You've probably seen them: 44.1 kHz and 48 kHz. But why are there two sampling rates so close together in the first place? John explains the origins of these two seemingly similar audio sampling rates.
Always enjoy your educational videos. This one reminded me of a video by Techmoan from about a year ago demonstrating an early digital audio recorder that used VHS tape and it seems a fitting complement to yours since it includes a little history of PCM recording. ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-WVDCxTtn4OQ.html
@@bradcole8191 That's a bit different, as it's multi-channel and doesn't have to retain compatibility with a normal video signal. I have ADAT and DTRS machines in my studio.
I have a "digital" VCR--Toshiba--made in the late 80s. It has analogue ins and outs, but the audio is PCM. My band recorded an entire CD with this format because we couldn't afford a DAT machine. When we went to master our mixes, we took the VHS tape and the mastering house was able to decode it digitally. It was amazing at the time. The VCR cost about $1100, and I still have it, and it still works.
I had this same machine: the Toshiba DX900. Known as the "poor man's DAT." If yours still works, consider yourself lucky! The most obvious thing to go wrong is the rubber idler wheel turning into a hockey puck and any of the rubber belts disintegrating - but that's common with any older tape machine. I mixed & mastered many home studio projects on it with good results (the audio specs of this machine probably exceeded what I was putting into it!). I'm pretty sure it used the Sony PCM-F1 format but was 14-bit instead of 16.
bobby sands You needed adaptor as the Sony PCM, too get it in PCM out of the VCR cassette. Many VRC recorders supported the PCM audio recording on VCR. You should be able to read the tapes you made, restore them, are they still good now?
Just as an anecdotal data point, I've been using 48k since the late 90s when I started using Cubase VST with a SoundBlaster Live! card and found that the card wouldn't sync properly at 44.1k, because it was internally 48k and doing an inaccurate downsampling to get 44.1. It's funny now remembering how I was annoyed at the extra disk space it used, totally irrelevant now. But back then, when I'd spent a lot of cash (for me) on a "massive" 17.4GB hard disk for audio recording storage, it was a significant overhead.
Haha, believe it or not, I am STILL using a Sound Blaster Audigy 2ZS for audio production, which relies on an evolution of the chipset used in the SBLive! and has the same "48khz or inaccurate downsampling" limitation (the original SBLive card relies on the emu10k1 chip, while the Audigy2ZS relies on the emu10k2 chip... there's even an Audigy Rx released a few years ago relying on an emu10k2.1 chip which still caries this limitation). Which leads me to usually work with a 48KHz workflow, and then manually downsample to 44KHz in software if need be. It also seems to me that, as video and audio become more intertwined (e.g. people listen to music on RU-vid), 48KHz is increasingly becoming a standard even for audio only.
@@Liam3072 Excellent! I've actually considered building a small PCI machine just so I can use my SBLives again. it feels to me that in many ways we've gone backwards in PC audio, I mean the SBLive is over 20 years old and was an absolute powerhouse. And I was actually running two in the same machine using modified drivers! My final production config was one using the Live! drivers and one using the E-Mu APS drivers which had some great effects. Good days.
@@ian_b Or you could get a usb audio interface that has no issues with whatever sample rate you use, better dac's, asio drivers, no ? is there a reason to use soundblasters/ non audio prod focused pci cards ? (this is a real question I'm not trying to be antagonistic)
@@escalator9734 and, if he insists on Creative, could use 0404 or, say, 1212M. Older recording audio cards can be often found for cheap. Especially the PCI ones. Or Firewire, though getting it to work reliably on Windows is not easy.
@@FilmmakerIQ Haha, not you but the nice explainer videos you used to do back then. I remember I binge watched these videos in a single day few years back. Keep it up Benjamin Button 🤪
I share a bunch of your stuff with my TV students, I hope you haven't minded ;). I've taught the 44.1 vs 48 lecture for ages but without the PCM background. This is another gem that I can add to help make my dull lectures more engaging. Thanks John Hess!
