@MF Nickster I don’t know about other Daws, but reaper lets you move audio clips on a sub sample level

hah. I have an old MOTU Ultralite mk3 kicking around the advertised 192Khz and in the day it was the new hotness I was just like O.O at ever wanting to record that

Holy moly you’re brave. About a year ago I started mostly working at 88.2/96. I’ve been blown away by how far computing power has come as performance is super smooth these days and the results are quality but man a decade+ later I still feel you on that storage battle. Can’t even imagine what you went through. Whenever I break and print my hardware inserts in the session it’s basically a “lol -12gb” button for my C drive

96khz is all you need even if you are a believer in this stuff. Not a fan of 192

It's interesting. Shot a concert recently with ≈ 10 cameras and multicammed it in the NLE. We used Sony Venice across the board and shot X-OCN ST. That's about 5GB per minute. We recorded ≈ 100 channels of 24/48 on the audio side. If we had recorded @192Khz we would have still only wound up with around 45% less data than the video. From a purely data storage and bitrate standpoint PCM audio is not really all that terrible. Sony X-OCN is pretty efficient actually. ProRes is a real hog on the other hand but notably unlike PCM neither X-OCN nor ProRes are "uncompressed". Undoubtedly the resource hog for audio is plugins because unlike video where VFX and color correction tend to be tertiary operations (i.e. entirely non-real time and done by someone else), audio engineers are typically focusing on editing and finishing within the same software environment and in real time.

Thanks, Colin! Sounds like you’ve worked on some awesome projects!

There is no "battle"going on, Dude. BTW, did you work on the analog or digital side of (Abbey) Road?

@@frankfarago2825 I said battles lines, I did not say there was a battle, I worked for Sony broadcast in the time when digital recording equipment like the PCM 3324 was being introduced, and remember conversations with engineers where they preferred analogue recorders, because they could get a better sound by altering it like bias levels, which to me always felt they were distorting the original recordings. I ran a team of engineers who installed, maintained and supported (sometimes during recording session (sometimes overnight)) these products at a time when the industry was starting to embrace the technology.

I remember this time when digital audio wasn't quite in the hands of the consumer yet and a guy who's name escapes me from Sony's "Digital Audio Division" as he put it brought a digital reel to reel deck into the studio of a radio station in San Francisco and played the theme to Star Trek Motion Picture/ Wrath of Kahn. It was awesome but the station was not properly set up for it and there was heavy audible clipping. They stopped and came back to it later and while the clipping was gone the solution just sucked all the life out of the recording. I wish I remembered the guy's name. I think it was Richard something.

But how many albums out there have been recorded to tape ? most all of them ! how many digital albums have i heard ? squat and if i have ? it was early adat !

There's a fun way to show off aliasing, because it actually results in NEGATIVE frequency. Take a circle and put a single dot on it, let's say at top dead center. Now, spin it at 250 turns per second, but take a picture 500 times per second. You'll see the dot jump between the top and the bottom of the circle. Increase this to 450 rotations per second. You'll see the dot move first 225/500 of the way around the wheel, or a bit less than half, then in the next frame it's gone 450/500 around, and so forth. Clearly, it's rotating clockwise. Now increase this to 550 rotations per second. In the first frame, these both start at 0. The second, your 450 RPM circle goes 225/500 forward, but your 550 RPM circle shows up with the dot at a position of 275/500 rotations forward. The third frame shows your first circle reaching 450/500 forward, while the second circle is at 50/500 forward. Yes, that's right: the second one just moved backwards from +275 to +50. In fact, that first step moved backwards from +500 to +275; or we could say the 450 RPM circle turned +225, then +450, while the 550 RPM circle turned -225, then -450. It's spinning at 450RPM BACKWARDS! This backwards spinning is just a sine wave shifted τ/2 out of phase, which still sounds the same. (Yes that's what a sine wave is, it's a circle.) No, you can't just intuit that it must be 550Hz because it's going backwards; that would be like silencing the first 1/1100 of a recording and suddenly it's at double the pitch.

