious VST collecting syndrome. I use these drum libraries by pro engineers and there is sometimes a lot of noise. Huge areas of pro studios, hundreds of meters of cable, patchbay, consoles, preamps. And then compressors ... you know. Sometimes I hear the drum chairs creak.
32 384 is harder to record for me , tends to sound overly bassy for some reason and most peoples gear wont play it. I think the happy medium is 24 96, I also like 16 88, I agree going up to 192 doesnt seem to help much but I also seem to hear more noise at 192, dont know why not being an expert. Steve Wilson masters everything at 24 96..
@nrezmerski Listen to the recent remaster of Abbey Road at 32-384, its not just me! If it sounds nice to you, let me know.. If you think its something that I haven't figured out, happy to hear from you! My gear is good, although admittedly I avoid 32 bit for the reasons I have given.. its trasparent, there is no noise, just the timbre is not nice.. regarding more noise at 192, its the recordings, they tend to be like that, same album at 24-96 just sounds cleaner. As I have said elswhere sometimes I choose the 16bit over the 24..If you have a technical explanation as to why this may be the case it would be interesting.. thanks..
Hi, really awesome your sarcastic comments in your videos. Ha! I (43) actually used Dolby S HX-Pro AND cleaned the heads recularly. I also used my dad's AKAI 4000DS reel-to-reel. Copying Paul Simons "Graceland" :-) My dad and I, we discussed those things like flutter and distortions factors when buying a tape deck. I guess, when dealing with digital audio (recording) the sampling rate will be more cruicial when using certain effects because of harmonics above the Nyquist frequency, and audible non-harmonic frequency components.
There is one benefit from tinnitus... the whole noise floor sorta becomes pretty much a non-issue. Recording 16 bit CD is plenty good for cassette, yeah I still do that. CDs transfer to cassette much better than vinyl. That being said that vinyl to cassette thing really doesn't work out very well. But it did enable tinnitus due to being able to turn the volume control to 11 without any feedback.
Just watched a video by an audio engineer for a record company. He stated that the only difference between 16-bit and 24-bit is dynamic range and you would have to hear the music at insanely high volumes to notice any difference.
With respect to noise floor, yes. But increased dynamic range preserves more detail and tonal separation at reasonable volumes. Just compare it for yourself. Those myths are ubiquitous, but they are based on misunderstandings. For example, no one is seeking 24 bits for the 48 extra dB of noise floor unless they are misinformed, or playing concerts.
@@blerblybliggots9801 i'm not sure how one could hear that sort of dynamic range. the louder sound masks the quieter one well before the bit limitation. it would be like trying to hear if the fridge is running while operating a chainsaw :)
Have there been any properly blind tests where people could reliably identify a 24 bit musical signal against the same signal, but degraded to 16 bits and dithered? I would suspect not, but if there was, I'd be more likely to reconsider.
I have always recorded in 24 bit. However as a consumer format, 16 is good but it is so much easier to maximize the quality of 16bit if you start with 24 and then convert the final product to 16.
Also, think that a very quiet room has a noise level of 20-30dB and you damage your ears if listening louder than 80dB. That is 50-60dB dynamic range. The 16-bit quantization noise will be more than 30dB below ambient noise. Even without dither it can be very difficult to hear distortion 30-40dB below noise level.
Many audiophiles conflate bits and sample rates used in the recording process with the bits and sample rates in the playback files. This is a mistake. High sample rates and more bits are used in the recording process to enable editing but are pointless for playback. If you actually had a 24 bit recording with a true 144dB dynamic range and the equipment able to handle it (which is impossible) and your quiet listening room had a 30dB background then a peak sound above background would be 174dB which would be lethal. No, I am not exaggerating, this sound level kills. Even a true 16 bit 96dB recording would be unplayable as the quiet passages would be inaudible or the loud passages screamingly loud. Most modern recordings can be delivered with 10 or 12 bits. A full symphony orchestra giving it all it's got (ƒƒƒƒ) peaks at about 104 dB SPL. Let's give the orchestra 105 dB, and 105 dB - 30 dB = only 75 dB real dynamic range. Most recordings are compressed dynamically to a sensible range so the listener is not constantly having to twiddle the volume control. Higher sample rates on playback produces ultrasonic noise and does not improve audio in the audible range.
I have heard arguments about ultra soninc wave you obv dont hear them but some argue that the soundwaves you dont hear interact with the ones you do hear causing interaction to the audible ones. XD
@@patrickharley1003 The quality of a recording is determined by the studio at the time of recording and production. A well recorded and skilfully engineered 16/44.1 CD can deliver audio quality that exceeds the limits of human hearing. Higher bit depth and higher sample rates are used in the recording process for editing purposes but are pointless for playback. 16 bits gives a huge dynamic range. You will never hear noise in a silent passage of a 16 bit recording unless it is tape hiss from a copy of an old analog recording. 24 bit playback does absolutely nothing to sound quality other than further reducing the noise floor which is already inaudible with 16 bits. Higher sample rates produce frequencies above 20KHz, which is not only above human hearing, but captures unwanted noise that can produce nasties in the audible range. These ultrasonic frequencies can get into a tweeter and cause problems. Higher sample rates on playback cannot improve quality in the audible range. Any effects from the brick wall filter was resolved 20 years ago by improved DAC technology such as over sampling. A modern $100 DAC can give superb audio. So called “hi-res” releases are marketing hype to sell the same music again. Loudspeakers and room acoustics have far more effect on sound quality.
@@kingzor100 If you let an inaudible wave interfere with an audible one the resulting change is still inaudible. The energy doesnt get magically transferred down to the lower frequencies.
Thanks for this explanation. Here’s an anecdotal story of why 16 bits isn’t enough, and why, as a professional video editor, I always do my final mixing at 24 bits, even if the tracks I’m working with start at 16 bits. A couple of decades ago, I put together a “reference” audiophile CD for an audiophile club. I ripped tracks from commercial CDs, assembled them, then burned a compilation CD. The copies sounded like the originals, but the live tracks had applause cut off, and levels of some tracks were too low. So I brought the ripped, uncompressed AIFF files into a Pro Tools project to fix the minor issues. I added fade-ins and fade-outs to the live tracks, and adjusted levels on a few tracks to make things more consistent. I spit the problem tracks out of Pro Tools at 16/44.1 and burned new CDs. I was proud of myself until I listened critically to the results on headphones. All the tracks that came out of Pro Tools sounded noticeably degraded. It was not subtle. When doing final mixing on my video projects, I typically use EQ, noise reduction and compression/limiting. I can clearly hear my processing degrading the 16/48 tracks, which limits how far I can push the processing if working at 16 bits. At 24 bits, the same mix with the same processing sounds cleaner. To me, it’s not subtle. Would my clients care when listening to my deliverables? Probably not. But I care because I know what to listen for and can easily hear the difference.
I'm an audiophile purest. I only listen to music played back on my "Webster-Chicago Model 7 Wire Recorder". It's the only way to truly experience the musical message that the artist is trying to convey.
As a HiFi guy I used to be totally convinced that 24 bit listening was better than 16 bit, but then I watched an interview with a mastering engineer. I heard him say that he masters differently for CD than he does for Hi Res tracks on HD Tracks. If that's the case for even some of the mastering engineers, then the argument that 24 bit sounds better (based only on it being more bits) is blown out of the water. Also I have more than one example of 16 bit masters of albums sounding clearly better than 24 bit masters (mastered by someone else). It's about all of the production, and not simply 16 or 24 bit. In general you might be better off with 24 bit releases, but that's only because more attention might have gone into making sure it sounds good, given the known consumer of "Hi Res".
I've had redbook 16-bit cd players with recordings (playback) that sounded better than a 24 bit DVD-A or dsd 64-bit SACD. The quality of the recordings is key.
Exactly correct. I have some 16 bit steely dan thats better then 24bit, it boils down to the recording. But a clean dsd 128 is going to be better then everything else.
The difference in masters is probably way bigger between vinyl and CD, some people like vinyl more than CD (not me) and make the assumption that its just the playback format in it self rather than being a totally different master (it must have a different master because of the physical limitations in the vinyl format) sure the reproduction source adds to it but it is not at all the only reason why it sounds different.
@@Whit-mh9nt Ah yeah, and if you would put the vinyl master on a CD or other digital format with god resolution it would likely sound very similar but without the crackle and pops.
Y'know what happens most of the time? Home audio listeners end up with a 16 bit recording sitting in a 24bit container with the lower 8 bits either zeroed off or randomly making noise.
@nrezmerski I am using pro-recorded drum sample libraries for my edrums and there are some libraries with ... really tons of noise. Like on a cassette. I contacted the support because I paid a lot of money and you know the adds of such companies... Nah, I got a $15 voucher for a MIDI Pack. Anyway ... so pro audio engineers and I hear humming actually on some presets on a normal volume level when I am playing. Ok, there are presets were there used compressors etc. and then ... So this discussion is needless for me. Really, if you see these huge consoles and patchbays and cables.
I think here is to compare 16-bit to 24-bit on the dB level is not mattering to much. The signal amplitude will stay the same, but basically you get more steps of the amplitude to recover the original signal. Where white lower bit rate you have to add additional noise in form of ditter that you don’t notice the difference, good examples comparison you can also find in pictures.
I've just recently discovered your channel within the last week and after several of your videos, I'm really enjoying the don't-take-it-so-seriously attitude you bring to your posts. Dispassionately examining the rhetoric is very helpful in dialing back the outrage on a topic like 16-bit vs. 24-bit. Even if I can hear the difference in bit rate, I'm more likely to hear glaring differences like room noise or poor mic placement, even when I'm listening analytically. Not to say there isn't a difference, but as you point out, there are some differences that don't make that much difference to the end user. But besides that, no single parameter on its own (bit rate, sampling rate, noise floor, etc.) is the silver bullet that provides the end-all definition of what makes a good recording. (To be fair, the misuse and abuse of some parameters can indeed yield a definitively bad recording, but that's not the same as the hair-splitting between 'near-perfect' and 'nearer-than-that-to-perfect'.) That said, from time to time I still listen to -- and enjoy -- CD's, LP's (to a lesser extent 45 and 78 RPM singles), cassettes, open reel tape and even 8-track cartridges. I call it the 'fidelity-to-what?' principle. Music is such a subjective topic when it comes to nailing down what makes it enjoyable for you, and it's mainly an emotional aspiration rather than intellectual. So if I'm willing, perhaps even prefer, to listen to a cassette from the 1980's or a cartridge from the 1970's, I may not be in pursuit at that moment of 'absolute fidelity to the original sound', I may be seeking the same feeling I felt the first time I heard that song, and remembering the people I was in company with at the time. Thinking of the car I drove or the place I lived. Or in the case of media that predates my own life, experiencing the sound the way contemporaries of that technology did. Fidelity to a memory or emotion, if you will. For example, I remember hearing certain songs on CD in the 1980's that didn't sound quite 'right' to me, because they didn't match my memory of hearing them the first time on AM radio in the early 1970's. Expanding from mono to stereo, losing the noise and static, and widening the frequency range emphasized different sounds in the recordings, but often the speed was off, seemed slow. I later learned that some Top 40 stations had a habit of running their turntables slightly faster, perhaps 46 or 47 RPM, so the music would sound a bit livelier (and by gaining a few seconds per song, you might have another minute or two per hour of advertising time to sell). But the point is, better audio fidelity actually interfered a bit with my enjoyment of some songs until I understood what was going on, simply because they didn't match my emotional memory. (I expect this generation may come to feel that way about 128K MP3's.) By the same token, as an audio engineer (and enthusiast), I'm not against creating and listening to the best we can possibly muster in the sonic arts; in fact, I think we should. Right now I work in a small studio that produces mainly podcasts and audiobooks, and I'm always looking for ways to eliminate audio distractions, especially noise. But I take the view that while we should obviously tend to the technical parameters, we should do so in service to the content we are recording without getting distracted into making the technology the main thing. Yet, in all things, charity. Whatever format a person likes and what they are willing to spend on it is fine with me. If I had elephant bucks to spend, I would likely purchase more classic gear, because it's fun! But I will also admit that even then, the grail for me would not be ALL about the sound alone...some of it would be visual, tactile, nostalgic, and again, emotional.
What brought me to watch this entire video, and its arguments, basically reinforces my experiences of digital in another arena… digital imaging… it’s the same technology, just catering to other senses. I was hoping for a clear cut resolve to a single question - are our eyes or ears better at distinguishing noise from detail. After seeing the video I am undecided, but it seems that our ears are actually better at resolving than our eyes… digital images at 16 bit are the absolute precipice of imaging today, well capture at least… scanners go beyond, but it matters little, as our displays are 8-bit based… which begs a different question, maybe more apt to the original… is it the displays which are inferior enough to limit our experiences of digital imaging? In this case the arena of digital sound and digital imaging, go complete circle… in the end, maybe it’s not the source sampling which creates issues, it’s the delivery of the source to us!
For digital recordings lower noise allows greater clarity and separation to be heard or percieved on a good system. The problem is that the sound is so good that you don't need to concentrate to hear it. On vinyl recordings however the very high noise level forces the brain the concentrate and strain to hear the detail behind the noise and therefore the hearer is encouraged to listen to the music more actively and often perceives this as better sound quality even when it is not. Experiment... take a top quality 24bit recording and mix in some snap, crackle and pop from vinyl and it instantly sounds better to an audiophile. Odd that.
i've listened to 24bit recordings on 24bit hardware (the high end soundblaster cards were very expensive.. so i got them second hand..) and i cannot tell the difference between a 24 and 16bit recording of the same song... might not be listening to the right music, but, old CD releases of movie soundtracks from disney, is i do believe, (disney used to know what they were doing) as good as home audio will ever need to be... listening to the Lion King on cassette i knew when the intro chant was coming... on CD it still makes me jump... The Pirates of the Caribbean soundtracks are really fun to use to demonstrate what 'dynamic range' means and what the 16bits of a CD can really do... as far as decently mastered audio goes you dont need more than 16bit.. i have no idea what kind of godlike ears one would need to require a medium that encompasses the entirety of the dynamic range of a jet engine starting from cold and going full throttle.. at that point you don't need 24bits; you need hearing aids...
I seem to remember early CD recordings or players only supporting 14 bits because 16 bits was a little ahead of the technology. What I do remember is that the Magnavox player I bought in the mid-80's made music sound brittle. Thankfully it self-destructed after a couple years.
We have SONY to thank for raising the audio CD spec from 14 to 16 bit. Philips' original _proposal_ for the CD spec was 14 bit. But by that time Philips was already too far advanced on the development/production of their single-channel 14-bit TDA1540 DAC. So early CD players based on the Philips TDA1540 DAC had to make do with the 14-bit DAC (+ 4 x oversampling thanks to the SAA7030 filter) for playing back 16-bit audio CDs. Being single channel it also needed to be multiplexed between the left and right channels. In 1985 Philips released their first 2-channel 16-bit Audio DAC on the market, the TDA1541, and carried forward the oversampling technology with its matching SAA7220 4 x oversampling filter.
