24 bits or 96 kHz? Which makes most difference?
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.
KZhead'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
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 KZhead 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 kzhead.info/sun/lq2Kaa2QioewqIU/bejne.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.
Quality, you do make my day!! Thx for sharing🎉😂😂
48k/24bits I looked at the options and those were the numbers that made most sense to me. Everything sounds great, I have flexibility.
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.
Excellent info. I always learn something new here. Thank you. 😊
Thank you very much for this highly interesting video!!!
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. 👍🏻
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!
Crazy 🔥🔥thanks for the info
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
Great video. That last bit was brilliant 😂 get it? Double denim tuxes aren’t for everyone, but certainly brilliant LOL
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.
Thank you sir!!! A good video to show the kids ( I've been making them read Mojo Audio's: 24 bit Delusion).
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.
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.
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.
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 KZhead's audio mangler will be an interesting question.
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.
I dithered around so long that I can only offer a truncated opinion that 24/48 is good enough for me.
Always providing top notch content with classy humor
Great video thank you 🙏🏽
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.
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.
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.
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!
"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
Appreciate the content and humour
I still use a Apogee Rosetta and record on 44.1-16 and 48-24 bit and happy with it .
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😅😉☺️….
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...
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.
funny thing is some 1980s famous tracks were recorded at 32khz 16bit pcm lol
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?
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.
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.
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.
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?
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
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.
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. @@JohnDoeEWI
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
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.😉
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
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 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.
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
My sampled instruments are 24/48 so I don't bother using anything other. Equipment is important and ensuring you have good AD-DA conversion and monitoring equipment. As for venue recording - keep it simple. Good tear down of the matter.
You are a healthy, intelligent man!
I went through a blind testing and I could clearly hear the difference between 24/96 and 16/44, at least on orchestral music.
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 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.
Wauw, there is some deafening truth on this channel. Love it! As I remember being told back then, that the record companies where affraid for their buisness when the DAT came out, since you could make a digital clone of a CD, hence the 48kHz. For a long time it was impossible to make a digital copy to DAT from a CD-audio, or record analog to a DAT on 44.1 kHz. Most of the video-broaadcast is still working on 48 kHz, and they won’t upgrade to 96kHz anny time soon, since the benefit of audioquality is minime compared to cost of doubling the data rate. But for acoustic music production I would strongly advise to record in 96 kHz 24 bits, mix in the DAW in 32 bit flow or better, and only SampleRateConvert after mixing. SRC can induce very nasty audible side-effects. Imho the benefits for recording and mixing at 48 kHz are lost with SRC if you need to go to 44.1 kHz for CD-audio. Our ears can only hear maybe up to 15 kHz or so, but what about the harmonics created by insruments above 20 kHz? Take two oscilators, one at 25 kHz and the other at 30 kHz. Now vary one of the two, and listnen to the harmonics you can perceive with your ears. I presume most fo these harmonics are lost when recording at 44.1 or 48 kHz.
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.
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.
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.
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 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.
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.
With regards YT's algorithms, I'm not so sure they're clever at all. They show videos on similar topics to the ones that I've watched recently but... they also persist in recommending items about Class 55 Deltic locomotives that my son used to love. Love that is... 10 sodding years ago! BTW... loving your channel :)
Unfortunately the algorithm heard you when you mentioned Class 55 Deltic locomotives and you will be seeing more of them soon. And now so will I.
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.
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.
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.
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.
You need high sampling rates to reproduce transients as these contain multiple harmonics of the original frequency. The harmonics in isolation are above our range of hearing but the effect they have on the shape of the waveform is detectable as a very sudden transient. As for bit depth it is just not about dynamic range but resolving power and again to reproduce music which is not a smooth sine wave but a messy waveform loaded with harmonic variations then 24bit does a better job.
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.
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.
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!😁
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.
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).
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.
When I was a much younger guy we only had analogue cassette tape, reel to reel or vinyl records. They were all analogue until we made digital tape recorders and players. Vinyl has always stayed where it belonged. In the dustbin. DAT and Mini Disc were tried but failed as mainstream alternatives to the CD. They then tried SACD and even DVD Audio but alas, not enough takers there.
I chose 48kHz, for technical reasons related to streaming. I can run run all devices at 48kHz, but not all of them at 44.1kHz. And dealing with multiple sampling rates can be a pain in delays caused by conversion. So while 44.1KHz is enough on the audio side of things, 48KHz is good on both, audio and technical handling.
Easy 32-Bit PCM or 64-Bit float with the highest khz either 192 or 320 forgot what it was has gotta make the most difference over both of them older formats. But then if hard drive space might go into it too so maybe sampling or bit depth may have to take a plundge whn that happens so the normal 44-48 khz for CD and Studio may be that plunge to save a little space or maybe even going old with 24 or 16 bits. Plus on the khz end many ouput formats don't support past old studio 48 anyway.
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.
semoga sehat selalu abah 😊
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).