We are really lucky how futureproof the 44.1 kHz CD quality standard was. It did, does, and will always sound good. Still higher quality than the audio of almost everything we stream today. It's hard to believe such a good quality standard came in the 1980s while video standards arguably didn't get futureproof until the Blu-Ray/HD DVD came out.
It was optimized by dithering on the one hand. On the other hand, one could use a higher dynamic at home. It's not allowed to disturb neighborhood, the room is noisy, consumer speakers can't handle high sound levels.
Yeah, they really knew what they were doing when they set that standard. On the recording side 24 bits is prefered, 16 bits is just fine for a mastered music track though.. I record, mix and master my band and have the equipment needed for that so i have both experience and knowledge of that, im also a service tech educated in the Audio/video area.
@Leif And they should be lucky that they cannot hear higher frequencies than what we mere mortals do as the so called high frequency recordings contains lots and lots of noise in the upper frequencies: ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-MOVT_4yO5hs.html
@@kaitlyn__L No i have never tried floating point, it seems to be just like safety net if you dont do your homework and record at proper levels (read: not to high levels) it wont make it sound any better than 24 bits and you will apparently have wav files that are 30% larger or maybe more, with more data it will also take more CPU power of your computer. If you dont clip the audio in the DAW when recording 24 bits is just fine, you dont have to worry very much about the recording level maybe being somewhat low either since 24bits has 256 times more dynamic range than 16 bits so there is plenty of range for audio production.
In the late 80's the amount of data you could put on a CD seemed unimaginable to us. The place I worked for put all of our dozen applications onto a single CD with space to spare and that was writing the contents of multiple hard drives onto this seemingly bottomless media. You could buy a 600 megabyte SCSI drive but the cost was horrific so we didn't.
I remember that hard drives only started to catch up with CD capacity around 1994. It's so weird these days to imagine optical media having much larger capacity than hard disks or SSDs. Even quad-layer Blu-ray tops out at 100GB, while you won't find a hard drive with less than 1TB these days.
I worked in a Electronic retailer here in Sydney, Aus (Dick Smith) in the early 80's... (A competitor to what you'd call radio shack, here known as Tandy) and we had a Tandy (RS) TRS80 clone called the System 80... I sold a 5Mb Drive for $15K (Aus), which also required a $1.5k I/o card option to use it! Fun times indeed. Thats 1 min mono/ 30sec Stereo in sample currency @44.1 / 16bit......
As a computer engineer with a background in signal processing, I would like to commend you on a very well presented summary of this issue. I found no issue with any of the facts you present. Sadly, that is very rare. You, sir, are a wonderful resource and I hope people appreciate you for it.
I knew about 44.1K but never got 48K, especially since 50K was already an established standard. Thank you! Side note, engineers of earlier digital recorders believed 60KHz would be ideal (you need room for the antialiasing filters to work) but it was pushing the tech a little too far, so they settled on 50K.
Decca used 50khz in a lot of early digital recording experiments of the late 70s in Europe. It might be a stretch to say it was a standard because nothing was standardize yet, but it was common.
@@FilmmakerIQ Soundstream, 3M and Mitsubishi also used 50K. Those were the most common digital multitrack systems in the late 70s. I forget what Denon used off the top of my head but it was somewhat less. BTW, I'm glad to see you posting again.
Despite, that CD Audio standard is 40years old, only few records has been able to utilize it's 96db of dynamic range. And people believes that if they don't have at least 32bits/ 196Khz, sound will be suffering on they precious soundbar. :-D
Takes very good front end equipment for anything past 19bits to not be lost in the noise floor, and very good back end equipment to be able to hear it. 20bits is really the effective minimum. 24bit audio does sound better than 18bit, but very few people have even seen equipment that can render it, let alone heard it.
