24 bits or 96 kHz? Which makes most difference?
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- Опубликовано: 5 дек 2023
- Which is more important, bit depth or sampling rate? What sampling rate is the best? What sampling rate is best for audiophiles? What sampling rate do I use? Featuring Audio Phil.
UPDATE
I made a comment about a Wikipedia page in this video. One of my viewers has since improved it.
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• 16-bit vs. 24-bit - Le...
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I love your sense of humor. You have a very clear understanding of the tech & you methodically dismantle the BS that so many pseudo-audio-experts spew. I record in 48k/24bits/chan, operate filters in 8x over-sample mode to minimize aliasing induced distortion. Final mix down to shaped-dither 48 or 44.1 in 16bits.
RUclips's algorithm didn't recommend me your videos until recently but it quickly learned that I will watch anything you want to make a video about 👍🏻. Love the content.
That name! lol bpt
I only like listening to the highest level of audiophile technical cheekiness and this channel is cracking! Another well done episode mate!
i think you didn´t understand the question
Quality, you do make my day!! Thx for sharing🎉😂😂
I dithered around so long that I can only offer a truncated opinion that 24/48 is good enough for me.
48k/24bits I looked at the options and those were the numbers that made most sense to me. Everything sounds great, I have flexibility.
Crazy 🔥🔥thanks for the info
I went through a blind testing and I could clearly hear the difference between 24/96 and 16/44, at least on orchestral music.
Excellent info. I always learn something new here. Thank you. 😊
In terms mastering from tape to CD, what makes most difference is: (1) the quality of the source tape (2) subtle changes to eq etc during transfer to bring "the breath of life" to the presentation (3) the quality of analogue to digital conversion (ADC). The limitations of early ADCs was recognized by Tony Faulkner, who modified Sonys for better performance & the team at Pacific Microsonics who were developing HDCD.
Finally another fantastic video about this topic besides Dan Warroll’s. We don’t think about needing to hear 96,000Hz frequencies when recording at 192k. What we are thinking about (in terms of auditory effect) is the blending of digital audio samples as they combine together in a virtual space. This “smoothing” effect becomes much more apparent on things like cymbals in drum overheads & instruments with longer envelopes that ring out.
When combined with digital (or virtual) signal processing like high shelves, you are able to use more EQ with less apparent artifacting and aliasing when compared to lower resolutions. Some plugins compensate for this by upsampling inside the plugin for processing, then downsampling on the way out, but not all of them do, and I haven’t heard of a DAW doing this with their console strip (please correct if I’m mistaken).
The argument listed here about CPU power becomes a bit more null & void as time goes on. 192kHz is also much better for any kind of time or pitch correction, since higher resolution gives more samples to stretch & blend. The last benefit I’d mention for 192kHz is the lower latency times on system buffers. On certain systems, this is extremely desirable and beneficial since desktop computer systems these days can handle the load with much greater ease than ever before.
Comparing me with the legendary Dan Worrall is almost as good as being awarded the coveted RUclips play button (which he has and I do not).
This seems to come from the "stairstep" myth of representing samples. Sampling theorem states that everything within the bandlimited signal is captured . Thus, 192k does not capture anything within the audible range any better than 48k. This is easily provable with analog equipment as was demonstrated in this video ruclips.net/video/cIQ9IXSUzuM/видео.htmlsi=KF5VEVHFq9ZM8885
"This “smoothing” effect becomes much more apparent on things like cymbals in drum overheads & instruments with longer envelopes that ring out."
Try it - but before you get disappointed: The most well payed audio-engineers whose sole job it is to validate audio quality can not head a difference where there is none.
@@ABaumstumpf I’ve gone back & forth a million times on it for the past 20 years and found slight, but very noticeable differences in such cases. Not everything requires such high sample rates, but it can transform how all of the tracks work together on a mix, and ESPECIALLY when hard quantizing drums.
Sounds more like you should try it, many, many times before relying on what someone else says about it.
@@JesterMasque Higher sample rates are massively important when doing any kind of intence time stretching in the digital domain. They can also help stop or reduce aliasing when you don't have oversampling avalible natively in a plug in that would benafit from it or in your DAW. If a top engineer can't hear aliasing in the audiable range, I'd question their position, and by using souly lower sample rates you will surely run into it if using diferant plug ins all the time.
Thank you very much for this highly interesting video!!!
Always providing top notch content with classy humor
16 bit 44.1kHz is the limit on a CD no matter what you up sample to there is no additional information.
its not about upsampling, but using high-res files from start
I think the low noisefloor of 24 bit is convenient for recording but at the same time I think my music does require high sampling rates as well so 24/44.1 serves me well
I’m also too 24/44.1 ✅ is enough. They say higher better. However, it is not always guaranteed that the oversampling results better performance😅😉☺️….
Great video and I agree with mostly everything you said. Whenever I multi-track, I use 24/48 as it's senseless to use anything else for the reasons you mentioned. When recording my vinyl records however, I always use 24/96 as that is usually what the master used in the process was given to the cutting engineer, on recent releases anyway. Anything vintage on vinyl I just use 24/48, but new releases on vinyl are truly magical and I don't want to miss a single harmonic.
"I always use 24/96 as that is usually what the master used in the process was given to the cutting engineer"
No. normal Vinyls were not even close to that. Commonly used cutoffs were in the range of 17-19kHz to not stress the cutter unnecessarily, with dynamic range rarely touching 16bit.
Recording with higher fidelity for preservation is fine, but don't kidd your self into thinking that vinyls are even close to being capable of reproducing those numbers.
I think they meant the masters that were given to the mastering engineer for vinyl were 24/96, not the final output. But to take away from your point, but interesting to see that the final output is so more constrained due to vinyl, crazy! But vinyl can still sound amazing.
