wow, just wow...the amount of thought that went into this. You take us through each concept so perfectly step by step. I honestly don't know if anybody could have explained this so well. Your teaching skills are through the roof. I just want you to get that this kind of teaching skill is just not normal. I just subscribed...because seriously...seriously!
Subscribed for the clear cut animations. After watching so many videos about digital audio, I only got the idea of sample rate. The bit depth was an enigma until I watch this video. Thanks a lot.
I've never heard any significant difference between the 16 and 24 bits, although some people claim they do. My classical music CDs, which usually display the widest dynamic range, sound fantastic and "uncompressed" to me. From a physics perspective, 16 bits give you a much wider dynamic range than what you'd encounter in everyday life. Anyway, with today's uber-compressed music, I suspect 8-bits will soon suffice :).
With a noise shaping 16 bits can give about a same dynamic range as 24 bits can. The only reason of 24 bits is to make a better result with a further DSP. For an output 16 bits are more than enough.
This is a fantastic tutorial. It's so nice to see a video like this with no trash talking in the comments and zero dislikes. Thanks for uploading. You earn one Karma Credit. :)
excellent explanation. finally someone who didn't just say "the higher the bitrate the better the quality" and went silent expecting the audience to take that as the best possible answer.
Thanks Alis. I think there are some of us out there who still rationalize our choices :). In the end however it's important to realize our ears are the real judges. I didn't know anything about bitrates or sample rates twenty years ago when I started playing, but I could immediately tell what sounded good and what didn't. Some people today create 8-bit music and it sounds awesome, although rationally speaking an engineer would say you "shouldn't" use 8-bits when you can use 16 or 32.
MangoldProject You are welcome. I understand now why 44KHz sample rate is standard in music production. But can you explain why is 48KHz and not 44KHz an audio standard in video production?
@guitarbasslover: generally, the answer is "yes". However, there are a few provisions. 1. Beyond certain thresholds our ears cannot tell the difference. There was a paper in the Journal of the Audio Engineering Society in which a 16-bit, 44.1 kHz AD-DA converter was inserted into the path of an audio track; several audio professionals were blind-tested and it turned out they couldn't hear a difference between the audio with and without the converter. (continued in next comment)
Thank you so much! I've always been a visual person and your visual representations and examples made it a lot easier for me to understand. Very well explained
Excellent video ! You not just explained what's sampling and bit depth but also proved it. The way you should how both signals match each other after adding more bit and frequency, it was so amazing !
Asaf.. im not sure if your aware of just how a good a teacher you are ! Even though you oviously have alot of knoledge about the subjects in your videos, you are still able to 'humble' the information and completely relate to the mindset of a beginner in the most perfectly straightforward and simple manner. Even for someone like myself who has some knoledge, this is an excellent way to re-cap on things. I can only wish you much success in your career as gratitude for these videos.
This was an excellent “conceptual” explanation of these otherwise very abstract recording sound ideas. I am definitely subscribing to your channel. I am a neophyte beginner when it comes to home studio music recording and the ‘pace’ of your explanations was perfect for me. Thank you so very much and you’re an exceptional instructor 👨🏫.
@mikevanstaveren Your math paper analogy is good, with one caveat: think of the soundcard as the "grid" on the paper, not the paper itself. You DO play connect the dots, BUT not the original waveform's dots - first, when doing Analong->Digital, you're "warping" the input dots to the closest positions on the grid. When doing Digital->Analog, you're connecting the RECORDED (warped) dots. This "connect-the-dots" does NOT match the original waveform. It may be very close, but not exact.
I never understood all this bit and sampling stuff and got so frustrated when setting up my recording kit to my lap top. Now I fully understand what it all means and it all makes sense now thanks to you. I think the biggest point I never grasped was that the computer had to break a sound wave down into NUMBERS . It never dawned on me although I know computers run with ones and noughts.
@MangoldProject As for why quantization is needed: it's a necessity. Ideally, we'd like to do away with it, but your computer has finite memory and processing capabilities, so it must store the voltage of the recorded waveform using a finite set of numbers. It's part of the price you pay for going digital (analog waveforms also have "effective" resolutions because of signal to noise, etc., but that's a different story.)
