How to get mind-blowing 110 dB dynamic range from CD-standard digital audio
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Thursday April 20, 2023
David Mellor
David Mellor is CEO and Course Director of Audio Masterclass. David has designed courses in audio education and training since 1986 and is the publisher and principal writer of Adventures In Audio.
When I could afford my first CD, what a revolution, I never came back to scratches, clicks, rumble, wow...
Thanks to digital audio creators 🙏
j en: the robot and robot voice are extremely annoying. the dulcit tones of your own voice suffice. please stop
Audio Masterclass replies to j en: Sure I'll stop. Kidding. DM
Precision Sound Works: Loving these vids, David!
Glade Swope: Would there be less noise (and/or more distortion) if you lowered the dither to half (or 75%, 90%, 99%, etc.) of the one-bit level?
Audio Masterclass replies to Glade Swope: As I understand it, there is a compromise here. Less dither, less added noise but more distortion. The right amount of dither then more noise but no distortion. Noise is random and distortion is correlated with the signal. DM
Rick: Cannot watch and listen to that AI voice/animation.. disgusting
Nic c: I'd be interested in an analysis of the HDCD format from you. I have a couple of players and 3 DAC's (Cambridge ISODAC and Audio Alchemy) that have this functionality, and a handful of CDs (Mike Oldfield remasters) which are HDCD.
Mark Hayman: Hi , really great and interesting video. Regards mark
NewGoldStandard: Amazing.
Great content, as always.
Editing SECRETS revealed!: I stand corrected. I thought triangle dithering could only add 2 bits worth of additional encoding, giving effective 18 bit range for 108 dB theoretical maximum. If real world measurement is that signal is happening and noise isn't at -110 dB, that would imply another bit's worth of information... from the frequency shaping of the dither perhaps?
Jonathan Vogel: you dont. the noise floor in your house wont allow a 110 dB range
Editing SECRETS revealed! replies to Jonathan Vogel: He didn't say it was a good idea to play back this signal through loudspeakers, only that the system could record and reproduce it accurately as an electronic signal.
Eddie Willers: Betty's voice definitely needs to be a little more euphonious; it doesn't match her appearance.
When Better Cars Are Built: So why is the effective dynamic range of music on CD is often only 10 or even 5db, when 20 or 30db is very pleasing and fun.
Glade Swope replies to When Better Cars Are Built: Because "professional mastering" has become an unfortunate misnomer term, along with "serious music".
Christopher Sanderson: Why do people care so much about the most quiet of sounds? The music people listen to doesn't come anywhere near these dynamics. This pursuit seems like a waist of effort when you really should be focusing on actual sound design skills.
Editing SECRETS revealed! replies to Christopher Sanderson: Dither affects the lowest used bit. If you have a full scale blast, the least significant bit is at -96 dB. If you are recording to a 12 dB dynamic range standard, then the least significant bit is only at -12 dB down and you can definitely hear what happens there.
Focuspix video & audio services: Sorry call me stupid but to clarify.. When delivering to a 16bit service in 24bit does the codec they use dither with any grace or is it preferable to pre dither...Also with a 24bit service in mind as well - you should have different masters depending on the service. I guess that's my point...
Thanks if you answer. And none if you don't..
Editing SECRETS revealed! replies to Focuspix video & audio services: Ask the service. When they drop the lowest 8 bits, they might just truncate the file to 16 bits (chop off the last 8 and leave the top 16 unchanged), or they might dither the bottom bit or two of the 16 bit output to encode some of the contents from the lower 8 bits. Depending what types of dither your system can produce, the result might be better or worse than what their system would produce. This would be a good conversation for you to have with a technical expert at your service.
MonguzTea: 90s cd players applied dither in the dac to boost dynamic range. It didnt make them sound better though.
Editing SECRETS revealed! replies to MonguzTea: It's only useful if it encodes a combination of noise and signal below the nominal cutoff, in this case from bits 17 and beyond of a greater than 16 bit wide sample. On playback, there's no additional information available to the player other than what's already encoded in the 16 bits.
Seiskid: What happens if you apply a 220hz pass band filter to your dithered signal. Obviously you can't do that for music. But I imagine it cleans it up considerably. Likewise running the non-dithered signal through the same filter would remove the harmonics. Interesting concepts.