One of the first consumer digital recorder made by Technics/Panasonic (based on VHS tapes no less) was using 14 bits - which back in 1979/1980 was considered good enough. Sony pushed for 16 bits as it was a round number and exactly two bytes. As explained in the other comments, Philips released it's first DAC TDA1540 as a 14 bits DAC. But thanks to the magic of noise shaping (it is not just over-sampling) it could achieve true 16 bits resolution. This is the path that later lead to 1 bit DAC - what Philips did in 1985 still amazes me to this day, and I have a couple of CD player based on the TDA1540 and they do sound nice !
More bits equals less quantization error. This applies across the entire dynamic range, not just in the inaudible portion that is also being captured, stored and reproduced digitally. I don't understand why people who defend the idea that there is no audible difference between 16 bit and 24 bit, cling to an argument that you can't hear any difference across the entire dynamic range because of the differing noise floors, then offer a test that takes place below the threshold of human hearing to validate their argument! The funniest thing about that "null test" is that it applies dither to the 16 bit signal, which further reduces the noise floor of the 16 bit signal and invalidates the test.
@nrezmerski In this video it is conceded that more detail is captured in a 24 bit recording. It is not categorically stated that there is no difference in sound between 16 bit and 24 bit. That's because there is a difference. What is being saterised here in this video is this very small percentage of the music consumer population that care to listen for the difference and can actually hear it. They are characterised in the video as being pompous audiophiles that believe their choice of chair while listening is also a prime consideration in the pursuit of audio reproduction excellence. I think it's a brilliant video, because he is saying that people like me who can hear the difference actually have a point, but it annoys him so much he has to be mocking while making the concession.
"The reasonable amount of noise from LPs..." I never found it reasonable. It annoyed me when LP was all we really had. I could not drop LPs fast enough when digital audio came out myself.
I suppose that depends on factors such as thew condition of the LP, etc, etc, but just knowing I was degrading the LP by listening to it (said stone on plastic...) drove me crazy. CD was love at first site for me, even though the early ones didn't always sound all that good. @@AudioMasterclass
Really? Must be something wrong with your records or the set-up. The majority of my LPs sound almost as quiet as CD. Clean records and well kept stylus is a major part of it - with a good deck and tonearm.
To add to this conversation: I recently listened to Houdini by Enimen on Tidal and Qobuz. Both were playing at the same 'quality' 96.0kHz. The Tidal version sounds markedly different to me, better.
Your screen name is Undead.Audiophile and you're trying to bash audiophiles with the same so-what argument as everybody with a so-what music player. Most people complaining about 'audiophiles' are taking themselves way more seriously than anybody I have heard in recent years calling themselves an audiophile. The 'snooty' people are now the ones who come to forums like this crying about 'my shitty system is good enough for me and it should be good enough for you, but you think you're better than me and I'm so mad about it'
@@jamescarter3196 😆😂 I'm not snooty or do I take myself seriously. I find it quite amusing when strangers react like this to a single post. Why am I (a stranger) lumped into a group that seems to aggravate you. I don't know you, and you don't know me. Audiophile is in my screen name because I like all the tech and gear, and how it makes my music sound different for each genre. I will never go broke for gear. I on average listen to music anywhere from 6 - 10 hours a day, 5 days a week while at work. I listen on various tws earbuds and headphones, and with numerous portable dac/amps and normal IEMs and headphones. Maybe I should be a musicphile instead.... Anyways my 3 sentences set you off.... And I'm the complainer...lol You do you...... I'm going to listen to some tunes....🎧
This was fantastic when you consider that the Audiophiles will in the same breath say that vinyl sounds far better than 24bit audio. So about half the detail of their 24bit sound file,, that they will say FLAC sounds worse than WAV as there is some form of data compression in haw it’s encoded, but that’s a separate comment. Glad we didn’t take into consideration the noise generated from not using medical grade electric plugs, and how we skimped out with consumer usb cable to our DAC we bought off Amazon.
It's about how many bits are being used as well as how many are used in the system. Low level signals use less bits, this adds distortion. This is more noticable when using smaller word sizes.
16 bits are adequate to make noise-free recordings. 24 bits gives us ‘slack’. For example, an engineer on a live date doesn’t get the opportunity to set gain before the converter and notes the channel is peaking at minus 30. No problem!! Feeding a 24 bit converter at -30 or even -40 is perfectly valid. Just leave it and enjoy the song. 16 bit would be deep into the noise at -40, completely unacceptable. That’s pretty much it. Detail doesn’t really enter the equation as long as you are not in the noise. A perfectly set 14 bit converter captures the same detail as a 24 bit converter because we can’t hear more than 85dB of dynamic range on a single channel contributing to a stereo mix. This isn’t just theory, this is memory. I recall those days when success was tracking at -2. It was stressful and resulted in overloads nobody needed.
1:30 in, You make cutting through the BS one of my most enjoyable listens on UA-cam. The fact that these chairs all have headrests make it the joke of the century...
dither is needed any time the bit depth is reduced to avoid quantization noise. combining multiple 16-bit pure digital signals into a single 16-bit output needs dithering if the bit depth is being reduced. I would love to have a listening environment available where -96dB against a max reference level (100dB SPL?) is even detectable; maybe I need to find a local audiophile who has this. from my experience, dolby C does a decent job on cassettes, and dolby S is comparable to noise levels on CDs (we'll ignore wow and flutter), but I'll still take dolby B over nothing at all, especially on type I tapes. I blame the current unavailability of new competent cassette decks and limited availability of used ones coupled with misplaced nostalgia for the sad state of no noise reduction at all on most new cassette releases. ("but it's analog!" yes, but my otari has a lot more hit points than your nakamichi.)
Good that you mentioned this, I have a dire straits boxed set which I play through a Marantz CD75, the sound quality is much better than all the others I wondered if it was in 24 bit Quality is amazing
In theory lower noise floor means more low level detail.. One can often easily tell the difference between 16 or 24 bit files just by listening loud to the last few bars when the track is fading out. Maybe you can provide a demonstration of this effect. This of course is of little consequence for normal listening.
@@triple_x_r_tard I know, and my point was, if you have to amplify the lowest amplitude bits to try and hear them, the test is moot. When normal amplitude are to loud for your ears, trying to figure out if you can listen to lower noise is just silly. You don't hear it in a regular playback...also its moot because at any noise level, if you crank it up enough you will always hear some noise.
@@AudioMasterclass You may be able to provide downloadable files as a way to get around YT. They won't need to be longer than a few seconds each. Although I feel if you increase the levels enough it will be heard regardless of YT shenanigans.
I've just moved from a 24-bit recording interface, to one that does 32-bit floating point, and my DAW software supports that. I haven't had it long, so I've only done some preliminary testing. Aside from the convenience of having so much dynamic range that you can essentially set your mic levels after the fact, there is a bit of an audible difference. I honestly wasn't expecting to hear a difference, but when changing only the interface, the 32-bit recordings sound more 3-dimensional, more dynamic, and like I can hear farther into the mix. It's a subtle difference, but enough that I noticed it when I wasn't listening critically at all.
@@BeatsbastelnI moved from Apogee to MOTU. Both interfaces have the same DAC chip. The difference in the AD stage is a 24-bit AKM chip in the Apogee vs a 32-bit floating point ESS chip in the MOTU. Both units have virtually identical THD, noise, and dynamic range specs on the input.
32 bit floating point is only 24bit mantissa... so if you think that you are doing yourself a favor there, then you weren't paying attention in your numerical methods class in university. You would be infinitely better off with proper use of a 32bit integer library for signal processing.
@@lepidoptera9337 Thankfully, I was a Computer Science major who paid attention. I’m glad to explain how 32-bit floats can describe lower-level signals, despite having a 24-bit mantissa. It’s because they can have negative exponents. So if the lowest-level signal that a 24-bit integer can describe is 1, a 32-bit float can describe a signal as low as 1.2x10^-38. A 32-bit float audio file has a theoretical dynamic range of -758 dBFS to +770 dBFS, and across that range, it can describe every amplitude with the same granularity that a 24-bit integer file can describe a dynamic range of just -144dBFS to 0 dBFS. This pushes the noise floor of a 32-bit float file off into complete irrelevance, and also means you can record at any volume level, and set mix your levels later.
@@lepidoptera9337 Just to follow up on the integer vs float portion of your comment, with PCM audio, each bit of integer gets you 6 dB of dynamic range, so with 32-bit, you can have 192 dB of dynamic range if you format it as an integer, or 1528 db of dynamic range if you format it as a float, both with the same resolution.
I will probably have to run some more experiments but it does not sound the same to me. I just downscaled some Led Zeppelin. Not a big difference but clearly more smooth and finer definiton on 24bit. And I listnen on low volume and notebook fan in front of me. IMO it all depends very much on the listening enviroment and loudness too. The more loud you play the more detail those extra bits shall reveal. As to enviroment the more quiet the more we can focus on music detail. Last not least the oversampling magic which is always there is designed to trick our hearing somehow too ... That should not be forgotten . I am professional musician so my ears are long time trained but by no means I can relate to those "pros"| stating that they can not distinguish between 16 and 24 bits.
@nicksterj That is simply not true. A good microphone has close to 120dB dynamic range. Very expensive condenser microphones are specified with 132dB. That's equivalent to up to 22-23bits. This gets even worse when you are producing multi-channel recordings, because now the noise averages out, but the signals may not, so you can actually get more dynamic range out of e.g. a multi-microphone orchestral recording than out of a single channel.
@nicksterj I have an audio system on my table right now that has close to 120dB SNR. I don't know what you are talking about and neither do you, it seems. If you actually read my posts then you will have noticed that I said that 16bits are enough AT CONSTANT GAIN. The production process is the opposite of constant gain and that's where you absolutely need higher resolution. That I don't turn the volume knob while listening to classical music is not true, either. Do you know why? Because I have a noisy apartment. My local noise floor drowns out the pianissimo, hence I have to raise the level during those passages to hear the music and that is when 16bit artifacts become quite audible.
@nicksterj If you are recording in 24 bit and you are producing in 24 bit, then why in the world would you ship in 16 bit??? I don't get the logic here. Is there some sort of pandemic of the inferior file format virus going on? Did people miss that one can get 512GByte USB sticks these days? What is wrong with all of you? ;-) Let me repeat this, again. 16 bit has horrible and audible distortion during pianissimo passages. If you can't hear that because you have a stereo system where the volume control knob has been welded to the front panel, then maybe you shouldn't be in a discussion about audio formats. ;-)
@nicksterj It's not my problem that Fauxtaku doesn't know how to design audio electronics correctly. I do. My systems are close to physics limits. I am not even using anything special. It's just the manufacturer's ADC and DAC test boards and a few home made amplifier stages with properly chosen low noise opamps. Stuff that every electronics design engineer should be able to do in his or her sleep. This ain't amateur hour over here. ;-)
@nicksterj I gave everybody the evidence. Get a recording with pianissimo passages and turn up the volume. It's as simple an experiment as one can do and it works. At this point you are just like an obstinate kid who doesn't want to eat his dinner, kid. ;-)
Disk size is cheap, BUT STILL, with often 192/24 disk space goes out of control. I decided, based on quality or importance of the material to bring the flacs down to 96/64/48 or redbook, also some to 19-22 bits via dithering.
I record location sound for TV/Video and getting the recording levels right is neigh impossible, so I have be very defensive and usually record with peaking at -12dBFS to -18dBFS. Thus I loose a lot of the potential dynamic range. Therefore I record in either 24bit or 32float to keep at least 96dB of dynamic range. After mixing and mastering I can utilize the full dynamic range of the output media and then 16bit is fine. But for some strange reason 24bit 48kHz have become some sort of standard in the video world.
In this day and age, there's a lot of room correction, digital crossovers, surround sound or other processing, which is applied to the final recording. Not just during production. God knows what other things there will be in the future. 24 bit recordings provide an extra layer of "rounding" to make sure the 14 or 16 bits' payload will still get to us as well preserved as possible, even thirty years from now.
Those systems can take the 16-bit input and work in 24 bits from there on, including for output. This gives them the room for rounding etc. The only issue would be the dithering added to 16-bit, which would become noise that the system would also treat like signal. Once you add dithering you're basically saying that it's finalized and should only be sent straight to a DAC. It's putting assumptions about the output system into the recording.
Back in the early digital days, I was at a friend's studio where he had a 12 bit sampler. I asked if that wasn't too poor quality to be used in making recordings. He told me (and it made sense to me) that any samples that were used would be inconsequential relative to the rest of the mix.
12-bit samples are noisy or grainy or both. But since this was all that was available at the time, producers accepted this and made music that worked with the 12-bit texture.
While the extra dB is great I prefer having the extended dynamic range of frequency range / response to play with with 24bit - over 48khz . But 1644.. it's fun to record to and people don't generally notice if you pull it off balanced.
I find your videos very entertaining and informative. I'm the kinda of idiot that would blow a load of money on high end gear just because on paper a number is higher than another number so it has to be better, even if to my ears there's no decernible difference. Your videos keep me more grounded.
Any audible differences between 24 and 16 bits are likely caused by the lowpass filter of the dac @44.1 kHz sampling rate, not by the difference in bit depth.
there is a diffrence in music 16 bits sound boxy 24 bits is still in some kind of box feeling but with 32 bits it dissappear the noise cliping is gone basically
I don't think I’d make the difference when I listen to music. On the other hand when I play (digital) piano I’m using an acoustic simulation program named PianoTeq that uses high resolution (32 bits 92000 Hz) which I listen with a 24/96 audio interface and headphones. When playing it’s important to forget that your instrument isn’t real so any realistic detail has its importance, be it the tiny reveberation of the sound once you’ve released the keys, or sympathetic vibrations of loose strings (PianoTeq allows you to adjust the wear of you piano to make it sound bit less brilliant and a bit more convincing, so there may be loose strings). I should try to downsample a recording and see what it does to those tiny details that have no importance for the listener but have some for the performer. Maybe there is a placebo effect or something.
I have spent many hours comparing 16 bit to 24 bit when using the same DAC, and same sample rate. The difference is often very significant when listened to on good gear. I am currently using a MOTU M4 to 2 hafler dh220s to drive kefXq9s. Most albums that were recorded in 24 bit will sound substantially dryer and less rich when played at 16 bit. There is also usually a lot of information from the decay of sounds that is chopped off with 16 bit. So, the the difference in 24bit is that it is not only less dry, more spacious, and richer, but there is more information about the echo and natural decay of sounds in the recording. The increased dynamic range is not always noticeable, but the attack of something both natural and abrupt, like a snare, for example -- will sound slightly different at dynamic ranges above 96 db, even at low volumes.
@nicksterj based on what? Have you actually taken time to compare this on proper gear with all other variables controlled? If you are confident that I am wrong, I am happy to meet to do a blind test that you administer. And I will bet you whatever amount you want that I can identify them correctly in an A to B test 10/10 times. And I won't even be paying attention to the noise floor. I'm also willing to bet that you would hear it, too.
@nicksterj the data is lost once you convert it to 16 bit. Have you actually just found a song that is recorded in 24 bit, and then toggled a setting on your system or the app handling it and listened to the change?
