The truth about Nyquist and why 192 kHz does make sense
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- Опубликовано: 11 авг 2016
- I have been attacked quite heavily for claiming that we need higher sampling rates to increase the time resolution up to that of our auditory system. Time to respond. (subtitled in English and Dutch - Nederlands ondertiteld - כתוביות עבריות).
The written English version:thehbproject.com/en/articles/5...
Dutch written version: thehbproject.com/nl/artikelen/...
Link to my book: thehbproject.com/en/news/116/f...
www.hansbeekhuyzen.nl
www.theHBproject.com/en
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This matches BBC Engineering's findings when they first introduced digital audio to the broadcast chain (I led a BBC group on digital audio testing). They were also concerned about the effects of cascading multiple A-D and D-A converters, as there was so much analogue equipment between the microphone and transmitter, eg a digital mixing desk, an analogue VT suite, analogue continuity mixer, digital transmission backbone (NICAM), analogue transmitter, etc.
We did a lot of measurements on the early DACs to test for artifacts and came to the same conclusions. Steep filters produce more ringing and have poor impulse responses, whereas a more gentle filter doesn't fully attenuate the signal until a much higher frequency, thus requiring a higher sampling frequency. If the sampling frequency is not high enough, any input frequencies above the sampling rate will reflect back down into audible spectrum causing artifacts.
Thanks Alastair for letting us know. BTW my opinion is based on the research that Stuart and Craven have published over the years. Since it coincides with my auditory observations, I felt confident to publish them in this video form. Extra conformation from you does really help.
Is that why video uses 48khz?
This is why 48khz is used for studio reporting, so that you can have a smoother low pass filter and then the final result can be saved at 44.1
You need to have equipment (including speakers) that can reproduce the the increased frequency range. Most consumer speakers cannot reproduce up to the 20k that 44.1K provides.
@@pilotavery 48 kHz and 44,1 kHz was to avoid digital copies. Eg DAT recorders used 48 kHz which prevents making digital copies of CDs.
Thank you for taking the time to request that all commenters use arguments and remain respectful, and for removing any that didn't. It seems to have resulted in an interesting and useful discussion, a rarity at RUclips.
Thanks for moving this discussion forward
Thank you, you have clarified some issues for me and answered arguments that there is no validity in sampling above 44.1 kHz.
I think you explained this really well. Very good video.
Hans, it's a double-edged sword. Yes, a less steep rolloff would be beneficial, but aliasing occurs in the analog world as well. With such a high sampling rate of 192kHz to rolloff to inaudible levels, there is a greater chance of high frequency noise to exist due to the highest possible frequency content (96kHz). The issue is that this noise is in the nonlinear range of some amplifiers, and nearly all transducers. Thus, the difference between frequencies in this range will be scattered throughout the spectrum, into the audible range. Then, all equipment must be capable of 96kHz playback, which could require items such as supertweeters to reproduce sounds you can't even hear, to prevent audible distortion.
To me there's no obvious solution, as a gentle theoretical rolloff is preferred, but high sampling frequencies causing intermodulation distortion is also undesirable. Your points are fair, but there are definite cons to 192kHz sampling, so no clear answer is available. At least we both agree that we can only hear 20kHz (or maybe slightly above when we were younger), so many people I see arguing for high sampling rates think they can hear 90kHz or something, and tend to seem quite delusional ;)
I agree with you, the higher the sampling rate the more probabilities of getting armonics at our reconstructed signal. What we could use in order to get a better quality is more bit deep in both adc's and dac's.
Jonathan, I think you might be confusing non-linear artifacts (distortion) with phase response problems. Drivers exhibit wonky phase response off axis at frequencies which are too high or low for the type of driver in use. You will hear “non-linear range” while they are referring to the phase response problems. This has nothing to do with harmonic distortion. Distortion is also said to be non-linear, which is where the confusion comes from.
Non linear artifacts happen when we clip a signal. New harmonics not originally in the signal are created. Speakers have specifications for maximum output: peak (clip level of a transient) and RMS (the output level at which constant energy clips). It is the roughly 8-10dB below the speaker’s specified clipping point that we want to avoid, in order to not create harmonics in the reproduced sound which weren’t in the recording.
Speakers and amplifiers do not automatically distort signals at all SPL levels within a certain frequency range (that is unless it is an incredibly bad speaker :). In this way there is no ‘non-linear (frequency) range’. Distortion-induced problems like intermodulation, happen near the top of the speaker or amplifier’s dynamic range.
Another aspect to all this: Music is not flat. Progressively less energy as we move up higher in frequency. This proves true when looking at music on an FFT or RTA. The fundamentals are lower; the harmonics are higher, yet the fundamentals have way more energy. The harmonics have progressively less energy the higher in frequency they go. A 96K high res recording for example, will show on an FFT that 99% of the energy in the music is in the audible band. Comparatively little above 20Khz. This makes it hard to image the ultrasonic range clipping. Everything below 20Khz would have to clip to an absurd degree in order to get the VHF up to the clipping point! You’d have turned your speakers down well before that point.
You refer to noise in the non linear range. You mean noise above 20Khz? For the noise floor to be near maximum output at any frequency, and therefor clip the driver or amplifier; there would have to be something seriously wrong with the amplifier! The kind of problems you are worried about only happen when we are saturating the signal at some gain stage (or at the driver). This does not happen with low level signals. I think some of the extra concern comes form SACD, which was a noisy technology. PCM does not exhibit the same problems in the VHF.
If we say that we shouldn’t have sound reproduction above 20khz because there could be distortion-induced problems, then we would have to say that we should not use speakers at all (because you can have distortion induced problems at any frequency). The difference is us the listeners. And wether a person can hear the difference between a 10,000hz square wave and a 10,000hz sine wave, with an analog oscillator, in a 100% analog environment, with a system that can reproduce VHF….
Never heard above 17kHz in my life, even as a teenager. No music up there anyway
I've done quite a bit of recording and playback at 192khz, and even some at 352khz, and I've never encountered said intermodulation issues. For decades we have heard theoretical concerns about intermodulation distortion, but I've never encountered anyone who actually identified it in a real world scenario (except trying to mix DSD64 on analogue). I have found that converters with higher jitter clocks do not handle 4x and 8x rates as well, but in the age of femto clocks, that is no longer an issue.
I would not doubt that there may be greater ease and audio quality for the listener with less ultrasonic content, but I just haven't heard any significant signal degradation that was greater than the benefit longer filter transition bands. And even so, if the recording was made with transformered mic preamps, or the DAC/Preamp uses transformers, there is generally a rolloff above 50khz and not much reason to even worry.
Thank you for an excellent and very informative video, I enjoy them immensely! I am sorry that you have been treated shabbily, the Internet can bring out the best and the worst in people and I applaud your standing up and demanding to be treated with respect as all humans should be treated with respect and dignity. Bravo!
