Why Higher Bit Depth and Sample Rates Matter in Music Production 

Fantastic breakdown. I went through a 192Khz phase about 15 years ago and suffice it to say.... lack of hard drive space and the computing power of the day cured me of that pretty quickly. 🤣

That video you mentioned last time absolutely blew my mind. I didn't have a clue that the aliasing around the Nyquist frequency issue was a thing at all. I had the feeling that higher sample rates were better for basic audio clarity, in the same way that a higher bit-depth helps with dynamic range. I just had no idea how or why.

Love it, I worked for Sony Broadcast including professional audio products, my team worked in Abbey road and the like, this take me back to those days when the analogue and digital battle lines were being drawn, I've always maintained digital offers a better sustainable quality, for the reasons you outline. Keep it up

Yep, I think that it's quite analogous to photography, where 8-bit colour channels work "pretty well" for printed output, but really fall apart for original scene capture, and just get worse when any kind DSP is applied to the "signal", where 'banding' shows up in stretched tones, and softened edges can get banding or artifacts introduced during processing... Great discussion.

I find a higher sample rate most useful when stretching audio tracks is necessary. Especially on drums to avoid “stretch marks.” But it's enough to bounce up from 48 (my standard) to e.g. 96 and bounce back when done.

I like high sample rates and I cannot lie. You other engineers cannot deny....

For me 24 bit, 48khz digital recorder with analog desk and outboard gives all I need. You get the balance of pushing the levels through the analog to create the excitement and keeping lower digital levels to capture it with plenty of headroom.

A fun fact: The exact same reasoning is used in professional video cameras. The Arri Alexa 35 - a camera often used in movie making - has a whopping 17 stops of dynamic range. So even if a scene is way under exposed or over exposed, the problems can be corrected in post-processing.

Thank you for showing that 24 bit is not necessary for audio playback however for audio production that makes a big difference in terms of the amount of buffer between clipping and not clipping the input audio that is being produced.

I always thought about the sample rate problem being that if you wanted to slow down a piece of audio with the highest frequencies being 20kHz, you'd lose information proportional to the amount you slow it down. So you need the extra magical inaudible information beyond 20kHz for the slowed down audio to still fill the upper end of the audible spectrum. That is something every producer will have probably experienced.

Something to add about bit depth and floating point for audio processing is the phenomena of rounding/truncation and accumulated error. If you are processing with 16 or 24-bit integers then every time you do a math operation, you are truncating to that length. Now that doesn't sound bad on the surface, particularly for 24-bit, what would the data below 144dB matter? The problem is that the error in the result accumulates with repeated operations. So while just the least significant bit might be wrong at first, it can creep up and up as more and more math is done and could potentially become audible. It is a kind of quantization error. The solution is to use floating point math, since it maintains a fixed precision over a large range. Thus errors are much slower to accumulate and the results more accurate. So it ends up being important not only for things like avoiding clipping, but also to avoid potentially audible quantization errors if lots of processing is happening. In theory with enough operations, you could still run in to quantization error with 32-bit floating point, since it only has 24 bits of precision, though I'm not aware of it being an issue in practice. However plenty of modern DAWs and plugins like to operate in 64-bit floating point which has such a ridiculous amount of precisions (from an audio standpoint) that you would never have any error wind up in the final product, even at 24-bit.

I've slowly been learning the benefits of oversampling for the last few years and before final mix export ill spend an hour or so applying oversampling on every plug in that offers the option on every mixer track. This video really solidified my knowledge and affirmed that me spending that extra time has always been worth it! The final mixes and masters do sound fucking cleaner by the end of it all because I do use a lot of compressions and saturation on most things.

Would be cool to see the importance of audio resolution in resampling!

Thanks for clarifying these things! Really useful for deciding on project bitrates and sample rates. Cheers!

Thumbs up for the Dan Worrall link. His, and Audio Universities videos are the top vids on YT to watch.

Both this and the previous video are great! Thanks for the great work!

This is an excellent and very clear explanation. Thank you so much! I've seen Dan Worrall's videos on this topic, and I agree they are also brilliant.

Awesome stuff. Keep 'em coming.

Same reason why graphic designers need high res and high bit depth. A 1080p jpg image is great for viewing, but will look terrible once you zoom or change the brightness. If your final image is composed of other images, they better be at a good resolution, or they'll look pixelated

This stuff is pure gold. Thank you so much.