Kalvinjj replies to Seiskid: If you add the band pass filter to the square wave distorted 200 Hz you'll also recover a perfect 200 Hz signal as well. Kinda how tuning a radio works.
Editing SECRETS revealed! replies to Seiskid: The term is noise shaped or frequency shaped dither, it's a real thing to try to push the noisy artifacts of the process out of audible frequency range.
Casual Introvert: Ironically, I have my airpods max wirelessly connected to my PC with a usb filter inserted into the chain before the bluetooth receiver, and if I listen with noise cancelling on, I can just barely make out the tone at -90db. I was expecting the bluetooth stream compression along with the youtube codec to pretty much get rid of it, but I could actually hear it, albeit with my headphones at max volume.
Brutus Undercroft: This brings up an issue I've been debating lately, and I definitely want your take on it. I constantly find people arguing about ripping CDs to higher bit depth and sample rates. Huge recitations of things people have read. Blah blah, copy-paste Nyquist to get the spelling right. But, opinions change or go silent when one mentions the high frequency data sometimes discovered when resampling. There's an opinion my best friend shares that the new wave data is harmonic distortion having come from the resample. But, peripherally, that would threaten the "math is pure" stance that materializes when resampling arguments come up. Now to the part I'm concerned with: I recently started wondering if that resampling phenomenon isn't creating harmonic distortion, but is instead revealing a particular dithering method/algorithm used on the album.
Editing SECRETS revealed! replies to Brutus Undercroft: True for a realtime capture. Audio pros usually tweak a system so as little is possible might block kernel audio processing. If a block at a time is read from the CD into a buffer, then written from the buffer to a drive, and that doesn't have to be realtime, a stray overtime interrupt will delay appending that block to the output file. But the file will be a perfect clone of consecutive samples of the source. On the input side, the CD format's error correction algorithm should be able to use the checksums and redundancies in what's on the disc to recover from an occasional misread pit.
Editing SECRETS revealed! replies to Brutus Undercroft: @Brutus Undercroft The CD player might have some small random jitter, when the reading laser doesn't perfectly catch every bit at exactly the right time, or the output buffer clock doesn't always increment to read out the sample at exactly the right number of nanoseconds between words. If you're using a personal computer to measure a live audio stream, then occasionally an operating system interrupt or hardware timing glitch could make this happen.
Editing SECRETS revealed! replies to Brutus Undercroft: For a perfect clone, write one output bit exactly for every one input bit. 16 bit linear 44.1 kHz PCM in, 16 bit linear 44.1 kHz PCM out. If you want to change the output to have different qualities than the input, such as pitch shifting or equalizing it, processing it at higher resolution will help reduce artifacts from the processing step. But this is internal to the processing step only. It has no more source information to work with than is already present in 16 bit linear PCM words at 44.1 kHz.
Brutus Undercroft replies to Brutus Undercroft: Right. But, I'm not concerned with questions of benefit.
Rather, it's the matter that there's a change to the high frequency audio content, and I have yet to find answers on why or how this is happening. The fact that it's inconsistent is all the more fascinating to me. (I'm ignoring older lossy formats with this, btw, and thinking more about FLAC and .wav.
But, yeah.) I've seen it happen a fair number of times. When it does turn up, it's usually inaudible, but shows up via spectrogram/spectrograph.
Obviously it's largely irrelevant at a common user level if it's typically inaudible. But, it doesn't change the fact that it's happened, and I'm just intensely curious to know how this happens. In 10 years of noticing this, nobody ever seems to have an answer.
Audio Masterclass replies to Brutus Undercroft: I would find it hard to believe that ripping a CD to anything other than a 16-bit, 44.1 kHz PCM file would provide any benefit. DM
Scot Peacock: Was the dither in this example simple dither? If so, I would have liked to see the results with noise-shaping. Apparently, with noise shaping you can get something like 120db dynamic range with 16bit.
Audio Masterclass replies to Scot Peacock: The simplest dither Pro Tools has to offer. I felt that anything other would overcomplicate this demo but I'll consider it for a future video. DM
Joseph Kosak: I don't even know if doing that makes sense. If a device's amplifier doesn't have a signal to noise ratio of at least a few dB more than that, the quiet parts will be masked by the amp's background noise. Also, I wonder what the dynamic range of an average person's hearing.
Hmmmmm...
