How does upsampling increase information?

How can you get more information by simply upsampling? Paul explains how digital audio works, what sample rate and bit depth means, and how computers count in binary in this simple short video. Have a question you want to ask paul? www.psaudio.com...
I am getting close to publishing my memoir! It's called 99% True and it is chock full of adventures, debauchery, struggles, heartwarming stories, triumphs and failures, great belly laughs, and a peek inside the high-end audio industry you've never known before.
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[@GodfreyMann]

Paul is the Richard Feynman of audiophile world due to his ability to effortlessly explain complex electrical engineering concepts in simple (yet accurate) terms that laypeople can understand. This video exemplifies this ability.

[@richardpells5974]

What a great explanation....I understood it, at least till I forgot it a couple of minutes later. But I can watch it again and I'm really grateful for your off the cuff explanation. Thanks Paul

[@itamarkas]

Thanks for sharing! Keep doing these great videos with explanations for newbies, even if it is 95% correct it is a free lesson!

[@mag-wp6yt]

'Don't get all pissey on me" Lol, nearly fell off my chair! Love it when Paul talks straight!

[@Tom_Losh]

*TWITCH* *TWITCH* *grimace* Sorry, I'm a retired digital engineer and several times I wanted to correct you (as you know, engineers are like that), BUT OVERALL you gave a very cogent, understandable, simple explanation of a very complex subject, just as you intended. Bravo! Well done!

[@geographicaloddity2]

EE/Control Systems Engineer myself and yeah, it's not perfect or exhaustive, but I seem to remember a similar print article from Julian Hirst being my introduction to DSP when I was in high school.

[@triple_x_r_tard]

i'm an electrical engineer. i watch these videos just because they make me angry.

[@ColinFox]

16 bits = maximum positive integer value of 65535. Since you're doing a sine wave, you generally want positive and negative, so a signed 16 bits yields a range of (-32768 to 32767). That describes the amount of accuracy you have in the amplitude (height) of the wave. The sample rate of 44.1 kHz means each sample represents .0000226s. To determine the maximum value of a certain number of bits, the formula is: 2ⁿ-1. For 16 bits, that's 65535. For 24 bits, it's 16 Million. For 32 bits it's 4.2 billion. The only advantage I see of upsampling is synthetically aiding the higher frequency harmonics in potentially getting a more accurate representation. A 20 kHz wave only gets 2.205 samples per cycle, which is terrible, though that's at the upper limit of human hearing, so it doesn't really matter. In the range of music, we get a lot more samples. The top key on a standard 88 key piano keyboard has a frequency of 4186.009 Hz. A 44.1 kHz sampling frequency will capture that top note with 10 samples per wave. You can get a pretty decent sine curve with 10 samples.

[@lucalone]

Paul, I can clearly see that you are a salesman for audio gear^ man, to keep it simple and true: the bit depth just gives you the possible dynamic range, which is 96 db with 16 bit and 144 db with 24 bit. the sample rate gives you the frequency, which is ca. 22 kHz with 44.1 kHz and 48 kHz with 96 kHz.

[@RussCottier]

Yep. This video is not ideal is it. If anyone wants to lean about digital audio properly they should probably first go and watch “digital audio show and tell with Monty Montgomery”. It’s available here on KZbin and explains how digital audio works

[@marianneoelund2940]

Needs Part 2: The Reconstruction Filter. The main reason for up-sampling a 44.1KHz-sampled signal, is to perform most of the playback reconstruction filtering digitally, and greatly simplify the analog filter which follows the DAC. This is economically important, because every playback device needs one, while the anti-aliasing filter used in front of the ADC for recording only needs to exist in the recording studio, so its cost is much less significant to the audio community.

[@Reticuli]

Yes. Nyquist Shannon theorem assumes ideal, impossible engineering, especially in the integration and anti-aliasing filter. Oversampling, interpolation, and especially using higher rates in the first place, allows more tolerance in variations in the hardware implementation. Totally separate of the aliasing issues, it's also useful not to have much extreme ultrasonic content on even hi res recordings, though, because analog gear performs worse downstream with it.