@@johnmoser3594 That's an excellent way to visualize it. It's kinda like how a camera's shutter effects can make a helicopter's blades look like they're standing still or going backwards very slowly.

It’s all you need. Your DAW might process audio as a 32bit float but your ADC is more than likely capturing 24bit. 48k gives a touch more high frequencies before nyquist sets it without essentially doubling file size.

That's really interesting. So why is everything on my TV so dark and hard to see!!!!!!

​@@jackroutledge352 Maybe modern filmmakers think underexposed means "gritty" and "realistic". Lol.

​@jackroutledge352 perhaps your TV doesn't have a good dynamic range.

​@@jackroutledge352yeah that sounds like an issue with your TV quality. Or maybe your settings are not optimized? I have a 4K OLED Sony TV and it has HDR. Looks gorgeous with the right material.

@@RealHomeRecording It's a well-known problem/phenomenon. Lot's of people are complaining about the darker TV/movie-productions these days. It is much darker now. I also have a 4k Oled TV (LG) but I can also see that some scenes are very dark in production.

Here's the simpler way to get the same effect. Check the settings for your time stretch algorithm. The default is usually the highest pitch accuracy. What increasing the project sample rate does is decrease pitch accuracy in favor of time accuracy. The alternative way of doing this is to set the time stretching algorithm to a lower pitch accuracy.

​@@simongunkel7457 Are you using REAPER? There is this setting to preserve lowest formants - is that what you mean?

@@PippPriss Sorry for the late reply, didn't get a notification for some reason. REAPER has multiple time stretch engines and for this particular application switching from Elastique Pro to Elastique Efficient is the way to go. You can more directly change the window size on "simple windowed", though Reaper actually goes with a time based setting, rather than a number of samples. Also note that stretch markers can be set to pitch-optimized, transient-optimized and balanced..

Haha, “Stretch Marks” never heard that before. I’m going to say that instead of ‘artifacting’ from now on

How exactly does upsampling from 48k make the stretching more transparent? Since it's not adding any new data, I'd imagine it would do nothing.

Rounding and truncation should not be a problem as long as the levels are not wildly out of range, either way too high (digital clipping) or way to low (down in the noise floor). The latter would be almost impossible, since recording at 144 dB below digital full scale would be obviously ridiculous -- even room sound and microphone preamplifier noise should be quite a ways above this level. However, there is one thing that needs to be watched, and that is proper dithering. Vanderkooy and Lipschitz did pioneering work on dither, and they recommend that triangular probability density dither should always be applied at 1/2 least significant bit whenever audio is sampled, or resampled (sample rate converted or gain shifted down, where the dither might be reduced below the current bit depth). Vanderkooy and Lipschitz did say that dither might not be necessary when working with more than 24 bits, especially if the master is going to be converted to 16 bits for distribution. It can be dithered for 16 bits when transcribing to CD Audio or whatever. The dither provides a digital noise floor that spreads the quantization error power spectrum across the entire audio band, effectively making the resolution greater than the dynamic range implied by the actual bit depth. This is the white paper, from AES, not free: aes2.org/publications/elibrary-page/?id=5482

Lol...itty bitty waist....round thing...face..😂🤘🎶

When the engineer walks in with some RME and a Large Nyquist in your face, you get sprung!

​@@Mix3dbyMark😂😂😂

FWIW I'm running 32 bit float and 48k on my DAW. That's my max bit depth with Studio One 6.1 Artist, it goes to 64 bit float in Pro. As for sample rates, it looks like S1 goes up to 768K. Good enough?

@@Mix3dbyMark nice!

It's not about resolution it's about compression. Not entering int eh details but actually how much you compress your JPEG and the trade off between image quality and file size is exactly what is discussed here : matter of sampling rate.