The video explains it perfectly from 4:51 The quantization of a sample, in A/D conversion for example, simply rounds off the sample to a set degree. So if you have a wavelength, you have high peaks and low peaks. With quantization, the peaks are rounded off to a set height/depth. 1 bit means all high peaks are rounded to the same level and all low peaks are rounded to the same low level. The higher the bit depth the less rounding off occurs, meaning the sample becomes more accurate in reprod
The way you explained is the best ! Very simple to understand. I can see that you worked hard to make this video. I really thought you used animation software. This is also a great skill to get what you want by just using a simple software.
@mikevanstaveren kbps means "kilobits per second". It doesn't tell you anything specific about the sampling rate or bit depth. For example, an uncompressed 8-bit, 16kHz recording is (2^8 bits) * 16000 (per second) = 4096 kbps. Same goes for a 9-bit, 8kHz recording (=2^9*8000=4096 kbps). Note these numbers are much higher than the average 320 kbps MP3s use. That's because MP3s use lossy compression to reduce the number of kbps (without significantly degrading the quality).
Hey Seskandari, I'm not sure I understood your question. Could you be more specific? If you're asking what we need quantization for, then quantization is a necessary step when digitizing audio (or any signal, really).
Good question. Unfortunately outside my expertise: I know what the differences are, but these have to be weighted in the context of digital audio, which is tricky and probably best done by a (very experienced) software engineer who specializes in DAW. Personally (just guessing), I'd be very surprised if the differences are meaningful. With today's 32-bit engines, I'm sure errors only accumulate in the first few bits, well below the noise levels and ears' dynamic range.
@guitarbasslover (continued): 2. Additionally, there is a "law" which states that, to reproduce frequencies up to X, you need to sample at a rate 2X (the so-called Nyquist's criterion). Since we're interested in audio only up to 20kHz, sampling at 40kHz should be enough. For practical reasons having to do with converter design, this is often extended above 40kHz. However, if you're asking me whether a human being could tell the difference between, say, 192kHz and 96kHz, my answer would be: NO.
@mikevanstaveren, if you are ever buying an mp3 online, remember: mp3 is a lossy form of compression. Generally, the recording doesn't have to be the same bit depth as the distributed recording; especially if it is online or on CD. Generally, online mp3 purchases are 16 bit and 44.1kHz, so that the average consumer may easily play the purchased audio using other CD and mp3 players if the wish to burn or play a copy off of a computer. If it is 24 bit, it will likely say hi-fi, or at least 24bit
Recording a 60dB signal with 40dB of noise in 16 bit is the exact same as recording it in 24-bit (except the noise and signal being more "detailed"), as long as your input is calibrated appropriately to the entire inputs' dynamic range. In general, although I don't hear a difference, many claim they do, so I'd recommend 24 bits (there's no "cost"). Most importantly, TRUST YOUR EARS and EXPERIMENT. Record the same signal in 16 and 24 bit and see for yourself! Don't let "experts" push you around.
You can never "trust your ears" as I know an audio engineer who's end result if his mixes are very "tinny" and he thinks they are fine. Obviously everyones ears are different. I have superb hearing in the high range. I can hear up to 16khz then it drops off then I can "hear" a sound around 17.5 khz. The audiologist says that my bones in my ear are resonating at 17.5 by my eardrum isnt actually hearing it. So with that being said, everyones ears are in fact different. This is why standards are developed I have worked in broadcast video for over 50 years and i can tell you that we align both audio and video using electronic measuring devices to adhere to a standard. This ensures that everyone is guaranteed of receiving the same signals and everything along the transmission path is the same. When it gets to your home devices this is where things sound and look different. For example a person once said "we broadcast a picture not some electronic waveform, so adjust the picture to the monitor thats on your video monitor." Of course that is PURE nonsense, as even the term "monitor" only means to monitor a signal, not to align it with a standard. We use waveform instrument & vector scopes to align the picture quality to a standard. Whatever is displayed on a "monitor" is irrelevant. Audio is adjusted to a calibrated VU meter to ensure proper audio level is being sent out. The video tape machines themselves are calibrated to a test tape that was recorded on a perfectly aligned machine this a standard in the industry. No one relies on thier own eye & ears for correct values. Your hearing may be vastly different than mine, but I am accustomed to hearing things a certain way, and that way is my own personal standard. As long as audio engineers conform to standards in the signal chain they only variable is the audio engineer themselves and they vary greatly. It would be interesting to see the results of these audio engineer audiology reports.