@nicksterj oh, i guess i am not paying attention to which video this is under. But I see that you are going far out of your way to find any comments that prove your youtube video wrong. Also, I am not married to this information. I haven't gone on youtube to make embarrassing videos about this. But that does explain why this is so important to you. I just confirmed for myself what the difference was, then went to sources that talked about this issue, and then was surprised when I saw so much objectively incorrect information in the threads that I made simple comments so that people won't just read the misinformation. But that apparently triggered you. lol
@nicksterj well, it sounds like we have the same goal, then. So, I am happy to bring this to a conclusion. And I am sorry, but no. Your videos only prove that the creator of said videos is incapable of a scientifically valid comparison on so many levels that it became a masterclass in how to fail at providing meaningful data. and that includes your video that you posted. You haven't address any of the rebukes of those videos, but you have doubled down on the misinformation with blind assertions. I'll address your points that you are making in the other video. But maybe it's easier to just have this discussion under your video? i know that might be embarrassing there, but honestly, I am doing you a favor if you finally realise how your video was flawed and take it down before more people see it. up to you, though. I wont pressure you to defend yourself on your channel.
Oh it was worth hanging on through all of this JUST for that ending. 😂 You brightened my day and lessened my DAW worries both at the same time with this video. Most grateful for your simplified explanation. It's pretty much all I needed. Now to chase down what the heck is causing my glitchy pops and crackles in mixdowns. Now I know it can't be my 16bit 44.1kHz limitations. Processing power maybe? Just bad software?
Firstly, let me say that this comment will make it fairly obvious that I can’t be considered to be anything approaching an audiophile or technically au fait with matters discussed regularly on this channel, but it illustrates the reality of the (my) aging process. One of the “joys” of getting old and turning 70, apart from developing a taste for droll, witty and, at times tongue in cheek UA-cam channels, is that you lose the ability to hear above and below certain frequencies. So when I sit down to listen to music, taking extreme care not to get too excited and turn it up too loud thus activating the tinnitus that’s always just lurking in the background in my ears, I just don’t know what I might be missing in those frequencies, what’s more I have no idea if there’s actually anything there in the recording and as naïve as my attitude might be, I don’t care, I still enjoy the music. As to any noise I may hear, which I admit I find quite irritating when I do actually hear it, there’s a chance it may not even exist in the recording but be a product of my less than audiophile system or possibly the “head noises” that old age brings on, I just don’t know - but some things you’ve just got to accept, like getting old or, say, never achieving perfection.
Yes, of course I freely admit I am neither "technically proficient" nor technically "up to date or fully informed" regarding the matters you discuss, but it's still a lot of fun to watch.@@AudioMasterclass
I think there may be another factor at play why sometimes 24 bit is perceived as better: ISC, (Inter Sample Clipping) or lack of it. 24bit usually goes along with higher sampling fq, at least 88.2. At 44.1 ISC does occur on tracks mastered too hot. at double or higher sampling fq, thus more samples, the DAW would show the signal exceeding 0dB F and thus one would lower the level, thus fas less ISC and the DAC's reconstruction filter can restore as it should. Obviously, at 192kHz even less ISC (and at frequencies where it wouldn't bother anyone). Perhaps that is the better detail, more precise effect one hears with higher bitrate recordings? not the the bit count, but the sampling frequency is what makes the difference. just a thought...
Intersample peaks leading potentially to intersample clips are an issue. However because it is increasingly common to use true peak metering we can soon hopefully stop worrying.
The cheapest and most effective way to improve detail and soundstage is cup your hands behind your ears 😂 whilst seated in your audiophile chair in a dark room😂
Actually possible in theory, but it is not a tape fault; it is either extremely poor tape/head alignment (I don't mean azimuth) or a faulty/worn actuator for the rotating head in an auto-reverse mechanism. I think in theory a badly worn capstan bearing could also cause this by pulling the tape at a skewed angle through the head tape guides, enough for the adjacent (B-side) track to just touch the edge of the head gap, but that _would_ go paired with a significant azimuth error and loss of most high frequencies.
I think it depends on the persons whole set up, I have a pair of Sennheiser HD800 headphones and I used them for a while directly connected to the head phone outs of my RME HDSPe AIO sound card in my studio, I used to get noise at 16 bit or 24 bit due to the connections, I have since upgraded to use the SPDIF out and a Mojo 2 Amp and I can't hear any noise on either, but detail in sound is very different in my experience, using 16 bit in cubase with VST synths sounds really poor, but switching that to 24 bit is a huge difference when producing music the virtual instruments just sound so much better and the extra head room is a blessing.
@nrezmerski yes I have done that before and I can tell the difference, tbh on any other headphones I have had that wouldn't be noticeable but it's really obvious on these things.
@nrezmerski the difference is in how it sounds not the numbers, use your ears, try working with 8D audio, the difference is so clear, 16 bit sucks, sorry for you that you can't hear it.
@nrezmerski I have just been A/B testing my self and the difference still seems clear to me, VST Synths stand out, and tracks that are fully mixed with 8D especially with sounds put in the 360 degree sound space about 127 degrees to the back left or right, the positioning of the sound isn't right when converted and high frequencies have a grittiness to them, also different conversion options with Dither and Noise shaping in my software have different results some better than others, but with out a blind test set up and a third person testing me I am going to have to concede I can't prove it and given the evidence of others being A/B tested and failing then I can't assume I some how have some super hearing :-D I am interested to set up a test for myself at some point but until then this will remain unresolved for me, I feel I can spot the differences more easily on tracks I have been working on for weeks vs someone else's music so I wonder if that can make a difference if someone knows the fine details of a track, anyways I will keep an open mind to this in light of your comments until I test myself properly.
Some of the best DAC's can only achieve 21bits if you are lucky a lot reach less so 24bits is lost anyway, the majority of recorded music doesn't even reach 16bit once mixed and mastered, if you were to hear the noise from a 16bit stored recording your system would have to be pretty damn loud probably beyond normal listening levels too, also your analogue components would have introduced more noise and distortion before it reaches you anyway.
Ummm ... no. The "21 bit" rating is for analog clarity and noise floor... It's another confusing spec. cooked up by confused people to confuse other people. A 24 bit digital sample always uses all 24 bits and a 16 bit sample always uses all 16 bits.
@nrezmerski Well yes. Low level noises recorded right into the files. But that lower 4 bits is at -120dbfs and would be totally inaudible anywhere outside an anechoic chamber. Certainly not audible in the average listening room.
A record has if you are lucky something around 50 db of dynamic range and it turns out that most hifi systems can't reproduce it because they will already start to compress and distort the signal so 16bit is already completely overkill and 24bit is not changing anything at all. It is like debating if Thor or the Hulk is stronger. Fun when you are 13 but at some point you grow out of it. And the correct answer is the Hulk of course ;-)
32bit mastering is the new no1, just in the same way as 32bit colour is in windows, this is EXACTLY the way you should look at this topic, less bits=less colour. with 16bit you just dont see the full picture. 16bit dithering is no different that a blur effect of a low res image to get rid of the artifacts. and no "audiofools" dont have better hearing, they simply have more money, in my vast experience of audio i can tell you the best is NOT the most expensive. also another thing we only listened to cassettes back in the day because there was no other option.
This is a very poor analogy. We can SEE more colors, we CAN’T hear the difference in 16 bit vs. 24 bit especially when you consider the noise floor in any room you can imagine. After all, no one listens to music in an anechoic chamber.
@nrezmerski Never the less, its still adding something to the audio that shouldnt be there. A sort of "band aid" theres absolutely no need to use 16bit today, this isnt 1982
But Audiomasterclass didn't go into 32-Bit PCM and 64-Bit Float and they are the new mainstays for 1-1 audiophile recording now. 16-Bit and 24-Bit are both out of date Wave recording technology in your latest Audacity. So where does the headroom change the quality in 32-Bit PCM and 64-Bit Float. lol
Non dithered 16-bit audio dynamic range is 90 dB. Not 96 dB. Full-scale 16-bit audio signal can only be reduced by half 15 times without going below the noise floor. Reducing it by half a 16th time puts it below the noise floor.
@nrezmerski The 96 dB range stated in the video is a reference to non-dithered 16-bit audio. My comment is in re: to that. Not to make a point of whether or not it's an issue.
@@xuser48 20*log(65536/2) = 90.3 The range of 16 bit digital audio is -1 +1 to -32768 +32767. Both the + and - are amplified the same. Thus there are 15 x 6dB steps. Not 16. The minimum level that is not below the noise floor is -1 +1. A range of 2. The maximum level that is not clipping is -32768 +32767. A range of 65535. So the max to min ratio is 65535 to 2. 20*log(65535/2) = 90.3. The 96.3 that people keep stating is a non-real world theoretical that could only exist if the noise floor was 0. In which case the dynamic range, and gain too, would be infinite. Not going to happen.
I spent 5 sec deciding on 16 vs. 24 bit for recording and hours obsessing over mic placement. Mic placement has made more auditable difference to my recordings that bit depth. Spend your time where it really matters.
Because when you were in your room she couldn't hear your noise, so she didn't give you dither. When I was 16, in our house the floor was noisy upstairs, so with the noise floor being up we had to have dither downstairs
Hi there, Thanks a lot for your instructive videos. I wonder to know as the CDs are still 16bit/44.1kHz, is it worth to record raw files with 32bit/96kHz ? Thanks for your feedback. Greetings
@@AudioMasterclass Thank you for your kind feedback. The CDs are still 16 bit. So I have to convert files 96/32bit to 44.1/16 bit. Why I hear people advising to record at 88.2kHz as the multiple of 44.1kHz ? I am still asked for to produce CDs which are still 16bit/44.1kHz... It's a jungle. Thanks again...
In the modern world of mp3, loudness war, music played on smartphones, I consider myself an audiophile, but 16/44,1KHz is enough for me, because the background noise present on the master recordings is always higher than the ground noise of the 16bit. Than why should I bother with 24/48 or 24/96Khz tracks, sacrificing more storage space? Not to mention that it is already difficult to find records that exploit correctly the entire 16 bit range. Maybe the difference can be heard on 24bit files of some classic or jazz music, but they are not my genre.
If we stay in the analogue domain first, before addressing the video's question, then let's first wonder what "bandwidth" is needed for a 440Hz music note. That's a loaded question, though. Play that central A note on a piano, violin, flute, clarinet, saxophone, etc. Each time it is 440Hz, right? But none of these have perfect sines as wave shape. So there is information on the 440Hz that we can analyse so as to recognize the instrument (voice, kind of sound), and we can place it somewhere in space. But to recognise individual voices/instruments must be learned and that takes many learning moments. If you never went to a live classical concert, then you don't know what is wrong with a recording of that (if there is something wrong). Imagine the clarinet produces a block wave and the flute sine wave, but a cello has a sone wave with a couple simple spikes in them. Here we have left the 440Hz as each of the deviations are in a much higher frequency range. Our ears are physically limited to about 20kHz, when measured in sine waves, but that is a coarse abstraction. Whatever the point of the 20 kHz, in electronic recording and playback, the question is not what we can hear - assuming an exceptionally well trained listener - but what bandwidth is needed in an amplifier, cross-over filter or speaker to play a perfect 440Hz block wave, or to get the two tiny spikes that make a cello sound like cello, sound like a cello in playback, because the tiny spikes are at the right place and in the right form. Because - phase fault, if the spike shifts it's no longer a cello we hear, and if the shape distorts, it's no longer a cello too. And the same applies to aunt Lucy's voice, or uncle Stephen's. I saw a pianist in a Steinway grand piano parking room of a concert building - 5 model D grand pianos. She had to select one that best matched her music in character and best matched her touché in its action. She played all five and about number 2 said, "I played this one last year". I saw a blind person in a new large room that asked, are there windows there, pointing at windows, and is there an open door over there, pointing at one. The auditive brain is much faster than the visual and it relies on wave shape processing in an acoustically relevant to human survival band. Electrical engineers underestimate how good it is. Listeners accept electronically mixed recordings that have "micro-dynamics" for left right but conflicting phase information. This all goes back to wave shape. If you ever saw how good an amplifier can play back a block wave at 20 kHz then may be aware to never have seen a block wave, but spiked versions with over/under shoot, or saw tooth, or sine approximations based on a perfect block wave input. My pre-amplifier as a bandwidth of 1MHz. The power amp "only" of 150kHz with peak current delivery capability of 60A per channel. An electrical engineer once responded to my reasoning about wave shape that Fourier analysis could create a block wave with only five sine waves. Well, you need five half sines each of different frequency to fill the positive block and there is no amplifier in the world that can do that. I guess. And, what signal source does the Foerier transformation? How many bits do I need? What sample rate? Lets go to the bandwidth requirement first. You tell me. Oh, a big problem in life and science is that people think something is not true when they cannot imagine it to be true. Remember the geocentric model of the cosmos? The problems are all over the place and humans are very capable of filtering the message out of the noise, but that doesn't mean it sounds nice or correct. If we throw speakers into the debate then it becomes hopeless almost. With lousy amplifiers and loudspeakers we do not need a lot.
True revolution or Musical Change can happen only if we make different Sounds Speakers and headphones with different response to frequency giving different output results and current situation with low quality, high end and studio sound reproducing devices Listeners are listening not critical like we do and context is different something like just another Track or a just another Story telling through the Sounds/ Music. Like Hz the more is enabling more accurate sound manipulation and processing but end results will be none ... but if devices become more quality then maybe we will hear noticable advantage. Cassettes were blurry for me compared to Digital Disc .
I have my aged DAC with 100dB Dynamic range that specifies SnR without THD. I turn it to maximum to have maximum Dynamic range (100dB) then I attenuate the output of DAC by 30dB analog potentiometer (variable resistor potentiometer) to pass it to my amp with more than 124dB full Dynamic range/more than 99dB at 1W signal output. No question I don't hear my amp noise from my licening position since my speakers have sensitivity only 90dBA/(m*W). 0dBA - is the lowest sound the human can hear, while the amp is -9dB below 0dBA level, so it is 2.83 times less signal volume that human can hear. But I also don't hear my DAC noise since it is -100 - 30 = -130dB below maximum signal my amp can produce, and -6dB = -130dB+124dB below my amp+speakers own noise level equal to -9dBA (-15dBA accumulated). Now if I play my CD-DA player via my aged DAC will I hear any noise? No, I will not since CD-DA dithering will be between -120dB and -130dB level - way out from any human hearing capabilities. Do I need 24bit music to level noise to -150dB level? I don't think so since the amp is still the main source of noise at -9dBA level.
I think you are mistaken about how 16bit/24bit conversion takes place. It is not one bit or the other, there is a conversion process based on the original Nyquist Theorem for sampled signal reproduction. In practice there will be computational compromises in the DAW, which is why many DAWs will use 32bit floating point internally regardless of the bit level of the sampled audio. In other words, there is no masked detail when converting between 24 bit and 16 bit with competent software and hardware.