It's like Ansel Adams 'zone system' for audio. Adams would prefog his film with light, then over expose the film in camera, while under exposing the film in chemistry, so as to get rid of the noise floor (full blacks) and get rid of the digital clipping (full whites), both of which he maintained "contained no information". This resulted in very pleasing photographs.

Great video, Kyle.

Thank you for all your great videos and subscribed.

Good stuff to know. Thank you.

!All points on point!

I guess having a high sample rate for when you need to e.g. slow recordings down is also useful because you still have that data. And to me that's pretty much the only reason to have things above 44kHz sampling rate

I'm surprised he didn't mention how higher sample rates decrease latency when live monitoring. PS. I would love to see videos about the future of DANTE AV and Midi 2.0.

More samples are a great thing for denoising as well. Temporal Denoising is quite a resource intensive task, but it works wonders in recodings of any type. Especially if you want to get rid of higher frequency noise.

Gracias amigo, saludos desde México 🇲🇽❤

great video!

It is good to see you back in action with your awesome videos!

you have done it again. i would love to see a video on square waves

Don't ignore clipping. Or it will sound like Golden hour by JVKE.

Compression you apply before any gain or od/ds is brought into the signalpath. It only might be applied again when mastering for different formats.

Got so excited when I saw you were using reaper

WRT noise floor and compression, when working with analog tape, it was (and I presume still is) much more common to compress and EQ on the way in to avoid adding noise by doing it later.

Modern converters work as 1bit sigma-delta anyway and convert the data stream after the fact, using digital filters with a dithering noise beyond the audible range.

Clear explanations

Excellent video Kyle. Sometimes I miss the analog tape days, till it comes to signal to noise. At least tape saturation sounds much better than digital clipping, which I'm sure nobody goes that hot. 🙂

Here are two other reasons to go with 96k: The ad/da latency of your system will be much smaller, and (if for some reason) you record to a file played back in the wrong sample rate you will notice it right away 😁

That was really great man! I didn't know this before! I was just using standard because I didn't know what would it change. But now I understand it! 🤘

I already say this but i say again, most plugin work better at high sample rates since most plugins doesn't have internal oversampling, so it's good to work at reasonably high sample rate like 96k or 192kHz, though i say this but im still working at 44-48kHz 😂

"...for any properly mastered recording." I long for properly mastered recordings. A thing of myth and beauty. Like a unicorn.

Succinctly, while human hearing has an upper frequency bound, targeting that limit when converting from analog to digital can (and usually does) result in literally corrupted digital representation because the contribution of the higher analog frequencies to the waveform don't just disappear, they get aliased into the lower frequencies.

Rather than for musical purposes, I think it is valuable as data for profiling or natural phenomenon analysis in the future.

Really nice stuff. But I disagree with the Nyquist alignment. You can defend it if you know the input, but if its misshapen like music, then you cant defend Nyquist. Misshapen frequency, volume and variance needs to be taken into account and you need mpre than 2x the frequency for that, as well the bit depth. Not to mention intentional saturation. Keep it up, I'm eager to watch the next vid!

AFAIK this method is built in to every audio interface nowadays. So obviously a sampling resolution higher than 44.1 is useful, but in principle you shouldn't record audio files at 96kHz or higher, because they just take up a lot of hard disk space and need more cpu power to play them, especially if you have a lot of tracks... but you don't gain quality.

almost nailed it although what you said about the pressure on anti-aliasing analog filter is only true with very basic converter topologies. So if you've only atanded mixed signal electronic 101 that's what you have seen. However we don't use that topology anymore for audio for several decades now and mostly for this exact reason. The "true" (i.e. : in terms of what is seen by the analog side) sampling frequency is several MHz.

well, the comments around noise floor are a bit misleading. a 24 bit signal doesn't have 144 db noise floor, that would be nice, as this depends on the noise floor of the conversion. 144 ( 6db per bit, ) is theory only.

The demonstration about analog audio gain and noise floor is exatcly how cameras work aswell. I'm actually shocked by how similar they are. Capturing images with a camera is a constant battle between distortion (clipping the highs) and the image being too dark (blending in with the noise floor) and bringing it up in post then causes the noise to come up aswell.