GREAT job !! i finally got a clear, precise, deep, and precise knowledge of what was once a jumbled jigsaw puzzle,,...THANK YOU VERY MUCH..( i subscribed of course !! )
No prob. I almost never do music technology videos, though, just harmony/piano ones. Nevertheless, I hope you will find other interesting videos on the channel.
This is a really good tutorial. I understand everything mentioned very clearly. This also explains why my sound always loses quality when I record my voice. >.>
(continued:) However, since analog tapes have a noise floor (more or less), you need to record close to it's limit to keep the signal-to-noise ratio as high as possible. So most analog recordings are recorded at the volume where the compression (and distortion) start naturally kicking in. Digital recording tries to record close to the limit aswell to make use of the bit depth. But if you record at 24 or even 32 bit (extremely rare, I think), you don't really need to care.
Quantization is pretty interresting and linear interpolation can resamble a better approximation of the original wave form,by following those points,my theory is that even 2 bit audio can be appoximated well trough linear interpolation.
(Continued 2:) Btw, analog tapes bit depth and sample rate is basically only limited by the density of atoms in the material (and so on, not to get too technical). They have almost infinitely higher bit depth and sample rate than any digital recording. That's part of why analog tapes actually sound better, not including the noise and compression/distortion, unless you like that.
I have to disagree with the explanation of the sample rate - it's wrong to represent it as a connection between the dots. It's actually a process of "interpolation" involved! The sampling (analog->digital) is just a recording of discreet values in period of time - so here you are correct. But the opposite conversion (digital->analog, so we can hear it from the speakers) is NOT "connecting the dots". It's interpolation. And 44,1kHz by the Nyquist sampling theorem are above 40kHz, which ideally reproduce the original sound wave. So any interpolation above 40kHz is producing THE SAME result. It does not matter if you sampled 44,1 or 96 or 192 - you get the same analogue output.
Thanks for your input Wattie. See my discussion below with Chris Kamper as to why I've made that choice while explaining. Basically, in my book, when teaching beginners, a clear mental picture which is 95% accurate is better than a complicated explanation involving interpolation, etc ... It's the same way Newtonian mechanics is taught first thing in physics schools, even though we all know it to be "incorrect" and an approximation of the more correct quantum mechanical picture of the universe.
MangoldProject The "kid oriented approach" must NOT leave the impression that the incorrect is correct. When we teach Newton mechanics in school we always tell the children that more complicated theories exist and they will be learned in university. At least we do in Bulgaria. Btw interpolation is not so much complicated - a simple example followed by "similar, much more complicated, model is actually used in digital->analogue conversion" will not hurt anybody.
***** Can you name just one interpolation method which is used in practice and does NOT interpolate the original wave at 100% lossless when the sampling rate is 44.1kHz or higher? I bet you can't :) :) :)
FYI: The "alternate" interpolation methods will differ ONLY in the amount of computing time that they consume. They do NOT differ in the results! The only difference between the methods is how much CPU time and memory they will use to produce the results. Have a good day :)
![[UNCUT] The Loyal Pin ปิ่นภักดิ์ EP.7 (1/4)](http://i.ytimg.com/vi/2rPGOyKs0VE/mqdefault.jpg)
Dude, more than 10 years later and the best video I have found on this topic!
that is no lie - I finally understand it!!
l'm OG like that.
@@MangoldProject lmao
wow, just wow...the amount of thought that went into this. You take us through each concept so perfectly step by step. I honestly don't know if anybody could have explained this so well. Your teaching skills are through the roof. I just want you to get that this kind of teaching skill is just not normal. I just subscribed...because seriously...seriously!
You are a lifesaver. Ultimate video, very crisp and clear. 👍👍👍👍 One of the best videos on the internet.
Subscribed for the clear cut animations. After watching so many videos about digital audio, I only got the idea of sample rate. The bit depth was an enigma until I watch this video. Thanks a lot.