As my mother-in-law used to say: if you can't' stand the heat, don't come to my kitchen. And about 95% of all comments are flattering, to say the least. Happy Holidays and thanks for your concern.
Good argument indeed. Thank you for bringing this up, and putting this swing on it.
The fact, that something sticks mathematically doesn't mean that it is technically achievable. The mathematics of the Nyquist theorem obviously doesn't take into account any implementation issues such as how to limit the bandwidth of a signal. In short, one should take into account the practical limitations of the technical implementation and then apply the Nyquist theorem. Of course, in the 80's when the Redbook standard was defined, this would prove too costly, hence corners needed to be cut. Luckily we now live in an era in which there is so much more understanding about digital signal processing and also a standard of hardware that allows us to no longer cut corners.
I wished it were. We still have to cut corners, less than in the 80's, but still... But you can cut corners wisely, dump, cheap and so on.
Thank-you for your videos. Very informative and mentally challenging on the theoretical level.
Very good approuch indeed, i have to agree with you on this issue, maybe the secret of better sound is not on sheer resolution but outside the range filtering.
I tested this, and you are absolutely right. Although I cannot confirm the theory of course
Great Video, perfectly hitting the spot. But one should consider that our hearing system (ear - nerves - brain) should be able to perfectly reintegrate even signals with distinct pre- or post-echoes, so in most cases these flaws should be inaudible, even for a trained ear, simply because our hearing system also "applies" a lot of filtering, but in psychoacoustic ways. The integrating effect of our hearing system e.g. enables tricky dynamic compression, where waves dont time-align any more, but sound won't change.
Very accurate and right to the point. Also very clear and easy to understand. Thank you very much
Doe je goed, Hans! Mooie uitleg!
Hi Hans what this this mean for music reproduction and the file properties needed? If I understand this video correctly, 192 kHz is the sampling frequency for use in the studio or wherever the recording is done to reduce artifacting by aliasing. After mastering by the audio engineer the artifacts is removed.
Is there any audible difference in buying a 44.1kHz/16bit and beyond that? Most digital stores are now beyond CD quality and have similar prices.
I am new to audio and would like to enjoy the music, but would like to understand before spending much money.
Thank You.
Thanks for great educational videos! Filtering is very expensive especially if you demand high quality sharp response. So it’s ingenious for DSD, MQA et al to use high sampling frequency to alleviate the need for high Q filtering.
Absolutely!
8 years ago I got my first ADC that could multitrack jazz trios at 192khz, this same debate was raging back then so I decided to find out for myself. Well let me tell you that my experience was that all things being equal the higher end of cymbals, snares, and female vocalists that I recorded at various rates at the time was preferable at 192khz, and overall music I recorded at that resolution was more pleasing to listen back to. However for practical reasons I can't always choose to use 192khz, but if I could, I would.
What are your thoughts regarding higher sampling rates (at least greater than 44.1) improving the imaging of the reproduced sound?
I have tried to express them in this video:-(
I think he is referring specifically to the effect on imaging that higher resolution audio may have vs. standard resolution (not necessarily MQA encoded). Improving the time domain with higher sample rates alone, may be advantageous to imaging, but how higher resolution will fix imaging (in particular) may not be obvious or part of the video above. Just saying. Again. Kudos on your awesome videos!
Thanks voor clearing this up. Trying to answer too many questions in too little time is claiming its toll, I am afraid. Indeed, the filtering artifacts do have a negative effect on imaging and placement. So reducing the artifacts does improve this. In what way and to what extend depends on where your started, what improvements were made , how these improvements were implemented and so on.
I'm a simple man, I see a video with a correct content and I upvote. also I'm an engineer who has worked with DACs/ADCs and I agree, the higher the sample frequency the better you can reproduce/replicate the signal, that is the reason I will not get an oscilloscope with sampling frequency less than 5x of the max signal I want to sample, merely to reduce aliasing.
I recall that in one of the DACs I worked we used a nominal oversampling frequency of 4.8MHz (yes MHz) for most audio, even for something as 16b@8KHz PCM, the only reason for the oversampling was to reduce artifacts in the output signal.
You always make a lot of sense to me Hans, due to your logical explanations (and your English is good too). I know it's easier said than done, but try not to let the attacks bother you too much.
They don't bother me, they come with the job. But thanks for your concern and your compliments.
Hello, i think that your explanation of the NOS tecnology was a little to short, any chance for you to go deeper?
Do you have any listening experience with later converters?
thank's
Francesco
Thank you.
I have been using digital audio editing software, and decided to use the Nyquist theorem (with Frequency Analysis) to detemine the highest frequency in my recording, then double that rate. I was trying to reduce the file size of samples so I could fit more on my hardware sampler (musical instrument). The result sometimes was aliasing, which sounded like distortion or smearing of the audible signal, so I think this is correct. Nyquist is the minimum you need, but a higher sampling rate may sound better and reduce aliasing.
I agree to a certain degree. If the really chain is almost optimal, the differences between 44.1 and higher become insignificant.
Remark to the band limiting during recording: it is already done by microphone. Almost any of the mics commercially available has lower frequency limit somewhere between 20-50 Hz and upper limit somewhere between 12-20 kHz. For both infra- and ultrasound recordings, specialty microphones must be deployed, which are almost impossible to buy for common people. A friend of mine was doing his diploma work about bioacoustics (bird voice signals, specifically) and even to find, not to buy mics able to go somewhere between 20-80 kHz was really pain in the ass.
Not talking about a need of having covered both "normal" and ultrasound ranges simultaneously.
Although I had Philips earphones able to reproduce ultrasounds up to 26 kHz I never had a mike able to go such high.
It is easily demonstrable: try to do a recording with any of your mikes at any sampling rate and calculate the spectrum at any moment. You will not see any frequencies above approx. 20 kHz with any mike.
(Actually, I can see a problem with the reconstruction of non-harmonic signals, but to find an answer how that is done, I need to do some maths and measurements with signal source - e.g. reconstructinng of square or triangle wave needs theoretically infinite Fourier series. Long story short, not all audio signals are superimposed sine waves.)
Thank you so much for this useful video. I have heard many different dacs from different manufacturers using different chips to do the job and i always preferred the NOS implementations. They somehow sound more natural to me in comparison, so i am using one in my own system.
Well, see this video: ruclips.net/video/EykkzxUstL4/видео.html
Hi, thank you for the nice explanation.
I hope I have understood (also from the other comments):
1) the recording and mastering is (hopefully) done at high freq (with low artifacts).
2) When creating the CD, the signal is bandwidth limited digitally (introducing artifacts)
3) then downsampled to redbook file.
4) The DAC at the consumer side takes the file and upsample before converting, to reduce reconstruction filter artifacts.
Upsampling in the DAC is key for a consumer...
The improvement in keeping higher sample rate end2end is related to point 2): avoid the digital bandwidth limit filter before downsampling. If only we could have perfect filters, redbook would be enough.
Did I understand correctly?