[@michaeltamburello]

Paul, you might someday want to explain how interpolation plays a function in error correction such as when data on a CD becomes missing where there is a physical scratch on the disc.

[@DandyAudiophile]

Paul, you made understand so much in few minutes. I love your videos. Thank you so much!

[@OGmolton1]

This is a great video. Thanks. I would love to see this kind of breakdown about PCM to DSD or about exactly how DSD noise shaping works with only 1-bit samples

[@LordAus123]

OGmolton1 DSD, instead of quantizing the amount of voltage in a signal at a giver sample rate, quantizes whether the voltage is rising or falling at any point in time, at a given sample rate. It turns out that if you sample the signal 2.8 million or 5.6 million times per second (as opposed to 44,100 or 385,000 times per second for mqa pcm) the signal can faithfully be reproduced with just that one bit of info: up or down, 1 or 0. My understanding is that DSD requires highly precise internal clocks because of the sample rate, but the 1bit info stream is less error prone in processing.

[@RadioHamGuy]

That sure made a lot of sense to me Paul, great job of explaining that, you made that much easier for me to understand.

[@connorduke4619]

Does all of this mean that the major sonic benefit of upsampling is on the higher frequency part of the sound spectrum, which is presumably most impacted by the low pass digital filter?

[@DanBoulet]

Nice explanation, Paul, but please stop drawing stair steps - it perpetuates the myth that the output from DACs are stair-stepped, which is not at all the case. It’s more fair to draw the samples as discrete unconnected points.

[@marianneoelund2940]

@Dan B Yes, it would be a much better representation for most of his discussion, if he used discrete points - because that's how the information exists in the digital domain. It's also much easier to draw. However, where on earth did you get the idea that the analog DAC output isn't stair-stepped? What do you imagine that it's doing between the time points where it's set? It's an analog signal, and exists continuously, so it has to have a value. The only deviation from stair-step, is the buffer amp's slew rate and settling response.

[@DanBoulet]

This video has a good explanation: kzbin.info/www/bejne/mXq0anyOiLqtq68

[@marianneoelund2940]

@Dan B Monty did a very nice job on that discussion. However, he completely neglected to mention one important point: Nowhere did he show the actual DAC output. Instead, he only showed the output downstream of the reconstruction filter, which is built into his A/D/A converter box. If he added an internal test point to the converter box at the actual DAC output, and connected it to his 'scope, you would have seen a stair-step - just like the one displayed by the app running on his ThinkPad.

[@DanBoulet]

I see what you’re saying, thanks for pointing that out, Marianne. I was only concerning myself with the signal which is sent out to the amplifier/preamplifier, which after filtering is not stair-stepped. In that context, I consider the filter to be part of the DAC.

[@crashtech66]

I think in this context, considering the filter at the end of the DAC would totally obfuscate the lesson that's being imparted here. getting those "stair steps" as small as possible can present the filter with a more accurate representation of the original sine wave even before smoothing takes place.

[@changedahanddlessss]

dont beat yourself up for getting old sir, we highly enjoy these videos.

[@julioestebanperezescudero6246]

My life is richer after hearing your. Thanks🙏

[@scottgordon1721]

You get an A for effort Paul!

[@user-tk7kz1fl2r]

M Scaler by Chord definitely makes the sound better. I understand that it's an upsampler? So it does increase resolution. Im not experienced in these gadgets, but for sure it improves the already excellent Dave and TT2.

[@moshet842]

If you are confused about the order of the bits where he flipped the place values, it is because he is doing Least Significant Bit first (LSB) which is how digital audio is usually represented.

[@Virdevir]

Thank you !

[@shrodingersman]

I’m waiting for dedicated 4K audio

[@stereopolice]

You are badass! My highest compellent.

[@topa1798]

thanks Paul

[@manudeteruel]

Thanks, I finally understand it...