Wow, this is one of the oldest videos on my website ... But welcome aboard!
I've never heard any significant difference between the 16 and 24 bits, although some people claim they do. My classical music CDs, which usually display the widest dynamic range, sound fantastic and "uncompressed" to me.
From a physics perspective, 16 bits give you a much wider dynamic range than what you'd encounter in everyday life. Anyway, with today's uber-compressed music, I suspect 8-bits will soon suffice :).
8 years later, TH-cam audio bitrates are far higher and capable of far better quality
With a noise shaping 16 bits can give about a same dynamic range as 24 bits can. The only reason of 24 bits is to make a better result with a further DSP. For an output 16 bits are more than enough.
I just arrive from my Basic Audio Course and this is the best way to explain the bit depth and the sample rate.. EXCELLENT
Thanks Jorvic!
this is hands down the best video out there on sample rate and bit depth.
This is a fantastic tutorial. It's so nice to see a video like this with no trash talking in the comments and zero dislikes. Thanks for uploading. You earn one Karma Credit. :)
excellent explanation. finally someone who didn't just say "the higher the bitrate the better the quality" and went silent expecting the audience to take that as the best possible answer.
Thanks Alis. I think there are some of us out there who still rationalize our choices :). In the end however it's important to realize our ears are the real judges. I didn't know anything about bitrates or sample rates twenty years ago when I started playing, but I could immediately tell what sounded good and what didn't. Some people today create 8-bit music and it sounds awesome, although rationally speaking an engineer would say you "shouldn't" use 8-bits when you can use 16 or 32.
MangoldProject You are welcome. I understand now why 44KHz sample rate is standard in music production. But can you explain why is 48KHz and not 44KHz an audio standard in video production?
I am a music technology stunt your video helped me understand this topic very easily. keep up the good work !!!
@guitarbasslover: generally, the answer is "yes". However, there are a few provisions.
1. Beyond certain thresholds our ears cannot tell the difference. There was a paper in the Journal of the Audio Engineering Society in which a 16-bit, 44.1 kHz AD-DA converter was inserted into the path of an audio track; several audio professionals were blind-tested and it turned out they couldn't hear a difference between the audio with and without the converter. (continued in next comment)
Thank you so much! I've always been a visual person and your visual representations and examples made it a lot easier for me to understand. Very well explained
outstanding... the best beginner material i hav seen in youtube regarding sampling and quantisation!!! G8 demonstration..
I had assignment about it. I watch all other videos. But your video satisfied me the most. Thank you.
Finally a fantastic easy to understand explanations. Great job.
I just can't tell you how amazing this is!!!!! Really, now everything is crystal clear!
This was such a hugee help, thank you very much, simple and effective explanation!
Happy to help out :)
After watching many videos , finally I understand sample rate and Quantisation .. Thank you so much for this video .
Excellent video ! You not just explained what's sampling and bit depth but also proved it. The way you should how both signals match each other after adding more bit and frequency, it was so amazing !
Asaf.. im not sure if your aware of just how a good a teacher you are !
Even though you oviously have alot of knoledge about the subjects in your videos, you are still able to 'humble' the information and completely relate to the mindset of a beginner in the most perfectly straightforward and simple manner.
Even for someone like myself who has some knoledge, this is an excellent way to re-cap on things.
I can only wish you much success in your career as gratitude for these videos.
This is by far the best explaination I found on TH-cam regarding this topic!
man this is awesome no one has ever explained mathematically and scientificallly like this ever bfore .keep doing this gud job ever bob..!
Very good explanation, just enough depth for a non-math person like myself to get a better understanding of the subject. Thank you for posting.
I love how chill your tone is. Bless you. Made it alot easier
This was an excellent “conceptual” explanation of these otherwise very abstract recording sound ideas. I am definitely subscribing to your channel. I am a neophyte beginner when it comes to home studio music recording and the ‘pace’ of your explanations was perfect for me. Thank you so very much and you’re an exceptional instructor 👨🏫.
Thanks. Your sound explanations go even beyond the keyboard.