Great video thank you 🙏🏽
The reason why you never see bitstream used in DAWs is they can't handle the processing that way. They would effectively have to convert it to PCM on the fly, process and then convert back to bitstream.
Appreciate the content and humour
Virtually all ADCs of the 90s were 1-bit 64fs. In other words, DSD is just recording the first stage of converting to PCM and does away with the need for brick wall digital anti-aliasing filters, as well as reconstruction filters on play back. Nowadays, most quality converters are 2-bit or even 3-bit, at 128-256fs but the real world performance isn't that much greater than can be achieved with 1-bit 64fs ADCs. The main difference is you can get away with not having an analogue anti-aliasing filter (usually 5-pole) in front of the ADC.
44.1 CD quality is all you need for audio. I'm old enough to be around when CD's came to be sold and how much better they sounded than records. Above 44.1 is in the realm of discussion only.
I have 1000s of CDs. I have nearly a dozen devices of various quality to play them on. I am not interested in other formats at this point in my life. It will be CDs until the end for me or until the power grid quits for all of us. Cheers.
I'm curious if the low-pass filter required for a flat frequency response using 44.1 is slow enough to avoid audible differences in the time-domain?
@@nicksterj When downsampling to 44.1khz though, a filter is required.
@@nicksterj Forgive me where I might be wrong. The filter for 44.1khz would need to have sufficient attenuation >22.05khz to avoid aliasing whilst having a flat response up to 20khz. It requires a fast roll-off over 2.05khz. Is the slope too steep (too fast/sharp) to avoid audible impulse response ringing? A slower filter could instead lead to either audible aliasing or attenuated treble.
You are so right!
Thank you for the video. Have you done a video on how the streaming services manipulate your music as far as loundness and compression when uploading your tracks to their platforms.
I have in mind at some point to compare one of my tracks on Spotify with the original master. Whether any difference will be heard through RUclips's audio mangler will be an interesting question.
Was entertaining to listen to and glad that it’s pretty much factually accurate up till the SACD as really you missed how the supper high frequency low but rate encoding actually works and why the recording industry would want to use this for archival and data storage. Well worth listening and even subscribing to. 👍🏻
As an electronics engineer, I really enjoy our down to earth technical videos, keep them coming! Regarding DSD & SACD, it's no coincidence that Mobile Fidelity Sound Lab use it for their digital releases... From a hardware engineering perspective, a bit stream D to A is much simpler than the alternative so it's much easier to do it right.
How do you mix the DSD streams? Or do you just release the recording without any mixing?
@@ianhaylock7409 conversion between bitstream and PCM is trivial and transparent.
Great video. That last bit was brilliant 😂 get it?
Double denim tuxes aren’t for everyone, but certainly brilliant LOL
One subject that some people ask me at times but unfortunately i lack the skill to explain as comprehensively as you is what happens when you
work in one bit/resolution rate and then export at another. Like what happens if you limit a project at 96khz/24bit and export at 44.1/16 and the other way around, what happens with the ISP etc etc. I think a video on the subject would be worth a lot
An correct optimal solution derives the sound wave using interpolation, e.g. cubic spline. Then it resamples the waveform to the new output format.
In a lot of DAWs, at least Adobe Auditoon, if you zoom in to the max you’ll be shown both the samples and the interpolated waveform.
There’s a lot of old misinformation about how very old stuff maybe worked with digital in the 80ies. According to that misinformation there’s a lot loss and strange dithering noise applied to counter act it. My friend tried to explain that voodoo stuff to me and I stared confused at him…
Pretty sure no DAW in modern times uses strange voodoo bullshit. Calculating the waveform and resample the to the output format is the same approach with barely any up/down sampling issues. Why you’d ever go away from the waveform in any application that prioritizes accuracy would be beyond my comprehension.
Maybe some hardware that requires absolute zero latency do weird voodoo still, but DAWs surely must just resample waveform at best accuracy possible.
@@randomgeocacher thank you for the elaborate answer. I’m asking because in the past somebody once told me that for one example, if you use a limiter work in one Bitrate , and then export to another, you might be ending up exporting the file without any limiting applied.I cannot, however remember the exact terminology what betray toward betrayed this applies
@@randomgeocacher A complication with doing things this way is that applying a brick-wall filter to a sampled signal may result in peaks which exceed the peak level of the original signal. As a simple example, if one has a signal which, after sampling at 1 million samples per second, looks like a 1000Hz square wave whose peak values are 0.01dB below saturation, and downsamples to 44.1kHz, each edge will have a certain amount of overshoot before the signal settles toward an equilibrium until the next edge approaches.
"I buy two copies of each CD so I have twice the resolution" ... that had me dying!
I genuinely used to be able to hear the difference between 48Khz and 96Khz when I was younger, but nothing more. Specifically when I would have a lot of high pitched distortion in a track with no LP cut off filter. But I'd be pushed to hear the difference now. My ears used to go up to around 22Khz, My left is down around 18-19k these days though and my right a bit lower than that maybe 17-18k (dam DJing headphones). Although sometimes I can tell if there's higher frequencies by how that top end range I can hear sounds, I can't actually hear them >20k frequencies anymore and there would have to be way too much of it for normal listening for it to even be noticeable to me in my audible range. (I've got speakers that go up to 25k, but I can no longer hear super sonic stuff at all sadly, dam age, I just have them so the cut-off is a bit further away from the range I actually can here currently).
you are full of sheet son
Thank you sir!!! A good video to show the kids ( I've been making them read Mojo Audio's: 24 bit Delusion).
Even CD quality is "only" 44.1 kHz (16 bit), enabling almost 22kHz bandwidth. That is what 10 year olds can rarely hear, even at higher volumes, and only when not masked by lower frequencies. If you are above 25 years of age - forget everything above 18 kHz.