Joseph Kosak replies to Joseph Kosak: I'm surprised EVH could say that, unless it was early on in his career. As time went on, I wouls say their onstage SPL would have destroyed at least some of his hearing.
Editing SECRETS revealed! replies to Joseph Kosak: You're right that the weakest link in the chain limits the whole system. Amplifiers can easily have noise and distortion levels WAY better than most recording formats. The recording is usually the weak link.
For most people the threshold of discomfort to pain is around 100 to 120 dB SPL, permanent major hearing damage above that. It's very rare to find a room with such sturdy construction or such a quiet setting that the background noise inside is less than about 30 dB SPL. Eddie Van Halen and some other producers have remarked on easily noticing a less than 1 dB difference in mix levels.
Unless you live out in the country or have massive walls, playback at over 100 dB SPL inside, in addition to being painful pressure for many people, might be enough to get a visit from the cops. So around 70 dB dynamic range is actually about as much as actually useful at home, which hi-fi vinyl through a good system can provide. Studio multitrack recorders, 2" 24 track, can provide that for the raw tracks. 16 bits gives some extra headroom when tracks are combined, the design goal intended to be able to accurately capture a full symphony orchestra blasting away without distortion.
No Nonsense Bennett: I love that Karen!
Editing SECRETS revealed! replies to No Nonsense Bennett: She's going to ask to speak to the manager in a little bit.
JulesC: Question: in a 16bit recording he says 6db/bit in 24bit audio are is this smaller?
Are there more steps per decibel?
Editing SECRETS revealed! replies to JulesC: @JulesC It'll be faster and more effective if you read and learn from those who know what they're talking about, rather than speculating inside your own mind based on an absence of information. The actual science, math, and engineering are available from many excellent sources once you're ready for it.
I'm sitting out from here. I feel any more back and forth in the comments, as an alternative to your making use of available articles, books, or courses, wouldn't support the educational goals of this channel. Good luck.
JulesC replies to JulesC: @Editing SECRETS revealed! I will go read of these references but in the mean time...
I do understand how it relates to the db but the db is different size depending on how far you get from the quietest available sample in the respective bit depth. Understanding it form the quietest being 0 db in 16bit and 0 db in 24 bit at +96db both db scales have the same number of digital steps but at the +144th db the 24bit has millions more digital steps, so the loudest db in 16bit has 1.6billion less steps in the sample than in the 24bit sample because its exponential. Normalize each the 16bit and 24bit to their loudest samples respectively then the 24bit's loudest db has many more steps per db over the loudest 16bit db.
i'm thinking of it from the river being empty rather than from being full following to your analogy.
Editing SECRETS revealed! replies to JulesC: @JulesC Apparently Youtube considered my links to be spam. So be prepared to type to remove the spaces.
Moulton Labs Dot Com slash more / bits_really_bits
" Happily, it is easy to express this resolution in decibels of dynamic range, to wit: each binary bit represents a power of 2 and approximately 6 dB of dynamic range. A 4-bit signal (2^4, or 16:1) has a dynamic range of 24 dB, while a 16-bit signal (2^16, or 65,536:1) has a dynamic range of 96 dB and a 24-bit signal (2^24, or 16,777,216:1) has a dynamic range of 144 dB."
Sound Devices Dot Com slash 32-bit-float-files-explained
"The maximum dynamic range that can be represented by a 16 bit WAV file is (0 dB – (-96.3 dB)) = 96.3 dB
16-bit WAV files, whether in a digital audio recorder or DAW software, call the largest signal captured 0 dBFS, meaning 0 dB relative to the full-scale (of the file). So, 16-bit WAV files can store audio from 0 dBFS down to -96 dBFS.
The dynamic range of a 24-bit (fixed point) file is (0 dB – (-144.5 dB)) = 144.5 dB
Just like for 16-bit files, audio recorders and DAW software call the largest signal in a 24-bit WAV file 0 dBFS. "
ccrma dot standford dot edu slash ~ jos / st / How_Many_Bits_Enough dot html
"each bit in a linear PCM format is worth about 20 log10(2), approx. 6 dB of dynamic range"
eit dot uni-kl dot de slash mpcs / grant / chapter2 / grant2_2 dot html
" the SNRQ in dB.... increases linearly with N at 6 dB per bit"
cmtext dot indiana dot edu slash digital_audio / chapter5_quantize dot php
"Therefore, a general rule of thumb: Every additional bit per sample size results in a ~6 dB greater dynamic range."
repository dot upenn dot edu slash cgi / viewcontent dot cgi ? article = 1144 & context = ese_papers
"Note that for each extra bit of resolution in the ADC, i.e., for every increment in N, there is about a 6 dB improvement in the SNR. Thus, there is a direct relationship between the resolution of an ADC and its SNR, and it is common to equate differences in SNR in dB to bits by dividing the dB value by 6."