I am afraid you do really need to watch my playlist on MQA: ruclips.net/video/r_wxRGiBoJg/видео.html
Thanks so much for keeping matters logical and scientific. I'd like to understand why the artifacts mentioned from sampling exist. Is there a simple example like there is with sampling images eg aliasing or maybe others. Perhaps you have a reference that talks about filters in both time and frequency domain with illustration as can be done with image DSP. It's kind of easy to understand 4k TV vs 8k TV as it's ultimately visual but very hard to relate to 96k and above audio sampling in a physical way... Also I wonder as an example the philosophy behind the design of filters in dragonfly cobalt and high end chord dac how do they find the filters that make it so good and how much work is needed. Is the work art or science and why do I have to pay 3K up for a DAC !
I have already covered a number of items you mention. Just shop around in my playlist page: ruclips.net/channel/UCR4tuhqPppVp-PD0q17sPEAplaylists?view=1&sort=dd&shelf_id=0
Pretty well explained and argued. I would add that the reason Nyquist works _in_ _theory_ but is difficult to implement _in_ _practice_ is that the Nyquist-Shannon theorems employ a perfect frequency window filter in the _frequency_ _domain_ , whereas practical electronic implementations are limited to imperfect filters in the _time_ domain. This gap between what filtering is required theoretically in the frequency domain and what is possible practically in the time domain, can be bridged by increasing the sampling frequency to a point where the required lower-order time-domain filters become well-enough behaved.
Agreed
In order for me to understand this video I would need another video explaining the jargon galore in this one, like - Time resolution - Impulse Response - FIR Filters - Filter with a Milion Taps - Pole Filter - Jitter - Reconstruction Filter - Band Limited Signal - Filtering producing artifacts - Band limiting reconstruction filter - Aliasing - Proprietary Filtering and Ladder converter.
So I'm afraid I did not understand much what it was about.
I understand that you don't understand. But as I said in the video, you really don't need all this to enjoy music. It was merely to put an end to the 'discussion' about higher standards of music files.
As I said Plexuss, there is no need to know all this just to enjoy the music. Tone it down a bit please. Let's keep this place a nice place.
Hi, recording and mixing with Sample Rate at 44,100 kHz and bit depth at 16 loses quality? Thank you.
Recording at any sampling rate and bit depth give losses. Higher sampling rates up to 384 kHz and bit depths up to 20 bits give less losses. Using MQA also gives less losses. But it all also depends on the recording equipment. A very good analogue-digital converter at 34.1kHz might sound better than a cheap analogue-digital converter at 192 kHz.
Thanks and congratulations for the channel!
Amazing video
My question is in regard to playback versus initial sampling. Let me give an example. I record a simple session at 96/24 of an acoustic guitar and a vocal using two mic's. Once it is finished, I do a finalized stereo bounce down into an mp3 file so I can play it on a digital medium that can only play mp3 files. I also do the exact same thing again, except this time when recording the original tracks, I use 48/24 instead of 96/24, and it just doesn't sound as good as the session that was done at 96/24 then bounced down to a stereo mp3 file. Both sessions ultimately end up in a mp3 stereo file, so why does the one tracked at 96/24 sound better than the one tracked at 48/24?
MP3 is a lossy way of storing music. The higher the quality of the source material, the easier it is for the MP3 encoder to decide what part of the information hurts the least when left out. It's the same with pictures. If you want a high quality, low file size internet photo, start with a very high quality losslessly encoded photo at very high resolution.
Thanks for your response. I am currently looking into Antelope Audio products because they seem to be the one of the only pro audio component producers that focus on making their products with 192 khz capability at an affordable price.
Great video... yes, I understand and agree with your points and I too, enjoy the music ;-)
An excellent explanation, thank you
🙏🏽
I want no debate here, we all have our experiences and backgrounds... it’s been awhile since I have looked into these designs, but I seem to recall that using to aggressive a ‘brick wall filter” (digital IIR or FIR filter) when you still had 44.1 kHz sampling can lead to some phase problems after the D/A stage plus the analog reconstruction filter used afterward (which had to have steep slopes). But, usually by late 80s to mid-90s in consumer CD players electronics the combination of an over sampled FIR filter (say 48 k or 96k sampling) combined with a gently sloping analog reconstruction filter was a pretty darned good setup and fidelity! So anyway I like how this gentleman was placing more emphasis on filtering. I like this video.
I totally agree and I will add.
The Nyquist theorem works, and is true, but only to identify sinusoidal signal long enough in time, and constant in frequency and amplitude. In case we approaching the upper limit of the band , and the pulse is short we may not have enough iterations to correctly identify the signal. Example is; the occurrence of one single wave in 19KHz band recorded with a CD standard 44.1KS/s. The real audio consists at most of fading impulses, rather than long in time and continuous in amplitude sinewaves, that is why I consider the CD format to be lossy and a higher sample rate will be beneficial, 48 or 96KS/s. The reconstruction of impulses by DACs is often an approximation. In CD digital format at 5KHz band we get only 4 samples per halfwave, enough to identify the 5KHz wave, but not enough to reconstruct it with accuracy in amplitude and frequency. Increasing the sampling frequency will increase the accuracy and definition, as a proof I can give the equivalent to CD format, stored in 1 bit resolution but iterated at 2.8MHz, the DSD format.
,
Hello, thanks for the video, It was really useful, I have a question, why is important the reconstruction filter in the dac if the frequencies higher than 20 khz are not audible?
The reconstruction filter is an essential part of digitising. It prevents aliasing, misinterpretation of the digital information.
HI, I recently recorded guitar & vocal (Roland R-26, Digital Recorder omni + x/y mics) 24 Bit with 96 kHz Samplingrate instead 48 khz. Voc & git seemed to be the same, but noise and reverb are much better & more subtil in my ears - in the fast mix (Cubase 7 Elements).
Wonderful explanation! Many thanks for sharing your knowledge with us. I will order your book forthwith!
Extremely wide bandwith is useful to 'shunt' artifacts out of the 20Hz-20kHz band. This bandwith gives us 'elbow room' working with signals and results in 'cleaner' sound. Some authors posit that after this is done the 'beyond 20kHz' area can be chopped away, retaining the clean part for music reproduction. That is, a properly crafted audio PCM file can be 16/44 or 24/48 after all work has finalized and we have a final master.
Would you be so kind so as to offer us your view on this?
Some people argue that all the 'spikes' and 'brush' in the ultrasonic range may hurt the sonic result if left there for the DAC to process. It wouldn't add anything and it might detract from quality. Once the 'main tranche' is cleaned up we may be able to attain a lower noise floor where it matters by filtering away the 'excess'.
Is this argument 'sound'?
An argument for 24/96 files is the broadening of dynamic range. (The non technical but commercial problem lies in 'the loudness wars' and severely compressed masters and 'remasters')
Modern technology permits widespread use of 24/96 which I hope will become a 'standard' of sorts, at least in the streaming services.