This is very interesting, thanks. I have 2 questions which are not related. Firstly, given that most, or at least many, new non classical albums are to some extent 'brickwalled' is there any point buying the more expensive 24 bit version (say from Qobuz) over the 16 bit version anyway? I don't know but I would love to. My feeling is that the potential increase in dynamic range from the 24 bit version will be more than wiped away by the process of brickwalling the album, so it becomes a waste of money going for the 24 bit version? What is the answer please? Then this ........ Secondly, if an album is available as either 48kHz/24 bit or 44.1kHz/16 bit, but not at all in 96kHz, then if we assume that it was not recorded in 96kHz then to arrive at the 44.1kHz from 48kHz, the bit rate must have been resampled, which will damage the sound at least in part (or visa versa). Would that act of resampling be audible enough to make the purchase of the 48kHz version a better choice just for that reason alone?
There's a simple way to find that out: buy both versions and compare them. If you don't hear a difference, you can spend less in future purchases. If you already have a Qobuz subscription, you can flip the switch in the bottom right corner to compare. Or better, let someone else flip the switch and make it a blind test. You'll be surprised, you may even prefer the mp3 in some cases.
I sometimes repurchase music in 24-bit format when it becomes available, and even though I definitely have the highly accurate equipment required, I perceive very little improvement (at best). Still I purchase 24-bit here and there, because (much as with 4K in the movie realm) I expect that careful remastering has taken place as part of the process, at least usually. The satisfaction of having the definitive version (I suppose that's a third dimension, David, besides noise and detail) of something that's worth having the definitive version of, is often tempting.
I use my ears and mark my CD albums from Red book astonishing down to utter trash. Out of my 13,000 CD collection I would have well over 10% that are trash. Another 30% at least are tolerable and maybe a few thousand are INCREDIBLE!!!!! @@AudioMasterclass
So what is best for me on a Computer PC. i have a sound card that can play 16-bits or 24-bits and betwen 44.1 Khz - 48 Khz - 88.2 Khz - 96 Khz - 176.4 Khz - 192 Khz ... Witch one is best do you think? Thx for a good video. / Sweden!
Close enough for most practical purposes in light of consumer product. For a more nuanced understanding, I recommend Digital Audio Explained for the Audio Engineer, by Aldrich. Technically, bit depth contributes nothing to detail above the bit in question. That is, 24-bit is more detailed than 16-bit - below about -100 dB full scale. Above that level, everything is identical. You can prove this to yourself with a null test. Take the best 24-bit file you can find, apply 16-bit dither to a copy. Compare the original and a copy in a 24-bit DAW with one channel phase flipped. There will be no signal AT ALL above the 16-bit dither noise. If you understand how digital audio works, that result and its meaning will be intuitive.
i told A Dj Friend About It, He Said It Dosent Matter People Dont Hear The Differnce But I Could Hear The Differnce Between Mixes The High And The Low Quality
OK, I confess, I'm a hard-core audiophile with a soundproof acoustically treated room, I don't have a fancy chair, I have a 3 seater on a small podium. My Audio note dac 1.1x signature is an has an 18bit AD1865 chip in it, sounds amazing. I've had dac with 24,32 bit and dsd 1024, in the end the Audio note suited me best. I remember as a kids I use to buy the best blank cassettes I could find, mostly Sony HF/UX ES 90/Metal master. I would use both the EQ and dolby b and c to tweak the sound to my liking. I loved my Sony Walkmans. I was 11 when I got my favourite headphones, the Sony mdr v7's. Love Audio Phil 😂
@@DarenLee I listen mostly to music from the 1950's-1980's so 16bit enough for me. I just enjoy the music for what it is however it was recorded back in the day. I keep away from all these modern remix's such as Re-mastered, deluxe edition,super deluxe edition and even hi-res.
@@ac81017 thanks for your kind reply. I think I get what you mean ;) I am into IEMs and sometimes with different sound signature really doesn’t goes well with old recording spending weird or fake I sound say.
Now U are Talking, MasterClasd is what Im her for!!!!! The spice of the channel some tech , My current faves 16/44>176/24 upsample same for 96/24 when 192/24 not available Nyquest/Shannon was right
I do not disagree with your axioms. There will always be people who want a good quality and they are really passionate about it, so why would I wanna disappoint them? Edit: But I do not agree that audio needs to be 24bit. My audio interface for example only supports 16bit, so I never listen to 24bit audio at home. That in turn also means I make music in 16bit. I'm fine with that for now. The music can sound great and if it was designed with this bitdepth, then it was also meant for that bitdepth. Only if someone converted a 24bit audio file to a 16bit one, while being able to listen to 24bit audio, I'd maybe feel like I might be missing out on something subtle
@nicksterj It depends what you want to do with the signal. 16 bits is not primarily more noise. It's mostly more distortion for low amplitude signals. 16 bit is worth roughly 96dB of dynamic range. 8 bits are only 48dB, so if you are listening to signals that are -40dB or less on a 16bit system, then you are getting the distortions of a 8-9 bit system... and they are awful. We are talking less than telephone quality awful. This is usually masked by the low volume, but if you want to run audio with 16 bit quantization through a compressor, then you are in for a nasty surprise. When would you want to do this? A typical application would be extended sustain on guitar samples by using a digital compressor effect. Not much of a problem at 24bit, but a serious challenge at 16bit quantization.
@nicksterj And I didn't argue that a dithered 16 bit system isn't useful as a distribution medium. It just doesn't work for much else in the audio chain. For that we need to do a lot better. I even gave you an example for which it does not suffice.
@nicksterj "Detail" is not an engineering criterion. Harmonic distortion, in particular intermodulation distortion is. 8bit quantization has absolutely horrible IM components, so every time you get down to the -40-50dB level your 16bit system is going to hurt your signal audibly and not just a little. If you don't believe me... just try it.
Any 24 bit DAC can play 16 bit files. Where you run into trouble with some older DACs that don't do 24bits is that playing the file as 16 bits strips away the least significant 8 bits.
Thank you for bringing the subject.
I happened to utilize dithering technic for low level signal detection (as well as I am highly experienced in audio signal restoration) so I'll try to simplify the explanation of dithering.
1) The reason you can't hear dithering noise is because usually a noise shaping is involved pushing the energy of the noise to the higher frequency where the ear is less sensitives.
2) Noise shaping also increases resolution of the low level signals.
3) signals that are at level lower than the least significant bit cannot be restored without dither. In order to be effective The peak level of the dither signal is usually higher than the LSB by 2-3 bits.
4) A lower than 16th bit signal is not only a signal that is at loudness level lower than -96dB it can be an harmonic with signal level lower than 1 bit, say low level violin harmonics played together with a contra bass. These harmonics are also lost (or heavily distorted) by the challenge to affect the LSB bit
5) when (only) a high frequency section of -90db level white noise is played, the LSB bit will change it's status from 1 to 0 randomly, we will get a noise at RMS level of 3dB with average of 0 volt DC' meaning and equal avenge number of "0"s and "1"s .
6) When playing a 1KHz signal @ -108db level with no dithering, the digital recording value will return a constant 0 voltage because the level is too low to affect the A/D LSB bit.
7) When adding the two signals (1kHz and White noise) together we will still get a DC average of 0 Volts, we will get a noise of about 3dB level, and the long time DC average will be 0VD. However, the digital statistics of the noise will have more "1"s during 1 KHz signal period is at the positive area and more "0"s when the 1Khz period is at the negative cycle.
8) This means that when adding an LPF for a 2KHz at the output of the DAC, we will get rid of most of the noise and can reconstruct the -108dB level 1Kz signal.
9) Luckily for us as humans, our brain can recognize the 1khz since we barely hear the noise shaped dither' so no need for adding an LPF in order to hear the 1Khz.
10) When FR is limited to 20Khz (at 44KHz sampling rate), we will be able to recognize only medium frequency which is fairly good result.
11) A High res sampling rate allow pushing the dither noise shaped signal energy to above the 20Khz range, ( beyond human hearing capability*) and allows for higher audio frequencies to be restored.
12) Is Dither needed for 24 bit Hi Res Signals? to my experience the answer is yes. it is not needed for the recording itself' but it is useful for the reproduction. not all systems are capable of accurately produce a 24bit quality of signals (due to poor DAC circuitry for example or other reasons), in other cases the recorded signal is passing through digital compression channels such as streaming or Bluetooth and Etc.
13) For my work I always use a 24bit 96Khz production with noise shaped dithering of 20 bits (4 bits for the dither).
14) * We can hear information above 20KHz: We can't identify tones above 20KHz (on our age ithe limit is lowered to above 12Khz) , this is what makes dither works: We can hear identify a "20Khz to 40Khz modulated white noise carrying a low level hi-mid frequency (it is not realty modulated, just a sum of the 2 signals)
Hope the explanation is not to detailed or too complicated.
Why not 32 bit in production?
Shure, However the subject here relates to final product. 16bits, 44.1KHz done properly, can sound amazing. so is there any added value for a 24/96 product?
@@FirstLast-is9xe
'too detailed' you're welcome, who needs a spellchecker when you're an audio genius.
@@FirstLast-is9xe 32 bit float is useful inside a DAW or similar, since it's 24 bits of data, shifted by an amplifier/attenuator that's capable of several hundred dBs. This allows you to work with a signal whose reference level is off by a bit, without losing quality; for example, if I've recorded a line level source using a 24 bit ADC while forgetting that my microphone pre-amp is turned on, I can keep the full 24 bit samples, and attenuate back down to line level; I can also take a microphone recorded with the wrong pre-amp settings and amplify it, keeping as much of the original sample set as I can.
It's also worth noting that your ADC is not just picking up signal; it's picking up thermal noise in the electronics and cabling, too. A UA-cam comment is not a good place to go into how Boltzmann noise affects an ADC's effective number of bits, but if you do the maths, you find that for professional line level signals, you've got around 20 to 22 bits of resolution before all you're picking up is noise, even with a top-quality 24 bit ADC (and similar on the output stage - the bottom 2 to 4 bits of your DAC are effectively obliterated by thermal noise).
As a result, the "optimum" is to accept that your signal has around 20 bits of information in it, and to build your workflow around that. One option is 24 bit integers and being careful with your chains of (digital) attenuation and amplification in the studio, so that you've always got 20 bits of information flowing through the system. Another is to use 32 bit (or even 64 bit) floats and then have less concern about foolish chains of attenuation and amplification, since you're unlikely to damage your 20-odd bits of source information even if you do something completely crazy like attenuate by 130 dB followed by amplifying by 120 dB (which can be replaced by a 20 dB amplification).
Thank you!
For me it's all about headroom. This is important for the recording and production phases. I was given a raw 24bit live recording where the levels had been incorrectly set for the first few minutes and I needed to bring the levels up to match the rest of the recording ... I feared the worst whilst remembering trying such things in the old world of analogue recording where the previously inaudible noise floor would intrude unpleasantly. I therefore was blown away when I simply selected the low level section and brought it up to the rest of the recording using Audacity. To my ears it was audibly perfect with no noise floor issues at all! However, an end product distributed in a 16bit format (once all the production tweaks have been done) is fine for my ears.
THIS^^^
There are also many reasons why you might lose bits while trying to reproduce the audio. For example, my Motu M4 will just throw away bits if it doesn't get the packet in time, and if you don't add latency, the sound will lose audible information. some cheaper DACs even intentionally lower the bit depth to lower the volume. 😳
I’m sure people are tired of hearing from me A EE degreed person who understands digital signal processing and audio as well as higher than audio frequency audio processing and equipment that I am involved with. The 24 bits does nothing but lower the already noise floor even lower or correspondingly gives you many more higher register bits for a more high dynamic range headroom. 24 bits is usually useful only internal to audio processing systems like DAWs and mixing console type systems because as you add any EQ reverb chorus Flanger delay, d-Essing special effects (to remove sibilants from vocals) it’s nice to have all that headroom as all those numbers get crunched together with those calculations getting on but on the final rendered output once you master it to 16 bits to make a CD there should be really no quality loss at all. I get so, so tired of the 24 bit vs 16 bit debate everybody has. Tsk, tsk, tsk. 🤨
I would just like to re-iterate, but I love this UA-cam channel master audio class it is or something because it gives you so many different viewpoints about what audio files and music can be. It lets engineers, audiophiles and just average music listeners come together to Appreciate music, and I guess we can debate and dawdle about the details, but it is just such a fun field to be a part of. I love this channel!
I'm in University for Comp Eng in my 5th semester. From what I understand the bit depth determines the quantization levels for each given sample. Now I didn't learn about quantization in systems class yet, but I can imagine that it is useful to capture either more fine grained, or a higher dynamic range with no loss of detail.
I came here cause I saw an add for some headphones that were boasting their 24 bit capability, and it just felt off to me and like an unnecessary gimmick. Also they were hinting towards having a higher spectrum which also didn't make sense to me.
I'm completely new to recording and I have a 16-bit dynamic mic with specs 44.1-48KHz and a sample rate of 192, but my mic audio is so muted compared to my other inputs.
I can level my inputs with my mic audio to make them more balanced, but then I have a quiet video. How can I boost the audio by 4-5x without risking static?
Nice video. I like your distinction between ‘enthusiast’ and ‘audiophile’. I’ve come to realise I’m the former as I don’t have the money to become the latter. I like being happy enjoying the music with my tech that is good enough. 16-bit 44.1kHz for me 😊
Why do we have to have a war on the the word "audiophile" ???
I know major recording engineers (real names) and they don't mind being referred to as "audiophile" because it sends a message that they are interested in doing quality word whereas other in their trade are just in it for turnover and money and that brought us the compression wars otherwise known as loudness wars. There is some really bad recorded stuff out there or good stuff that was then wrecked in the mastering process. Trust me, I am speaking from experience.
I don't mind being called and "audiophile" and I have good ears and claim no more. Don't make war on words, make war on bad recordings and rubbish.
@@D800Lover Agree, I don`t even know where there is this urge to attack the people in the first place. Why not just leave them alone? Sure, you can tell them that they get ripped off, but why make fun of them all the time?
@@johannalvarsson9299 - I can be as critical and skeptic as anybody, just don't throw the baby out with the bathwater, right?
@nrezmerski - And are you a music lover? What great sin to call yourself an "audiophile" because you like music to sound like real music in your own home? I say, keep it up!!! And audiophiles often spend within their budgets and still call themselves audiophiles.
@nrezmerski - The reason people laugh at other people is bad behaviour. My mother thankfully taught me right. I am not rich and I can describe myself as an audiophile without somebody making videos at my expense? That's pompous and clickbait and deposits in the bank. So I decided, that deserves some pushback. Most audiophiles are not rich dudes.
This is anecdotal at best so take from it what you will. Years ago, we're talking Pro Tools7 era, I was recording a group and they were going to get the tracks mastered elsewhere. We were about finished when they finally mentioned that their mastering guy wanted 24/48. I said, "Sorry guys, I recorded 16/44.1" This is followed by a little bit of grumbling from the mastering engineer. A few weeks later I get a call from said engineer saying, "This is the best 16/44.1 recording I've ever heard." To me that just came across as silly. I'm sure you've heard better. I'm recording in a project studio in a log cabin with very minimal gear. My only goals were the right mics in the right place and capture the best performance. That same band came back to me for their next album as well.