I very much enjoy your educational videos. Would you consider making a video about harmonics or ultrasonics? Frequencies from instruments higher than humans can hear, that may or may not add to the perceived sound of music when played live or when listening to 96khz or above recordings through resolving audio hardware and or Hi Res headphones. I keep reading and hearing of this perceived sound that subconsciously makes the sound we hear more real. High frequencies that our body can sense but not necessarily hear, like sub bass below 20hz. I'm wondering about the benefit of adding a pair of super tweeters to my home hi-fi system. Would love to learn your thoughts. Zach
Broadcast audio is moving toward 48khz as the recording standard. Used to be 44.1 for a very long time. For playback, it's 48khz, but that is changing. The final link from the digital audio processing to the transmitter exciter is becoming 192khz, so that stereo multiplex and ancillary data can ge generated by the computer doing the processing.
What if we took the single-bit sample to an effective infinite sample rate. We could use amplitude to modulate a single stylus moving across a spinning cylinder coated with wax so that the number of samples is limited only by the number of molecules in the wax.
High-frequency response would be limited by the inertia of the stylus. 1 bit sampling at a high frequency is essentially what a class D amplifier does.
The first CD players when they came out in the 80s, had a characterstic “metallic” sound in the high frequencies, because of that steep curve filter needed to eliminate aliasing, which was destroying the phase. Then, overampling came along, generating non existent samples between the actual recorded ones using math, and a much phase friendly 3db/octave for example could be used to eliminate the aliasing, instead of an 18db/octave that was usually needed for the 44.1kHz. The oversampling would make the signal 4, 8 or even 16 times the 44.1kHz original signal, but only to use that kind of filter afterwards. TLDR: Our ears are much more sensible in phase shifts in the higher frequencies than listening to the individual frequencies themselves.
"Our ears are much more sensible in phase shifts in the higher frequencies than listening to the individual frequencies themselves"
This aspect is often ignored in those discussions. Consumers do not need higher sample rates for better frequency response, and if that is the thing people focus on when trying if they can hear a difference between 44.1khz and higher sample rates, there shouldn't be much of a difference, if any. But it can make for quite a difference in perception of placement of sounds. Both timing and level differences play a big role in that, and the timing requirements for that are much stricter than one would think when just looking at what frequencies we can hear.
@@nicksterj An article with a fair bit of hand waving, which requires very carefull reading to get some details.
For example, your claim that timing resolution really doesn't depend on sample rate... his article suggests that, but also claims: "Notice that the calculation of the time resolution does not include the sample rate. Nevertheless, it can make a difference due to the frequency of the signal being part of the formula. A higher sample rate permits higher-frequency signals, which means smaller time shifts can be measured at a given sample resolution"
That is simply deception, and while glossed over in the remainder of his article, it contradicts the exact conclusion you took from his article.
Additionally, even he is honest enough to point out how his timing resolution claim is very much affected by the choice of the signal he uses, and how a still unrealistic input which is a bit closer to what we'd typically see in music, will cause an order of a magnitude bigger time error.
When doing a quick read, his article is highly deceptive, and you for example have been mislead into believing sample rate has no relation to timing resolution, while a much closer reading tells you it totally does, even according to the article you reference.
The real points he makes is that it is too simple to equate sample rate and timing resolution, and that bit depth also affects timing resolution, and that is not wrong as such. This is actually well known, but neither of those result in the conclusion that sample rate and timing resolution are not related, merely that the relation is more complex than a simple 'sample rate equals timing resolution'.
@@nicksterj " But as I understand it, if you shift the phase of the input signal by some fraction of a degree, the sample values will change by a certain amount - at some point you reach a small enough shift that all the samples will be quantized the same way as before, putting a limit on the difference in timing that you can resolve."
That is correct, and explains why bit depth also affects timing resolution and the relation between sample rate and time resolution is not at all the same as the interval between 2 samples.
But it does not mean timing resolution doesn't also depend on sample rate.
Rather, both affect resolution, and not just timing resolution.
This is evident from very high sample rate 1 bit streams like DSD, which have very low sample width, yet produce excellent resolution both in 'levels' and 'timing'.
As a general rule, you can reduce sample width as long as you increase your sample rate enough, and the other way around, you can reduce your sample rate, as long as you increase sample width, but within reason. Your sample rate must at the very least be high enough to cover the frequency range you want to capture (ie, 2x the highest frequency you want to deal with).
@@c128stuff I agree with your comments. What I was referring to, considering the first audio CD players, was the analog filters that were used at that time to cut down any frequencies above 20kHz, in order to eliminate aliasing: i.e. two frequencies passing from the exact same samples you have recorded, when you attempt to to play them back. From a pure mathematical point of view, having a window of sound sampled at 44100Hz, can contain a spectrum of frequencies up to 22050Hz if you perform a spectrum analysis, but the same spectrum mirrored after the 22050Hz is also satisfying the same samples. Even though our ears obviously can’t hear those high frequencies, the differences between them create harmonics in the audible spectrum, and that’s why you want to cut them off. So if you wanted to have a frequency response from 20Hz to 20kHz from that sample, you’d need an analog filter which is flat in that frequency range, and then drops dramatically from 20,000Hz to 22,050Hz. That was the one introducing that phase shift I was referring to back then, considering the analog electronics of the ‘80s too.
funny thing is some 1980s famous tracks were recorded at 32khz 16bit pcm lol
If you are doing sampling where you are going to slow down the audio to pitch it down or make drones, higher sample rates make a huge difference.
So two ends of the range are Niquist sampling with infinite bits per sample giving infinitely accurate quantization , and on the other end is infinite sampling rate with 1 bit per sample. And we choose to work somewhere in between, giving best quality / size tradeoff
Thanks for sharing. Can't say I understand all of them but have some ideas of it. Someone told me before that 44.1 is the best.
If you record and mix in 44.1 or 48, then I suppose that's typically OK provided that the digital effects are oversampling to remove any aliasing from the application of those effects?