@mikevanstaveren Your math paper analogy is good, with one caveat: think of the soundcard as the "grid" on the paper, not the paper itself. You DO play connect the dots, BUT not the original waveform's dots - first, when doing Analong->Digital, you're "warping" the input dots to the closest positions on the grid. When doing Digital->Analog, you're connecting the RECORDED (warped) dots. This "connect-the-dots" does NOT match the original waveform. It may be very close, but not exact.
Thanks so much for this, you are very talented at making difficult concepts easy to understand! Very much appreciated!
This is one of the best tutorials on this topic. Well done.
This has always remained a mystery for me until now. Thanks for enlightening me sir!
+Koen Vanbelle You're welcome Koen.
Super explanation! I knew it was like this, but had problems explaining it to a friend! Your video made my day! Kudos!
mrsarun Thanks! I always say the best test of one's knowledge is trying to explain something to someone else. Glad I could help out!
I never understood all this bit and sampling stuff and got so frustrated when setting up my recording kit to my lap top.
Now I fully understand what it all means and it all makes sense now thanks to you.
I think the biggest point I never grasped was that the computer had to break a sound wave down into NUMBERS .
It never dawned on me although I know computers run with ones and noughts.
@MangoldProject As for why quantization is needed: it's a necessity. Ideally, we'd like to do away with it, but your computer has finite memory and processing capabilities, so it must store the voltage of the recorded waveform using a finite set of numbers. It's part of the price you pay for going digital (analog waveforms also have "effective" resolutions because of signal to noise, etc., but that's a different story.)
This helped with my homework/assignment for college. Thank you for the great and simple explanation.
The video explains it perfectly from 4:51
The quantization of a sample, in A/D conversion for example, simply rounds off the sample to a set degree.
So if you have a wavelength, you have high peaks and low peaks. With quantization, the peaks are rounded off to a set height/depth.
1 bit means all high peaks are rounded to the same level and all low peaks are rounded to the same low level. The higher the bit depth the less rounding off occurs, meaning the sample becomes more accurate in reprod
Thank you very much mangold..i was struggling with my text book..Your video and graphics made things easy...
Happy I could help out. Good luck with your studies.
Thank you so much! You are brilliant! This is the best, by far, explanation I have ever heard in my life on this and I've heard lots.
Great video, I really never fully understood Bit and sample rate until now..
The way you explained is the best ! Very simple to understand. I can see that you worked hard to make this video. I really thought you used animation software. This is also a great skill to get what you want by just using a simple software.
Thanks Advait. All those years slaving making slides for my work finally paid off :)
:)
Fantastic tutorial. Well-paced and clear and with effective graphic accompaniment. Thank you.
Pure clinical explanation. Great work!!! thanks a ton!! cleared with my doubts.
@mikevanstaveren kbps means "kilobits per second". It doesn't tell you anything specific about the sampling rate or bit depth. For example, an uncompressed 8-bit, 16kHz recording is (2^8 bits) * 16000 (per second) = 4096 kbps. Same goes for a 9-bit, 8kHz recording (=2^9*8000=4096 kbps). Note these numbers are much higher than the average 320 kbps MP3s use. That's because MP3s use lossy compression to reduce the number of kbps (without significantly degrading the quality).
What an excellent way of explaining a concept. Very well done.
Awesome explanation, thank you. Exactly what I was looking for
thank you man. as the compulsive ballbreaker said, there could be some inconsistency in the details but the concept pass perfectly
Compulsive ballbreaker ... I'll remember that :)
I Usually don't....but when i read people's comments on the internet....Jesus...in a few seconds i lost all hope in our species survaival ;)
man its amazing how you put to much useful information to 9min long video.. thanks!
Karel Potužník Awww shucks, I do my best :)
Very well explained, Sir. Thank you :)
Hey Seskandari,
I'm not sure I understood your question. Could you be more specific? If you're asking what we need quantization for, then quantization is a necessary step when digitizing audio (or any signal, really).
one of the best explainations i have ever seen...
Finally got this under my belt. Thank you.
Good question. Unfortunately outside my expertise: I know what the differences are, but these have to be weighted in the context of digital audio, which is tricky and probably best done by a (very experienced) software engineer who specializes in DAW. Personally (just guessing), I'd be very surprised if the differences are meaningful. With today's 32-bit engines, I'm sure errors only accumulate in the first few bits, well below the noise levels and ears' dynamic range.