If you know better than all of them, take it up with them, not with me.
Editing SECRETS revealed! replies to JulesC: @JulesC I'm starting to fade a little so if this doesn't work we can try again in the week.
Each bit of linear quantization measurement corresponds to 6 dB change in the input or output voltage. I'm too tired to run the math for you right now, maybe someone else can chime in on why 1 bit LPCM corresponds to 6 dB. I can assure you it's been well documented since I first read the CD specs in the 1980s.
If equipment was perfect, the top 16 bits of a 24 bit sample would be identical to a 16 bit sample of the same voltage.
Below the bottom of what the 16 bit converter can measure, the 24 bit converter can measure an additional 8 bits further down. If equipment were perfect, the 24 bit sample would be 256 times more detailed at capturing the tiniest tiniest tiniest extremely low level signals.
Imagine measuring water levels underneath a pier. 0 dB = the pier is flooded, full scale. Below 0 dB full scale = the water is this much below the pier. 16 bits measures down to the lowest level the water ever typically gets. 24 bits measures even lower than that.
In practice this won't happen during analog to digital conversion, because random thermal noise inside the electronics is at around -100 to -120 dBFS anyway. It's like having your water level measurement below the bottom of the pier, but the bottom of your ruler is stuck in the mud.
If you set an analog to digital convertor measuring signal coming in from a microphone, and calibrate speakers so that the same level is generated on playback, so that 0 dB FS matched 120 dB SPL, about the threshold of pain for most human beings, 96 dB below that is around 30 dB SPL which is quieter than almost anywhere on the planet that's not an anechoic test chamber. 16 bits really is good enough to capture and play back just about any content that's musically useful in real world conditions.
Doing the processing inside the software algorithms at 24 bits lets you have headroom and gain staging for mixing and processing. The 16 bit input recordings are extended with 8 more zeroes to fill out the least significant bits of the 24 bit word going into the processing steps. Then on output to CD truncate or, better, dither the output to 16 bits for delivery.
And as pointed out, however careful the audio engineering, most listeners will use the cheapest lossy compression, tinniest phone speakers, most mediocre earbuds on sale, or awful speakers that made a big boom in the store.
If you really want to learn the math and computer science of overtones, bits, and audio processing, 80s books "Horns, Strings, and Harmony" and "Musical Applications of Microprocessors" are great starters.
JulesC replies to JulesC: @Editing SECRETS revealed! Please explain how there are 65 536 digital linear steps in 24bit to get to -96db and in 24bit there are 16 777 235 liniar steps to get to - 144db?
A programmer on compufile explained pcm 24bit being better because there were 16 billion more digital steps a sample had.
Maids and Muses: You are only looking at 16 bit PCM correct? Whilst not exactly the same, I think a similar theory underlies the workings of 1-bit DSD. There you get a high dynamic range out of only a 1 bit signal but at a very high pulse-modulation frequency, effectively integrated over time with noise shaping. This works because the binary pulse train frequency is so very much higher than the highest audio frequency that needs to be reproduced.
I suspect in this demo the 220Hz can still be reproduced to some extent at the low level here as the dithering used can employ the 44.1kHz sample frequency, effectively doing a similar job to that of the binary pulse modulation used in 1 bit DSD. Thus the noise dithering can be shaped such that it has an integrated residual of the 220Hz sine wave buried in the noise.
However, I would bet that trying the same with a 10kHz wave wouldn't work so well as it is getting so much nearer the 44.1kHz digital quantisation noise frequency (ignoring oversampling); then filtering out the 44.1kHz quantisation noise from the 44.1kHz dithering/modulation that tries to integrate to a 10kHz wave becomes nigh impossible. Thus this increased dynamic range for 16bit 44.1kHz PCM would only work for low frequencies like the 220Hz used here, you couldn't do it to the same extent for much higher frequencies. Am I correct?