Correct. Filtered on the way in, filtered on the way out.
youtube didn't bring me here - i saw you before (admittedly among audio-engineering vids, but are the "list by" columns in my file explorer an algorithm?)
Well said indeed !!!
The detailed exploration of sampling rates from 44.1 kHz to DSD's 2.8 MHz raises questions about the impact of these rates on phase coherence in multi-microphone recordings. Could you discuss how different sampling rates influence phase relationships between tracks, especially in complex recording setups? Furthermore, how does phase coherence at higher sampling rates contribute to the spatial imaging and depth of a mix?new subscriber here
What may seem crazy today might just be every day in the future. I’m not sure if this is a version of Moores law but I know my ears are not going to double in frequency response any time soon. Great humour with Audio Phil a supporting cast who knows his thang.
As for the "utility" of it, well to each his/her own. I certainly see a use for a medium capable of storing more information than apparently necessary which is down to documentation/curating purposes.
Audiophile William here, I've tried and tested many highend DAC's over the years, i never paid any attention to all the numbers just the sound. I borrowed a MSB Premier DAC that does 44.1kHz to 3,072kHz PCM up to 32 bits and from 1 to 8xDSD, in the end i decided to keep my old modified Audio Note DAC from 1998 which sounded just as good if not better than the MSB Premier. My old Audio note dac manual says it has no over sampling, no jitter reduction, no noise shaping and no re-clocking and uses the highest grade AD1865, 18bit stereo converter chip what ever that all means?? Audio Phil tickles me every time 🙂
Thanks for that , Audiophile William !! I don't feel bad now, about really liking my 2004 Apogee Mini DAC ! 😂. I think the clocks and analog sections of DACS have much more to do with sound quality rather than the DACS used .. Almost ALL DACS today are perfectly capable on specs ...
Love the Audio-Note dacs. I have ANK 5.1 Signature dac kit 24/96 nos. best dac I ever heard.
All modern "brand name" DACS like TI (formerly Burr Brown) are absolute overkill for "listening" to CD's. I recently purchased a Rotel RCD1572MKII that uses the TI PCM5252. However it's more than just a $10.00 chip to do a wonderful conversion. The OP amps and output side design and components are also extremely important. In my opinion most US$1,000+ CD players are more than satisfactory to most ears without needing an external DAC. @@LukeSchneiderEWI
the super high oversampling does not make the aliasing issues go away. they have been already made in the ADC when recording the music.
I just listen to music and my latest CD player is even more "sensitive" to crappy recordings/mastering's than anything I've owned before. It is utterly merciless and as analytical as my B&W speakers are in a similar manner. @@stefanweilhartner4415
24bit 48Khz for recording, mixing, mastering
(24bit 96khz only for raw recordings and source archive purposes;
for special applications, scientific, we might go for 32bit 96kHz but those are not really for music)
If we have 24b\48kHz, then DSP (digital sound processing) should include 'over-sampling' (×2 or ×4 the orig. freq. 48kHz in our case).
Record near 'hot' levels (test with low + percussive sound for worst case scenario of a small 'headroom VU' - eaten 'volume units' by the low freq. test sound, and louder Peaks - the percussive hit), if clipping occurs for a few samples DO NOT overthink it - they can be restored in post-recording\pre-mixing production!
New advances in computing are looking at replacing digital processing with analogue processing -or digital analogue hybrids- due to being more energy efficient, etc. Well sound processing and storage may go back to analogue again?
I just love how you cleverly are able to push all of the right buttons, without actually having to push them. LOL Another very well-done and informative discussion sir!
The other point about filters that almost got talked about is that steeper (higher order) filters generally introduce unwanted phase shifts. I think a lot of people can notice this.
This is a good point. I mentioned phase very briefly here and it's a topic I will cover in more detail in future.
@@AudioMasterclass YES, YES! You did drop an ph-bomb and I noticed it, thanks!
@@AudioMasterclass As Captain Kirk often said, set phasers to stunning.
The thing that is nice with higher sampling rates is that you can line in feed directly a very good 44,1 khz recording and upsample a higher signal thru a ok Dac.
Then pass the output to a console and finally to the recording input of your sound card.
The console is really interesting because here you can modify the sound and induce a surround effect to the dac output signal and makes it actually produce audio to those frequencies above 22khz all the way to 48khz, within a 96 khz recording sample.
I have digital samples of each pipes sounds of my vitual pipe organ recorded at 96 Khz/24bits. Up sampling from 44.1khz won't give the same detailed timbres as shown by waveforms.
When recording with High quality sound card, the ADC is in hardware so it won't affect much CPU cycles.
OP says he can't hear the difference. But OLDER people like him has "presbycusis". That's why opera is mostly attended by older people since young people finds opera irritating.
I believe that some people used the 48khz rate because that is what film and television production requires. It fit into their standard of frame rates. SMPTE frame rates, etc.
The reason I heard for the 44.1Khz was because the human hearing has a frequency range of 20Hz to 20KHz and, according to Nyquist, to convert an analog signal to digital without any loss you have to sample it at twice the rate of the maximum frequency.
By sampling it at higher than 44.1KHz you're just losing storage space.
Yeah, its due to that as well but the exact sampling frequency apparently originates to that audio recorded to a video recorder that he spoke about, i havent heard about that before this but it was common earlier on to use the same formats for better compatibility betweens systems, nowdays things are often far more flexible. I agree, 44.1KHz is already "full res" anything over that is mostly wasted space and CPU power, i record my band in 44.1KHz 24 bit in my DAW for our albums.
Oh, my hard drive stuffed full of 96 vinyl rips begs to differ it's just a waste of space LOL.