Let's hope quality 'uncompressed' masters find a place in the market. Is LP vinyl dynamic range too much to ask for in contemporary times?
I didn't know about the 'reverse echo' from digital wave filtering! However I have noticed a great difference with sound quality while working in the 96 khz 24 bit environment of the MPC Live...
I'd like to address a problem that I myself have while doing audio signal processing, and it is that sometimes it needs to be louder in either amplitude or sampling rate for computing purposes, it's not the same to have a sinewave at 16 bit 44.1khz than it is to have a 24 bit 96khz sinewave compressed down to the dynamic range of 16 bit 44.1khz (with a compressor, not reducing the sampling rate) the harmonics obtained out of the process are impossible to replicate from the bitrate of 16 to 8 after normalizing back to 44.1... do I make sense? Think about it like those CSI scenes where they try to resample the face of a man from a very low resolution image back into the crisp and clear high res image. The responsiveness of effects is way bigger while working on a higher bitrate, and higher resolution environment. Other than that, the song is going to sound pretty much the same once it gets it's final mix and gets mastered for distribution. So for production purposes, 24 bit is greater, provides with more virtual headroom, and 96000 khz should be enough, unless the intention is to destroy the sound to extract the very deep harmonics of it, in which case, having more fabric to cut from is always better. :D
So I don't know about going all the way to the 192 khz, or the 32 bit-rate but I assume there could be a benefit in digital audio processing, if the equipment was there to produce the amplitude of such signals at all!
yes, for me as a rookie in this topic, sounds same like i am thinking.
I'm just getting into the deeper side of audio, and what you say makes absolute sense. I always figured it was analogous to higher wattage=more headroom in an amplifier, since I wasn't aware of the issue of artifacts. So, higher frequency sampling doesn't remove the problems, just raises them higher than we can hear anyway. It's like filming video: someone could be chatting on their cell phone right outside the frame in a horror movie, but it's outside the frame, so it doesn't matter!
I’d like to know if there are differences between natural instrumentation and completely electronically-produced music in relation to this? Are the benefits just as justified if the instruments are digital to begin with? Is this just for analog instruments (which I know can include the human vocals)? I want to argue for higher bits and sample rates from electronic music but some artists don’t seem to believe in it. It might be linked to the human frequency response thing. Do you still get time smearing in electronic music? Thanks.
Digital instruments vary in quality depending on their working principle. Old analog synths use analog oscillators and filters. They often use the same oscillators for the entire spectrum they cover, making them sound very dull. Although there are those that love the old analog synth's. Nowadays often samplers are used that use pieces of recorded instruments. So there is no synthesis. The quality of those instruments depend on the sampling rate and bit depth, the quality of DSP's they might use and the quality of samples the user loads. If musicians choose a given electronic instrument, they specifically want to use the sound it gives. Which might include time smearing.
love it! thanks for the education. This makes sense
Higher sample means more room to have more audio bands.
Thank you Sir Hans!
I've heard hi-res files played on some sophisticated gear and yes, I found the sound richer.
Enjoy!
It would be great if you explain why more than 16bit resolution is needed as well.
I’m not talking about noise. It’s the damage the reconstruction filter causes to the time information.
Sure if the test is set up properly. But that’s not easy
In love listening to this guy, and trying to keep up!! Smarter than me, he is..
I want to understand how time resolution works in a analogue system say a vinyl. Does a 20khz single from a vinyl and flac over pcm have the same time resolution? And if analogue is continuous without any interval then shouldn’t it be infinitely superior to digital? Also maybe u can make a video elaborating on time resolution. How does it affect the sound etc. thank you
Perhaps this helps: ruclips.net/video/nFWYM_ZQN2Y/видео.html
I like hans if you come to Australia i would love you to visit. I could listen all day.
If I ever visit down under, I'll drop you a note.
Great video! This concept makes perfect sense to me. The bain of speaker design is the Crossover. So why would it be any different with source files? I have vast experience in Car Audio, where we attenuate crossovers readily. From my experience the best sq comes from a more relaxed filter roll off.
Another thing, blending a subwoofer with mids is a real pain in the ass! Why?? Crossovers! Cheers
With crossovers, you can actually do them in the digital domain and have seperate DAC+amplifier for the different bands (bass, mid, treble, or even more of them).
Using a somewhat powerful DSP should give you access to a lot of tricks & math to deal with gain flattening over frequency, phase e.t.c.
For storage, I personally don't see any need for more than 48 kHz 16 bit-ish.
If you want to downsample with digital filtering at recording (using a good filter-algoritms) then go for it.
Same for playback, if you want to upsample and interpolate, by all means. At least you can keep the filters digital and rather complex if so desired.
For me it seems like a discussion about filter type/algoritm vs the other ones.
Firstly thankyou🌟✨🤛
Is it possible to dedicate a 12 core mac pro using Logic X as a dedicated 192 kHz sampler to then record 192khz stems from another mac logicX arrangement?
〽️🕶
It has been too long since I worked with LogicX that I can't answer your question.
The Hans Beekhuyzen Channel
FairPlay🌟✨🤛
Are there any other 192 KHZ solutions multichannel solutions like ADAT or indeed any all as I found your video attempting to find a solution as Ive never personally been fooled by the digital music industry proposed limits a back the Apple Ipods & hard drive apace would have financially had a tremendous impact on how readily this maths theorem was accepted as Gospel. And I know top end hi fidelity Analogue is incomparable to standard or even fidelity CD players like Meridian compared to Linn record player as I have owned both.
Cheers〽️🕶
Fully agree!
I know from experience that switching from 44.1 to 88.2 and 176.4 (or 48-96-192), opens up and sweetens the sound, also giving it much more room. I suspect that’s not the high frequency content (can’t hear that anyway), but the accuracy of the timing, that better reflects reflections in the room, and less filtering artifacts. All using proper gear (DPA mics, Jensen/John Hardy mic pres, UA 2192 ADDA).
So I’m interested in the timing issues, and especially in attack. Some instruments (think harpsichord, guitar), have a very sharp attack, which could occur right in between two samples. What happens then? Are you going to miss out on the attack, or will it be time shifted?
Steep filters are always a big problem. Microphones will solve the filtering issue when recording and playing back at high sample rates, but there will be plenty over 20kHz content with good mikes. And when you have to put that on a CD, during downsampling filtering is needed in the end…
I actually chose my DAW based on the quality of the downsampling, which had much to do with timing and filtering. I ended up with a certain German product, which sounded much warmer and retained more than any other product I tested the rhythmic pulse of the music. And sure conversion takes its time, it’s terribly slow.
👍🏼
@MF Nickster Thanks for the hint, I’ll study it.
why do we need to attenuate high frequency content on playback if it is already attenuated in recording/mixing?
good question. I might just make a video on this. Can't do it in a short answer., sorry.