As an amateur Photo(GIMP)-shopper, I figured this out a few years ago. Always start with a higher resolution than you think you'll need. It's easy to reduce the final product but a pain to go back and redo it with higher resolution.

@@lolaa2200 Ehh no, audio sampling rate is directly analogous to image resolution. We're not talking about image compression nor audio compression here.

@@MatthijsvanDuin I reacted to a message talking about JPEG. The principle of JPEG compression is precisely to give a different sample rate to different part of the image so yes it totally relate. JPEG compression IS a sampling matter.

Yes please

I love Dan’s videos! Thank you for the kind words, Sven!

yeah if its essential for your workflow to slow some audio down then yes by all means. otherwise im happy with 48 or 44.1 because it sounds good. I like to export any files before mastering as 32 bit files tho, saves you issues from downsampling (as most DAWs run a 32 bit master fader now)

Great point, Zadar Leader!

Not something I think is neccessary, unless you specifically want ultrasonic content and make bat sounds audible. Now if you think time stretching sound better with a higher sample rate, you might be right, but you are using the most computationally expensive hack I can think of. Time stretching and pitch shifting algorithms use windows of a particular size (e.g. 2048 samples). But their effect depends on how long those windows are in time. So a higher sample rate would just decrease the time window. All of these effects make a trade-off though: The longer the window, the more accurate they get in the frequency domain, but the shorter the window the more accurate they get in the time domain. Most of them default to maximal window size and thus maximal accuracy in the frequency domain, but the errors in the time domain lead to some artefacts. So instead of increasing your project sample rate, which will make all processing more costly in computation, you could just opt for the second higherst frequency domain setting for your pitch shifting or time stretching algorithm. Which means window size is decreased, which actually reduces computational load for pitch shifting or time stretching.

@@simongunkel7457 I think he means the simpler resampling version where things get pitched down when playing slower, then the higher sample rate still has content to fill in the top of the spectrum when pitched down.

@@5ilver42 That's the ultrasonic content he referred to, wanting it to be captured so when you lower the rate it drops into the audible range.

@@5ilver42 It depends on the application, but if you're just slowing down for effect you actually want the top of the spectrum to remain vacant rather than shifting (previously inaudible) ultrasonic sounds into the audible part of the spectrum. Obviously if you want to record bat sounds you need to use an appropriate sample rate for that application, regardless of how you intend to subsequently process that record.

Who would I do this process in photoshop

I guess because the digital buffers fill up sooner?

Good point, Zyon Baxter! It’s a balance in practice though, as it’s more processor intensive so using a higher sample rate might lead to needing a larger buffer size. If anyone reading this is interested in learning more about this, check out this video: kzfaq.info/get/bejne/sOB9Z9ycmK-cpJc.html

Actually with a given computing power, and assuming you make full use of it, higher sample rate mean higher latency.

@@lolaa2200 lower latency, even within the ADC/DAC, feedback loop is reduced

I think this is kind of a myth in all honesty. In every other way, doubling samplerate means doubling buffer sizes. You have a delay effect? You'll need twice as many samples in the buffer at 96k. Same for the playback buffer: if you double the samplerate, but keep the same number of samples in the buffer, you've actually halved the buffer size.

Actually almost all plugins these days have internal oversampling.

My DAW (Reaper) has external oversampling per plugin or plugin chain, which means it takes care of the upsampling and after processing the filtering and downsampling. To the plugin it looks like the project runs at a higher sample rate, while for plugins where aliasing isn't an issue it can still run at the lower sample rate.

most plugins like 10 years ago had oversampling allready built in them.

No. It’s not useless. You can even bounce out the multitrack instead of combining sections into stems.

That’s an interesting question, would be curious too, but I assume you’d be able to hear the difference, what you think?

Check out this video: Digital Show & Tell ("Monty" Montgomery @ xiph.org) kzfaq.info/get/bejne/i9eZda2Tt6u5l4k.html Monty runs a square wave through the system and illustrates something called the Gibbs Effect. Although, the frequencies that make a triangle wave or square wave perfectly triangular or perfectly square exceed 20 kHz. So the sound should be the same theoretically!