Interesting video. Thanks for that. Making a jump to the real world. Our hearing is analog and we use digital as a bridge. I went to a Roger waters where he performed " the wall" and during the act they build up a wall. at one moment you don't see the band anymore and you are looking at that wall. I made the joke that the band is likely backstage having a drink while we now listen to a tape. But that's in essence the best thing that can happen. If I can't hear the difference between a live recording and the playback I'm 100% satisfied for I'm there when I close my eyes.
And what comes closest to that ? is it still tape ? is DSD the best thing ? or PCM ? I don't know for I don't have that reference.
Thanks for this piece, I really enjoyed it.
Having been trained as studio engineer back in the 80's, back when we were getting our heads round digital and just learning to edit without putting some tape on the cutting block, I feel I know many technical matters in regards to sound reproduction and recording techniques, but I learnt several things from watching this.
In addition to my more formal training, I've also been an 'audiophile' since I bought my Roksan Xerxes turntable as a young bushy tailed 20 year old, and have spent the last 40 years enjoying fine home reproduction and keeping abreast of the developments in the industry. 40 years of this has clearly shown me there is still so much we dont know and dont know how to measure. Whilst I'm a scientist by nature and training, Science has always been about explaing the observable to me, not dismissing observations and so I'm happy to acknowledge that amplifiers sound different and that cables sound different. When its as clearly observable as I hear it I accept my experiences, guided by many years of both home and Studio Audio.
I'm almost 60 now and I'd be lucky to hear anything much over 14Khz. But given a known decent digital mastering, I can wholeheartedly tell you I can hear a difference between a 24/96 and 24/192 version of the same master, and its not a small difference. This is not expectation bias as I can get my wife to swap between the recordings (easy with Roon and access to all the files via Qobuz or similar) and identify the recording sampling rate with very high accuracy. My wife just like a good sound, but she has regularly commented on some studio masters sounding astonishing too even though she does not undertsand why they are different. I cant explain why I can hear it so clearly, but I accept it.
It is of course the mastering that makes the most difference to the quality of a recording and I can also point to many remastered Albums in 24/192 or 24/96 which sound much worse than an earlier master on a Japaneses CD at 16/44.1.
With so much confusion and so much we still dont know, its no wander the industry is plauged by those peddling 'snake oil' and their very exsistance gives those of us happy to work with the observable, some resistance from those that confidently state 'if you cant measure it it doesnt exsist'. Its always been a conundrum for those of us who indulge in some audophilia whilst still seeking the scientific answers of how to measure what we can observe.
Phillips proved that sampling rate matters and bit depth is overkill just a few years ago - back around 1980 when they introduced 4X oversampling (I bought one of the first Magnavox players when it came out - FD3040). Those convinced that 24 bit or even 16 bit resolution is crucial are ignoring the long, long history of digital audio. The product of bandwidth times power is the key metric that matters in any system that deals with sound. Bit resolution mostly affects the power resolving capability and sampling rate affects the frequency resolving capability. With digital audio, 14 bit resolution comfortably handled enough dynamic range to surpass human hearing capability. And 44khz sampling was more marginal in its ability to handle the 20khz bandwidth requirement. This was done to reduce the amount of data storage requirement for an hour's worth of music. And Phillips cleverly dealt with the "brick wall filter" D/A conversion issue by employing 4X oversampling. There's not much else to say about this issue. When it comes to human hearing, 16 bit is overkill and 24 bit is ridiculously overkill. No microphone in existence has the ability to cleanly generate signals that take full advantage of what a 24 bit system can deliver - so what's the point?
Since modern AD and DA converters work with pulse density (1 bit), the precision must decrease with increasing sampling frequency while the chip master clock remains the same. However, at 96 KHz you have twice as many samples with less precision and an anti-aliasing filter with less phase rotation and better transient response. I would consider 96 KHz to be the best compromise for real analog sources, although for my ears it ends at 18600 Hz. I've never been to a disco.😉
I used to mix in 48Khz, while this maybe enough to convey the mix to any listener, the math behind it or rather said the simulation of analog gear will be less acurate compared to 96Khz
And for those who work like people in the analog domain, comiting to a sound or dymamic feel is mandatory to keep the computer fast. I always commit virtual instruments with loads of plugins and bounce them to an audio track. If I could i would rather mix on 192Khz that is what most commerial studios do and they are not wrong. Mr. Neve has said this and he is right, there is an psychoacoustic ellement in the high frequencies that is experienced by listeners, this was one of the revelations mr. Neve had during his life.
At 192Khz reel2reel archiving, it also captures the 50-70Khz BIAS oscillator faintly from the original recorder.
Now this oscillator drifts as the recorder warmed up (tube era), either its mass, or surrounding environment.
If it was stable, it could have been a better source to calibrate the motor/mechanical speed shifts shift on original deteriorated tape, but many times there is faint 50hz hum, assuming that was more accurate, I use that.
Thus, the Plangent Process works so well at correcting wow and flutter.
I believe its "Let's get started!"
96kHz is very popular in live sound, as it halves signal propagation (processing) delay, which improves fold back to performers.
Regarding DSD & SACD, to my aged, & admittedly rather abused, hearing; I find it sounds far more natural, open & clean (best words I can think of) but that might just be down to how they were recorded & mixed and that they have not suffered from being limited to death as is current CD mixing & mastering practice.
Another DSD/SACD enthusiast like me. I agree with you wholeheartedly regarding the cleanness and openness of the sound. I also find it true with regard to transients such as cymbals and drum thwacks, Keyboard fingering on the piano is more evident too. I have been downloading 5.6448MHz DSF (DSD128) albums from NativeDSD Music for some time now and most of them sound wonderful.
Native DSD has much more care taken in its engineering and capture than studio PCM for popular music. Upsampling Redbook PCM to DSD is objectively worse and silly. Invest in better reconstruction filters or accept the source was a poor recording.