I'm a bit rusty on this, but there is an issue with the Nyquist frequency. Going from analog to digital. You want to "brick wall" band pass the signal at half the sampling frequency. Brick wall is a perfect low pass filter, which doesn't exit. There are very good low pass filters. Going from digital to analog, you again want to brick wall filter the signal to recover the analog signal from the sampled signal. Even more confusing, there are digital low pass filters, but they have to obey Nyquist as well.

I record 24bit because of the noise floor. But I record on 96KHz because of the round trip time! My system actually has a lower latency when it's set to 24bit 96KHz.

Waiting this stuff...

That was super clear. You’re a great instructor. Is it useless to record in 96k and then bounce stems down to 48k to give my logic session a break?

I use 192kHz multi tracking then master to DSD for amazing replay via a decent DAC

Love this, always great content. Where have you studied?

I wish there was a higher sample rate option for highmid to higher frequencies that keeps a 48hz sample rate on the lowmid-low frequencies but targets a higher sample rate for the rest.

Perhaps an Idea to consider (and make a video) that compares DSD to PCM and the differences between PURE DSD recording mastering output and the ones that use PCM in between ... Nevertheless, DSD128 or DSD256. PCM 24/96 vs DSD128 ... Is it really that close ? Or is there some "hidden difference" ;-) ...

Is there any advantage to upsampling when applying parametric EQ, crossfeed, filters, volume leveling etc? Also do some DACs work better with higher sample rates if you are able to offload the conversion in the digital domain in a pc? I am a roon user and curious your take on this.

🔥🔥🔥

say whatever you want to ...believe whatever you want to .... the difference between DSD recordings and Lossless wav or flac on my Fiio DAP is NIGHT vs DAY Its really about the recording just as much as the file type / resolution

5:55 that is cool

sample rate does more than help with anti-aliasing. rupert neve was convinced that capturing and processing the ultrasonic signal that came with the audible actually contributed to the perceived pleasantness of the sound, and the emotional state it communicates. so, even if you can't hear it, per se, it counts in the overall timbre and feel - you can easily argue that, in the analog domain, ultrasonic signal - for instance harmonics - actually changed the behavior of compressors, to say the least - and that, multiplied by x number of tracks. so higher sample rates also allow for a wider bandwidth into the ultrasonics, which seems to matter for the quality of the signal. the downside is the processing power, and storage space.

Word of the day: Nyquist 😉💥

Finally I have strong arguments to argue with audiophiles! 😅

Weird how limiting was not discussed. It's one of the applications of high sample rates that actually make actual sense in practice most of the time. At least in a mastering context that is. The sample peak level has a higher chance to match the intersample peak level (true peak) when higher sample rates are used even if high sample rates have no effect on the d/a. That's the main working principle behind true peak limiters.

Yep, frequency and volume range is enough. But what about resolution? In 16 bits 48 kHz signal is just so many data paints, which will wipe any difference between very close, but slightly different signals. For example, digitizing short enough 15 kHz and 15.001 kHz sine signals would result in the same binary file. Moreover, DAC is not looking at the whole file, only at a short part of it, meaning that we will likely have frequencies changing over time. Compare this it to image sensors or displays. Having 1 inch HDR sensor gives enough size and depth. But we still want it to be 4K or 8K.

Here is the right level to render to for audio uploaded to RU-vid: The ideal volume limit level is -5.9 Db. (RU-vid automatically normalizes volume to that level) All instruments should be below this level with the peak spikes reaching -5.9 Db. Just put all instruments at around -18 Db and then increase accordingly between -18 and -5.9 Db.

it's sort of the same with visual production with pictures and video files. The average joe posts jpegs in 8bits and maybe a png with alpha channel every sunday. But in production we use 32bits EXRs everywhere because you can play with high dynamic range in comp and it's fast it can store layers and metadata you haven't even heard about and deep data and ....

02:02 it worths noticing that it is not because you record in 24 bits audio that you have 24 bits of dynamics : hardwares have noise level as well. But sure, digital intefrace are still way lower in noise than analog.

I heared Spotify is rumored to offer higher fidelity audio, probably with less compression or lossless audio using codecs like Flak instead of mp3. My audio equipment probably isn't good enough to hear the difference though, but maybe it will be good for music producers.