Don't get me wrong here, I wouldn't argue that 16/44.1 is objectively better than 24/48 but to me the difference is minimal. Prior to that time period I was working with ADAT XT20 machines. I subscribe to the concept that if you record garbage at any bit depth and sample rate you'll still have garbage.
So, as I said, take from that what you will. I'm not saying that what I did was right/wrong or better/worse, only at the end of the day everyone was satisfied.
If you record multitrack in 16 bits you can mix to 24 and get 24-bit resolution because most or all of the tracks will be lower than 0 dB on the fader. Also reverb tails.
@AudioMasterclass Pardon me if the following question seems a bit naive, I know what sounds good but I don't really worry about the technical side of things that much.
So, you're saying that I can multi-track in 16/44.1 but bounce out 24/48 stereo file assuming I'm not clipping the the whole thing out? (which, of course, I would not) What benefit, if any, would that provide? Just a little bit of extra wiggle room for the mastering engineer?
The benefit of 16-bit during production is that the computer doesn't work so hard, therefore you can have more tracks / more processes. That's the theory anyway but I doubt if many producers do this.
@@gabrielgodwin9953 When you apply effects there are math operations applied to samples, many involving multiplications (division is just multiplication by the reciprocal). These tend to product values with fractions of a whole sample step. If the project is in 24 bits or higher, these fractions don't need to be rounded off (rounding produces distortion). When the 16-bit mix is made proper dither will be applied. As mentioned, if you take a 16-bit track and halve the volume, you lose a bit if you stay in 16 bits, but in 24 bits you keep the full resolution. Or consider reducing to 75% volume. You can't reduce 65536 possible sample values to 49152 values without merging some pairs into the same output value. Distortion.
24 bit vs. 16 bit is dynamic range 144db vs. 96 db. Noise is from YOUR recording system, from the ambient noise of the room through the microphones through the mic preamps and then the rest of the mixing desk, including the AC system of your room or building. Cassette decks needed Dolby B because of the tape hiss inherent in the mylar tape medium itself which was noise at about -50db from the peak of 0db. With Dolby B, C, S; A for the pro recorders and on. You could get to a noise floor of -60db to -80db, maybe. In my DAW I can see the noise floor of MY recording system in my meters. With my blower motors off in my HVAC system and AC line filtering I can get, usually -70db to -80db down from 0db with my mics open in the recording chain. I can see it. Even with 16bit (-96db of dynamic range) I have still not used up all of the dynamic range and my noise floor is higher than -96db of the 16 bit system. I have not reached the bottom of the theoretical. Detail is a matter of sample rate so if I go above 44.1khz I can record more and I like 24 bit/96khz for all my masters. I have tried 16 bit/96khz and cannot hear the difference, but use 24 bit anyway. I read of some studios in NYC using 16 bit/96khz for their masters. I can't hear the improvement of 192khz or 384 khz using 32 bit float. It is just more work to get back to the 16/44.1 of CDs. In my DAW I have pulled some "professional recordings" from CDs and looked at them at the beginning of a track. They usually leave about 2 seconds of blank space before the music starts. I have seen in some commercial recordings a -40db to -50db noise floor before the music starts. They could have used a noise gate, but did not. This tells me their room is noisy from open mics through their recording chain.
Yeah, that's what I am saying in a comment above. And I am hobby drummer with a serious VST collecting syndrome.
I use these drum libraries by pro engineers and there is sometimes a lot of noise.
Huge areas of pro studios, hundreds of meters of cable, patchbay, consoles, preamps.
And then compressors ... you know.
Sometimes I hear the drum chairs creak.
32 384 is harder to record for me , tends to sound overly bassy for some reason and most peoples gear wont play it. I think the happy medium is 24 96, I also like 16 88, I agree going up to 192 doesnt seem to help much but I also seem to hear more noise at 192, dont know why not being an expert. Steve Wilson masters everything at 24 96..
@nrezmerski Listen to the recent remaster of Abbey Road at 32-384, its not just me! If it sounds nice to you, let me know.. If you think its something that I haven't figured out, happy to hear from you! My gear is good, although admittedly I avoid 32 bit for the reasons I have given.. its trasparent, there is no noise, just the timbre is not nice.. regarding more noise at 192, its the recordings, they tend to be like that, same album at 24-96 just sounds cleaner. As I have said elswhere sometimes I choose the 16bit over the 24..If you have a technical explanation as to why this may be the case it would be interesting.. thanks..
@nrezmerski but the point is the 32 bit abbey road sounds awful, tr bass is boomy and the timbre in the mid is unnatural
The human ear has a dynamic range of about 90 dB and no more than 20 kHz of bandwidth. So anything above 16-bit 44,1 kHz is a waste.
CD IS ENOUGH.
Hi, really awesome your sarcastic comments in your videos. Ha!
I (43) actually used Dolby S HX-Pro AND cleaned the heads recularly. I also used my dad's AKAI 4000DS reel-to-reel. Copying Paul Simons "Graceland" :-)
My dad and I, we discussed those things like flutter and distortions factors when buying a tape deck.
I guess, when dealing with digital audio (recording) the sampling rate will be more cruicial when using certain effects because of harmonics above the Nyquist frequency, and audible non-harmonic frequency components.
There is one benefit from tinnitus... the whole noise floor sorta becomes pretty much a non-issue. Recording 16 bit CD is plenty good for cassette, yeah I still do that. CDs transfer to cassette much better than vinyl. That being said that vinyl to cassette thing really doesn't work out very well. But it did enable tinnitus due to being able to turn the volume control to 11 without any feedback.
Just watched a video by an audio engineer for a record company. He stated that the only difference between 16-bit and 24-bit is dynamic range and you would have to hear the music at insanely high volumes to notice any difference.
With respect to noise floor, yes. But increased dynamic range preserves more detail and tonal separation at reasonable volumes. Just compare it for yourself.
Those myths are ubiquitous, but they are based on misunderstandings. For example, no one is seeking 24 bits for the 48 extra dB of noise floor unless they are misinformed, or playing concerts.
@@blerblybliggots9801 i'm not sure how one could hear that sort of dynamic range. the louder sound masks the quieter one well before the bit limitation.
it would be like trying to hear if the fridge is running while operating a chainsaw :)
Have there been any properly blind tests where people could reliably identify a 24 bit musical signal against the same signal, but degraded to 16 bits and dithered? I would suspect not, but if there was, I'd be more likely to reconsider.
I have always recorded in 24 bit. However as a consumer format, 16 is good but it is so much easier to maximize the quality of 16bit if you start with 24 and then convert the final product to 16.
Is there any quality concerns when converting from 16-bit tracks into 24 later on? or vice versa?
Also, think that a very quiet room has a noise level of 20-30dB and you damage your ears if listening louder than 80dB. That is 50-60dB dynamic range. The 16-bit quantization noise will be more than 30dB below ambient noise. Even without dither it can be very difficult to hear distortion 30-40dB below noise level.
Many audiophiles conflate bits and sample rates used in the recording process with the bits and sample rates in the playback files. This is a mistake. High sample rates and more bits are used in the recording process to enable editing but are pointless for playback.
If you actually had a 24 bit recording with a true 144dB dynamic range and the equipment able to handle it (which is impossible) and your quiet listening room had a 30dB background then a peak sound above background would be 174dB which would be lethal. No, I am not exaggerating, this sound level kills. Even a true 16 bit 96dB recording would be unplayable as the quiet passages would be inaudible or the loud passages screamingly loud. Most modern recordings can be delivered with 10 or 12 bits. A full symphony orchestra giving it all it's got (ƒƒƒƒ) peaks at about 104 dB SPL. Let's give the orchestra 105 dB, and 105 dB - 30 dB = only 75 dB real dynamic range. Most recordings are compressed dynamically to a sensible range so the listener is not constantly having to twiddle the volume control.
Higher sample rates on playback produces ultrasonic noise and does not improve audio in the audible range.
I have heard arguments about ultra soninc wave you obv dont hear them but some argue that the soundwaves you dont hear interact with the ones you do hear causing interaction to the audible ones. XD
@@patrickharley1003 The quality of a recording is determined by the studio at the time of recording and production. A well recorded and skilfully engineered 16/44.1 CD can deliver audio quality that exceeds the limits of human hearing. Higher bit depth and higher sample rates are used in the recording process for editing purposes but are pointless for playback. 16 bits gives a huge dynamic range. You will never hear noise in a silent passage of a 16 bit recording unless it is tape hiss from a copy of an old analog recording. 24 bit playback does absolutely nothing to sound quality other than further reducing the noise floor which is already inaudible with 16 bits. Higher sample rates produce frequencies above 20KHz, which is not only above human hearing, but captures unwanted noise that can produce nasties in the audible range. These ultrasonic frequencies can get into a tweeter and cause problems. Higher sample rates on playback cannot improve quality in the audible range. Any effects from the brick wall filter was resolved 20 years ago by improved DAC technology such as over sampling. A modern $100 DAC can give superb audio. So called “hi-res” releases are marketing hype to sell the same music again.
Loudspeakers and room acoustics have far more effect on sound quality.
@@kingzor100 If you let an inaudible wave interfere with an audible one the resulting change is still inaudible. The energy doesnt get magically transferred down to the lower frequencies.
Thanks for this explanation. Here’s an anecdotal story of why 16 bits isn’t enough, and why, as a professional video editor, I always do my final mixing at 24 bits, even if the tracks I’m working with start at 16 bits.
A couple of decades ago, I put together a “reference” audiophile CD for an audiophile club. I ripped tracks from commercial CDs, assembled them, then burned a compilation CD. The copies sounded like the originals, but the live tracks had applause cut off, and levels of some tracks were too low. So I brought the ripped, uncompressed AIFF files into a Pro Tools project to fix the minor issues. I added fade-ins and fade-outs to the live tracks, and adjusted levels on a few tracks to make things more consistent. I spit the problem tracks out of Pro Tools at 16/44.1 and burned new CDs. I was proud of myself until I listened critically to the results on headphones. All the tracks that came out of Pro Tools sounded noticeably degraded. It was not subtle.
When doing final mixing on my video projects, I typically use EQ, noise reduction and compression/limiting. I can clearly hear my processing degrading the 16/48 tracks, which limits how far I can push the processing if working at 16 bits. At 24 bits, the same mix with the same processing sounds cleaner. To me, it’s not subtle. Would my clients care when listening to my deliverables? Probably not. But I care because I know what to listen for and can easily hear the difference.
Good example.
Editing in 16 bits is sort of like editing with analog. Every operation degrades things.Maddening.
I regularly show Monty Montgomery’s Digital Show and Tell to audiophiles. It’s here on UA-cam.
I'm an audiophile purest. I only listen to music played back on my "Webster-Chicago Model 7 Wire Recorder". It's the only way to truly experience the musical message that the artist is trying to convey.
As a HiFi guy I used to be totally convinced that 24 bit listening was better than 16 bit, but then I watched an interview with a mastering engineer. I heard him say that he masters differently for CD than he does for Hi Res tracks on HD Tracks.
If that's the case for even some of the mastering engineers, then the argument that 24 bit sounds better (based only on it being more bits) is blown out of the water.
Also I have more than one example of 16 bit masters of albums sounding clearly better than 24 bit masters (mastered by someone else).
It's about all of the production, and not simply 16 or 24 bit. In general you might be better off with 24 bit releases, but that's only because more attention might have gone into making sure it sounds good, given the known consumer of "Hi Res".
I've had redbook 16-bit cd players with recordings (playback) that sounded better than a 24 bit DVD-A or dsd 64-bit SACD. The quality of the recordings is key.
Exactly correct. I have some 16 bit steely dan thats better then 24bit, it boils down to the recording. But a clean dsd 128 is going to be better then everything else.
The difference in masters is probably way bigger between vinyl and CD, some people like vinyl more than CD (not me) and make the assumption that its just the playback format in it self rather than being a totally different master (it must have a different master because of the physical limitations in the vinyl format) sure the reproduction source adds to it but it is not at all the only reason why it sounds different.
@@Stefan- Ya I get that.. my best recordings are vinyl ripped to dsd
@@Whit-mh9nt Ah yeah, and if you would put the vinyl master on a CD or other digital format with god resolution it would likely sound very similar but without the crackle and pops.
Y'know what happens most of the time?
Home audio listeners end up with a 16 bit recording sitting in a 24bit container with the lower 8 bits either zeroed off or randomly making noise.
@nrezmerski I am using pro-recorded drum sample libraries for my edrums and there are some libraries with ... really tons of noise. Like on a cassette.
I contacted the support because I paid a lot of money and you know the adds of such companies... Nah, I got a $15 voucher for a MIDI Pack.
Anyway ... so pro audio engineers and I hear humming actually on some presets on a normal volume level when I am playing.
Ok, there are presets were there used compressors etc. and then ... So this discussion is needless for me. Really, if you see these huge consoles and patchbays and cables.
The size of the file should be a clue.
@@Whit-mh9nt
a 16 active bit file held in 24 bit containers will be the size of a 24bit file.
@@Douglas_Blake_579 So an album that at 16 bits is 1.8 gb is going to look like its 2.6gb? As an example?
@@Whit-mh9nt
Yes.
Think about it ... 16 active bits, in a 24bit sample... 8 bits of wasted space.
I think here is to compare 16-bit to 24-bit on the dB level is not mattering to much. The signal amplitude will stay the same, but basically you get more steps of the amplitude to recover the original signal. Where white lower bit rate you have to add additional noise in form of ditter that you don’t notice the difference, good examples comparison you can also find in pictures.
Should an audiophile listening chair have spikes like speaker cabinets do?
Neither, one should float in the air as to not cause any weight or interact with any physical object
I've just recently discovered your channel within the last week and after several of your videos, I'm really enjoying the don't-take-it-so-seriously attitude you bring to your posts. Dispassionately examining the rhetoric is very helpful in dialing back the outrage on a topic like 16-bit vs. 24-bit. Even if I can hear the difference in bit rate, I'm more likely to hear glaring differences like room noise or poor mic placement, even when I'm listening analytically. Not to say there isn't a difference, but as you point out, there are some differences that don't make that much difference to the end user. But besides that, no single parameter on its own (bit rate, sampling rate, noise floor, etc.) is the silver bullet that provides the end-all definition of what makes a good recording. (To be fair, the misuse and abuse of some parameters can indeed yield a definitively bad recording, but that's not the same as the hair-splitting between 'near-perfect' and 'nearer-than-that-to-perfect'.)
That said, from time to time I still listen to -- and enjoy -- CD's, LP's (to a lesser extent 45 and 78 RPM singles), cassettes, open reel tape and even 8-track cartridges. I call it the 'fidelity-to-what?' principle. Music is such a subjective topic when it comes to nailing down what makes it enjoyable for you, and it's mainly an emotional aspiration rather than intellectual. So if I'm willing, perhaps even prefer, to listen to a cassette from the 1980's or a cartridge from the 1970's, I may not be in pursuit at that moment of 'absolute fidelity to the original sound', I may be seeking the same feeling I felt the first time I heard that song, and remembering the people I was in company with at the time. Thinking of the car I drove or the place I lived. Or in the case of media that predates my own life, experiencing the sound the way contemporaries of that technology did. Fidelity to a memory or emotion, if you will.