Editing SECRETS revealed! replies to Maids and Muses: @Melanie Zette DSD is just the raw oversampled bitstream without it being binned into 16 bit Linear Pulse Code Modulation words. With only one bit measurement, theoretically every sample could be 50% quantization error noise relative to the signal, as opposed to 16 bit theoretical quantization noise down at -96 dB, 1/64-thousandth of the signal. But with that quantization noise at 16 times the frequency of CD, theoretically the fuzz is at such extremely high frequency that it either won't get passed through the electronics, won't get pushed out of the speakers or headphones into the air, or won't be perceivable by humans.
As you mention, in practice most people can't hear any of the theoretical difference. As the math is harder than with LPCM and requires different algorithms throughout the mixing process, it never really caught on in either production or consumer levels beyond a tiny number of audiophile enthusiasts.
Melanie Zette replies to Maids and Muses: When I look at frequency response graphs, Dsd noise is closer to 20Khz than PCM. Some argue that it's the reason why some prefer dsd sound. I have an sacd player and frankly I've never been able to tell any difference
Editing SECRETS revealed! replies to Maids and Muses: "There you get a high dynamic range out of only a 1 bit signal but at a very high pulse-modulation frequency, effectively integrated over time with noise shaping. " Exactly!
Morbid Man Music: Your assistant may need to see a doctor. She shows signs of hypertension
Editing SECRETS revealed! replies to Morbid Man Music: It's because her boss keeps her dithering just a bit
Allen Cavedo: I loved the “See you soon” from the fake gal.
Tim Miller: "Wikipedia the source of all knowledge that is righteous and true" I hope this comment was ironic cause nowadays wikipedia is quite far away from that.
MF Nickster replies to Tim Miller: Sarcasm: The Best Thing Ever!
Marco de la Peña: Anyone knows which piece of software/web page is he using to bring virtual AI assistant Betty ?
Mark Fischer: I have a collection of over 3000 CDs mostly classical music, some with exceptionally wide dynamic range. I have yet to find a single where any part challenged the dynamic range of a CD. Here is a disc that is a good test for dynamic range capability. The Disney Soundtrack Pirates of the Caribbean Dead Man's Chest. It includes a symphony orchestra, a pipe organ, and a men's chorus all at the same time. Try first the very soft opening of Track 1 and then track 2 The Kracken. This will test not only the limits of these musical instruments but the dynamic range and frequency limits of a sound system.
Editing SECRETS revealed! replies to Mark Fischer: Not sure if it's true but I read (don't have citation for this) that CD format designers used a recording of Mahler's 9th Symphony as the dynamic range reference.
GumonX: Assistan is AI generated? 😔
Audio Masterclass replies to GumonX: @GumonX The scary part is that these kinds of things will look 100% real within a couple of years or so. Maybe they are already... DM
GumonX replies to GumonX: @Audio Masterclass indeed it is, a sad future were everything is fake, sorry for being an antiquated guy! And thank you for the awesome video…
Audio Masterclass replies to GumonX: Yes. In the future, I will be too. DM
Eddie Cucumber: The redook dynamic range is all very well and good, but unfortunately most music commercial music is compressed to between 10 - 15DB dynamic range. The same goes for SACD. This is a commercial mastering practice. So dynamic range on media is a moot point in the real world.
Editing SECRETS revealed! replies to Eddie Cucumber: @Todd Sauve I'm in an apartment. I have a neighbor who I wish would trade in his powerful subwoofer for a bad AM radio!
Thanks for the fun chat.
Todd Sauve replies to Eddie Cucumber: @Editing SECRETS revealed! I don't think they will ever dispense with the mastering process until AM radio is gone and people can only can only buy stereos of better quality than what is currently sold. The lowest common denominator sucks. But the bass region is where things get lost very badly in mastering, in my experience. Though I suppose it does prevent some nasty confrontations between neighbours, LOL!
Editing SECRETS revealed! replies to Eddie Cucumber: @Todd Sauve Sorry I didn't make clear that I was only trying to go for a laugh.
Auratones for the win! I wonder if Atmos will spark a new era of audiophile listening, much closer to what was intended from the full-range mix. Does Atmos dispense with the mastering step? I don't know whether it does.