Probably the later. But to the serious stuff. I don't really know if I am correct, but Hemholtz theorized a bandwith--acoustic and non-recorded of course, of around 46khz, because from his experiments he felt that young teenagers could hear up to 23khz, or it might have been 22.5khz. So he wanted a frequency rage twice that-- 46khz-- so that all harmonics below 22.5khz would be able to interact with the information above it and produce combination harmonics in the audible band. Say 20khz combined with 25khz would produce a combination harmonic of 5 kHz, though of course at alower level than the original two tones. So you need to get to 46khz, so that 46 and 22.5--the highest audible frequency--would produce a combination or difference tone of 23.5khz, which would be out of the audible range of hearing for a young teenager with no hearing damage. That was his theory and was why he wanted a bandwidth of 46khz or so. It is also why some manufacturers in the past, especially Harman/Kardon, and I'm sure some British brands, emphasized large bandwidth amplification and the good reproduction of square waves, which is an indication of bandwidth. Essentially, they wanted flat response up to 46khz or beyond. The great Stuart Hegeman engineered some design that went up to 300khz. And the square squarewaves measure by hi-fi magazines showed that. This was in the 60s and 70s. Harman/Kardon still designs this way even though the Harman group, which included a large professional and recording division, and notably, AKG of Austria, was purchased by Samsung, the giant South Korean manufacturer, about 8 years ago. Sad but true. But getting back to the central point. I know I can't hear that high, and I'm not really sure if this matters or if Hemholtz was correct. But the issue is interesting, or intriguing, as the British would say. Maybe we do need a bandwidth and a sampling rate of 88.2 to get all those harmonics into the audible range. I really don't know. As for bit depth, "whew, that's a complicated matter". I guess it can get you lower noise and a larger dynamic range. I can't say if it is of practical use.
I use a FiiO K9 Pro (AKM version, a Dac/Amp), with 32bit 96kHz... The 32 bit becomes possible by downloading a driver. I can hotswap between 48kHz and 96kHz while listening to a piece of music using the FiiO driver, and the main difference i'm noticing is a slight difference in loudness in the treble region... It's very small, i'd say like 1 dB difference, with the 96kHz being the louder one of course... It doesn't make the music sound like it has a treble spike, it kind manifests as nuanced/dynamic contrast (with a lack of a better term), it tends to make music sound kinda... lively, aggressive? Give the music a nice "attacking" quality, which sounds great with electronic music! :) Though like i said, the difference is small, and likely wouldn't matter much for the average listener... Truth be told, if i couldn't hotswap between them, i likely never would have noticed the difference either! : /
Entertaining, and actually, very strangely, DSD (SACD) does sound quite good.
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
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 😂
If you are producing music, the delivery format defines the sampling frequency and bit depth. 24/48k is the norm these days. Multitracking with 16 bits will save on disk space at the expense of a higher noise floor. If you’re applying any non-linear processing then higher samples rate (or oversampling) will help combat any aliasing.
It is absolutely true that 16-bit has a higher noise floor. However it's likely that most or all of the faders will be lower than 0 dB so there's less noise than there would otherwise be. If you're mixing to 24-bit of course.
Don’t forget the lofi of us out there .. I’m sampling at 13bit (Ensoniq eps-m sampler) 28-39khz and it sounds fantastic
Yes this is a thing for music creation. As well as low bit-rate, I rather liked the aliasing in the Sequential Circuits Prophet 2000 and was rather disappointed my Akai S1100 didn't have it.
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.
Great topic, it seems to me it doesn't matter that much the Sample Rate while recording, but then when processing is when 96Khz and + actually shine, as mentioned in the video - yes, you'll have your CPU working more and using practically half as many digital processes as if you were working at 44.1 or 48. But that said, it is precisely those processes that can damage your audio, and not your recording. You may be adding aliasing over aliasing each time you put harmonic creating processes one on top of the other, not that you would be adding distortion, but if you are adding emulated preamps, emulated compressors, or emulated saturations of any kind... you may very well be adding unwanted aliasing. So for those cases, it is wiser to use 96Khz: 1. You are obligated to use less processing. And 2. Those processes that you use, will send potentially aliasing signals to the inaudible spectrum, which you can LPF down to hell where they belong xD. It of course depends as well on your workflow and music style.
If it doesn't matter while recording, and only makes a difference for the internal calculations of the DAW, then the solution is for the DAW to do its internal calculations at a higher sample rate, not for you to needlessly use twice as much hard drive space to capture ultrasonic frequencies.
@@AnthonyFlack Yes, that's called oversampling, many plugins have that feature, but not all of them and even when they use it, it is not a problem-less solution, you'll need to dig which plugins have the feature, meanwhile it seems that Reaper is the one DAW that has the feature for all the session or the tracks as needed, not sure. A good example are the Plugin Alliance Elysia plugins, they oversample whatever signal you have to 172.4 or 196Khz depending on your source (multiple of 44.1 or 48Khz), very good work with low latency. There are more of course but those are a good example. And nowadays space is less a concern with bigger hard drives at lower price.
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.