[@saaie]

I am so glad i quid all that shit, i went back to the ladder dac from Metrum and i don’t feel i am missing out on detail or dynamic☺️

[@georgebedorf7950]

Sort of explains why vinyl is still around. Smooth analog sine waves but mechanical limitations in theoretical frequency response limits and dynamic range. However the cost of a really good turntable, cartridge and preamp can be out of reach for most people vs. a decent CD player.

[@musicman8270]

George Bedorf Hey, a lot of turntables are over engineered, I bought a ATLP120 on sale for 230 dollars,because I like direct drive. In spite of what they tell you spinning a platter at 33 1/2 RPM is not that hard, major differences in TT's is the cartridges

[@rabarebra]

Exactly. This is why vinyl records sounds so much better.

[@melissamybubbles6139]

Thanks. Basic explanations help beginners like me.

[@rabokel]

Never thought about it, but does 44khz mean a full wavelength of a 22k sine wave is approximated by two samples? Sounds pretty drastic. Makes me wonder why it sounds anything like the original wave

[@johanvanderpulst5250]

You are correct.

[@justthebeginning1448]

I learned so much thanks 👍🏾

[@coffeexfx801]

Was all of that explained in one take? Cause that is impressive.

[@ZERO-CAPACITANCE]

That was an excellent explanation, thank you!!

[@chrisrussell5498]

Learnt a lot there Paul. Thanks 😊🙏

[@pauldemara7633]

A fun watch. The good news is with the right playback solution Redbook CD's or equiv. digital 44.1/16 files can sound really good. IMHO, it's all about the DAC and how it recreates the analogue signal.

[@krishnanarayan5612]

You could also compare bits as pixels in a camera ?

[@rabarebra]

And he could cut out 12 minutes of the video

[@steveg219]

Good intention and overall a solid channel, but, I think the point of the Nyquist theorem is that sampling at 2x the highest frequency does in fact produce the equivalent output ( not stair-stepped). The issue is more around the accuracy of the electronics and the impact of filters which he does cover and that is accurate and helpful in understanding oversampling on playback

[@timharig]

Not quite. Specifically for the Nyquist-Shannon theorem to work, the channel must be bandwidth limited . This is where the low pass filter comes in. Creating the stair step signal injects very high frequencies on the signal in order to make the straight lines and sharp corners of the stair steps. It is only when all of those high frequencies are removed that the smoothing takes place. Without the high frequencies to generate the sharp edges of the quantization noise, the signal is smoothed back into its original rounded shape. What the Nyquist-Shannon theory really does, is determine the cutoff frequency between the frequencies needed to accurately reproduce the signal and the frequencies that only generate quantization noise.

[@steveg219]

@@timharig that is why I called out the accuracy of the filters. There is no stair step in the final output, that is the whole point of the Nyquist theorem- if you sample at twice the rate of the Top frequency of a band limited signal, the output IS equivalent to the input, i.e. it is not stair stepped

[@timharig]

@@steveg219 I'm seeing a lot of confusion in the comments from people who are failing to understand the purpose of the filter. Paul didn't properly explain the need for the lowpass filter. It is important for this explanation for them to understand that Nyquist-Shannan only produces an "equivalent output" AFTER the quantized "stair step" signal from the DAC resistor ladder (or pwm or whatever) has gone through the lowpass filter to return it to a smooth signal. Until it goes through that lowpass filter it is still a "stair step" PCM or a square wave width pulsed signal with sharp corners. Paul's entire point is about what happens BEFORE the "final" output. If they assume that the signal already came out as a smoothed then they see no benefit of oversampling because they are only seeing the finished signal after the lowpass filtering has already been accomplished. Without understanding that the quantization noise ever existed in the first place, how can they understand the benefits of moving that noise up the spectrum by increasing the sampling frequency? They assume that any filtering is done on the signal itself rather than on the artifacts created by the DAC modulation. Finally, the Nyquist-Shannon theorem is not simply about sampling at double frequency. It is much more general than that. It determines the lowest frequency of the quantization noise. Sampling at double frequency is simply the special case at which the sampling frequency can no longer be reduced before the quantization noise occurs in the frequencies of the signal itself. Paul's whole point is that by increasing the sampling rate above the signal rate, you move the lowest noise frequency up the spectrum. This creates a gap between the highest signal frequency and the lowest frequency of the quantization noise. If you increase the sample rate enough then you can you create a gap large enough for a the lowpass filter to have a long runoff before it has to attenuate the high frequency quantization noise. You cannot fully understand that if you only think Nyquist-Shannon is as simple as doubling the sampling rate. The fact that the sampling frequency must be double the signal frequency is a consequence of the theorem. Not the theorem itself.