I have just learn an essential building block, thank you very much for the articulate explanation.
Thank you so much for this video - I was so lost before watching this!!
Thank you very much this was amazing. I was struggling a little bit with this and you made it all clear in a glimpse. Thank you again
the best explanation I have seen.
Very helpful! Thanks for taking the time to create this great explanation.
You're welcome Dave.
Mr.Assaf thank you for that great explanations,they are very clear.
This was a great explanation that I am using to study for my exam. Thank you!
@guitarbasslover (continued):
2. Additionally, there is a "law" which states that, to reproduce frequencies up to X, you need to sample at a rate 2X (the so-called Nyquist's criterion). Since we're interested in audio only up to 20kHz, sampling at 40kHz should be enough. For practical reasons having to do with converter design, this is often extended above 40kHz. However, if you're asking me whether a human being could tell the difference between, say, 192kHz and 96kHz, my answer would be: NO.
@mikevanstaveren, if you are ever buying an mp3 online, remember: mp3 is a lossy form of compression.
Generally, the recording doesn't have to be the same bit depth as the distributed recording; especially if it is online or on CD. Generally, online mp3 purchases are 16 bit and 44.1kHz, so that the average consumer may easily play the purchased audio using other CD and mp3 players if the wish to burn or play a copy off of a computer.
If it is 24 bit, it will likely say hi-fi, or at least 24bit
Superb explanation! I couldn't have done it better myself.
Great video! It was just what I was looking for. Cheers!
steelfire819 Thanks mate!
That is true, but which part of the video are you referring to?
My knowledge just went deep....makes total sense
You are an amazing teacher, Thanks.
Very well explained. Well done.
Excellent explanation!
Congratulations!
***** Thanks Bruno! :)
Recording a 60dB signal with 40dB of noise in 16 bit is the exact same as recording it in 24-bit (except the noise and signal being more "detailed"), as long as your input is calibrated appropriately to the entire inputs' dynamic range. In general, although I don't hear a difference, many claim they do, so I'd recommend 24 bits (there's no "cost").
Most importantly, TRUST YOUR EARS and EXPERIMENT. Record the same signal in 16 and 24 bit and see for yourself! Don't let "experts" push you around.
You can never "trust your ears" as I know an audio engineer who's end result if his mixes are very "tinny" and he thinks they are fine. Obviously everyones ears are different. I have superb hearing in the high range. I can hear up to 16khz then it drops off then I can "hear" a sound around 17.5 khz. The audiologist says that my bones in my ear are resonating at 17.5 by my eardrum isnt actually hearing it. So with that being said, everyones ears are in fact different.
This is why standards are developed
I have worked in broadcast video for over 50 years and i can tell you that we align both audio and video using electronic measuring devices to adhere to a standard. This ensures that everyone is guaranteed of receiving the same signals and everything along the transmission path is the same. When it gets to your home devices this is where things sound and look different.
For example a person once said "we broadcast a picture not some electronic waveform, so adjust the picture to the monitor thats on your video monitor." Of course that is PURE nonsense, as even the term "monitor" only means to monitor a signal, not to align it with a standard.
We use waveform instrument & vector scopes to align the picture quality to a standard. Whatever is displayed on a "monitor" is irrelevant. Audio is adjusted to a calibrated VU meter to ensure proper audio level is being sent out. The video tape machines themselves are calibrated to a test tape that was recorded on a perfectly aligned machine this a standard in the industry. No one relies on thier own eye & ears for correct values.
Your hearing may be vastly different than mine, but I am accustomed to hearing things a certain way, and that way is my own personal standard. As long as audio engineers conform to standards in the signal chain they only variable is the audio engineer themselves and they vary greatly.
It would be interesting to see the results of these audio engineer audiology reports.
The best ever explanation. Thanks.
A fantastic explanation!
This really helped me with my class. I thank you kind stranger.
+fsiagia You're welcome!