Todd Sauve replies to Eddie Cucumber: @Editing SECRETS revealed! I did recognize the paraphrase from "A Few Good Men" but could not tell that you were joking. Now I do! 😉👌 Mastering engineers usually do the best they can and are hampered by the low quality of some of the listening devices people own. I remember the first time I ever set foot in a recording studio and they showed me the lower quality speakers they had to use in the final stage of the mixes they sent for mastering. This was back in late 1970s or early 1980s and they had to use these tiny little cube shaped speakers in order to keep the frequency range within these parameters. Compared to their JBL studio monitors with their individually amplified drivers it was poor, to say the least. 🤷♂
Editing SECRETS revealed! replies to Eddie Cucumber: @Todd Sauve No offense taken! I agree with you! I meant to twist Jack Nicholson's famous cinematic rant from the film "A Few Good Men" into a remark on mastering's role in the "loudness wars." If you aren't familiar with the film, then the attempted joke must have landed with a thud well below the noise floor.
Mark Hayman: Very interesting and informative video, I understand dynamic range on comms and radio equipment. Regards mark
William Palminteri: Interesting presentation, I've done these tests also, without the dither.
I personally prefer the term 'usable' or 'listenable' when discussing S/N ratios or any other audio phenomenon.
A sine wave buried in white noise is actually more annoying than the square wave of 1 bit sampling.
Maybe.
Both are pretty unusable or otherwise unlistenable.
Also, at a pleasantly loud 80 dB SPL listening environment, a signal at 2 bits quantization (-84 dBs). is inaudible.
But being 1/4 Welsh and 75 years old, these tests make perfect sense to me.
All the best,
Bill P.
Jerry Taylor: I really like Betty, I have no idea what we are talking about or the relevance of this noise but I enjoy your delivery!!
Melanie Zette replies to Jerry Taylor: 😃
George Ogrady: Better noise free
George Ogrady replies to George Ogrady: @Morbid Man Music thank you sir
Morbid Man Music replies to George Ogrady: Noise is your friend. Truth.
George Ogrady: 18k tweeter 20k no good don't work even in 31 band
George Ogrady: Perfact sound 320 bits audio and in peak
Sherrill Shaffer: I notice that the distorted 200Hz tone is much louder than the dithered tone when both are boosted by the same amount (doubtless because the dither noise contributes its own acoustic power to the total, in the dithered case). So, although the original un-boosted tone was at -90dB, the tone that remains in the dithered case must be softer than -90dB, implying that dithering not only removes quantization distortion but also reduces the level of the remaining signal below its original level - thus introducing a type of dynamic nonlinearity. More tradeoffs... I haven't seen this discussed anywhere.
By the way, what happens if you apply noise reduction in your DAW to the dithered sample? I suppose, if you had boosted the dithered soft tone back to a normal level before noise reduction, you could recover a clean tone.
Kalvinjj replies to Sherrill Shaffer: The distorted tone reaching the same amplitude as the original 200 Hz tone will have a TON of square wave harmonics, those all adding up to the sound, hence the perceived added loudness.
Now, I'm too lazy to calculate (integrate the waves and compare) but I would also assume that the distorted one also does indeed carry more energy on it by being at the same maximum amplitude.
Iain F replies to Sherrill Shaffer: I think the issue is with the normalisation. Adding dithering probably adds some peaks higher than the sine wave peaks. This means that the original level is not restored by normalisation.
Audio Masterclass replies to Sherrill Shaffer: Yes this is interesting. The peaks levels are the same but if I measured the subjective levels in LUFS I imagine I'd get something different and I might try that. As for noise reduction in these very low levels, maybe it will work but I'd more likely suspect some new kind of hell that we would be better offer staying out of. DM
Brian O'Brian: This stumbled across my feed and you so remind me of Paul McCartney, like his cousin or something or uncle
Audio Masterclass replies to Brian O'Brian: I just wish I had his talent. DM
Arty F Hartie: The best sounding cds are the ones marked AAD meaning they were produced using an analogue source, edited and mastered using analogue technology and put on cds as the final digital medium. Loud does not mean better. Huge dynamic range does not mean better. The best music reproduction is by using tape decks and reel to reel or cassette tapes. The equipment must be cleaned and aligned meaning the heads must be cleaned using isopropyl alcohol with q tips demagnetized amd lubed and the heads aligned. Not for lazy or people with preconceived ideas. The sound is like the sound of a live studio, music club or concert hall played using good well kept tapes.