I read up on wikipedia. It is because a frequency component in the discrete domain can be represented with a combination of multiple frequencies in the continuous domain. Low pass filtering on reconstruction increases the likelihood that the frequency components is represented correctly. Eg. 20 Hz in the discrete domain with a sampling rate of 48KHz can be represented as 20, 47980, 48020Hz, etc in the continuous domain. Oversampling, filtering and mechanical limitations are what makes 20Hz discrete really become 20Hz continuous.
Yes, but do you understand it now?
Somewhat, I don't know how the spurious frequencies spawn in the first place but I just assume it is a phenomenom in physics.
You might want to watch ruclips.net/video/z8dIzTZaRFY/видео.html
a psychologist would say its a mistake to moderate bad behaviour. The rude man will most likely help you see the answer you seek is in your question.
Is this somewhat connected to bad or low quality DACs question that I have sometimes wondered?
If one routes 44kHz CD-signal digitally to low quality DAC for example in cheap av amp. It will sound in my opinion worse. If one uses same cheap av amp as only amplifier and feeds analog signal from high quality CD player (which has better DACs) It will sound much better.
I've allways wondered why this is? Even though digital signal is the same and bit is a bit. Is this issue connected to DAC or DAC output filters?
I really can't say
Yes there can be many other factors in play. I just wondered when I saw this video. That if it would be the DAC output filters that cause the difference.
Then it would simple way to improve audio by running the digital signal at higher sampling frequency. With higher sampling the output filter is not that critical because it would be out of hearing range, right? And so we would get more out of low end systems, maybe?
This would be interesting to test. I unfortunately don't have equipment to do so.
It was very pleasing to hear this subject explained so clearly. I decided to digitize my music collection, and yes, to compress much of it for my retirement pleasure. I had a pretty good system, and I just wanted to find the rate beyond which I could not tell the difference. I ultimately settled on 192 kHz. There was nothing technical about the decision, just a desire to get files as small as possible without audible changes. Now I know what I was hearing at lower sample rates. Excellent video!
Thank you Sir! Most, just love to learn new information... Keep it coming.
I will admit one thing Hans and you are perhaps one of the best and most competent reviewers on Hi-Fi currently on RUclips. Your ability to perfectly describe complex phenomena and concepts is unparalleled. Moreover one thing I've always enjoyed about your reviews is they seem to be non-partisan, they don't favor one particular thing or point of view over anything else.
As well I seem to recall hearing that you had a health scare not long ago and I hope you're doing well.
Blush😊
Yes. Exactly.
Hello Hans, I am interested in this topic and would like to know if this is true, why NOS DACs are so much praised. By the way, if you know any foobar2000 plugin that contains a FIR filter to reconstruct transients I would like to know.
Thanks for your videos, I learn a lot from them!
AS I said in the video, NOS DAC's are another approach to the same problem and not neccesarily a better one. Some might NOS DAC's, others might like oversampling DAC's. It also depends on how good each principle is implemented. I don't know Foobar plugins but please realize that just applying a FIR filter will reconstruct transients. Anodizing filters might, but then again perhaps not every anodizing filter to the same degree. Good audio technology isn't easy and doesn't come cheap.
Thanks for the response. I will keep investigating, this is very interesting. By the way, if I remember well its been some months since you received an iPower for testing. Will you upload a review? No pressure, just curiosity.
Have a nice weekend! ;)
Did you mean apodizing?
Great videos, by the way! Some of the best I've found. Thank you!
There is another thing you are not talking about that's the microphone, I don't know a mic with that higher frequency range spec. I google it and all I found mic's from 10-140 khz response.
Microphones that have such a high frequency range must be condenser mikes and have very small diaphragms. As a result they will have extremely poor signal to noise. Apart from that there is little to no content above 20 kHz in acoustical music and if there is, the energy is very low. That energy will get lost while traveling through the air. So in a concert hall it is unlikely frequencies above 20 kHz will reach you and if they do it is unlikely your hearing will register them.
Verry interesting vid, thanks for posting this, there is 1 more thing I wanna know. The more hertz the deeper the bass or the lower the hertz the lower the bass ? It is a bit confusing for me . I am looking for woofers to replase but I want woofers who can handle verry low basses. To what I schould look ??? high or low hertz ? thanks Hans
Low hertz/ 30 Hz is lower than 40 Hz.
@@TheHansBeekhuyzenChannel ik spreek engels tegen jou maar ik denk dat jij nederlands spreekt aan je naam te zien. Bedankt voor je berichtje terug . Dus hoe lager de HZ van je basswoofer is hoe lager je je woofer bassen kan produceren , heb ik dat juist begrepen ?
@@peterderidder2655 yep
@@TheHansBeekhuyzenChannel Thanks Hans
I always hear the same brittleness from PCM digital audio no matter what sample rate. The only thing where I hear a significant difference is in albums recoded in DSD DAWs. In the studio I flip the live analog return from the console to the digital return of the DAW through the Digidesign 192 converters and there is a world of difference.
See my video coming Friday.
Technogeek cerca 1984 in the midst of a discussion about the merits of digital audio vs analog audio:
"Anything digital is superior to anything analog."
What a brilliant engineer and mental giant! (Not!)
This video makes a solid point about implementation. I've been saying similar for 35 years.
Danke!
Your written english version only shows the first chapter.
Better sound using higher sampling rates have nothing to do with hearing higher frequencies. No one will argue that 44.1khz is too low. You will get all the frequencies up to 20kHz but unfortunately the low pass filter of 44.1khz can and does cause problems. However a well designed 44.1khz low pass filter will sound better than a cheap poorly designed 96khz filter.. 192khz low pass filter will sound better than a lower one but only if it's done properly. On a cheap $100 U.S. U.S.B interface it is better to record at 48hz than say 96khz. What you say is 101% true but only if done properly.
You make a lot of sense. You are the first person I have heard to describe the real reason why high sampling rates are needed. I was very impressed. This area, although I understood it, was always a little mmmmm.....foggy. thank you for clarification.
I think I would want to put an easy way to understand Nyquist theorem. It simply says that the system needs two samples to know that there is a sine wave one on the positive side and one on the negative side. Now I could fit any continuous sample on those two samples. But given that you don't allow any of the higher frequency of those two delta samples to appear, They are pure sine waves when passed through that LPF. Lower frequency components have more samples reducing the source of error and increasing the ease of interpolating the points to a nice clean sine wave by filtering.
I have covered this in several videos, like this one: ruclips.net/video/geaoEt-9V-w/видео.html
Bedankt meneer 👋
One correction: In mathematics, "theorem" is an explicitly proven, factual statement. It is a very different thing than a scientific "theory" where there is an explicit question about the universal provability. The Nyquist-Shannon theorem has been perfectly proven mathematically; there is absolutely no question to it's complete accuracy.
There is potential validity to your statements pointing out that it is for perfectly band limited systems, and so aliasing is an issue. But that is a very different thing than implying there is ANY question about the validity of the theorem.
I do not question the Nyquist theorem.
Do digital amplifiers will solve problems with D to A conversion ???