​@@BenCaesar Above around 5k, no audible difference between a sine and square wave tones. Try it yourself!

I believe KZfaq videos have an audio Bitrate of 128kbps.

The two reasons you listed are indeed valid points. Pitch correction or pitch manipulation would be another.

@@RealHomeRecording Many thanks, that's helpful!

did you test it in an ABX-setting? Otherwise your perception just might have fooled you! 😀 #beenthere

@@paulhamacher773 a lot of the time its difficult to find a higher res version of the same mastering

@@crapmalls ...then just record your own stuff at different settings and let somebody else play it for you without telling. also, use more than just a few examples. you will see, you got fooled. there isn't more info, you can't hear above 20kHz.

@@mb2776 yeah thats what i mean. I know theres literally no difference because the higher sample rate just goes into higher frequencies. The file size is the giveaway. Apparently it can help with timing in the dac but thats an oversampling issue and a dac issue IF the dac is even good enough for it to matter

I just use the stock system, but I’d love to upgrade someday! Thanks.

The way modern converters work just amplifies his point: the higher the sampling rate, the easier the filtering is.

Glad to help, 3L3V3N DRUMS! I still use 48kHz most of the time because the processing power and storage I save outweigh the tiny bit of aliasing that might occur. (In my opinion)

@@AudioUniversity Great to know. That's actual the standard in Ardour while I'm recording my drums. So I'll let it like that!

It will require 50% more storage capacity, so be sure to factor that in on your projects.

@@AudioUniversity Where would it occur though? Your converter on the hardware side always uses the maximum sample rate it can support, because that makes the analog filter design much easier. Then if you record at lower sampling rates it will apply a digital filter and then downsample - both are hardware accelerated DSP that get controled via the driver. If you set to record at 48k, your converters don't switch to a different filter design and a physically different sample rate, they just start to perform filtering and downsampling before sending the digital signal to the box.

@MF Nickster I agree.

Sadly, most are like flat earthers. They just won't listen.

Yes.

@@AudioUniversity I make predominantly reggae music and for some reason in I normally create music in Ableton now mixing in Luna at 24 has given me the desired sound I was looking for. All out API and studer Tape.

It's risky to record frequencies above 20KHz, even when the original sample rate is above 88.2khz. Ultrasonic frequencies in this band are susceptible to being digitally folded down into the audio range, producing extremely unnatural-sounding aliasing distortion. While this hazard can be carefully avoided within a pure 96Khz+ digital processing chain, any side trip to an external digital processor may involve resampling that can run afoul of ultrasonic frequencies. Why take such risks when the speculative benefits have never been shown to be audible?

In the 16-bit days keeping the digital noise floor below analog meant going in hot and thus risking clipping. With 24 bits, you gain the headroom and your analog noise floor will be louder than the quantization noise. So it a non-issue these days, but only because we moved to recording at 24 bits, where the bottleneck becomes the analog chain in front of the ADC.

@@simongunkel7457 Do you have the order of magnitude? I feel like 90dB of SNR is already extremely good for your whole analog chain.

@@jonasdaverio9369 If you wanted to make use of the 96dB provided by 16 bits, you'd have no headroom. I tend to be cautious and leave at least 12dB between the loudest peak I got during soundcheck. There are plenty of mics that can beat 90dB, and dynamic mics don't generate noise on their own, so you only get the preamp noise. 100dB SNR isn't that uncommon even at quite low price points and I just measured 90dB on an old Behringer interface I have lying around.

@@simongunkel7457 Thanks for the details!

You get an increase in noise as you add effects and tracks in the digital domain, thus it's not just capture, but also editing that needs a lower noise floor. Even just adjusting gain in the digital domain adds noise.