Matthew Saunders replies to Arty F Hartie: @Editing SECRETS revealed! Very well said. Your comment cannot be argued with.
Editing SECRETS revealed! replies to Arty F Hartie: @Arty F Hartie It's after midnight my time, so I'm signing off now. If you have a further reply I'll see it tomorrow.
Editing SECRETS revealed! replies to Arty F Hartie: @Arty F Hartie I've no problem with that.
I also agree that different types of music production equipment usually result in different types of tonality, which some people enjoy more than others. If someone has found some music that consistently brings them joy to uplift their life, that's wonderful.
There are two things I don't get.
One is why someone would use a comment system designed to make it effortless to reply and have a dialogue, but then get upset when people do exactly that.
The other is why someone wouldn't feel it's plenty good enough to just say "I happen to enjoy this art more than that art" and leave it at that, but would try to explain personal preference as a matter of physics and engineering... even after multiple well informed people, with the original references readily at hand, clarify that no, the physics and engineering actually work differently than that.
Arty F Hartie replies to Arty F Hartie: @Editing SECRETS revealed! At the end of the day, you are you and others are others. What you like, others don't
Editing SECRETS revealed! replies to Arty F Hartie: @Arty F Hartie In Youtube comments, "leave me alone" is spelled "don't comment if you don't want anyone to use the Reply button automatically placed next to every Youtube comment."
Eric Austin: Was there a point?
Editing SECRETS revealed! replies to Eric Austin: There was, but it was 100 dB down so nobody could hear what it was.
Audio Masterclass replies to Eric Austin: Yes. Use dither if you're mastering to 16 bits. DM
Tomekichi Yamamoto: So, it becomes a digital soup.
Nice 👍
Marley Pumpkin: Betty… I love you ❤
Adam Machin: For me it didn’t sound any louder than the dialog.
Jeff Christian: Love your assistant, very cool.
walthaus: I dunno, sounds to me like you've moved the goalposts a bit here. If a system's noisefloor, being louder than the actual signal is not considered to be the lower end of the signal-to-noise range because the signal is still audible in some form or another, then it follows that the S2N ratio of cassette, tape, vinyl or wire recordings is much larger than previously assumed, because somewhere in that noise there is signal that can possibly be heard, if it's the right kind of envelope, attack, pitch or whatnot. I'm not sure I'll buy that, but to each his own.
Editing SECRETS revealed! replies to walthaus: Eventually the signal level is so small that it can't magnetize distinct clumps of rust or individual spots on the wire. Analog recording actually is digitized at the microscopic level: a magnetic domain is magnetized, or it's not. But as manufacturing processes don't spread magnetic particles perfectly evenly, every millimeter has a slightly different effective sample rate. I don't have the reference handy right now but I seem to recall a paper explaining there are about a thousand domains per millimeter. Multiply that by tape speed and you get the equivalent sampling rate, then by track width and you get the equivalent bit depth.
Audio Masterclass replies to walthaus: Your comment is correct if you substitute 'dynamic range' for 'signal-to-noise ratio', using dynamic range in the sense of including signal that is lower in level than the noise. So yes, even a wire recorder will have audible signal that is lower in level than the noise. DM
Teilo: So this is really the whole point of dither, and it does not, under any circumstance, increase the dynamic range of your material, but it does prevent quantization errors. You cannot get a 110db range out of 16-bit in any circumstance. That's simply impossible. The reason there is any discernible signal in the dither at all is because you have retained that -110db signal in the 32-bit float internal processing of your DAW, and when that signal is converted to 16-bit with dithering, the signal is quantized into the dither noise. The result is still a 16-bit file with 96db of dynamic range, with the formerly -110db signal quantized into the lowest 4 bits of the dither-induced noise floor. Because of dither stochastically smoothes out the quantization errors that would otherwise result, the signal is still detectable. But the resulting file is still limited to 96db. In fact, because of the dither, it's actually closer to 84db of dynamic range. The dither eats up some of that range.
MF Nickster replies to Teilo: @Jacob This is correct. With shaped dither you can reduce the noise floor to effectively zero in the most sensitive range of hearing, at the cost of raising it in the higher frequency range, but it would have to be VERY loud to be heard up there. You should check out Mike Clements's article on "Digital Audio: Bit Depth vs. Resolution," where he generates a signal at -114 dB using dithered 16-bit. You have to turn the gain WAY up to hear it, but it's clearly there, buried in the noise floor!