[@steveg219]

@@timharig good additional information, I think we agree here that his point was good but was lacking some relevant info. Thanks for the additional clarification. Btw, I do all my recordings at 96k!

[@hnipen]

Nice video as always Paul! You have two kinds of people, those who can count binary and those who don’t :-) Maybe it’s more intuitive to show it as 1 and 0 instead of the stair steps

[@rabarebra]

Exactly, Paul doesn't know how to properly set up the bits on that board, the most significant bits to the least significant bits. Also, he doesn't understand nibble, byte and word terms -> 4-bits, 8-bits and 16-bits.

[@ChedCuaresma]

Great explanation. Worth the tangent :-)

[@JPBlades]

Just watched the video - I like the tank of water analogy in the comments below - its like a square ledge in a tank of water - a wave will reflect off the sharp edge - creating ring harmonics. I presume this is referring to the wave-fronts leaving the speakers (chopped at 22khz) acting in this way. A cd (44.1khz) can capture a signal upto a frequency of 22khz so if you sample at say 176khz you can see a signal at 88khz signal. I assume by interpolating the signal between the existing signal points we can sample at 176khz and construct a reading at 88khz. Which allows the filter to be softened. So we are generating a wave at 88khz not 44.1khz and the chop off distortions are at 88khz not 22khz so they have much less effect

[@luomoalto]

I think the math around 9:50 is wrong? With 4 bits you would have 16 steps, not 4

[@davidsuzukiispolpot]

Rick Yarussi Yes, and I also cringed when he thought 16 bits would be around a million levels instead of 65536, but his preface meant that it was not for us. Also, when he counted the levels per numbers of bits he went 2,4,6,8 instead off 2,4,16,32. I am glad he was trying to make a simple explanation.

[@allenholdway8683]

Wow great job I really understand your explanation,learned a lot today. That is cool. Thanks

[@kwstasg]

Excellent sir so well said

[@petersagi275]

Hi Paul! I always wondered what is about with the lower part of the frequency response of an audio CD. Is the lower limit 1HZ or is there some filter also in the low end, say between 1 and 20HZ?

[@andrasnelhiebel6726]

Péter Sági No. The Nyquist theorem puts only an upper bound. EVERYTHING below the half Nyquist frequency can be reproduced correctly i.e. 1Hz as well. There the limit is actually the physical capability of your reprodiction system (speaker, headphone).

[@petersagi275]

Thanks for your response Andras, but it's not 100% correct. According to the Nyquist theorem you could put frequencies up to 22KHZ on a CD. The reason it only goes up to 20KHZ is that there's a 2KHZ region to put gradual rolloff filters into - as Paul describes in the video. My question was that is there a need for such a filter in the low end region.

[@n.shiina8798]

I don't think you need a high pass filter. it will works fine at low frequency since the sample rate only limits the highest frequency possible.

[@timharig]