For example, I remember hearing certain songs on CD in the 1980's that didn't sound quite 'right' to me, because they didn't match my memory of hearing them the first time on AM radio in the early 1970's. Expanding from mono to stereo, losing the noise and static, and widening the frequency range emphasized different sounds in the recordings, but often the speed was off, seemed slow. I later learned that some Top 40 stations had a habit of running their turntables slightly faster, perhaps 46 or 47 RPM, so the music would sound a bit livelier (and by gaining a few seconds per song, you might have another minute or two per hour of advertising time to sell). But the point is, better audio fidelity actually interfered a bit with my enjoyment of some songs until I understood what was going on, simply because they didn't match my emotional memory. (I expect this generation may come to feel that way about 128K MP3's.)
By the same token, as an audio engineer (and enthusiast), I'm not against creating and listening to the best we can possibly muster in the sonic arts; in fact, I think we should. Right now I work in a small studio that produces mainly podcasts and audiobooks, and I'm always looking for ways to eliminate audio distractions, especially noise. But I take the view that while we should obviously tend to the technical parameters, we should do so in service to the content we are recording without getting distracted into making the technology the main thing.
Yet, in all things, charity. Whatever format a person likes and what they are willing to spend on it is fine with me. If I had elephant bucks to spend, I would likely purchase more classic gear, because it's fun! But I will also admit that even then, the grail for me would not be ALL about the sound alone...some of it would be visual, tactile, nostalgic, and again, emotional.
I couldn't agree more, excellent text
I love your perspective on audiophiles
What brought me to watch this entire video, and its arguments, basically reinforces my experiences of digital in another arena… digital imaging… it’s the same technology, just catering to other senses. I was hoping for a clear cut resolve to a single question - are our eyes or ears better at distinguishing noise from detail. After seeing the video I am undecided, but it seems that our ears are actually better at resolving than our eyes… digital images at 16 bit are the absolute precipice of imaging today, well capture at least… scanners go beyond, but it matters little, as our displays are 8-bit based… which begs a different question, maybe more apt to the original… is it the displays which are inferior enough to limit our experiences of digital imaging? In this case the arena of digital sound and digital imaging, go complete circle… in the end, maybe it’s not the source sampling which creates issues, it’s the delivery of the source to us!
For digital recordings lower noise allows greater clarity and separation to be heard or percieved on a good system. The problem is that the sound is so good that you don't need to concentrate to hear it. On vinyl recordings however the very high noise level forces the brain the concentrate and strain to hear the detail behind the noise and therefore the hearer is encouraged to listen to the music more actively and often perceives this as better sound quality even when it is not. Experiment... take a top quality 24bit recording and mix in some snap, crackle and pop from vinyl and it instantly sounds better to an audiophile. Odd that.
i've listened to 24bit recordings on 24bit hardware (the high end soundblaster cards were very expensive.. so i got them second hand..) and i cannot tell the difference between a 24 and 16bit recording of the same song... might not be listening to the right music, but, old CD releases of movie soundtracks from disney, is i do believe, (disney used to know what they were doing) as good as home audio will ever need to be... listening to the Lion King on cassette i knew when the intro chant was coming... on CD it still makes me jump... The Pirates of the Caribbean soundtracks are really fun to use to demonstrate what 'dynamic range' means and what the 16bits of a CD can really do... as far as decently mastered audio goes you dont need more than 16bit.. i have no idea what kind of godlike ears one would need to require a medium that encompasses the entirety of the dynamic range of a jet engine starting from cold and going full throttle.. at that point you don't need 24bits; you need hearing aids...
Depends on the dynamic range of the source. Not everything has 1812 overture dynamic range.
I seem to remember early CD recordings or players only supporting 14 bits because 16 bits was a little ahead of the technology. What I do remember is that the Magnavox player I bought in the mid-80's made music sound brittle. Thankfully it self-destructed after a couple years.
We have SONY to thank for raising the audio CD spec from 14 to 16 bit. Philips' original _proposal_ for the CD spec was 14 bit. But by that time Philips was already too far advanced on the development/production of their single-channel 14-bit TDA1540 DAC. So early CD players based on the Philips TDA1540 DAC had to make do with the 14-bit DAC (+ 4 x oversampling thanks to the SAA7030 filter) for playing back 16-bit audio CDs. Being single channel it also needed to be multiplexed between the left and right channels. In 1985 Philips released their first 2-channel 16-bit Audio DAC on the market, the TDA1541, and carried forward the oversampling technology with its matching SAA7220 4 x oversampling filter.
One of the first consumer digital recorder made by Technics/Panasonic (based on VHS tapes no less) was using 14 bits - which back in 1979/1980 was considered good enough. Sony pushed for 16 bits as it was a round number and exactly two bytes. As explained in the other comments, Philips released it's first DAC TDA1540 as a 14 bits DAC. But thanks to the magic of noise shaping (it is not just over-sampling) it could achieve true 16 bits resolution. This is the path that later lead to 1 bit DAC - what Philips did in 1985 still amazes me to this day, and I have a couple of CD player based on the TDA1540 and they do sound nice !
More bits equals less quantization error. This applies across the entire dynamic range, not just in the inaudible portion that is also being captured, stored and reproduced digitally. I don't understand why people who defend the idea that there is no audible difference between 16 bit and 24 bit, cling to an argument that you can't hear any difference across the entire dynamic range because of the differing noise floors, then offer a test that takes place below the threshold of human hearing to validate their argument! The funniest thing about that "null test" is that it applies dither to the 16 bit signal, which further reduces the noise floor of the 16 bit signal and invalidates the test.
@nrezmerski In this video it is conceded that more detail is captured in a 24 bit recording. It is not categorically stated that there is no difference in sound between 16 bit and 24 bit. That's because there is a difference. What is being saterised here in this video is this very small percentage of the music consumer population that care to listen for the difference and can actually hear it. They are characterised in the video as being pompous audiophiles that believe their choice of chair while listening is also a prime consideration in the pursuit of audio reproduction excellence. I think it's a brilliant video, because he is saying that people like me who can hear the difference actually have a point, but it annoys him so much he has to be mocking while making the concession.
"The reasonable amount of noise from LPs..." I never found it reasonable. It annoyed me when LP was all we really had. I could not drop LPs fast enough when digital audio came out myself.
I say reasonable because the noise level surpasses reasonable expectations of a lump of rock ploughing a plastic groove.
I suppose that depends on factors such as thew condition of the LP, etc, etc, but just knowing I was degrading the LP by listening to it (said stone on plastic...) drove me crazy. CD was love at first site for me, even though the early ones didn't always sound all that good. @@AudioMasterclass
Really? Must be something wrong with your records or the set-up. The majority of my LPs sound almost as quiet as CD. Clean records and well kept stylus is a major part of it - with a good deck and tonearm.
Yt 320kb is great through a half decent 2.0 or 5.1 system. Just my personal humble opinion.
To add to this conversation:
I recently listened to Houdini by Enimen on Tidal and Qobuz.
Both were playing at the same 'quality' 96.0kHz.
The Tidal version sounds markedly different to me, better.
Amazing...😊
I love your videos. Always make me laugh.😆😂😁
Most audiophiles take themselves and overpriced gear too seriously.... In my opinion.
Your screen name is Undead.Audiophile and you're trying to bash audiophiles with the same so-what argument as everybody with a so-what music player. Most people complaining about 'audiophiles' are taking themselves way more seriously than anybody I have heard in recent years calling themselves an audiophile. The 'snooty' people are now the ones who come to forums like this crying about 'my shitty system is good enough for me and it should be good enough for you, but you think you're better than me and I'm so mad about it'
@@jamescarter3196 😆😂 I'm not snooty or do I take myself seriously. I find it quite amusing when strangers react like this to a single post. Why am I (a stranger) lumped into a group that seems to aggravate you. I don't know you, and you don't know me. Audiophile is in my screen name because I like all the tech and gear, and how it makes my music sound different for each genre. I will never go broke for gear. I on average listen to music anywhere from 6 - 10 hours a day, 5 days a week while at work. I listen on various tws earbuds and headphones, and with numerous portable dac/amps and normal IEMs and headphones. Maybe I should be a musicphile instead.... Anyways my 3 sentences set you off.... And I'm the complainer...lol
You do you...... I'm going to listen to some tunes....🎧
This was fantastic when you consider that the Audiophiles will in the same breath say that vinyl sounds far better than 24bit audio. So about half the detail of their 24bit sound file,, that they will say FLAC sounds worse than WAV as there is some form of data compression in haw it’s encoded, but that’s a separate comment. Glad we didn’t take into consideration the noise generated from not using medical grade electric plugs, and how we skimped out with consumer usb cable to our DAC we bought off Amazon.
It's about how many bits are being used as well as how many are used in the system. Low level signals use less bits, this adds distortion. This is more noticable when using smaller word sizes.
16 bits are adequate to make noise-free recordings. 24 bits gives us ‘slack’. For example, an engineer on a live date doesn’t get the opportunity to set gain before the converter and notes the channel is peaking at minus 30. No problem!! Feeding a 24 bit converter at -30 or even -40 is perfectly valid. Just leave it and enjoy the song. 16 bit would be deep into the noise at -40, completely unacceptable. That’s pretty much it.
Detail doesn’t really enter the equation as long as you are not in the noise. A perfectly set 14 bit converter captures the same detail as a 24 bit converter because we can’t hear more than 85dB of dynamic range on a single channel contributing to a stereo mix. This isn’t just theory, this is memory. I recall those days when success was tracking at -2. It was stressful and resulted in overloads nobody needed.
Is dither worse than the noise of the audience during a live performance?
I've heard that some audiophiles are turning to Atmos to reproduce the full all-round audience experience.
YOU ARE AMONG THE FEW SOUND SPECIALIST WHICH EXIST IN ONLINE COMUNITY.
1:30 in, You make cutting through the BS one of my most enjoyable listens on UA-cam. The fact that these chairs all have headrests make it the joke of the century...
dither is needed any time the bit depth is reduced to avoid quantization noise. combining multiple 16-bit pure digital signals into a single 16-bit output needs dithering if the bit depth is being reduced. I would love to have a listening environment available where -96dB against a max reference level (100dB SPL?) is even detectable; maybe I need to find a local audiophile who has this.
from my experience, dolby C does a decent job on cassettes, and dolby S is comparable to noise levels on CDs (we'll ignore wow and flutter), but I'll still take dolby B over nothing at all, especially on type I tapes. I blame the current unavailability of new competent cassette decks and limited availability of used ones coupled with misplaced nostalgia for the sad state of no noise reduction at all on most new cassette releases. ("but it's analog!" yes, but my otari has a lot more hit points than your nakamichi.)
3:20 Acoustic band? Is that abdominally fitted, and tightened to prevent noisy emissions?
Good that you mentioned this, I have a dire straits boxed set which I play through a Marantz CD75, the sound quality is much better than all the others I wondered if it was in 24 bit Quality is amazing
In theory lower noise floor means more low level detail.. One can often easily tell the difference between 16 or 24 bit files just by listening loud to the last few bars when the track is fading out. Maybe you can provide a demonstration of this effect. This of course is of little consequence for normal listening.
I might try. Getting it through UA-cam's audio mangling system will be the tricky part.
and why would you want to listen loud to a fading out? that is not a normal or desired listening experience.
@@rugaliz to tell if a person can hear the difference between 16 or 24 bit files through listening. he wrote it all right there in his comment.
@@triple_x_r_tard I know, and my point was, if you have to amplify the lowest amplitude bits to try and hear them, the test is moot. When normal amplitude are to loud for your ears, trying to figure out if you can listen to lower noise is just silly. You don't hear it in a regular playback...also its moot because at any noise level, if you crank it up enough you will always hear some noise.
@@AudioMasterclass You may be able to provide downloadable files as a way to get around YT. They won't need to be longer than a few seconds each. Although I feel if you increase the levels enough it will be heard regardless of YT shenanigans.
I've just moved from a 24-bit recording interface, to one that does 32-bit floating point, and my DAW software supports that. I haven't had it long, so I've only done some preliminary testing. Aside from the convenience of having so much dynamic range that you can essentially set your mic levels after the fact, there is a bit of an audible difference. I honestly wasn't expecting to hear a difference, but when changing only the interface, the 32-bit recordings sound more 3-dimensional, more dynamic, and like I can hear farther into the mix. It's a subtle difference, but enough that I noticed it when I wasn't listening critically at all.
it could be that your new interface has a better A/D or D/A conversion, which would mean it's not the bitdepth
@@BeatsbastelnI moved from Apogee to MOTU. Both interfaces have the same DAC chip. The difference in the AD stage is a 24-bit AKM chip in the Apogee vs a 32-bit floating point ESS chip in the MOTU. Both units have virtually identical THD, noise, and dynamic range specs on the input.
32 bit floating point is only 24bit mantissa... so if you think that you are doing yourself a favor there, then you weren't paying attention in your numerical methods class in university. You would be infinitely better off with proper use of a 32bit integer library for signal processing.
@@lepidoptera9337 Thankfully, I was a Computer Science major who paid attention. I’m glad to explain how 32-bit floats can describe lower-level signals, despite having a 24-bit mantissa. It’s because they can have negative exponents. So if the lowest-level signal that a 24-bit integer can describe is 1, a 32-bit float can describe a signal as low as 1.2x10^-38. A 32-bit float audio file has a theoretical dynamic range of -758 dBFS to +770 dBFS, and across that range, it can describe every amplitude with the same granularity that a 24-bit integer file can describe a dynamic range of just -144dBFS to 0 dBFS. This pushes the noise floor of a 32-bit float file off into complete irrelevance, and also means you can record at any volume level, and set mix your levels later.
@@lepidoptera9337 Just to follow up on the integer vs float portion of your comment, with PCM audio, each bit of integer gets you 6 dB of dynamic range, so with 32-bit, you can have 192 dB of dynamic range if you format it as an integer, or 1528 db of dynamic range if you format it as a float, both with the same resolution.
I will probably have to run some more experiments but it does not sound the same to me. I just downscaled some Led Zeppelin. Not a big difference but clearly more smooth and finer definiton on 24bit. And I listnen on low volume and notebook fan in front of me. IMO it all depends very much on the listening enviroment and loudness too. The more loud you play the more detail those extra bits shall reveal. As to enviroment the more quiet the more we can focus on music detail. Last not least the oversampling magic which is always there is designed to trick our hearing somehow too ... That should not be forgotten . I am professional musician so my ears are long time trained but by no means I can relate to those "pros"| stating that they can not distinguish between 16 and 24 bits.
@nicksterj That is simply not true. A good microphone has close to 120dB dynamic range. Very expensive condenser microphones are specified with 132dB. That's equivalent to up to 22-23bits. This gets even worse when you are producing multi-channel recordings, because now the noise averages out, but the signals may not, so you can actually get more dynamic range out of e.g. a multi-microphone orchestral recording than out of a single channel.