@@joshua43214 It makes mostly sense in phase of recording in studio, as amater musician I'm sampling at 44.1khz@24bits from my old synth rack modules. But if you are good in mastering you can make excellent 44.1khz@16bit downmix (not myself :-)), which will blown your mind on good D/As, amplifier, and speakers. I don't forget to hear Vangelis album DIRECT (CD) on Mcintosh mc2120, trebles and basses has been on 0,0, almost showcase dynamic range.
yeah recording at 24 makes sense because having the extra headroom gives you a lot more leeway with your recording levels while avoiding clipping, but there's not much reason to make your final rendered output more than 16bit. I've never met anyone who's been able to actually tell the difference between a 16bit and 24bit recording on a proper A/B/X test, or even hear the difference waveform between the 16 and 24-bit version without cranking the volume up so high that the original audio would destroy their eardrums.
@@joshua43214 As long as proper dithering is used the only difference an increase in resolution makes is a decrease in quantization noise floor, so it's a bit weird to refer to the difference being "lost in the noise floor" ... though you presumably mean the analog noise floor as opposed to the quantization noise floor.
Great video! It is very interesting to learn why CDs use 44.1kHz. My job involves processing audio recorded from telephones. Historically, that's usually involved a sample rate of 6 or 8 kHz (due to limitations of the old hardware), although, more recently, there's been an increase in the use of 16, 24, 32, or even 48kHz. From that perspective, 44.1kHz audio is a major pain, because a lot of software used in the telephony sphere (e.g. speech recognition) still tends to expect 8kHz. If someone records some audio externally and then wants to import it into the system, it's a lot easier to deal with 48kHz than 44.1kHz if and when it's necessary to convert it to 8kHz!
I once wrote a resampling library for a phone system. I did that by filtering the signal (so it wouldn't have anything over the nyquist of the target rate) then making sure it was oversampled by at least 4 * the target rate (sinc filters for oversampling), then using splines to interpolate down. That's a method that works for arbitrary rates. It's been used in synthesizers.
Love videos like these! A surprisingly in-depth review of some foundational material along with historical and contemporary context. Learning disguised as a good time, great job!
Thanks John. I have missed these tutorial videos, which seem to have been replaced by live stream content and similar. This is what I subscribed to the channel for, and I am delighted to have watched this one. I hope you will be able to make more of these very educational and interesting videos in future. Thanks.
The live streams are a way to stay relevant in the RU-vid algorithm. And they are also personally fun for me to do. The problem is these videos take a tremendous amount of research to pull off and RU-vid favors people that consistently add content.
@@FilmmakerIQ I’ve also heard they can actually tank your Algorithm Recommendation Likelihood (or whatever it’s really called) if the streams have low view counts, allegedly it considers them similar to video view performance. So they can be a real double-edged sword depending on various factors. I’ve seen a lot of other channels make a separate streaming channel and then promo their videos in the streams. Not suggesting you have to do any particular action, but I figured I should share the knowledge!
Your videos are quite simply the best, most informative free online resource for the technical aspects of film making I’ve come across. Always a delight and your presentation is top notch too
Also note that 48khz is still the universal standard in all HDTV formats, UHD as well. Standard HD frames rates are 23.98hz, 24hz, 50hz, 59.94hz and 60hz. Note the 59.94 still hangs on even though it's need is obsolete in ATSC. But because of conversion to and from NTSC, it lives on.
Damn, every subject i think I know well, you teach a ton more. I remember marathoning your education series in Feb2020 when on holiday and learnt a ton. Thanks!
That was fascinating! It's one of those things I just took as a given, and never realised I wanted to know, but then you popped onto my RU-vid feed, and I realised I really wanted to know this. Great explanation! Subbed.
Going to digital does require that any frequency above the Nyquest limit must be filtered out or aliasing will occur. However, the best analog Low pass filters we can make are a sixth order filter with-36 dB/Octave. (Roughly 1/64 the voltage per octave) That's just not enough and any ultrasonic sounds will appear as lower frequency sounds due to aliasing. The higher the sampling rate, the greater the reduction in ultrasonic noise. No surprise that some recording studios like to sample at even higher rates of 96,000 samples per second and higher. Digital filters have an impressive reduction of over -100dB per octave, which puts all ultrasonic sounds into the floor. After that the digital audio can be reduced to a lower sample rate.