I upsample pcm to DSD1024 using very powerful modulators and filters from a powerful computer to a 1 bit discrete dsd ladder dac, easily destroy any pcm I've tried. DSDAC1.0 can also do this without a PC is partly why it sounds much better than anything else in its price range.
Neither matter. 16 bit/44.1kHz is the maximum that makes sense. Unless you have bat ears and live in a completely isolated room in the countryside.
Jup you don`t need to compress and push to the limiter life as much as the PA is loud enough to kill your ears anyway. But the highest frequency you can reconstruct is at half the sample rate. Infact 44kHz and 96kHz were chosen to get a clean 20kHz or 40kHz signal where the rolloff is neglectable for the human ear.
The real issues are tied to the low pass filters and aliasing which you briefly mentioned. The filters are not there to filter the sample frequency,but the signals above the sample rate. Very steep filters cause their own problems both in recording and reproduction.
Quite. The likes of our esteemed host never seem to grasp the problems with such steep filters so close to the audio band. He (and countless, like-minded reductionists) never stop to wonder if the "crystal clear" sound of digital is actually realistic.
Also the digital synths sounds drastically different on different samplerate
Depending on how SRC works in all daws
When you render project in different sample rate in Ableton you will hear the difference in mix
While in cubase/nuendo you will not
You should not worry about filters at all!
@@nicksterj sure, but the quality of over sampling is another part of the reconstruction filter which still needs to be done properly. The same goes for down sampling.
@@nicksterj I agree yet some systems still sound better than others. 16 bit 44.1 kHz has enough information if processed correctly for home audio. 20 bit should match pro audio dynamic range. If you have insights on why higher sampling rates often sound better I would be interested. As a note I have worked on DSP systems and have seen examples of poor application of theory so I am not convinced it doesn’t get screwed up in audio from time to time.
I won't get into the "what bit rate should you record at" debate. I mix at whatever rate it was sent to me. I've done 24/96 since around 2004 for the simple reason that it (seems) that my plugins create less undesirable artifacts at that bitrate. But I wouldn't use processing power as a reason to use BIT/khz. Back then, I'm pretty sure I was on a dual 1.2ghz G4, PT HD2 and cheap 4 drive RAID 10 IDE. I'm pretty sure I got 32-48 audio tracks with lots of plugins with no issue. The cheapest mac mini you can buy right now can run circles around that without any DSPs.
Brilliant! Ya old bloody bastard! Well done... Ya know it all comes down to a human ear. Just like modern TV's that claim a "billion" colors, when the human eye on average can only see about one million. The same applies here (or should I say "hear"). Keep doing your stuff. Love it...
I still use a Apogee Rosetta and record on 44.1-16 and 48-24 bit and happy with it .
Thank you for the ever exilarating but serious presentation. I use, if I can choose, 48KHz sampling rate and 32bits floating point resolution (mostly to avoid internal clipping). And I export my masters in 24bit 48KHz AND 16bit 44.1 properly dithered. I have tried 96 KHz and I DON'T like it unless it is for Jazz or Classical recordings. For Pop and Rock I feel is losing some grit (It might be some sort of psich-acoustic impression on me knowing the sampling rate. In general I am fully satisfied with 48KHz and if I can use oversampling in som plugins for extra resolution in some cases I am fine. The most important fact is that if it sounds good (to me) then IT IS good and that's it.. And if in 30 years from now they are still reprinting my recordings, then, good luck. I don't think any sample rate difference will make a difference. But who knows... When I built my first home studio around an 3340S Teac in 1975 I was dreaming to have a way to have a full 24 track studio in my bedroom. And in 1992 I was dreaming about a software I named "de-blender" that was capable of separate the individual instruments from a store recording... And here we are... both are a reality now. How impressive. The next step is a software that mix the tracks with A.I. (hahah... ) But I hope not as good and as artful as myself. Thank you for your contribution to the demistification of sound engineering.
We should consider this debate over, and start discussing lossy/lossless compression formats more! Audiophiles are saying it's all "mp3" but modern compression can achieve way better sound while using less bandwidth/storage. The Opus compression kinda blew my mind, as it is maybe 3x more efficient than mp3 (and used by youtube for its audio).
Should we? Says who?! 😂
@@fernandofonseca3354 I say. Opus is indeed amazing.
@@fernandofonseca3354 says me, an audio professional and huge nerd
Good, then go ahead and by all means hijack the thread! Enjoy your christmas day. Next!😁
There are more reasons why 24b96k is better than are listed in this video but really I’m here to say that I’ve been producing performance based recordings (live instrument pushing air at microphones) of over 24 tracks per song at 24/96 for years. Yes I had some cpu load issues on the 2010 Mac but on my new M2 MacMini it breaks NO sweat.
I would recommend folks stick to 24/96 if possible and never go below 24/48.
Gotta love audiophile Phil.
I feel as if I've known him all my life!
I know he is just - as it were - your alter ego.
So every time I see him, In my head he's audiophile Shill.
I record tracks in Reaper (DAW) and have always been a bit unsure as to what frequency and bit rates to choose. When I master (usually electronic analogue music) in Ozone 7, I always opt for 24 bit at 96khz. The mastering figure are what producer (Tubular bells) Tom Newman told me he opts for.
Recording 24 bits has its pros, sometimes, and just maybe, but working with 96khz is totally unneeded unless you need to record some material to pleasure the bets. Using 48khz is enough, jsut use oversampled plugins if you use saturators, compressors, and any other plugin that creates aliasing.
semoga sehat selalu abah 😊
Well, there are benefits of working with higher frequencies and higher bitrates for a DAW internal audio engine, allowing more accurate multiple tracks mixing and audio processing.
Obviously afterwards, for whatever actual output format, 44.1/48 kHz are good enough.
Working with higher bitrates also allows a lower input latency while still having a decent/large buffer size (avoiding buffer under runs).