@@petersagi275 No. DAC's do not need low pass filters. I think that you are confusing the purpose of the low pass filter in this instance. This might be because of the simplified explanation or because you are associating with the purposes used for low pass filters in other places -- such as in a speaker crossover. In other cases, you are trying to remove unwanted parts of a signal. In this case, you are effectively doing something more akin to a mathematical function. You are removing something that was produced by the DAC and never part of the signal itself. If you look at the stair step wave at 12:00, you will notice that the stair step structure produces sharp angles and straight lines. These sharp angles and straight lines mathematically require very high frequencies to produce, so the part of the DAC that produces the stepped voltages is effectively inserting a bunch of very high frequencies onto the signal that were never actually present in the digital stream. This can be referred to a quantization noise. The low pass filter acts as a mathematical operation to convert that stair step pattern with all of the sampling noise back into the original smooth waveform that was recorded. It does this by removing all of those frequencies that were injected by the DAC. Without those frequencies, it is impossible to recreate those lines and sharp angles and so the waveform is smoothed back into its original rounded form. So that is why you absolutely need a low pass filter out of the DAC.Without it, you would end up with the quantization noise in the output. But, you would not want to pass the output through a high pass filter. The quantization noise is all above the Nyquist-Shannon sampling frequency. If you were to use a high pass filter, you would be removing parts of the original signal that were part of the recording.

[@shrodingersman]

A smooth curve has a lot of information, the smoother you make the curve through interpolation the closer it approaches the analogue in terms of presentation, not information. I would use this analogy, it is an improvement, in that it becomes more bio available to the analogue brain system.

[@starbase218]

There’s a lot more to tell about this subject in general, but suffice it to say the 44.1kHz is all you need to capture every bit of information in the audible frequency spectrum. Nyquist/Shannon tells us that a band-limited signal only has to be sampled at twice the upper frequency of the band. The stair-steps are not present in the analog output of a digital source because these stair-steps are made of frequencies beyond the band limit, and are filtered out. The only thing I kind of missed in this video is the fact that 44.1 is enough, as the question implied it wasn’t because “you only have so many samples”.

[@bobnixon4015]

Really good one Paul.

[@Snipersounds]

YEAH! I really enjoyed that video too!

[@rabarebra]

😂 Really?

[@MarcelOoms]

Are you talking about /advocating upsampling at the source (e.g. in software) or in the DAC? Most DACs do that already.

[@mareknygus329]

I hate upsampling, there are so many algorithms and almost all of them make sound upleasant, maybe cleaner, but there is always something wrong. Worst of all is dual upsampling, like 44.1 to 96khz on PC with unknown algorithm and then 96khz to 192khz in DAC... I think this is reason so many people hate computer as device to play music... sometimes there is also digital volume control on PC which can degrade sound even more... I also have NOS DAC, sound nice, but there much less detail in highs... New 96khz recordings sound best for me on modern DAC-s (like ESS DAC-s) without upsampling, but 99% of my music is old... so what to do?

[@FliskerX]

Holy s.. this is amazing. Thank you so much for all the videos Paul.

[@simonsmith1685]

Paul, are you talking here about increasing the bit rate of the ADC i.e the recording apparatus or the playback apparatus?

[@simonsmith1685]

Thanks, makes sense

[@glenncurry3041]

Very difficult subject. Lots to get across even in basics. But I think you missed the context of the question. I think the questioner was asking about playback, not record. While an understanding of the record process and problems would have a correlation to those complimentary problems on the playback side. The needs and process are reversed. We would start with a CD that already has the audio stored digitally on it with 44.1K/16 bit. If the CD player did a 44.1k sample, it would get one sample of the data for that word every 44.1 thousands of a second. Output filtering would have to be very sharp to remove anything above 20K. If you double the clock, double the number of pulses from the laser to the CD, you can sample that physical data twice. Thus if there were errors in the first sample, they error rate could be reduced with the second sample. 4 times over sample, 4 chances to get the sample correct. Or at least average the 4 samples to produce one output voltage more accurately. Now, with a 4 times sampling rate, if your DAC (digital to Analog converter) in the CD player will reproduce a 196.4K data rate/ frequency, then the OUTPUT filtering does not have to be as dramatic for the eventual 22K analog output. Technology advanced allowing lower cost circuitry with higher sampling rates and bit depth. But the delivery method that is CD was locked in. So they designed around the delivery method.

[@seagoat651]

Thank you Paul.

[@thegrimyeaper]

I actually understood that. Whoa.