@nicksterj I have an audio system on my table right now that has close to 120dB SNR. I don't know what you are talking about and neither do you, it seems.
If you actually read my posts then you will have noticed that I said that 16bits are enough AT CONSTANT GAIN. The production process is the opposite of constant gain and that's where you absolutely need higher resolution.
That I don't turn the volume knob while listening to classical music is not true, either. Do you know why? Because I have a noisy apartment. My local noise floor drowns out the pianissimo, hence I have to raise the level during those passages to hear the music and that is when 16bit artifacts become quite audible.
@nicksterj If you are recording in 24 bit and you are producing in 24 bit, then why in the world would you ship in 16 bit??? I don't get the logic here. Is there some sort of pandemic of the inferior file format virus going on? Did people miss that one can get 512GByte USB sticks these days? What is wrong with all of you? ;-)
Let me repeat this, again. 16 bit has horrible and audible distortion during pianissimo passages. If you can't hear that because you have a stereo system where the volume control knob has been welded to the front panel, then maybe you shouldn't be in a discussion about audio formats. ;-)
@nicksterj It's not my problem that Fauxtaku doesn't know how to design audio electronics correctly. I do. My systems are close to physics limits. I am not even using anything special. It's just the manufacturer's ADC and DAC test boards and a few home made amplifier stages with properly chosen low noise opamps. Stuff that every electronics design engineer should be able to do in his or her sleep. This ain't amateur hour over here. ;-)
@nicksterj I gave everybody the evidence. Get a recording with pianissimo passages and turn up the volume. It's as simple an experiment as one can do and it works. At this point you are just like an obstinate kid who doesn't want to eat his dinner, kid. ;-)
Disk size is cheap, BUT STILL, with often 192/24 disk space goes out of control. I decided, based on quality or importance of the material to bring the flacs down to 96/64/48 or redbook, also some to 19-22 bits via dithering.
I record location sound for TV/Video and getting the recording levels right is neigh impossible, so I have be very defensive and usually record with peaking at -12dBFS to -18dBFS. Thus I loose a lot of the potential dynamic range. Therefore I record in either 24bit or 32float to keep at least 96dB of dynamic range.
After mixing and mastering I can utilize the full dynamic range of the output media and then 16bit is fine. But for some strange reason 24bit 48kHz have become some sort of standard in the video world.
In this day and age, there's a lot of room correction, digital crossovers, surround sound or other processing, which is applied to the final recording. Not just during production. God knows what other things there will be in the future. 24 bit recordings provide an extra layer of "rounding" to make sure the 14 or 16 bits' payload will still get to us as well preserved as possible, even thirty years from now.
Those systems can take the 16-bit input and work in 24 bits from there on, including for output. This gives them the room for rounding etc. The only issue would be the dithering added to 16-bit, which would become noise that the system would also treat like signal. Once you add dithering you're basically saying that it's finalized and should only be sent straight to a DAC. It's putting assumptions about the output system into the recording.
Fantastic, the end! Thank you for your explanations on 16 and 24 bit
You’re welcome. There’ll be more to come.
My brother is a professional DJ. He can absolutely tell the difference playing them loud through decent club systems.
Back in the early digital days, I was at a friend's studio where he had a 12 bit sampler. I asked if that wasn't too poor quality to be used in making recordings. He told me (and it made sense to me) that any samples that were used would be inconsequential relative to the rest of the mix.
12-bit samples are noisy or grainy or both. But since this was all that was available at the time, producers accepted this and made music that worked with the 12-bit texture.
Dolby C and chrome tapes made cassettes just about acceptable to me.
While the extra dB is great I prefer having the extended dynamic range of frequency range / response to play with with 24bit - over 48khz . But 1644.. it's fun to record to and people don't generally notice if you pull it off balanced.
Love your presentation style: it's entertaining and informative :)
I find your videos very entertaining and informative. I'm the kinda of idiot that would blow a load of money on high end gear just because on paper a number is higher than another number so it has to be better, even if to my ears there's no decernible difference. Your videos keep me more grounded.
I concur with your axioms.....are you an enthusiofile per chance?
Enthusiophile. I think I like it.
Any audible differences between 24 and 16 bits are likely caused by the lowpass filter of the dac @44.1 kHz sampling rate, not by the difference in bit depth.
there is a diffrence in music 16 bits sound boxy 24 bits is still in some kind of box feeling but with 32 bits it dissappear the noise cliping is gone basically
I don't think I’d make the difference when I listen to music. On the other hand when I play (digital) piano I’m using an acoustic simulation program named PianoTeq that uses high resolution (32 bits 92000 Hz) which I listen with a 24/96 audio interface and headphones. When playing it’s important to forget that your instrument isn’t real so any realistic detail has its importance, be it the tiny reveberation of the sound once you’ve released the keys, or sympathetic vibrations of loose strings (PianoTeq allows you to adjust the wear of you piano to make it sound bit less brilliant and a bit more convincing, so there may be loose strings). I should try to downsample a recording and see what it does to those tiny details that have no importance for the listener but have some for the performer. Maybe there is a placebo effect or something.
I have spent many hours comparing 16 bit to 24 bit when using the same DAC, and same sample rate. The difference is often very significant when listened to on good gear.
I am currently using a MOTU M4 to 2 hafler dh220s to drive kefXq9s.
Most albums that were recorded in 24 bit will sound substantially dryer and less rich when played at 16 bit. There is also usually a lot of information from the decay of sounds that is chopped off with 16 bit. So, the the difference in 24bit is that it is not only less dry, more spacious, and richer, but there is more information about the echo and natural decay of sounds in the recording. The increased dynamic range is not always noticeable, but the attack of something both natural and abrupt, like a snare, for example -- will sound slightly different at dynamic ranges above 96 db, even at low volumes.
@nicksterj based on what? Have you actually taken time to compare this on proper gear with all other variables controlled?
If you are confident that I am wrong, I am happy to meet to do a blind test that you administer. And I will bet you whatever amount you want that I can identify them correctly in an A to B test 10/10 times. And I won't even be paying attention to the noise floor.
I'm also willing to bet that you would hear it, too.
@nicksterj the data is lost once you convert it to 16 bit.
Have you actually just found a song that is recorded in 24 bit, and then toggled a setting on your system or the app handling it and listened to the change?
@nicksterj oh, i guess i am not paying attention to which video this is under. But I see that you are going far out of your way to find any comments that prove your youtube video wrong.
Also, I am not married to this information. I haven't gone on youtube to make embarrassing videos about this. But that does explain why this is so important to you.
I just confirmed for myself what the difference was, then went to sources that talked about this issue, and then was surprised when I saw so much objectively incorrect information in the threads that I made simple comments so that people won't just read the misinformation. But that apparently triggered you.
lol
@nicksterj "never bothered to reply" from within 24 hours ago? lol yeah, I am not losing sleep over this and watching for your responses. 🤣
@nicksterj well, it sounds like we have the same goal, then. So, I am happy to bring this to a conclusion.
And I am sorry, but no. Your videos only prove that the creator of said videos is incapable of a scientifically valid comparison on so many levels that it became a masterclass in how to fail at providing meaningful data. and that includes your video that you posted. You haven't address any of the rebukes of those videos, but you have doubled down on the misinformation with blind assertions.
I'll address your points that you are making in the other video. But maybe it's easier to just have this discussion under your video?
i know that might be embarrassing there, but honestly, I am doing you a favor if you finally realise how your video was flawed and take it down before more people see it. up to you, though. I wont pressure you to defend yourself on your channel.
"BEAUTY - Philips 3000 nose hair trimmer" - LOL - (It worked!) - Made my night 🤩
Oh it was worth hanging on through all of this JUST for that ending. 😂
You brightened my day and lessened my DAW worries both at the same time with this video. Most grateful for your simplified explanation. It's pretty much all I needed.
Now to chase down what the heck is causing my glitchy pops and crackles in mixdowns. Now I know it can't be my 16bit 44.1kHz limitations. Processing power maybe? Just bad software?
Firstly, let me say that this comment will make it fairly obvious that I can’t be considered to be anything approaching an audiophile or technically au fait with matters discussed regularly on this channel, but it illustrates the reality of the (my) aging process. One of the “joys” of getting old and turning 70, apart from developing a taste for droll, witty and, at times tongue in cheek UA-cam channels, is that you lose the ability to hear above and below certain frequencies. So when I sit down to listen to music, taking extreme care not to get too excited and turn it up too loud thus activating the tinnitus that’s always just lurking in the background in my ears, I just don’t know what I might be missing in those frequencies, what’s more I have no idea if there’s actually anything there in the recording and as naïve as my attitude might be, I don’t care, I still enjoy the music. As to any noise I may hear, which I admit I find quite irritating when I do actually hear it, there’s a chance it may not even exist in the recording but be a product of my less than audiophile system or possibly the “head noises” that old age brings on, I just don’t know - but some things you’ve just got to accept, like getting old or, say, never achieving perfection.
Oh, man! “Technically au fait” is my new favorite phrase of the month. Thanks.
Thank you, a handy phrase I threw in as I had no idea what I was talking about@@johnfisher3929
"Techniquement compétent" if you'll pardon my French.
Yes, of course I freely admit I am neither "technically proficient" nor technically "up to date or fully informed" regarding the matters you discuss, but it's still a lot of fun to watch.@@AudioMasterclass
Well OK, but the French don’t care what they do actually, as long as they pronounce it properly.
I think there may be another factor at play why sometimes 24 bit is perceived as better: ISC, (Inter Sample Clipping) or lack of it. 24bit usually goes along with higher sampling fq, at least 88.2. At 44.1 ISC does occur on tracks mastered too hot. at double or higher sampling fq, thus more samples, the DAW would show the signal exceeding 0dB F and thus one would lower the level, thus fas less ISC and the DAC's reconstruction filter can restore as it should. Obviously, at 192kHz even less ISC (and at frequencies where it wouldn't bother anyone). Perhaps that is the better detail, more precise effect one hears with higher bitrate recordings? not the the bit count, but the sampling frequency is what makes the difference. just a thought...
Intersample peaks leading potentially to intersample clips are an issue. However because it is increasingly common to use true peak metering we can soon hopefully stop worrying.
The cheapest and most effective way to improve detail and soundstage is cup your hands behind your ears 😂 whilst seated in your audiophile chair in a dark room😂
@nrezmerski brilliant👍
What about playing "cassette tapes" so much you can hear the other side of the tape playing "backwards".
You jest, and your jest is amusing.
Actually possible in theory, but it is not a tape fault; it is either extremely poor tape/head alignment (I don't mean azimuth) or a faulty/worn actuator for the rotating head in an auto-reverse mechanism. I think in theory a badly worn capstan bearing could also cause this by pulling the tape at a skewed angle through the head tape guides, enough for the adjacent (B-side) track to just touch the edge of the head gap, but that _would_ go paired with a significant azimuth error and loss of most high frequencies.
I think it depends on the persons whole set up, I have a pair of Sennheiser HD800 headphones and I used them for a while directly connected to the head phone outs of my RME HDSPe AIO sound card in my studio, I used to get noise at 16 bit or 24 bit due to the connections, I have since upgraded to use the SPDIF out and a Mojo 2 Amp and I can't hear any noise on either, but detail in sound is very different in my experience, using 16 bit in cubase with VST synths sounds really poor, but switching that to 24 bit is a huge difference when producing music the virtual instruments just sound so much better and the extra head room is a blessing.
@nrezmerski yes I have done that before and I can tell the difference, tbh on any other headphones I have had that wouldn't be noticeable but it's really obvious on these things.
@nrezmerski the difference is in how it sounds not the numbers, use your ears, try working with 8D audio, the difference is so clear, 16 bit sucks, sorry for you that you can't hear it.
@nrezmerski that sounds like a colour blind person trying to convince someone there is no green and brown.
@nrezmerski no one can prove that can't either 😀
@nrezmerski I have just been A/B testing my self and the difference still seems clear to me, VST Synths stand out, and tracks that are fully mixed with 8D especially with sounds put in the 360 degree sound space about 127 degrees to the back left or right, the positioning of the sound isn't right when converted and high frequencies have a grittiness to them, also different conversion options with Dither and Noise shaping in my software have different results some better than others, but with out a blind test set up and a third person testing me I am going to have to concede I can't prove it and given the evidence of others being A/B tested and failing then I can't assume I some how have some super hearing :-D I am interested to set up a test for myself at some point but until then this will remain unresolved for me, I feel I can spot the differences more easily on tracks I have been working on for weeks vs someone else's music so I wonder if that can make a difference if someone knows the fine details of a track, anyways I will keep an open mind to this in light of your comments until I test myself properly.
Some of the best DAC's can only achieve 21bits if you are lucky a lot reach less so 24bits is lost anyway, the majority of recorded music doesn't even reach 16bit once mixed and mastered, if you were to hear the noise from a 16bit stored recording your system would have to be pretty damn loud probably beyond normal listening levels too, also your analogue components would have introduced more noise and distortion before it reaches you anyway.
Ummm ... no. The "21 bit" rating is for analog clarity and noise floor... It's another confusing spec. cooked up by confused people to confuse other people.
A 24 bit digital sample always uses all 24 bits and a 16 bit sample always uses all 16 bits.
@nrezmerski
Well yes. Low level noises recorded right into the files.
But that lower 4 bits is at -120dbfs and would be totally inaudible anywhere outside an anechoic chamber. Certainly not audible in the average listening room.
A record has if you are lucky something around 50 db of dynamic range and it turns out that most hifi systems can't reproduce it because they will already start to compress and distort the signal so 16bit is already completely overkill and 24bit is not changing anything at all. It is like debating if Thor or the Hulk is stronger. Fun when you are 13 but at some point you grow out of it. And the correct answer is the Hulk of course ;-)
Seems like Audiophiles have some sort of inverse placebo effect - "I know there is noise/distortion deep in the mix, therefore I can hear it"!
@nrezmerski I'm glad I'm not that fussy!
32bit mastering is the new no1, just in the same way as 32bit colour is in windows, this is EXACTLY the way you should look at this topic, less bits=less colour. with 16bit you just dont see the full picture. 16bit dithering is no different that a blur effect of a low res image to get rid of the artifacts. and no "audiofools" dont have better hearing, they simply have more money, in my vast experience of audio i can tell you the best is NOT the most expensive. also another thing we only listened to cassettes back in the day because there was no other option.
This is a very poor analogy. We can SEE more colors, we CAN’T hear the difference in 16 bit vs. 24 bit especially when you consider the noise floor in any room you can imagine. After all, no one listens to music in an anechoic chamber.
@@mabehall7667 Not really, digital is digital no matter how you look at it
@nrezmerski Never the less, its still adding something to the audio that shouldnt be there. A sort of "band aid" theres absolutely no need to use 16bit today, this isnt 1982
I'll be making a video soon on why 32-bit mastering is a dreadful idea.