With the newer CPUs, I feel 96 kHz or even 192 kHz shouldn't be an issue but unfortunately there are plug-ins (VST64/VST3) which won't work if your DAW is set to such high sampling frequencies...
I found myself having to limit my expectations and stick to 48 kHz with a pretty high buffer size (2048 bytes) if I want to be able to stack many synthesizers with no crackling...
The input latency is pretty bad but I can live with it.
Doesn’t a high rez like 24bit/192kz help the mix tracks in the DAW down to stereo?
Please give me your thoughts. I'm running a HD 3 system using protocols 10hd. My computer has 1tb hard drive and a 500g ssd. I want to multitrack bands up to 24 channels simultaneously at 192k 24 bit. Then when doing my mix add plug ins. Do you think that I have enough processing power? Do you think I will run into problems with lag, or anything else?
You'd have to test it. In particular test it for reliability because no-one wants to lose a take because the DAW has stopped. If it works OK, then it should also work for mixing. Most DAWs these days have a freeze function so you don't need to have all of your plugins running all of the time.
I use a streaming service that has both CD and Hi-Res versions of songs/albums. In some cases the Hi-Res versions do sound better. I've been wondering if this is because the Hi-Res versions have been mastered better, perhaps not being victims of the Loudness wars. Do you have the ability to see if there are any differences between different versions of streaming music (44.1k/16bit, 48k/24bit, 96k/24bit, 192k/24bit)?
@@nicksterj I have compared many LP's to the CD version and some sound better on LP, others on CD. One LP, Just One Night by Eric Clapton sounds worse on the CD, but sounds just as good as the LP in 192k/24bit version.
You may be right! I have tried all sorts streaming, but it never beat the old cd-sacd!
Debunking the Digital Audio Myth: The Truth About the 'Stair-Step' Effect
ruclips.net/video/cD7YFUYLpDc/видео.html
I'm inclined to call this the myth that never existed, rather it's a hangover from how the sampling process was explained in the early days of commercial digital audio. I might make my own video on this at some point.
Correct me if i’m wrong, but was 44.1 not a compromise for both NTSC AND PAL? That it was the ‘common’ frequency for both systems?
There is no common frequency between pal and ntsc. Umatic is obviously analogue composite, not that much better than VHS. Hi band Umatic was better.
they only match for NTSC B&W. PAL is 294 scanlines * 3 samples per line * 50Hz = 44.1k, B&W NTSC is 245 scanlines * 3 samples per line * 60Hz = 44.1kHz. Color NTSC is 59.94Hz = 44.056.
The Umatic machines used to record CD masters were specially manufactured ones to run at 30fps and only recorded black and white.
There was an alternative sampling rate of 44.056kHz, that enabled NTSC colour video recorders to be used, as NTSC colour actually runs a at 29.97fps.
Most DACs will turn PCM into a high frequency stream of single bits, prior to filtering down to analogue. SACD just skipped that whole step and saved some hardware.
So if I only listen to CD quality music, my upsample setting should be 88.2 on DAC?
This question came to me in the mid 90's when Pioneer developed a "Wide Range" series. At the time I bought the CT-95 cassette deck (10Hz to 30KHz) and the DAT D-07 which recorded at 96KHz but at 16 bits. A few years later I bought the Tascam DA-45HR which recorded in 24 bits... but at 48KHz. I used both to record vinyl. Well, the comparison of the two is never a pure comparison between 96KHz and 24bits, as there are many other factors that influence sound quality. It is difficult to choose between one and the other. I still have both DATs.
Sony also had DAT that recorded with SBM= super bit mapping. then they also had a CD recorder with SBM, apparently they also had a professional one
could we record at 8 bit 192k and then master in to 24/48k ? (later) oh i think you just answered this
The guy at PS Audio said on RUclips that they have started using DXD when making recordings which they then transfer to DSD. That is, significantly higher frequency than 48 kHz, they have a record label called Octave Records
Note that guy also answers questions from people on youtube . If I understood correctly, he is the founder and CEO of the company
I will never get tired of listening to Audio Phil.
I had trouble putting my finger on what it is about Phil that I like so much but I think it's the confidence he exudes while he speaks his over the top audio-science jargon. He's like a skillful politician who makes you want to believe that what he's saying is true.
Higher sampling rates are good for heavy processing in daw. Warping stretching pitching and so on. Especially for slowing down or pitching down.
That makes sense to me :) also if you pair with a specialty small condenser mic that can capture beyond human hearing range (there’s a few that aren’t crazy expensive) you could even recover human inaudible higher frequencies. In theory it could be cool to hear animals like bats well, or recover unheard ranges of chaotic sounds like explosions or crashing glassware..,
I read not too long ago in some forum that 96 kHz sampling is the thing if you are thinking on releasing vinyl records (analog). I don't know if there's any truth in that statement or it's just BS as per usual.
Discussing 32 bit float soon? Just subscribed to your channel. Thanks for the video. Cheers.
32-bit float... Yes, yes, yes, then no. More in a future video.
I don't care about bits or KHz or sampling rates. All I know is that digital music sounds so much better than records and tape. While I still love my retro component audio system from the 80s, my "go to" listening experience is my tiny Sony MP3 player and a pair of Logitech THX speakers and subwoofer or a good pair of Sennheiser headphones. No hiss, no rumble, no clicks or pops, perfect channel separation, bass I can feel even at low volumes, highs so crisp I swear they could break glass.
Mp3 is mainly sacrifying high frequencies (to obtain its low bitrate of max 320kbps). With 320kbps you can loose 60% of the information in the audiofile..... that s a lot and you would hear that on any decent audioset (not a tiny mpr3player).