[@Reticuli]

I've never heard down sampling decimation make something better unless it's with content with a bunch of ultrasonic garbage you're trying to get rid of, but I have heard good oversampling DACs and especially interpolative upsampling, both from Burmester Audiosysteme and on the Denon DJ DN-X1700 inputting 44.1khz and letting the mixer run at 96khz. The 44.1khz strait mode was closer to what other DACs produced with 44.1khz, but I found the interpolation to be more pleasing and seductive sounding. I've never heard any of the plug-ins for Winamp or Foobar for upsampling that did that type of pleasing interpolation, though. Maybe someone knows a good one? I realize you have to use a USB DAC that's capable of exploiting it. I also am a big fan of HDCD, though that's apparently a little different.

[@jonnyb2532]

Thanks for this. I would argue that the interpolation does *create* more information but this is neither here nor there. I now understand why upsampling could be good. Thanks for this.

[@380stroker]

It's artificial information, not real information.

[@rabarebra]

@@380stroker artificial cold (read wrong) information. Correct!

[@jonnyreverb]

This is very good. Thank you.

[@kurtlane6059]

Fascinating!

[@lwdp74]

Guess you lose a lot of phase shift with a lower order filter. I listened to your talk about regulating power supplies. Although the output stage doesn’t use voltage regulation it can provide a huge buffer against output power rippling the supply voltage. I believe using a voltage regulated power supply could trigger harmonics within the regulation, introducing some low level noise. A lot of film bypassed capacitance and a high current transformer may allow the output voltage to wander a bit with ac variations but should create pretty smooth and stable road for the signal to ride on, not introducing any power supply artifacts. A cathode follower tubed power supply is awesome but the price does add up.

[@silverado884]

I have a cary cdp1cd player that has a choice of upsampling rates you can choose. Is there an ideal upsampling rate?

[@rabarebra]

My advice. Do not use any. It just created an artificial sound.

[@rgandblue]

Why they didnt realize this brickwall problem when designing the format?

[@BetoSurvivorMax]

And then there's delta sigma and multi bit dacs to complicate this even more. Maybe you can do a video on that?

[@billtait6457]

can not thank you enough for this Paul

[@johneygd]

There’re so many factors to got good audio quality, it’s not only bit depth & sample rate,, but it’s also the amount of khz, the amount of kbps and the s/n to noise ratio, it drives me insane, one 16bit audio sample is not the othef 16bit audio sample,,, for instance an 16bit audio sample at 11khz at 64kbps doesn’t sound better then an 8bit sample at 22khz at 128kbps ,as a result we consumers can get easily fooled & confused by those audio equipment company’s, aaarrrggg.

[@johneygd]

Indeed, you cannot got more info out of 16bit but you can use linear prediction to add more samples between existing samples , being based on the next and previous samples to get a smoother audio, 100hz smooth motion works similar , if in the 1st image the hand was rased down while in the next frame the hand is all the way rased on top, then the system can presume that hand was in the middle first and so on,resulting in smoother motion, in fact you could multiply new samples & pictures as much as you want based on predictions by creating new predictions based on previous predictions and so on. So eventrough the information wasn’t there but by using prediction, we can asume what’s supposed to be there, albeit not 100% but still, it’s incredible what you can do nowaday’s with such advanced AI technology👍

[@rabarebra]

Prediction is not the way to go to reproduce the truth in sound. This will always sound artificial. This is why I stay away from upsampling.

[@changedahanddlessss]

yeah its some what difficult to remember that number especially if you dont use it everyday.

[@ArnoldVroomans]

In short: every digital signal is a snapshot were something has to be prophetically predicted on the assumption that what happens next is coherent with what happened before. More samples makes it possible to smooth things out even if there is not any actual data.... why does this look a lot like what an mp3 codec does....

[@Beyondabsence]

Was attacked on a facebook audio group today for saying upsampling is wonderful and I hear it.