To be honest, there's not much reason to go over 48kHz when you're using somewhat modern professional plugins. First and the most obvious is, 96kHz almost as severe aliasing issues as 48kHz. Sure, you have 2 extra octaves of headroom which will get rid of the aliasing issue when using saturation in one stage, but adding stages will introduce just as much artifacts as 48kHz. 192kHz would get rid of most of the aliasing issues for first two stages, but then again, it's not very practical and still not enough. You can add lowpass to every stage and get rid of the aliasing, but having 192kHz audio files being processed in three or more stages with lowpass in every stage is very inefficient. You get much more efficient results using 48kHz and 4 or 8 times oversampling in plugin (some even has as high as 16 and 32 times oversampling without being too hard on CPU) and it's also much more efficient in reducing aliasing. Then there's the 44.1kHz vs 48kHz and when using oversampling, the difference should be very minimal, right? With 44.1 filtering has to be extremely steep when removing stuff over 22.05kHz without affecting signal at 20kHz. With 48kHz you basically have twice the headroom in filtering out the unwanted stuff that creates the aliasing. So basically 48kHz is twice as good as 44.1kHz and as I've stated it's very inefficient to use higher than 48kHz. Then again, not everyone uses plugins with internal oversampling. Also, someone might as well have Threadripper and billion terabytes of space to utilize, so might as well record, mix and master at 192kHz.
@@eliteextremophile8895 Clearly 48kHz is better than 44.1kHz sample rate. Even at 48kHz, the Nyquest Frequency is 48kHz / 2 = 24kHz. Once you exceed the Nyquest Frequency, the frequency appears to go down. That is 24kHz is 24kHz, but 25kHz appears as 23kHz. This is due to aliasing. Assume that we start a Sixth Order Filter at 12kHz, it would be -36dB (1/64 the signal) at 24kHz. Just not enough to fully remove the high frequency noise. I've never staggered a bunch of 6th Order low pass fillers at different frequencies, like 12kHz, 15kHz, 20kHz, to see what happens, so I'm not sure if that's the answer. The reason I suggest 96kHz sample rate is that there's so much head room for a 6th Order filter. The input frequency would have to reach 48kHz before the Nyquest limit kicks in. Assuming the Low Pass is set to 19kHz, 48kHz is 2.5 octaves higher in frequency and offering a -90dB drop in signal level. From that point, a digital low pass filter can be over -100dB per octave and remove anything above 20kHz. Now if staggered low pass filters do the job, the 96kHz sample rate is unnecessary.
Great points Bob. 96k is a common choice in the studio, as you mention. For example, I use the DEQ2496 as a DSP and it unsurprisingly operates at 24bit 96kHz. That model has been around since at least 2003 and 96k was in use long before then. There are so many benefits of higher sample rates, especially since common hardware can decode at ridiculous rates like 32bit 384k and disk space is limitless in the studio. Though in my opinion, for (non-hifi) consumer audio 16bit 44k is good enough. Edit: I should note, this does depend on the studio and software/setup in question. Sometimes there are reasons to avoid higher sample rates.
Excellent. Just excellent. My 50 year media production career started with broadcast TV in the 1970's golden age of analog and rode through the evolution to digital audio and video production (starting with the introduction of character generator computers and the Ampex Digital Optics devices circa 1980). Always been fascinated with the tech of the various format standards and how the analog legacy dictated a lot of the digital video format decisions, but I didn't know about the direct influence of analog video legacies on digital audio standards until I saw this video. Thanks! Liked and Subscribed!
it's just pointless to me IMO compared to actual analog, as the same bit-depth exceeded "0 to 100" distortion nature of digital and lack of actual artefacting (saturation, compression) to the audio signal itself leads it to not have any kind of sonic aesthetic/ FX use case in audio production/ mastering. All though, perhaps some hardware has some means of A/D/A that may be coloured, and quite complex ie exiting, stereo processing. I'm sure it has other uses that people would dig up though.