@mfnickster The issue isn't processing power so much as ADCs MUST have group delay in order to have linear phase anti-aliasing. DACs must also have group delay for the reconstruction filters. The processing power within the console's DSP is fast, but nothing is instantaneous, so every place once can reduce the latency must be considered. Oversampling also requires group delay, so pick your poison. In a computer environment, the plug-in can report it's internal latency so the DAW can compensate by pre-reading the track, not so in a mixer.

The aliasing occurs at the conversion, so it would probably be too late at that point. The good news is that your converters already have anti-aliasing filters built in! If you mean using low pass filters within the DAW signal chain, I think Dan Worrall does a similar thing in the linked video. Check it out!

@@AudioUniversity Hmmm ok I got you. But how steep is that filter on my converters? Is it steep enough to give 48K that anti aliased sound...? Perhaps... it is better to run the higher sample rates during recording to give the audio the best chance at avoiding aliasing... Then perhaps using some high end SRC to get the 48... But SRC is a problem too Have seen Dan's video... I watch it a few times a month to refresh 😂

When it is already digital, it is too late to run the AA filter. So, not a good idea. Unless, say, you sample at 192 kHz, then run the steep digital low-pass filter suitable to be the AA filter for your 48 kHz recording, and then instantly after that downsample it to 48 kHz before saving (or to 44.1 or 96 kHz, whatever you wanted, but 44.1 kHz needs a bit more intelligent downsample here to work perfectly, but any decent software and any non-ancient hardware will easily handle even that in real time with minimal latency). “Best of both worlds.” Well, kind of. The ultrasound is lost... which hurts if the application was not just for music.

@@L2M2K2 "But how steep is that filter on my converters?" pretty steep, depends on the spec of your audio interface. also it is not steep enough to give 48k an anti aliased sound, whatever the hell that means, it is steep enough to prevent aliasing due to blocking frequencies above the niquist frequency which is half the sample frequency. and it's probably more than one filter, you can cascede them together to get a steeper filter. you have to keep in mind, just cause u can't hear super high frequencys doesn't mean they aren't there.

@@mb2776 The main problem with sampling directly at 48 kHz (or 44.1 kHz) is that it requires a fairly extreme analogue filter. This causes all kinds of phase shifts and what not, which can be (marginally) audible. A cascade of low-order low-pass filters will sum up their phase problems like attenuation before end of the pass band, and as a higher-order analogue filter will have its own set of quirks; a game with no perfect “winning move” at least outside the digital realm (where “perfection” can be obtained by introducing a small amount of latency for a non-causal filter with perfect phase response, or where the higher-order causal filters will also not have any analogue quirks). The scheme I suggest is by the way nothing new... The very first Philips CD player used such oversampling on its output (to create the correct, smooth output waveform from the CD through its DAC, a problem pretty much identical to the “to sample without aliasing”). It oversampled at 176.6 kHz (which is a “4×” oversample as back then it would have been hard to have a non-integer oversample in real time). Consequently, it sounded better than the Sony's first CD which did not and rather used a very steep analogue filter with all its problems, or as they call them a “brickwall filter” ... it sounded almost like a good modern CD player. The second added benefit for Philips was that they got the full 16-bit noise floor with a 14-bit DAC available at the time. I would assume that most current DAC also internally oversample even if it is no longer needed to get the noise floor to CD levels... it just works better with the waveform generation overall. Not at all sure which ADCs employ such a scheme is internally in hardware... there is probably less demand for that now that we can sample at 192 kHz even with cheaper equipment, but you never know if the older generation that could not output such a high sample rate actually produced better output at the low sample rates because of internal oversampling and optimised internal downsampling... With cheap modern 192 kHz equipment, I have not dared not to sample at anything but the nominal (maximum) sample rate and then downsample myself digitally after the fact when appropriate knowing what I will get.

Thanks, Professor O. I studied audio production at Webster University. I’ve also learned a lot from mentors, of course!