There are hardly any digital amplifiers. I think you mean class D amplifiers, but these are usually analogue switching mode types. And no, they don't solve problems with D to A conversion.
The Hans Beekhuyzen Channel yes! I meant digital amplifier, not class D :) I know about one, very expensive. I am not an expert that is why I asked if digital amplifier would bypass any problems with digital to analog conversion. Best regards
It won't. A real digital amp is in fact a D to A converter with very high voltage and current capabilities. It does have the same problems and perhaps even more since the output level has to be so high. I have no hands-on experience, though.
The Hans Beekhuyzen Channel thank you very much. Once again: kind regards, and carry on! I enjoy watching your channel! Thank you!
The biggest issue with class D amplification is the potential for altered frequency response since the speaker load forms a RC circuit with the output stage of the amp. And the frequency response can vary with changes in impedance. And of course, all speakers vary their impedance with frequency. Amp designers have to be able to compensate for those variations to keep the relative response flat.
Danke Hans!!!
I concur with you fully on Nyquist and the validity of higher Sampling Rates and Higher Bit Resolution.
I use 192KHz 32 Bit Float on all my critical recordings and store all my music in that format.
I use software Applications like Audacity for Listening and Export to Established Standards.
I am Not an Audiophile.
I am a Nashville Recording Engineer and Pro Audio Consultant and Studio and Facilities Designer since 1969.
If you check out the etymology of the word audiophile you must conclude you are one😄
I may be a “Closet” Audiophile.
But I am not prepared to come out.
My taste in Music Produced by others tends to follow those classic projects of the 1970-2000 Period except for the Classical Works.
Modern Composers like Aaron Copeland I like very much.
I have always been focused on the end product and I have left the Art to The Artists.
Thorough knowledge of the Technology is essential to my Life’s Work.
I have received the AES Lifetime Achievement Award and I am one of The Architects of the Nashville Sound.
My works with John Hartford and his Dobrolic Plectural Society Band including Norman Blake, Vassar Clements, Tut Taylor, Randy Scruggs, Sam Bush and others are my Favorites including “Aereo-Plain” the First NewGrass Album along with “The Ballad of Calico” by Kenny Rogers and The First Edition.
My work is all Analog using Flickinger and MCI Consoles, MCI, 3M and Scully Tape Machines with Dolby A Noise Reduction and JBL and Audicon Monitors.
thank you Hans for this video
I was always impressed by the sound of hqplayer when upsampling red book material to 24/192 but never had an answer to question of my friends who would ask why upsample?
they always think upsmapling is always a no. i guess I can show them your video to know how upsampling moves artifacts in sound outside the audible range making sound more clean
I would d wait some more weeks. I will publish a video on upsampling and oversampling that might give more clarity on those subjects.
+The Hans Beekhuyzen Channel
thanks Hans, do try to include software upsampling also from popular software like hqplayer as more and more seem to be using it.
+Rikhav Shah I will not include HQplayer. They have not responded to my request to provide me with a review sample and I won't buy equipment or software just to review it.
+The Hans Beekhuyzen Channel I agree with you that you can't keep on buying software and hardware just to review it. hq player is general free for one month . basically as trial you can use whole software and after month is over you will need key to run it
The point is that I can't spend all my time on one subject. Furthermore I sometimes need some time to think about the experiences. So doing a review in one month - surely when the pressure is on to do it in one month - is impossible for me. So I think you have to find other reviews. Sorry.
Thank you for your explanations Hans. You are a very good teacher and as far as I can see one of the only few if not the only person explaining this!
When you make copy, the way you judge how good the copy is to compare with original and not the two copies done with different quality or with your ears.
Not only does the Nyquist Theorem require a band-limited signal that you discussed, it also is only for continuous tones. It can be for a multitude of tones of different frequencies and amplitudes, but they must be unchanging over a number of samples. Thus the theorem is not mathematically sound with respect to most music.
Yes, but we are talking music here. Ever heard music that is only a few samples long?
Hmm... I though CD players solved this already in the 1980s. I mean by having a couple extra bits (say 18 bits) in the D/A-converter and using these to create new interpolated values between the recorded 44KHz 16-bit samples. Hence moving the need for filter cut off as well as filter steepness up (two octaves in the case of 16->18 bits).
The bit depth works in the amplitude domain and the reconstruction filter in the frequency domain. So I'm afraid you are incorrect.
@@TheHansBeekhuyzenChannel Of course, but the time and amplitude domains can be closely connected regarding physical implementation.
The extra bits enables a few extra levels needed to define some interploated values between the recorded samples. Synthetic sample values with shorter duration and thus at a higher frequency. This moves the digital artifcacts up in frequency when fed to a D/A-ladder (with these extra bits) and so relaxes the need for a steep and phase distorting filter. That's the connection between amplitude resolution and the frequency domain here.
A guess you get the point? This principle is what they called "oversampling [filter]" in the late 80s and early 90s.
@@herrbonk3635 Smooth ringing is better than rough ringing... :) It still rings though. The problem arises at the A->D conversion level, extra information is added that cannot be corrected in the D->A process. It's not a lack of bits, but the presence of added HF ringing that is the issue.
@@realworldaudio Ringing? I thought it was the phase distortion itself that was the main problem with steep filters close to (and therefore into) the actual audio band. If you move the filter to 44.1x4 or 44.1x8 (as I described), you would not have any filter effects whatsoever on the audio part of the signal. That's simply why "oversampling" (misleading term) works.
But from your nick name, I suspect you perhaps is the type of guy that don't even want a treble control in your HiFi (a low pass filter as well). These dudes (often called "gold ears" here) somehow seem to belive recorded acoustic music itself is clean from all kinds of artifacts... or that all artifacts always sound really bad, and must go, despite the fact that we have listened to them as long as we have had musical instruments and concerts :D
Thumbs up from me!
You are the best Hans !
PalominoKid you're too kind
Happy Holidays, and enjoy the music !
To you too
I have no back ground in the science but I'm adj and the logic of higher sample is solid more and resolution is what matters not the science
The question is then why not sample at even higher rates. That will lay a burden on all the electronics. It's like carrying all your money all the time in stead of putting it on the bank. That way you can have access to your money all the time? Makes sense, doesn't it. Well apart from practicalities like where to store it, you are risking losing it or being robbed. So it makes sense to carry only the money you foresee you will need. It's the same with sampling. You should only sample the information you need.
Aren't we arguing about only a part of the problem? When we record, we don't typically record with a single mic what will eventually be the entirety of an output channel. Has there been a discussion of the benefits of oversampling in a world where multiple digital streams will be mathematically combined (mixing) to produce the output? Even if your final output is Redbook, working at higher sample rates and bit depths intuitively seems to give the algorithms better data to work with before the final mixdown and downsample.
Then again, intuition isn't always my friend when it comes to higher math.
To omit the discussion: just listen.
That's very Zen, my friend. I learned a lot from your video, thanks!