@@AudioMasterclass Well, kraftwerk think its a good idea
Any recommendations for a good upgrade for my tinnitus, so I will get a noice-free experience? ¯\(ツ)/¯
But Audiomasterclass didn't go into 32-Bit PCM and 64-Bit Float and they are the new mainstays for 1-1 audiophile recording now. 16-Bit and 24-Bit are both out of date Wave recording technology in your latest Audacity. So where does the headroom change the quality in 32-Bit PCM and 64-Bit Float. lol
Hate to break it to you but it is all PCM recording, even WAV files are PCM.
PCM is a type of modulation, not a size.
Criteriously human, scientific and simple. 👏👏👏👏👏👏👏👏👏
Non dithered 16-bit audio dynamic range is 90 dB. Not 96 dB. Full-scale 16-bit audio signal can only be reduced by half 15 times without going below the noise floor. Reducing it by half a 16th time puts it below the noise floor.
@nrezmerski The first two words in my post are "non dithered".
@nrezmerski The 96 dB range stated in the video is a reference to non-dithered 16-bit audio. My comment is in re: to that. Not to make a point of whether or not it's an issue.
20*log(65536) = 96,3.
@@xuser48 20*log(65536/2) = 90.3
The range of 16 bit digital audio is -1 +1 to -32768 +32767. Both the + and - are amplified the same. Thus there are 15 x 6dB steps. Not 16.
The minimum level that is not below the noise floor is -1 +1. A range of 2.
The maximum level that is not clipping is -32768 +32767. A range of 65535.
So the max to min ratio is 65535 to 2.
20*log(65535/2) = 90.3.
The 96.3 that people keep stating is a non-real world theoretical that could only exist if the noise floor was 0. In which case the dynamic range, and gain too, would be infinite. Not going to happen.
@@RazorStrap - Learn some electronics!
I spent 5 sec deciding on 16 vs. 24 bit for recording and hours obsessing over mic placement. Mic placement has made more auditable difference to my recordings that bit depth. Spend your time where it really matters.
Indeed, other than the music and performance, mic placement is No. 1.
Whenever I misbehaved my mom said to go to my room without dither. Take that for what you will.
Because when you were in your room she couldn't hear your noise, so she didn't give you dither.
When I was 16, in our house the floor was noisy upstairs, so with the noise floor being up we had to have dither downstairs
There are advantages of24 bits for the mixing etc, but for a perfectly set gain, aren't the extra bits just recording the noise in more detail?
Hi there, Thanks a lot for your instructive videos. I wonder to know as the CDs are still 16bit/44.1kHz, is it worth to record raw files with 32bit/96kHz ? Thanks for your feedback. Greetings
Yes record in 96 kHz 32-bit float. 24-bit for delivery though.
@@AudioMasterclass Thank you for your kind feedback. The CDs are still 16 bit. So I have to convert files 96/32bit to 44.1/16 bit. Why I hear people advising to record at 88.2kHz as the multiple of 44.1kHz ? I am still asked for to produce CDs which are still 16bit/44.1kHz... It's a jungle. Thanks again...
What about 32bit for gaming?
In the modern world of mp3, loudness war, music played on smartphones, I consider myself an audiophile, but 16/44,1KHz is enough for me, because the background noise present on the master recordings is always higher than the ground noise of the 16bit. Than why should I bother with 24/48 or 24/96Khz tracks, sacrificing more storage space? Not to mention that it is already difficult to find records that exploit correctly the entire 16 bit range. Maybe the difference can be heard on 24bit files of some classic or jazz music, but they are not my genre.
If we stay in the analogue domain first, before addressing the video's question, then let's first wonder what "bandwidth" is needed for a 440Hz music note. That's a loaded question, though. Play that central A note on a piano, violin, flute, clarinet, saxophone, etc. Each time it is 440Hz, right? But none of these have perfect sines as wave shape. So there is information on the 440Hz that we can analyse so as to recognize the instrument (voice, kind of sound), and we can place it somewhere in space.
But to recognise individual voices/instruments must be learned and that takes many learning moments. If you never went to a live classical concert, then you don't know what is wrong with a recording of that (if there is something wrong).
Imagine the clarinet produces a block wave and the flute sine wave, but a cello has a sone wave with a couple simple spikes in them.
Here we have left the 440Hz as each of the deviations are in a much higher frequency range.
Our ears are physically limited to about 20kHz, when measured in sine waves, but that is a coarse abstraction.
Whatever the point of the 20 kHz, in electronic recording and playback, the question is not what we can hear - assuming an exceptionally well trained listener - but what bandwidth is needed in an amplifier, cross-over filter or speaker to play a perfect 440Hz block wave, or to get the two tiny spikes that make a cello sound like cello, sound like a cello in playback, because the tiny spikes are at the right place and in the right form. Because - phase fault, if the spike shifts it's no longer a cello we hear, and if the shape distorts, it's no longer a cello too.
And the same applies to aunt Lucy's voice, or uncle Stephen's.
I saw a pianist in a Steinway grand piano parking room of a concert building - 5 model D grand pianos. She had to select one that best matched her music in character and best matched her touché in its action. She played all five and about number 2 said, "I played this one last year".
I saw a blind person in a new large room that asked, are there windows there, pointing at windows, and is there an open door over there, pointing at one.
The auditive brain is much faster than the visual and it relies on wave shape processing in an acoustically relevant to human survival band. Electrical engineers underestimate how good it is. Listeners accept electronically mixed recordings that have "micro-dynamics" for left right but conflicting phase information. This all goes back to wave shape. If you ever saw how good an amplifier can play back a block wave at 20 kHz then may be aware to never have seen a block wave, but spiked versions with over/under shoot, or saw tooth, or sine approximations based on a perfect block wave input.
My pre-amplifier as a bandwidth of 1MHz. The power amp "only" of 150kHz with peak current delivery capability of 60A per channel.
An electrical engineer once responded to my reasoning about wave shape that Fourier analysis could create a block wave with only five sine waves. Well, you need five half sines each of different frequency to fill the positive block and there is no amplifier in the world that can do that. I guess. And, what signal source does the Foerier transformation?
How many bits do I need? What sample rate?
Lets go to the bandwidth requirement first. You tell me.
Oh, a big problem in life and science is that people think something is not true when they cannot imagine it to be true.
Remember the geocentric model of the cosmos?
The problems are all over the place and humans are very capable of filtering the message out of the noise, but that doesn't mean it sounds nice or correct. If we throw speakers into the debate then it becomes hopeless almost. With lousy amplifiers and loudspeakers we do not need a lot.
True revolution or Musical Change can happen only if we make different Sounds Speakers and headphones with different response to frequency giving different output results and current situation with low quality, high end and studio sound reproducing devices Listeners are listening not critical like we do and context is different something like just another Track or a just another Story telling through the Sounds/ Music. Like Hz the more is enabling more accurate sound manipulation and processing but end results will be none ... but if devices become more quality then maybe we will hear noticable advantage. Cassettes were blurry for me compared to Digital Disc .
idk why i have 16 and 24 bit files but 24bit sounds lower volume than 16 bit.
Good content, and Audio Phil had me laughing really hard. Well done, good sir!
I have my aged DAC with 100dB Dynamic range that specifies SnR without THD. I turn it to maximum to have maximum Dynamic range (100dB) then I attenuate the output of DAC by 30dB analog potentiometer (variable resistor potentiometer) to pass it to my amp with more than 124dB full Dynamic range/more than 99dB at 1W signal output. No question I don't hear my amp noise from my licening position since my speakers have sensitivity only 90dBA/(m*W). 0dBA - is the lowest sound the human can hear, while the amp is -9dB below 0dBA level, so it is 2.83 times less signal volume that human can hear. But I also don't hear my DAC noise since it is -100 - 30 = -130dB below maximum signal my amp can produce, and -6dB = -130dB+124dB below my amp+speakers own noise level equal to -9dBA (-15dBA accumulated).
Now if I play my CD-DA player via my aged DAC will I hear any noise? No, I will not since CD-DA dithering will be between -120dB and -130dB level - way out from any human hearing capabilities. Do I need 24bit music to level noise to -150dB level? I don't think so since the amp is still the main source of noise at -9dBA level.
I think you are mistaken about how 16bit/24bit conversion takes place. It is not one bit or the other, there is a conversion process based on the original Nyquist Theorem for sampled signal reproduction. In practice there will be computational compromises in the DAW, which is why many DAWs will use 32bit floating point internally regardless of the bit level of the sampled audio.
In other words, there is no masked detail when converting between 24 bit and 16 bit with competent software and hardware.
This is very interesting, thanks. I have 2 questions which are not related. Firstly, given that most, or at least many, new non classical albums are to some extent 'brickwalled' is there any point buying the more expensive 24 bit version (say from Qobuz) over the 16 bit version anyway? I don't know but I would love to. My feeling is that the potential increase in dynamic range from the 24 bit version will be more than wiped away by the process of brickwalling the album, so it becomes a waste of money going for the 24 bit version? What is the answer please? Then this ........
Secondly, if an album is available as either 48kHz/24 bit or 44.1kHz/16 bit, but not at all in 96kHz, then if we assume that it was not recorded in 96kHz then to arrive at the 44.1kHz from 48kHz, the bit rate must have been resampled, which will damage the sound at least in part (or visa versa). Would that act of resampling be audible enough to make the purchase of the 48kHz version a better choice just for that reason alone?
It would possibly depend on the type of music, but that 99.9% won't hear it, I'm sure.
There's a simple way to find that out: buy both versions and compare them. If you don't hear a difference, you can spend less in future purchases. If you already have a Qobuz subscription, you can flip the switch in the bottom right corner to compare. Or better, let someone else flip the switch and make it a blind test. You'll be surprised, you may even prefer the mp3 in some cases.
@@stevebodner3729 but then I am one who would .....
@@mvv1408 obviously so. I don't have Qobuz - they sound better as ripped CDs/downloads from them
Love your videos...I'm a cd listener and love it.soumds good to me😊
I sometimes repurchase music in 24-bit format when it becomes available, and even though I definitely have the highly accurate equipment required, I perceive very little improvement (at best). Still I purchase 24-bit here and there, because (much as with 4K in the movie realm) I expect that careful remastering has taken place as part of the process, at least usually. The satisfaction of having the definitive version (I suppose that's a third dimension, David, besides noise and detail) of something that's worth having the definitive version of, is often tempting.
Chasing the definitive version is a quest in itself. I watch Parlogram www.youtube.com/@Parlogram so I can quest vicariously.
I use my ears and mark my CD albums from Red book astonishing down to utter trash. Out of my 13,000 CD collection I would have well over 10% that are trash. Another 30% at least are tolerable and maybe a few thousand are INCREDIBLE!!!!! @@AudioMasterclass
How about 32 bit? I have the option to select on my sound card an output of up to 32bit @96Khz.
32-bit floating point is commonly used in production. I wouldn't recommend it for finished product though and I may cover this in a future video.
So what is best for me on a Computer PC. i have a sound card that can play 16-bits or 24-bits and betwen 44.1 Khz - 48 Khz - 88.2 Khz - 96 Khz - 176.4 Khz - 192 Khz ... Witch one is best do you think? Thx for a good video. / Sweden!
Close enough for most practical purposes in light of consumer product. For a more nuanced understanding, I recommend Digital Audio Explained for the Audio Engineer, by Aldrich. Technically, bit depth contributes nothing to detail above the bit in question. That is, 24-bit is more detailed than 16-bit - below about -100 dB full scale. Above that level, everything is identical. You can prove this to yourself with a null test. Take the best 24-bit file you can find, apply 16-bit dither to a copy. Compare the original and a copy in a 24-bit DAW with one channel phase flipped. There will be no signal AT ALL above the 16-bit dither noise. If you understand how digital audio works, that result and its meaning will be intuitive.
i told A Dj Friend About It, He Said It Dosent Matter People Dont Hear The Differnce But I Could Hear The Differnce Between Mixes
The High And The Low Quality
OK, I confess, I'm a hard-core audiophile with a soundproof acoustically treated room, I don't have a fancy chair, I have a 3 seater on a small podium. My Audio note dac 1.1x signature is an has an 18bit AD1865 chip in it, sounds amazing. I've had dac with 24,32 bit and dsd 1024, in the end the Audio note suited me best.
I remember as a kids I use to buy the best blank cassettes I could find, mostly Sony HF/UX ES 90/Metal master. I would use both the EQ and dolby b and c to tweak the sound to my liking. I loved my Sony Walkmans. I was 11 when I got my favourite headphones, the Sony mdr v7's.
Love Audio Phil 😂
So in short is 24 better than 16?
@@DarenLee I listen mostly to music from the 1950's-1980's so 16bit enough for me. I just enjoy the music for what it is however it was recorded back in the day. I keep away from all these modern remix's such as Re-mastered, deluxe edition,super deluxe edition and even hi-res.
@@ac81017 thanks for your kind reply. I think I get what you mean ;)
I am into IEMs and sometimes with different sound signature really doesn’t goes well with old recording spending weird or fake I sound say.
Now U are Talking, MasterClasd is what Im her for!!!!! The spice of the channel some tech , My current faves 16/44>176/24 upsample same for 96/24 when 192/24 not available Nyquest/Shannon was right
Both Sony and Phillips said they were going to improve the CD including sample rate.....but CDs are not the popular .mp3 put the kibosh on that
I do not disagree with your axioms. There will always be people who want a good quality and they are really passionate about it, so why would I wanna disappoint them?
Edit: But I do not agree that audio needs to be 24bit. My audio interface for example only supports 16bit, so I never listen to 24bit audio at home. That in turn also means I make music in 16bit. I'm fine with that for now. The music can sound great and if it was designed with this bitdepth, then it was also meant for that bitdepth. Only if someone converted a 24bit audio file to a 16bit one, while being able to listen to 24bit audio, I'd maybe feel like I might be missing out on something subtle
@nicksterj It depends what you want to do with the signal. 16 bits is not primarily more noise. It's mostly more distortion for low amplitude signals. 16 bit is worth roughly 96dB of dynamic range. 8 bits are only 48dB, so if you are listening to signals that are -40dB or less on a 16bit system, then you are getting the distortions of a 8-9 bit system... and they are awful. We are talking less than telephone quality awful. This is usually masked by the low volume, but if you want to run audio with 16 bit quantization through a compressor, then you are in for a nasty surprise. When would you want to do this? A typical application would be extended sustain on guitar samples by using a digital compressor effect. Not much of a problem at 24bit, but a serious challenge at 16bit quantization.
@nicksterj And I didn't argue that a dithered 16 bit system isn't useful as a distribution medium. It just doesn't work for much else in the audio chain. For that we need to do a lot better. I even gave you an example for which it does not suffice.
@nicksterj "Detail" is not an engineering criterion. Harmonic distortion, in particular intermodulation distortion is. 8bit quantization has absolutely horrible IM components, so every time you get down to the -40-50dB level your 16bit system is going to hurt your signal audibly and not just a little. If you don't believe me... just try it.
Headroom and post production :)....Nice video as usual. Thanks :)
What is happening when you play 16bit or even 24bit on device that don't support for example 24bit playing 16bit ?
Any 24 bit DAC can play 16 bit files.
Where you run into trouble with some older DACs that don't do 24bits is that playing the file as 16 bits strips away the least significant 8 bits.