[@FazerOnStunn]

Without getting way too complicated and fast Fouriertransform explaining here .... let me try to state it simply as I know how to: If you apply a digital FIR filter a finite impulse response filter in the process of interpolating data between the samples you were effectively controlling the main lobe shaped and sidelobe shapes or stopband of the frequency response. That is you are controlling the shape of the main passband response how rounded is how quickly it falls off after 22,050 Hz and how low the sidelobes or stop band a bit higher frequencies between let’s say 23,000 to 44,000 Hz what those are. If you make the assumption that there are no more higher frequency signals than 22,050 Hz that you are interested in because the human ear cannot Hear them .... then you have lost no information and you have not gained any information by up sampling - that is what you have done is really controlled some of the unintended distortion and aliasing effects that can happen if you were not perfectly filtering (in your output or reconstruction filter after the D-to-A stage) right after the 22.05 kHz (44.1 k sample rate) limit. Or You have made it easier on yourself to design the output with filter if the sample rate was double that, that is you were not having unintended noisy noticeable effects between 22,051 Herz to 44,100 Hz (in the case of 88.2 kHz sample rate. ) or you may find it’s easier to design a linear phase filter in case where the sample rate has been moved up higher. If I could try to boil this down as simply as possible : digital FIR filtering and creating interpolated samples can just give you only a more faithful representation of that original smooth or jaggy time-varying signal but cannot create any more information of the band-limited single (say 20 Hz and up to 22,050 Hz) you started with in original signal (there is always inherently a band limited filtering or an actual filter before the sample and hold A-to-D stage). If people want to get into debates about how you CAN hear the harmonics of percussion and brass and other instruments higher than 20,000 Hz in the human ear that is totally another discussion, folks.

[@FazerOnStunn]

yeah, I know: I am known to be verbose. i get passionate about engineering, dsp, audio and similar principals. I just hate it when people claim that there are more higher frequencies that the human being and its ear are supposed to “hear” which is simply just not true :)

[@garymiles484]

Paul, a 4 bit word is a nibble, an 8 bit word is a byte.

[@rabarebra]

and nor is 4 bit word a word, nor is an 8 bit word a word you got the other things right, though

[@cp070476]

Hell it's easier just to stick the CD in the player and press play! ha.

[@its1110]

DEVO Hat Stairway to Heaven

[@NipperDog]

It's no fun being stuck with an analog mind in a digital world. 🤔

[@kingtrance6826]

Yup. I think Joe Walsh agrees!

[@funkymonkey1198]

Lessons on binary digital are either good or bad. There is no in between.

[@KoreytheFunkyRayda]

Thank you, Paul. I'm one of those people who don't get it all.

[@bedrosdaoudian8927]

A hard day Paul! ... Good effort... But I think that there is one more thing about oversampling that can help in an issue you and an engineer at PS Audio discussed in another video... It is much easier to get good sound at 192 KHz than at 44.1 KHz... Because a dac is not entirely accurate while reading so if you give it more samples e.g. at 192 KHz it is more likely that the DAC will balance the errors it's making with the more samples... and give an overall better and truer sound to the original recording... If upsampling is avoided then a femto clock is required to ensure that each sample is accurately portrayed and an error by the DAC is completely avoided ...so upsampling... Or a femto clock or both! ... Thanks A lot Paul... Your effort is just great! Thanks again

[@noob9103]

what the hell is he talking after 14:00? I am amazed how people understood that.

[@FirstLast-ih6ec]

Paul, regarding the 'we can't make up information', this is not the case any more with machine learning (AI) algorithms. They have built in 'knowledge' of the music domain and can make up a reasonable missing information. To learn more search for 'Audio Super Resolution with Neural Networks'.

[@rabarebra]

This will always sound cold and artificial.

[@TheCpuCrew]

Great!

[@its1110]

== Upsampling lets you use a filter with lower resonance -- less expensive, sounds better

[@nickthequick]

If you had prepared better, created a structure beforehand, this would have been so much better

[@pspicer777]

The word is resolution.

[@worldsyoursent.1635]

💪

[@changedahanddlessss]

65535 also pertains to tcp/ip ipv4 which is also 16bit addressing... haha dam computers.. once again telecommunications and audio are aligned.. source en.wikipedia.org/wiki/Port_(computer_networking)

[@alecgrolimond1678]

Upsampling works likes when you over scan a document it uses a calculation to average between the missing information.