@@allyemeraart in the scheme of things, yeah they're pointless because we have even low tier stuff that outperforms it. But as a standalone recorder or if you need so IO via ADAT in a pinch or in the rare case you get someone looking to transfer tapes. They're good to have around
One little thing to add: one might wonder "Why not just use 40 kHz these days, given that we don't record on video tape anymore?" Well, to avoid aliasing artifacts, you need to use a low-pass filter on the recording to remove all frequencies that are higher than what the Nyquist-Shannon sampling theorem allows. The steeper the filter, however, the more artifacts the filter itself introduces. An infinitely steep cutoff would be a disaster, therefore you want to have a bit of give in the sampling frequency. Which also means that in principle, 48 kHz _could_ sound better than 44.1 kHz, even though no human can hear those frequencies.
Dan Worall, the Attenborough of audio engineering, goes very in depth about the samplerates from audio engineering standpoint. I couldn't recommend it more. Video is called "Samplerates: the higher the better, right?" and it's on FabFilter's RU-vid channel. Even if you don't have any prior knowledge about audio engineering, it will demystify the whole topic very efficiently.
@@eliteextremophile8895 I will have a look later but it's important to note that I was just talking about sample rates close to the minimum, not about 192 kHz and stuff like that, that's not doing anything for you other than to overwhelm your audio equipment. Unless you're a bat researcher, you don't need that sample rate. More than 16 bit, however, can be quite useful and at least has no detrimental effect.
@@eliteextremophile8895 very good video indeed. As I'm not a music producer myself, I definitely learned something about that part of the tool chain! Here is the link BTW ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE--jCwIsT0X8M.html
@@unvergebeneid I completed a Diploma of Audio Engineering at SAE back in 1994 and embraced digital audio almost immediately when it was released. While I agree that anything over 20k is not audible to the human ear (most people are lucky if they can hear much above 16k) it is interesting to note that frequencies above our hearing threshold still have an impact on the end result due to harmonics and intermodulation distortion. This is one of the main reasons that MP3 compression never sounds as good as a raw wave file, because the frequencies that it eliminates for the sake of file space may not be in the general hearing range but have an indirect effect on the rest of the audio that is in the range of your ears. Please don't take this as a condescending or rude comment it's just something that I learned a little about in my course and have also researched a little over the years, I run a small Music and Video production studio here in Melbourne Australia and generally use 48k as my standard sample rate and 24bit for that little extra dynamics and lower noise floor even though my audio interface is capable of 192k sample rate.
@@paulstrahan4670 isn't intermodulation distortion a result of imperfections in audio equipment and should ideally not occur at all? So if you lowpass those inaudible frequencies away, wouldn't that bring the signal closer to the ideal? As for compression, from my experience, mp3 is most noticeable in percussion, not in anything harmonic.
This was very interesting to watch and very informative. Definitely going to subscribe and check out the rest of your videos. I’m interested in learning how to edit video, and learning basic film making technique/technology. Just got a Mac mini and a friend of mine is giving me a version of final cut software for me to use and start on
I personally choose 48khz for my game engine since I can implement millisecond-based timing quite easily with that sampling frequency, otherwise it can stretch 44.1khz samples to 48 if needed.
I couldn’t discern a difference between 44.1 and 48 in various settings, so besides switching to match my music (half the FLACs I have from Bandcamp are 48 while the other half and my own CDs are 44.1) which is a PITA I just allow for resampling. But I think I left it in 44.1 as default so that CD rips wouldn’t need resampled. But if it reduces audio packet latency, I’d happily put it in 48 since it sounds the same to me anyway.