There is only one solution in those cases too. So long as it’s below the Nyquist frequency. If the fastest “wiggle” that matters is the 20 kHz, it and all lower frequencies will be accurately reconstructed as shown in the video. Did you watch the last part about square waves in Monty’s full video?

Just a detail: no speaker can reproduce above 20khz, the square wave is composed of many sines above the capability of the speaker everything gets lost either way, and all amps have low pass filters (also around 20khz) to not destroy speakers. If I remember well a d/a converter also has a 20khz-22khz low pass filter. The real limitation in frequency are our ears.

Look up Fast Fourier Transform Tutorial by Karl Sims. It lets you model an arbitrary waveform live and shows all the sine/cos components that produce it. I notice you get ringing if you have sharp edges, just as you would in analog when the bandwidth is limited.

@@Miisamm Indeed a DAC will have a low-pass filter and this is key to deducing the correct "wiggle" as @davidcooper8241 pondered. Any way for the signal to wiggle _aside_ from the desired one will contain >=Nyquist frequencies, so when you pass the signal through the low-pass filter, getting rid of those overly high frequencies, the resulting signal is mathemagically the desired one.

kzfaq.info/get/bejne/hcWdmJZ_17axmGw.html (most visible on the back wheel of the carriage going right). That's aliasing on a film from 1903 and that's not digital.

@@simongunkel7457 aliasing in the realm of analog Audio processing

​@@eitantal726 Well BBD delays can alias and a classic piece of gear which can do this is the Moogerfooger MF-104M. But you could mod any BBD effect to do it - it's just the Moog has controls that allow you to go to aliasing mode without any further tinkering.

Aliasing is an issue whenever something is discretely sampled, which it why also applies to motion film (with each "sample" being a frame of video)

This sounds like a good method! Record at a higher sample rate to use a more gradual anti-aliasing filter. Then save processing resources in the mix by down sampling to 48kHz, only over sampling when necessary. I think the professionals have differing opinions on the best method, but I think your way makes sense!

@@AudioUniversity Thank you really much for the answer.

skip all those steps, just record at 44,1kHz 24bit. Why do you want to go that far, are u using high end converter, super acoustic treated room or is it just you and your bedroom production? the "pro's" just use whatever they feel like and whatever they are most familiar with. sometimes it's the latest best gear, sometimes old stuff. if u really think all that stuff matters in your production, go, do that extra mile. but, I guess if u have to ask this stuff at a random youtube video, just stop. get rid of plugin doctor or whatever the hell that is, also get rid of the meta plugin. the easy route is 44,1kHz cause that's enough and 24bit just to make it easier. Most daw allready are able to export mp3 or other lossless formats.

It's odd that so many people bring this up. It tells me that many systems are poorly designed and don't adjust the sample buffer size to the sample rate, e.g. they are a fixed number of samples rather than a fixed amount of time. Or people just don't know how to adjust the buffer size to reduce latency (at a cost of higher chance of dropouts).

Depends on the computer, setup, configuration, etc. I have seen latency go up as the computer 1. Uses a larger buffer for larger amounts of data 2. struggles to process all the data for basic mixing tasks which are now 2x to 24x more taxing 3.your drive (if recording) and signal/effects processing (live or recorded). Sure, technically the samples come in faster, but they are typically chunked the same to transfer to the computer, and after the buffer(s), the result is usually slightly more delayed. I personally have a lower bitrate setup for better latency when doing timing-sensitive recording where I don't expect much post-processing. It is used more often than you would expect.

@@nobodynoone2500 Eh, yeah it depends from computer to computer, as you've said, but luckily most computers nowadays are pretty well suited for data rates that 192 demands. I'm talking mostly about a simple setup anyway - just a few VSTs and nothing much more. As I've said - for things that need to be done in a jiffy, not something you'd have in a production environment necessarily.

You using Pyramix for this? If not, then what?

Increased bitdepth and sample rate address different issues. More bitdepth means lower noise. More sample rate means fewer issues with Nyquist aliasing.