It seems like a better way to record audio is with a high-quality analogue medium rather than digital after watching some of your videos. If there was a method that didn't degrade that is, as copys would still degrade it.
analogue recorders have their losses too. For instance, due to the so called 'head bump' you never get as tight a bass as with digital. And there is more.
I agree.
I think a lot of the animosity against higher resolution audio stems from poor performing built in audio on pc motherboards and their high samplerates being used mainly for marketing since the actual analog output sucked. Once again that's what I think.
Or perhaps it's because bit depth controls nothing but noise floor and sampling rate needn't be any higher than 44100 or perhaps 48000 to capture the full audible range with ample room for low pass to take place without detectable aliasing. Hi-Res audio in the listening phase is a waste of space.
So 2L's DXD should be even better and offer less time smearing?
In theory at least.
@@TheHansBeekhuyzenChannel Thanks, Hans. Like Stephen Weiss I'm really sorry that you have been verbally abused for your opinions concerning higher sampling rates, which I wholeheartedly share. High-end digital audio isn't cut and dried like some people tend to believe. It's very complicated with so many factors contributing to its success.
The arguement seems to have merit. Using a higher sample rate ADC so that a more gradual filter can be used makes sense; however storing the audio as 192kHz doesn't. FFT, remove components above 44.1kHz, inverse FFT, and use the standard format. What's the point in storing the data that you don't want and won't reproduce. Even the DAVE claims the standard frequency response of 20Hz to 20kHz
When you downsample to 44.1 kHz you have to use a brick wall filter again and if you want to upsample again it's a reconstruction filter. The clever thing about MQA is that it uses about the same bandwidth as LPCM 44.1 kHz but does keep the higher sampling rate while throwing in compatibility for 44.1 kHz equipment.
A very good point - I watched this initially thinking "this will be a bunch of nonsense" but actually, everything you're saying I can't argue with - at least on an oscilloscope. I'm very skeptical that low pass filter artifacts are going to have an audible impact on the signal in a double-blind test, but I appreciate your video and the fact you based it on sound science. Anyone up for arranging a double-blind test? :)
Hearing the artefact differences between a bad and a very good product is rather easy. The difficulty for you is to set up such a comparison since it might be difficult to get the right equipment. In DAC's predominantly two kinds of artefacts cause clear sound degradation: jitter and the reconstruction filter/upscaling filter. So you need proper low jitter in both DAC's to compare only the filtering. About double blind testing audio, watch this: ruclips.net/video/QG6LS9VDZlg/видео.html
There's definitely a difference between the sounds in a DAW. Render and null and discover the difference lies in higher frequency content in higher sampling rates being maintained . Indisputable since its a null of the same material. I only wonder where it matters more since everyone simply starts talking about the subject without clarifying ; are you recording, mixing or listening. In mixing it's quite clear.
You might be right, but do you know what caused the difference. Might it be the reconstruction filter in your DAC's have an easier job? 2L and Channel Classics both offer free downloads on several sampling frequencies and these will not differ from what was in their DAW. When I play them on very high quality gear, the difference is so small that it might just be the result of the downsampling.
Wonderful video sir. I have watch it a few times and I believe I understand what you're getting at- that we need better/more gradual filters above the cutoff. What I don't quite understand is why the filters need to extend so far? All the way to 96khz??? If there is not even any relevant information much higher than 20khz, where are the artifacts coming into play? Put another way- if the filter starts well above the cutoff, does the filter still effect frequencies BELOW the cutoff?
Yet another question I have is.... are there not downfalls to recording/playing back information well above the auditory realm? If I remember correctly Lavry suggests playing back ultra high frequencies is more likely to cause distortion and other issues, rather than improve audio quality. I realize this refers back to the old arguments of higher samplerates... but my point is, isn't it a trade off??? Which is better- to have fewer artifacts from filtering? Or fewer artifacts from introducing ultrahigh frequencies that are not even needed?
The audibility of filters is complex but the steepness of the filter plays a role and the cut off frequency plays a roll in where in the frequency band the most audible artifacts occur. So choosing a higher cut-off frequency and a less steep filter is beneficial for the sound in the audio band.
Thank you for the well thought out response and for the videos. Well done! I'm still skeptical as to the audibility of these filters, especially well above the audio band, and whether or not the benefit/cost ratio is even close to being worth it.. But I would love hear some more descriptive words as to what the artifacts sound like and what to listen for. That said, I'm all for better audio if and whenever possible.
See it like this: the speed limit in my country is 130 km/h. A car that just does 130 km/h needs a lot of time to get at that speed. A car that can do 200 km/h not only gets to 130 km/h faster, it also is quieter, has less vibrations and so on.
First: yes, filters at higher frequencies do have problems that fold back into the audio band. That is the essence of my video. Choosing a higher filter frequency makes sure the artefacts are still above the audio band.
Second: you should not record more than necessary. For us humans the extreme highest frequency young people can hear is 20 kHz. As said, many mikes don't even get that high. So we're effectively recording thin air when recording at 192 kHz. All true. But it does reduce the anti-aliasing and reconstruction filter problems to a very low level.
Fascinating video and enlightening comments. While I truly am a fan of seemingly esoteric topics (if for no other reason than intellectual curiosity) I think there is a salient point missing here. The number of people for whom a higher sampling rate (and/or bit depth) matters is fairly small - perhaps one in a million. Whether we like it or not, the vast majority of people consume music on either their smartphones with cheap headsets, car stereos, their laptops or, now, digital assistants like Amazon Alexa. Furthermore the music is listened to on busy streets, crowded public transit, noisy offices or other equally noise-ridden areas. For these people, who I would argue make up 95% of all music listened to, the distinctions being discussed here are totally lost. That's not to say we shouldn't strive to decide on the best standards but we just need to be realistic that for most people it just doesn't really matter.
I don’t know where you got your figures but I can assure you they’re wrong. My country has 17 million inhabitants and I surely have more than 17 people in my social circles that love higher quality audio.
The Hans Beekhuyzen Channel I think you are taking me too literally. I was thinking this discussion was broader - discussing what the best recording format and storage is for digital audio. In that light, the large commercial record companies and distribution services like iTunes will make decisions based on what the masses want. Whether the number of audiophiles is one in a million, one in a thousand or one in a hundred, it is certainly less than 5% and I suspect less than 1%. At these numbers, the dominant companies are not likely to be driven to change no matter how compelling the technical reasons. That was the only point I was making.
Remember Betamax? It was by far the better video tape format but died because the general public didn’t care and preferred VHS.
Nyquist is right
You are right.
I am right.
192khz is good for some applications.
44.1khz is mostly good enough for me.
I couldn't personally hear differences between the two rates on my studio Quality headphones. But I vaguely remember in my college, very steep filters having a very awkward graph. I agree with your theory on that as well.
Thanks!
Totally agree with these principles Hans. I must confess I didn't believe in Hi-Res audio since the 90's until few years ago.