@@kaitlyn__L I would store data in the original format. If all you do is listen to audio, latency does not matter that much, and most media players will upsample in real-time. It is however good to configure your audio server ie JACKD to run at 48 kHz.
Simply put, incredible video! Even the slides are awesome. I've been text messaging pics of them to my buddies in real-time as I've been watching. Sure, I could have sent a RU-vid link, but hey, what's one more conversion standard? As they say, the good thing about standards, there's so many to choose from!
It really helps me to understand technology when the history of the development explained so clearly. For example what was an arbitrary, unconstrained choice versus what was guided/dictated by legacy. Thank you.
I remember some high quality formats that really didn't stick around. SACD & DVD audio. I have Natalie Merchant "Carnival" on DVD audio. Only a few DVD players would play the digital audio. I never purchased any SACD discs. I have "Las Ketchup" & "Axe Bahia" on a video CD. This was the forerunner to DVDs. Again, only a few DVD players played this oddball format or it only played on a computer. XP at the time. I remember the DAT tapes & Sony mini discs. Cassettes had chrome ıı or metal ıv & Dolby B or C. Later came S. To erase the metal required strong magnets in the cassette recorder. TDK & Teac made some studio version cassette tapes with ball bearings. (not for retail sale) That was extremely clean with headphones on. Later came MP3s. Standard was 128. Some car CD players would play those. Then you had the up to 320 & variable bit rate. WMA & AAC was out there too. Real Player had some oddball AT8 format.
Y aserejé ja de je De jebe tu de jebere Sebiunouva Majabi an de bugui An de buididipí Aserejé ja de je De jebe tu de jebere Sebiunouva Majabi an de bugui An de buididipí Aserejé ja de je De jebe tu de jebere Sebiunouva Majabi an de bugui An de buididipí Holey Underwear, I actually have this version. Don't ask me why. And I said a hey, a hah ... dammit I don't recall... but I _do_ know it's not jibberish, it's almost a patois of english pushed into spanish sounds. "My hobby on the boogie and the booty ..." And the DJ that he knows plays the hymn (anthem) at twelve (midnight), for diego it's his favourite song, and he dances and he likes (enjoys) it and he sings it... I said a hey, ahah...
I have the Sony PCM-601ESD PCM adapter, I recorded a lot of albums from my computer to VHS and Video8 tapes, a lot of fun. But there are computer programs that can do that now if you have a video card that can pull the video off of the tape and convert it to PCM inside the computer using special app or even record a PCM audio back into the video tape without needing the PCM adapter.
First video, looks ok. You're not the Frugal filmaker, but... you might get me hooked. Oh, and a shoutout for mentioning Umatic, which is the second time in a couple of days to be mentioned, since I had been talking to a friend about some old editing equipment and flying erase heads. No mention of "white-a-key" - the technique we used to get a clipped area out of the picture without a fancy switcher (piece of white paper, careful positioning, clip to white which dropped out and you see what's in the other channel/camera/playback source). Of course, you still needed the TBC. Only criticism: you dropped 'kilo' twice, just before the moire moment. Nobody but me noticed.
It's interesting to see digital audio data being saved as picture information on videotape. Just as digital data for computers used to be stored as binary code audio-signal like on Commodore 64 Datasettes or ZX Spectrum tapes. A clever concept. Something similar was later used in the 90s for VHS tapes to save digital files from harddrives to a Videotape. The software was sold for PC and came with a video-output card that had to be installed into the computer. You would then select all the files you wanted to backup on VHS and it would then pack everything into a archive dump and export it as a Videosignal which was a black & white video signal (series of blocks that looked like animated barcodes and QR codes).
Similar to using dialup modems to transmit data over public phone circuits which were designed to carry only the narrow band of frequencies necessary to convey intelligible human speech. Those ZX Spectrum tapes sounded much the same as a modem.
@@owendoconnor Yup. You could even hear if a turbo was used or not, which removed the many many redundancy blocks in the binary data, that was often checked multiple times to assure data consistency.