@@thepuma2012 i think he wants to say that even an mp3 sounds better than any old audio medium from the 80`s
which i agree with tbh. I mean there are some good Vinyl Decks but they dont sound „better“ they just sound „different“ imho and
i have a decent Deck and enjoy Vinyl once in a while but if you compare for example bass heavy, (relatively) modern music
on Vinyl and the same Song-Track on mp3 320 its day and night. You cant push Vinyl too much when it comes to stereo information. You need to stay much narrow-er in the lows and in the highs on productions meant to be cut on vinyl.
Vinyl simply cant keep up on a technical level.. But i think it`s predominantly a matter of taste.
Tape also sounds cool and i basically grew up with vinyl and tape but technology has simply moved on. I had the best and most expensive sony walkman (the ones covered in chrome) in the 90`s and then Minidisc came and i had a friend saying „The worst Minidisc walkman is tenfold better than any tape walkman“ and at first i did not want to believe him until i compared them. It was simply day and night
To your "60% loss of information“ i can say that i have encountered very very few people who can actually hear a difference
of 320kbit to Cd resolution given that you will use a good D/A converter for both.
My cousin for example will throw all my arguments and say vinyl sounds better. Its a funny subject thats for sure
@@saardean4481 ok. I understand all you say about the vinyl and 80s audio. But not hearing mp3 to CD difference with a good DAC i find surprising. Indeed a funny subject as you say 🙂
@@thepuma2012 Surprises are the salt of life. I am worried more about the increasing lack of dynamic range and abundance of distortion in modern music rather than mp3 artifacts. As for your reference " But not hearing mp3 to CD difference“ maybe it would help to read again. I said " i have encountered very very few people who can actually hear a difference
of 320kbit to Cd resolution“. If you on the other hand have encountered many people that can easily distinguish quality differences between 320 and cd in a blind test then you are a lucky person surrounded by gifted people. You should treasure this 😉
@@saardean4481 Absolutely agree about that "loudness war" issue! I do have a Blu-ray player, but i refuse to buy any audio blu-ray because all of them have compressed DR - even so that an LP version of the same music can have more dynamic range than the disc version. Also on stream-media that problem.
Why would one use a “ roll-off type filter to eliminate the sampling frequency when it is fixed. If I were designing a system I’d be inclined to use a notch filter which would likely have less effect on the desired band pass frequencies.
The critical part in the whole chain from analog sound to samples on a cd is the anti alias filter. What is the cut off frequency, how steep is it and what is its phase spectrum. This is an analog filter, applied before sampling. So how does this work in practice, that is what I like to know.
Audio Phil is my new hero...
Send pics when you have your double denim tux.
What do you think a sinuswave look like at a 44 KHz samplerate? Thats why you need higher rates.
48K for tracking - as my channel expanders are ADAT, I'd lose half the outputs on each 8 channels at a higher rate, but 48 still sounds great for me. One benefit - as I understand it - with higher sample rate capture above 48, is the files work better with elastic audio processing ITB.
The last sentences definitely for me
Is it true that Audyssey and Dirac downsample hires music to 48khz/24bit (or 32? ) ? So if I´m using calibration, does hires music make any sense at all?
I don’t know the answer to that but it is a topic I might consider for a future video.
Building a intercom so pretty interested in audio quality :)
I guess there could be issues with the digital audio stream during rotating videoheads switching as well as with multiplexing the stream with sync/blanking video pulses to cheat the video recorder.
The problem, as I understand it, was the dropout compensator which had to be deactivated. The Sony 1600 wasn't the first digital audio recorder to record onto video tape, nor was it the last, but DAT - also with rotating heads - mostly put an end to it (apart from CD mastering).
My late father (who did a lot of voice-over work) was acquainted with a prominent sound engineer (in Australia) who did a lot of work recording/engineering orchestral music (classical and sound track) who 'bounced' multi-tracks to stereo _through an analogue desk_ (back to digital) because he found the results to sound _considerably_ better than using a DAW's 'sum and differencing'. A digital zealot and reductionist like our host will, of course, find the idea preposterous. BTW, 2822.2 kHz is the sampling rate for 'DSD64' as it first appeared nearly 30 years ago (as an industry archival format, not an "audiophile" one - most people seem to be ignorant of this) higher frequency/resolution DSD128, DSD256 (and even higher still) have long since been available (thanks to those nutty audiophiles, presumably).
Those are some nice glasses. 👍
www.tigerspecs.co.uk/item/jelli-multi-coloured-reading-glasses
Damn! So I am not the only one, who thought ... What are these colours ... Reflexions?
Nice!
@@AudioMasterclass 😱, I thought those were some pricy designer glasses. They sure look like It. Thanks for the link, they might not look as great on my face though 😂
Entertaining, and actually, very strangely, DSD (SACD) does sound quite good.
Why does dolby atmos youtube music, and non atmos youtube music (skrillex etc), sound better on a samsung galaxy s8 android over a fo48u and x670 gaming x motherboard using kns6400 and 8400 headphones? The s8 is 32 bit 384khz while the montor and motherboard are both 24 bit and 48khz and 196khz respectively. Does this not prove that 32 bit is superior to 24 bit for those with the correct speakers/headphones?
Just can't get over the modest title this nobody has given his yootoob channel.
'Audio Masterclass' as a brand dates back to around 1998 and has covered a wide range of my activities. Considering what I'm currently doing on RUclips I'd like to change it but I'm worried that if I do the algorithm will no longer like me.
two-channel 16-bit PCM encoding at a 44.1 kHz sampling rate per channel, with four bit ovesrsampling and error correction
from heaven is fantastic.
One reason for 96 kHz was the 90's discovered or before Dess had discovered that record players gave off sounds that were above 40 kHz. Pioneer even had CD players that artificially created higher frequencies. if I remember correctly, refer them to some survey that showed where people heard sounds more clearly if there were higher frequencies Even on AM radio
As audiophiles can hear what no mortal who have participated in scientific experiments can hear, perhaps they could donate their ears and brains to scientific research; preferably _ante mortem_ .