The biggest advantage of Hi-Res is that the D/A process becomes significantly less critical in the system, in other words, any decent DAC will do a great job when fed with Hi-Res stream and the bottlenecks will be somewhere else in your HIFI system. One downside is the bigger size of the file, but this is becoming less critical with time, and new Codec developments are also helping (e.g. MQA). Another downside is that the "player-making-industry", including the DAC chip makers, have a problem.....how will they differentiate their products if they all perform the same when loaded with Hi-Res?
Pau Rey No argument there. But Hi-Res done right. Not upconverted from a compressed 16/44.1 file which has happened. Or taken from an over Equalized compressed 24/96 file. Hi-res music should be a straight copy from the 32/384 (yes, they exsist) 24/192, 24/176.4, 24/96, 24/88.2 or the 24/48 files. NO COMPRESSION, EQ, OR ANY OTHER AUDIO ALTERATIONS. Expect for edits and I suppose fade outs.
Getting copies of the original master file is what we need to strive for.
I been working in a mixing/mastering studio for 16 years now. And I can tell you that there is a big gulf between what you hear in the studio and what the customer ends up with.
For example: The CD you have of say, release "X" sounds nothing like the 16/44.1 CD production file of "X." They are bit for bit the same but they sound nothing alike. And we don't know why. I have heard 16/48 masters mixed back in the early 80's and I can tell you they sound incredible. Way superior to their DCC or M.F.S.L. counterparts - Not even close!
Don't judge 16/44.1 by commercials compact disks.
In mathematics, a theorem is a statement that has been proved, or can be proved.
Philips and Sony already know of the aliasing problem when designing CD RedBook, that's why they bump up the frequency to 22050Hz, and the sampling rate to 44100KHz. They already take into account the filtering errors. The 44.1 kHz sampling frequency allows for a 2.05 kHz transition band, that is more than enough. I can accept 48KHz that will allow a bit more of headroom for the cutoff filter, but above that leave it for mastering and mixing, but will not improve the sonic performance of the final record.
I know what a theorem is. The reason for choosing 44.1 kHz was primarily to be able to use a U-Matic VRC to store the digital audio signal in a video signal using the Sony PCM-1600 encore (In the US is was even 44.056 kHz because of NTSC). The difference between 44.1 and 48 kHz is negligible. The real reason is that according to Nyquist and Shannon the the attenuation at 0.5fs should be 96 dB (for 16 bit). That means that within 22.05 - 20k = 2.05 kHz 96 dB attenuation is needed, meaning a filter that does 96 dB/prime (single white note on the piano), about 8 times steeper than 96 dB/oct. Just try to find someone that can make such a filter that sounds right!!
All this is going well above my head and until the other day i didnt even know there was such in depth discussions about this kind of thing. Despite that , from what i gather from the video and reading some of the comments, as i understand it your argument for 48kHz+ rests on the fact that by limiting the audio play back devises to 48kHz as well cause extra noise that 'echo down through the frequencies' that either add extra noise to the playback for somehow distort the frequencies of particularly the higher frequencies of the transmitted sound. To fix this you propose that we create audio play back devise that can play the full range of sounds from a song recorded at 192kHz and by doing this the 'echos' from the cut off only distort the frequencies above those that we can hear.
Is this at all correct as i really only have basic knowledge of this kind of thing. I have also put all relating to the Echos in quote marks as this is really the thing that i dont understand at all, how such a cut off can alter the frequencies of lower harmonics. Thanks for your help
Well, I think you got the essence, although 'echos' is a somewhat misleading term for they are not discretely audible signals. Using 192 kHz makes the filtering needed in digital audio easier to design well sounding (in a nutshell).
But HOW does this cut off affect the lower frequency's?
That's difficult to explain in a few words. Perhaps I make a video later this year.
ok thanks
Actually, next week there will be a video on time smearing and that's the effect we're talking about.
Goedenavond,
I must agree ... bit I am of the opinion that you cannot accurately describe a 20khz wave form with a 44.1Khz sample rate (you can perfectly describe a pure sine wave at 22,050 hz with a 44.1Khz sample rate ... so long as you know its a pure sine wave, but unless your sample rate is always exactly twice the frequency, you are going to loose information .... maybe variable sample rates are the answer, not sure how you'd poll that one off) So I also believe sample rates need to go up an order of magnitude to genuinely reproduce the sound.
Sample resolution (16 bit) I'm OK with, I don't feel there is much need to increase the bit depth of the signal after it has been mixed and processed at the studio and made available for public consumption, but I do think the sample rate needs to increase.
Anyways, the question I wanted to ask you .....
it is kind of old technology, and forgive me if you have explained it in another video (I'm sure if this is the case another follower of your channel will respond to this post with a link) ... 1 bit dac ....way back in the dark old days (when cd players were mostly 14 bit pretending to be 16 bit) there emerged '1-bit dac' players ... I have a Mission/Cyrus CD Player from the dark old days that boasts '1 bit dac' .... now I've been programming computers, and have been doing so since early 80's, and I know you cannot simply break 16 bits into 16 single bits and process each bit without consideration of the power of that bit. and if 1-bit dacs are truly that, then they are only capable of either outputting +v or 0v, and nothing in between.
So .... can you explain (demystify) what is 1-bit DAC, and how does (how did) it work, because I've never figured it out.
dank je wel
Best wishes
Duncan
You can perfectly describe a 20 kHz waveform using a 44.1 kHz sampling frequency as anyone with a scope can show you. You can not describe perfectly a 20 kHz square wave at the same sampling frequency since that is built from all sine waves from DC to eternal frequency. Since the signal is filtered at 20 kHz, this is impossible. As it is impossible to record that square wave on an analogue tape recorder since that is band limited by design too. 1 Bit audio converters are comparable to what is used in DSD technology. Watch DSD Explained at ruclips.net/video/hXFIq11JAas/видео.html
A very interesting video (and part 2), thank you
Brilliant. Clear and succinct. Excellently explained. It's a shame the level of investment in high end vinyl replay is so high, but I have yet to hear a digital system which can compete. Many thanks for a truly informative video, keep them coming.
Well, I can assure you digital can sound as good or better than analogue.
Thanks for this explanation! I appreciate the description of band-limiting and the problems with proper filtering. I can add this to my own teachings of how Nyquist works.
You're very welcome!
I am used to hear a lot of non-scientific praises of high sampling rates in consumer audio, but this video actually touched on a subject I never heard addressed, and it is a real thing. Thank you! I am not sure if it's a big problem though. With 48 kHz sampling rate (this is what I sue in my home studio) we have quite a lot of leeway to use less steep anti-aliasing filters. I guess that even steep elliptic filters could be doing hardly any harm to the signal in the audible range at that point. Also: there are other ways to filter digital signals, for example Fast Fourier Transforms (but that probably introduces pre-ringing as you mentioned, same with linear phase filters).
Dig into it and listen. Really!
Thanks Edo Amin for the Hebrew subtitles!