I have one of Sony's PCM to video adapters, not from the PCM-1600 line, but the later, consumer oriented PCM-501ES model. It's similar to the Technics PCM deck that Mark Hesse posted, but it lacks a transport. Instead, it has a composite video input and output, designed to be hooked up to a VCR (be it VHS, Betamax or something else) It uses a different video encoding method than the 1600 series, but the three samples per line are still clearly visible. Of course, it does record and play back at 44.1 kHz. I wasn't sure about the origin of 48 kHz. I thought it might had to do something with DAT tapes, since that was the earliest widespread use of the 48 kHz sampling rate, outside of the film and video world.
Oh, and Sony's PCM-501ES (601ESD and 701ES) offered 16 bits of resolution, while the Technics deck could only do 14 bits. Here's a video of my PCM-501ES playing back some music from a VHS cassette: ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-MLDEwUDf5wE.html
48k also makes circuit design just a bit easier, you can use less aggressive anti-aliasing filtering.. and it simplifies the math a bit in digital side. The other number, 24bit is even more important specially in broadcasting as it allows for more headroom. 16bit is what we need but when you are working with live signals that are often unpredictable you will enjoy having those few bits extra. This works well also in other audio production, 24/48k is what vidiots wanted and it happens to also make things better elsewhere. So, thanks.. i guess.. There is no need to go higher, we can prove it quite definitively, also when it comes to processing but with one crucial detail added: 48k requires oversampling and filtering at each processing stage (384k project sample rate is equivalent to 48k if latter uses oversampling scheme, it also uses 8 times the bandwidth and file space and has a higher processing load. 192k is worse and 96k is damn right idiotic, it is considerably worse to a point where it can be audible..). So, turn all of those "high quality" and "oversampling" buttons on in your plugins, they are there for that specific reason. Especially important with all processing that affect the time component of the signal in anyway, this includes compressors.. but not stuff like linear saturation or simple gain, distortion or even delays (repeating is not modifying the time component.. ). Chorus and phasers are especially problematic and oversampling is absolutely mandatory. 24/48k is perfect format for all audio production.
assuming there is an abundance capacity to process/store that higher density data, wouldn't it be better to ditch oversampling (i.e. interpolation/guess) but use original recorded data at the equivalent sampling rate? (48kx8=384k, work for noise shaping, editing, downsample for various consumption)
The issue is computational processing power. Audio mixing involves often hundreds of simultaneous tracks with long effects chains. Where as there's some flex and give with sync when it comes to video sync (there's about 100ms wiggle room before you can tell something is out of sync), there is almost zero wiggle room when mixing audio tracks. Add on top of that oversampling a recording to 384khz is just downright pointless as it introduces more sampling errors and virtually no microphone can generate the response that could effectively utilize the frequency space. Now there might be some filters that might benefit by the oversample but they can do that on their own.
My understanding is limited but basically the underlying concept is that going beyond say 88 or 96khz, all your doing is capturing frequencies we aren't going to hear anyways. An in doing so we introduce intermodulation errors. Digital sampling theory states that we can recreate perfectly any signal that is has half frequencies that are half the sample rate, we need extra headroom to band limit the signal to prevent aliasing. But after that, we're just eating resources that's won't really get us any more benefit and potentially leadimg to degrading the signal.
it's about collecting (record) as much audio PCM/PDM data as you can, convert/process/edit using as much precision as you can, then downsample (depends on your targeted viewer/audience and expected playback equipment) for the final release, whether it be 16/48 or 44.1 or 24/96 or DSD64 or DSD128 or ...
I had a 'portable' Beta recorder which could be used to record sound as PCM. Real ahead of its time. It was only mono though. BTW, I noticed the candle stick phone on you desk. Do you know the back story of the ringer current for telephones? It is interesting. Ninety volts @ 20 Hz. I know crazy.