[@hxhdfjifzirstc894]

This is a false question because upsampling can't increase 'information', only noise. Anything that's not part of the original signal is noise, regardless of what it sounds like.

[@rikirex2162]

benevolent critic...dont open canes of worms when improvising a video...you should take it down, think it over ..find the best way to put across your lesson and do it again,, because is a very interesting and important subject but in the way you presented it ..very painful..cheers.

[@shaunhw]

Your bits are back to front lol. Least significant digit should be on the right, just like a decimal number. EG in decimal we have THTU columns etc. - ( IE thousands, hundreds, tens and units ) similarly in binary we have ( for five bits) 32's 16's 8's 4s 2s and finally 1s columns. The left most bit can be a sign indicator giving positive or negative polarity if you want or need to interpret as being so. So a left hand bit set would mean a negative number if you need that (a signed number). For unsigned 16 bit, the range of values is actually 0 to 65535. All 16 bits being zero to be all sixteen bits set. Music would need signed numbers, so (in standard twos compliment) you could have minus 32768 to plus 32767, or using the MSB (most significant bit) purely for sign then +32767 to -32767 So each half of the sine wave can have different 32767 levels for 16 bit audio. Each extra bit added doubles the range of numbers possible.

[@rabarebra]

Paul also mentions 4-bits as a word. hahahaha

[@milkman100001]

i get so wrapped up in listening to you going on about what normal people would call boring.

[@astra004]

With a 16 bit word you can count to 262.144. Given a voltage of 2,5 mV, the difference between each pair of adjacent bit word equals the difference of 0,000.00138 mV.

[@astra004]

Paul McGowan Ok, than the smallest difference between two voltage values is 0,00003815 mV, that is 0,03 microvolt. Can we call it something like voltage sensitivity of the signal transportation process? And doesn’t that means, that there are 65,535 steps in the stair?

[@EddieKMusic]

I really want to correct you, but it's not that important to the topic as a whole, and you asked nicely not to, so I won't:)

[@robh9079]

Brilliant! btw; there are 10 sorts of people in the world - those that understand up-sampling and those who don't...

[@Enigma758]

I you understand how sampling works, then you only need to watch the last 30 seconds of the video.

[@gizmothewytchdoktor1049]

:-) upsampling helps kill the hash! got it.

[@MrMarantzman]

I actually didn't really understand most of what was said in this video... Sorry

[@kurtzcol]

magic pixies of course

[@insulter4625]

4 bits represent 2^4=16 values, not 4 (which is 2 bits). Just for correctness.

[@ThinkingBetter]

Well, it doesn't.

[@al702893]

man, that was almost painful to watch...

[@rabarebra]

4-bit word. 😂 Guy don't know what he's talking about. Paul, what is your degrees?

[@error079]

perhaps Ted Smith should have helped answer that question.

[@aabb5283]

By adding noise, you increase the resulting information, but decrease the original information, that is the quality is normally reduced.

[@Uncle_Herman]

My 1984 Sony CD player and my 2018 Yamaha CD S1000 sound the same on my Klipsch speakers. A lot of it is woo woo hokus pokus snake oil marketing BS. Do blind tests and find out yourself. Many claim to have the ears of dogs or bats and can hear sonic differences and taste what vineyard the grapes came from in their wine. But blind tests are the great equalizer. I prefer my Yamaha CD player because it plays SACD and I have some albums that are SACD and not CD. As long as you can hear it clearly without skipping or jitter, it is all good. No human being will ever tell the difference from mp3 320kbps or SACD. I have done tests and had several friends do the blind tests. Nobody can tell the difference. So enjoy the music and hope that better music will come in the future.

[@laurentzduba1298]

You have a still functional CD player from 1984? The laser diodes of a typical CD player seldom lasts more than 5 years and if this goes kaput, the dealer more often than not suggests that you go buy a new unit. Unless of course you have a top of the line Sony that costs at least 3,000 USD - then at this price range the dealer will provide a "reasonably priced" laser diode replacement every 5 years.