GREAT job !! i finally got a clear, precise, deep, and precise knowledge of what was once a jumbled jigsaw puzzle,,...THANK YOU VERY MUCH..( i subscribed of course !! )
No prob. I almost never do music technology videos, though, just harmony/piano ones. Nevertheless, I hope you will find other interesting videos on the channel.
Thank you. I got what I wanted and loved the way you explained.
Congratulations Professor !!!
Great Explanation !
Perfect explanation!
Thank you for the visual presentation. I find this very helpful for understanding!
+Myles Sphere You're welcome Myles. I'm also a visual learner!
that helped so much thanks for the great visuals! it really makes things clear and easy to understand!
This is a really good tutorial. I understand everything mentioned very clearly.
This also explains why my sound always loses quality when I record my voice. >.>
I just looking that info for so long time. Thx alot
very clearly explained, thanks!
You're welcome.
Finally understand it now. Thank you so much
Excellent tutorial mate.
best explanation on this subject. thanks
And oldie but still good!
Great video! Finally understood after struggling with my book
Mrsmartdude13 Glad I could be of help.
Very clear explanation. Thanks
That was great. Very good explanation.
Thank you.
Excellent explanation. You'r good man!!!
(continued:) However, since analog tapes have a noise floor (more or less), you need to record close to it's limit to keep the signal-to-noise ratio as high as possible. So most analog recordings are recorded at the volume where the compression (and distortion) start naturally kicking in.
Digital recording tries to record close to the limit aswell to make use of the bit depth. But if you record at 24 or even 32 bit (extremely rare, I think), you don't really need to care.
Thank you so much this is the best explanation on youtube! I hope that you will post other more video!
Simona (:
I have quite a few extra videos. About 200 of them by now :).
Anything in particular?
Great summary!
Finally I'll remember it. Great. Thanks! Nice and simple.
+D NC Thanks! I do my best to simplify ...
Quantization is pretty interresting and linear interpolation can resamble a better approximation of the original wave form,by following those points,my theory is that even 2 bit audio can be appoximated well trough linear interpolation.
very interesting, and very well explained.
Very good explanation
Thanks!
very nice explanation!
Nice explanation sir!
Whenever I don't get sleep. I watch this video.
Very good explanation, thanks :)
(Continued 2:) Btw, analog tapes bit depth and sample rate is basically only limited by the density of atoms in the material (and so on, not to get too technical). They have almost infinitely higher bit depth and sample rate than any digital recording. That's part of why analog tapes actually sound better, not including the noise and compression/distortion, unless you like that.
I have to disagree with the explanation of the sample rate - it's wrong to represent it as a connection between the dots. It's actually a process of "interpolation" involved! The sampling (analog->digital) is just a recording of discreet values in period of time - so here you are correct. But the opposite conversion (digital->analog, so we can hear it from the speakers) is NOT "connecting the dots". It's interpolation. And 44,1kHz by the Nyquist sampling theorem are above 40kHz, which ideally reproduce the original sound wave. So any interpolation above 40kHz is producing THE SAME result. It does not matter if you sampled 44,1 or 96 or 192 - you get the same analogue output.
Thanks for your input Wattie. See my discussion below with Chris Kamper as to why I've made that choice while explaining. Basically, in my book, when teaching beginners, a clear mental picture which is 95% accurate is better than a complicated explanation involving interpolation, etc ...
It's the same way Newtonian mechanics is taught first thing in physics schools, even though we all know it to be "incorrect" and an approximation of the more correct quantum mechanical picture of the universe.
MangoldProject The "kid oriented approach" must NOT leave the impression that the incorrect is correct. When we teach Newton mechanics in school we always tell the children that more complicated theories exist and they will be learned in university. At least we do in Bulgaria.
Btw interpolation is not so much complicated - a simple example followed by "similar, much more complicated, model is actually used in digital->analogue conversion" will not hurt anybody.
***** Can you name just one interpolation method which is used in practice and does NOT interpolate the original wave at 100% lossless when the sampling rate is 44.1kHz or higher? I bet you can't :) :) :)
FYI: The "alternate" interpolation methods will differ ONLY in the amount of computing time that they consume. They do NOT differ in the results! The only difference between the methods is how much CPU time and memory they will use to produce the results.
Have a good day :)