Interesting stuff! Now what about bit rates? Last I heard it was still 16, and low end toys and such were 8bit. But now with 32bit and 64bit computers and such, is audio taking advantage of that as well?
On a similar subject, can anyone tell me why the Amiga used 8363hz as a standard sampling rate? It seems so odd, why not something round like 8000 or 10000hz?
Interesting. I couldn't find anything online about it other than it was the interrupt speed of the Paula chip (whatever that means). I also checked that 8363 is a prime number so that might have something to do with it
The process of recording sound stored in the form of thousands of indivdual measurement each at a discrete unit of time called.. very thanks for a super video....
Excellent contribution!! You will come across a lot of mythology surrounding the choice of 44.1Khz as the sampling frequency for the Compact Disc, but this is the true explanation. Rotating-head video machines were the only recording format we had at the time with the recording density needed to record continuous program at a sampling frequency above the Nyquist/Shannon rate of 40Khz, to deliver 20 Khz audio - and the U-Matic machine chosen for the Sony PCM 1600 - 1610 and 1630 mastering process delivered 44.1Khz using three frames per sample. Other formats did exist; As early as 1980, Sony's F1 EIAJ format allowed recording of 20Khz audio on VHS tapes at a lower cost, and a sample rate of 44.056Khz, but the PCM 1600, later to become the 1610 then 1630 had already established itself as the industry standard for mastering of CDs.
Hey John great content and information, your my TV dinner show. Just asking what's your opinion for You Tube talking head type videos using a real well appointed back ground or using a Green Screen? Scott
I have been recording/rendering my personal electronic music projects at 48k/24bit or 48bit for yours now. With kind of powerful workstations like the one I have, I can pretty much throw whatever I want at it.
So then how to edit and mix video with music overlays which may come from CD or sources with unmatched sampling rate? How is the resultant video production file mixed so there's no aberrations in the audio & video stream?
Lots of digital wizardry - probably some upsampling and resampling going on. Processing power really solves of a lot of those problems these days. Remember with Nyquist everything from 0-22.05 KHZ is EXACTLY the same be it recorded at 44.1 or 48... the only difference is in the high range (from 22.05-24 KHZ). So soundwise, there is no difference in the range we can actually hear.
Nice video! To add to this: Nowadays with good conversion algorithms you won't notice it if you convert from 48 to 44.1 if you don't do it over and over again. Thus I do all recording in 48khz, even for CD productions. So in the end I always get a video version and a CD version. But seriously, CDs are almost dead so the very most audio I produce goes into video anyways. One can discuss 96khz sampling, and people do. But also there is a difference between 96khz processing which basically is a 2x oversampling on the entire process, while still having recorded at 48khz, and 96khz recording. But anyways, most videos on YT are being watched on smartphones probably, so anything will be good enough.
That's why I'm thinking everything is going to start to move to 48... Though strangely, RU-vid is 44.1 and I don't think anyone notices the difference.
@@FilmmakerIQ I think YT goes 44.1 because in the end it saves them data traffic. As simple as that. Most likely re-encoding lossy formats will give you more damage than resampling from 48 from 44.1 once in the entire process. This is why I like to upload videos with 24bit PCM audio @48khz to YT. Most times I upload CineForm so that also no bad video (h264) will be transcoded to other bad video (VP9 or yet another h264). So good video to bad video only. :-D Sad thing is archived video live streams that get messed up by YT quite noticeable, at least the video. The audio less noticeable. But of course you can't stream CineForm, your internet uplink would simply break down. For audio there's platforms like BandCamp that actually do deliver 44.1 FLAC (lossless audio). Luckily.
@@kaitlyn__L The video bitrate is… don't know exactly, I think it's 1 to 2 gigabytes per minute, depending on the content. The audio bitrate is 24bit 48k PCM stereo, so 2.250kbit/sec if I used my calculator correctly.
I'd really appreciate it if you'd talk about surround sound; the specs, the standards, the encoding, and what it might take to record the audio and put it on a DVD.