Usual day I track at 176.4 as in listening tests all plugin processing and the tracks themselves sounded much better. I rarely use more than 64 tracks on a session even with reverb returns but occasionally have to make do with 16 when tracking to tape anyway so used to being sparing. IMO and experience only track at multiples of 48 if you have analogue mastering planned for music distribution. Digital conversion from 48 to 44.1 is not a nice way to treat a mix. However if you target video then 48 multiples are the way forward. If you really are recording music at 48 for a bit of extra quality and targeting CD or streaming music then try 88.2 or 176.4 and see how far your cpu will get you. You can easily down sample and switch the session frequency down with all the plugins to see how they sound. You may be surprised. I have never thought to try higher sample rates at 16 bit. Personally wouldn’t advise it other than for fun. Higher bit depth does produce better sounded compressed formats such as mp3 so probably worth sticking with 24 bit or even 32 float for as long as you can down the production pipe. Analogue mastering houses will usually do a good job of 48 to 44.1 conversion though and if you can have that extra fidelity for video usage then can’t hurt.
In my experience 96k gives less sibilance, less need for de essing and corrective high end EQ, better sounding pitch shift and warping. The best place to listen is in a vocal or a snare. If I can’t hear it immediately, after processing and limiting and on multiple tracks it becomes night and day. If I need to save space with other tracks il make sure the vocal and snare at least are 96
1. It's not about how high a frequency you can hear, it's about the resolution of the recording and the smearing of frequencies. 2. Interpolation cannot recreate something that was not there ie. 44kHz to 48kHz etc. Or how a DAC works. 3. Depending on headroom and dynamic range of the recording more than 16bit can make sense, not so much on the final highly compressed. 4. Most modern masters in the audio industry have been so highly compressed that you could get away with 8 bit. But don't only take my word for it. I'm an acoustic engineer and musician.
Well said indeed !!!
Did he even answer what the title says - 24 bit or 96k hz which is more important? He just went on and on about sampling rate.
I don't think the question is what sampling rate is overtly better but how the audio is interpreted and reproduced. With a lower sampling rate information about the shape of higher frequency wave forms is lost. For instance consider a square wave at 16khz. At the simplest a square wave requires 4 points. If you sample at 32khz that leaves only 2 points to depict a square. This is a problem even at higher sampling rates (up to 64khz for our 16khz square example), due to the waveform being represented by too few points. This translates into different frequencies in the Fourier transform that make up the signal, which change the harmonics and how the tone is perceived. It's possible to argue the limitations of a speaker system and general consumer use. However, to ensure the original intentions of the sound are most accurately portraied the highest fidelity source should be used.
@@nicksterj On most systems sure, speakers will dither the waveform. But the idea isn't what your speakers can produce, but the fidelity of the signal you provide them. Work backward from 44khz, the best possible square signal is only 11khz. And for a 32khz sample rate, 8khz. Now also consider how the fidelity of any audio signal can be expressed as the difference between the sum of infinite sinusodal frequencies and what the source is capable of reproducing. In any case lower sampling rates always reduce the ability to reproduce a signal.
kzhead.info/sun/ptSOftSxfWdudZE/bejne.htmlsi=TREI8CXTpvmN0lLx
explanation or that the marketing for DSD what that you put a piece on the conveyor belt At a certain speed instead of having 16 pieces a Wheelbarrow and someone or a few fall off on the way. With DSD, the sound would also sound more analog
Yes, sampling rate only needs to be twice the frequency of the sound you wish to capture but it will not capture the sound accurately. A sine wave of 22 kHz sampled at 44 kHz will effectively be captured as a pulse wave and sound distorted. This undersampling also affects high frequencies that are audible by the human ear. This is the reason that digital 'highs' tend to sound harsh: their smooth sine waves have become distorted. So yes, higher sampling rates are better.
no this is not how it works. the dac has a reconstruction filter that turns the 22khz "square wave" back into a sine. this is day 1 signal processing stuff...
@@Bestmann3n But how does the DAC know whether the original signal was a sine, triangle, pulse or more complex waveform? IMHO the 44kHz resolution is too low to capture complex high frequency signals
@@6db It doesn't need to know about those things. Does you screen need to know if you are looking at pictures of dogs or reading a document? 44kHz is enough to capture all the frequencies a human being can hear.
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 KZhead I'd like to change it but I'm worried that if I do the algorithm will no longer like me.
The guy at PS Audio said on KZhead 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
Some arguments in favour of 96 kHz I heard over the years are as follows: 1) your hears and your brain does not consciously detect any signal above 20 kHz (until age 20 maybe, then it is 15 kHz, 12 kHz ...) but you body does and if that high-frequencies are missing, you brain is not fooled by an Hi-Fi system and knows it is a reproduction 2) ultrasonic frequencies can beat with each other and generates other frequencies into the audible spectrum which our brain expect in a real live performance. I have no idea if there is any valid science behind those affirmations, but it *might* be possible until proven otherwise (and maybe someone already did). The real con of ultrasonic frequencies is that most of hi-fi equipment (at least the vintage ones I like the most) are not designed to handle anything above 20 kHz and might introduce unwanted distortions. As for the anti-aliasing filtering problem, I think it was solved 40 years ago, first with over-sampling and then with sigma-delta (1 bit) DAC. Anyway, doubling the data rate just to ease the work of the filter is non-sense to me, oversampling does it very effectively.