Jacob replies to Teilo: My understanding is that you can get far more than 16 bits of dynamic range out of 16 bit audio with noise shaping. Specifically in audible frequencies.
Audio Masterclass replies to Teilo: Respecting your point of view, I believe I covered this very early in the video. Wikipedia gives two definitions of dynamic range, one down to the noise floor, the other down to what is discernible as signal below the noise floor. I'm happy with either definition, preferably stating which one. Of course, perhaps a higher authority - AES, EBU, BBC whatever - may have a more definitive definition, if that's not a tautology. Whether one prefers distortion or noise, I guess that's a personal choice. DM
EgoShredder: I think she knows more than she is letting on. Just sayin' 😉
Alan M. Thornton: 🤯
Chunksville: We are talking of an audible sound even at 96db that cannot be heard unless you dare to turn up your amp way beyond usable listening levels, to over complicate it your analogue circuitry chain and components connected will have more noise being created than any digitally converted sound/noise anyway
Glade Swope replies to Chunksville: That may also be true of the input when recording. If the microphone has at least -96db of Johnson noise, that already causes a dither.
Dr Broncanuus: I didn't understand a word but would like to meet a real life Betty...
Editing SECRETS revealed! replies to Dr Broncanuus: @Dr Broncanuus She's kind of busy right now with all that pressure to come up with a new catchphrase
Dr Broncanuus replies to Dr Broncanuus: @Editing SECRETS revealed! hopefully Betty , will help me get over 7 of 9...
Editing SECRETS revealed! replies to Dr Broncanuus: @Dr Broncanuus Some says she's obtuse but David thinks she's acute.
Dr Broncanuus replies to Dr Broncanuus: @Editing SECRETS revealed! so she has the Right Angle on things ?
Editing SECRETS revealed! replies to Dr Broncanuus: If you get to know her more closely, you realize that at a low level she's really a bit square
Citizen: So, dither is removing the distortion on something I can't hear anyway? Ok, I appreciate this example is not representative of "normal" content, but now I'm going to have to set up test - how audible is the improvement made by dithering on regular content? I suspect I won't be able to tell the difference.
Glade Swope replies to Citizen: Trading that harsh square-wave-ish distortion for white noise improves it greatly at that level threshold that is "felt" rather than heard. The noise added is inaudible, but it cures the brittleness of the overall sound. It serves a similar purpose to the ultrasonic AC bias on magnetic audio tape.
Editing SECRETS revealed! replies to Citizen: Dither affects whether the lowest level bit that is used sounds clean or distorted. If you record a full scale 16 bit signal, the lowest level bit that is used is at -96 dB and you're right that you probably won't hear it. If you have an 18 dB dynamic range, fairly wide for modern recording of an expressive singer or trumpet player, you're only using the top 3 bits of the format and it's all zeroes for the remaining 13 bits. With the least significant USED bit at only -18 dB, you probably can hear a difference if that drops off smoothly or with quantization noise distortion.
Zbyszek Olko replies to Citizen: Undithered tails auditioned loud sound like a torn paper. Dither is used along with noiseshaping what further improves listening experience.
basspig: I did manage to get 86 decibels of dynamic range squeezed into a 16-bit audio on a Blu-ray disc. It's my ultimate fireworks Blu-ray which is a fully uncompressed natural sound recording of fireworks from the launch location we had to have special access passes and sign in Insurance waiver with all our recording crew but we made one heck of a recording. Only 1% of the sound systems in the world can play it effectively.
Editing SECRETS revealed! replies to basspig: @Carmine Dambrosio Fireworks show included for free with the audio track!
Carmine Dambrosio replies to basspig: The remaining 99% of sound systems will blow up their speakers !😁
basspig: Sure you can just use a dbx4bx in the signal chain after the CD player.
The Eyles: Is the noise shaped into a particular frequency band, or is it plain white noise. Does that make any difference?
Editing SECRETS revealed! replies to The Eyles: It does, frequency shaped dither is a thing.
SwishaMane420: Now put an entire track below the noise floor, then use AI to remove the noise, then normalize.
W Yuhasz: Thanks. Chesky records produced test cd in 1994(volume 3 in test series) showing impact on positive sound quality of using dither as well.