Could you do a video on dsd, what it is, pros, cons etc

Another reason to use higher sample rates is it decreases latency

One thing often overlooked in the sample rate argument is digital mixers. The converters are often run in low-latency (high speed) mode in order to keep the round trip through the console low enough that it doesn't affect people's performances. This is done by simplifying the digital anti-aliasing filters to reduce processing time. This is not trivial stuff, I'm talking on the order of 40dB attenuation at .6fs vs 100dB. In other words, if your console runs at 48KHz, an input of 28.8K at full scale will come out the other side of your console as 19.2K at -40dB. That's enough to cause some issues, especially since a lot of manufacturers trying to meet a price point completely leave out the analogue anti-aliasing filters (Sony suggests 5-pole analogue filters in front of 48K ADCs). Running a digital console at 96KHz effectively means around 90dB stop-band attenuation even with the ADCs in low-latency mode. Of course, you also reduce aliasing caused by internal processing as you say.

👏🏽👏🏽👏🏽 👏🏽

Dan Worrall entered the chat!

Aliasing definitely isn't a problem unique to digital audio recording, I'm a geophysicist and a geophysical data processor aliasing is also an issue in geophysical data in exactly the same way except it ends up being a visual issue.

I'm watching all the videos on the channel, thank you for sharing your knowledge with us. One question: what is the setup of the sound equipment installed in your car? Is it a High-End system? I'm curious to know what system (equipment) you use in your car...

Real music is not single static sine waves but a whole spectrum that varies with time. I would like to see this mathematical argument extended to spectra, because the error on each frequency component would surely accumulate? Real music is very very processed, being encoded and decoded multiple times from various streaming services and codecs, so I think adding a bit of headroom in terms of frequency and bit depth is quite sensible to keep the artefacts down.

Nice to see that this stuff is understood properly by the younger engineers that didnt live through the evolution of analogue and digital recording. So much nonsense spoken about hgh bit rates. Well done.

Excellent Vídeo!!!! One question I have is if the higher sample rates would help to draw a more complex wave , like a mastered song, with many instruments playing at the same time, more accurately. I'm supposing that a "complex" wave may be a combination of various frequencies and even sine and not perfectly sine wavez. May be worth testing.

I understand about oversampling and why it's used internally. However, sometimes I think of potential reasons to *record* at higher sample rates - but I'm no expert and I wonder whether this is ever justified. Two such reasons I can think of right now: 1. Field recordings that you might want to slow down later on to half or quarter speed. 2. Recordings made in adverse conditions that might need noise reduction processing (I've heard some people say that higher sample rates can help with NR quality). Do you have any comments on either of these? I'd be interested to hear your advice. Thank you!

96 is the sweet spot, think of sample rates as the display quality on your monitor, 1080p is going to look worse than 4k because it has less pixels, it’s the same thing in music, more samples equals more detail.

This is why the USMC Sony and others need to make DSD more available and not guard it so much. It's a lot easier to work with when the editing program supports it. Almost never have to worry about noise floor and you can do almost all processing on a core duo with ease.

Could have just linked the fabfilter video from the start

So, wrt sample rates (5:01), when a more gentle low pass filter to Nyquist of 48k is surely within the realms of DSP and CPU now, it begs the question: why do anti-aliasing filters remain at 24kHz for processing at 96k, or when over-sampled 2x-4x?

Great video. In audio production, another beneficial effect of using higher sample rates, apart from getting rid of aliasing, is that doubling the sample rate cuts latency in half...

Hey, what microphone are you using in this video? It's really nice

Higher sample rates reproduce higher frequency. There is no more info in the audible range. The clue is in the file size, double the frequency double the size. More bits is lower noise floor which most dacs cant reproduce out the audio port. And yet it sometimes sounds better to me 🤷‍♂️

Dolby S was introduced in 1989.

If your signal to noise ratio is below 96dB (including not only mic and preamp but also room), recording with 24 Bits only makes sense for the manufacturer. ;-) Unless you like to record 8 Bits of noise...

🙏👏

So in purely playback scenarios, is it recommended to oversample above 44.1....for example my streamer allows me to oversample thru its USB output to my DAC. Does it make sense to oversample to 88.2 or higher in order to get the smoother roll off above nyquist?

Question: Is the 20bit HDCD mastering any good for playback? Is it recognizable (compared to normal CD), given a good enough audio system? Thanks for answering.

Bit depth matters, assuming you are going to change the dynamic range of what is on the file by a lot. Sample rate does not (assuming you only care about audible frequencies!)… if you stretch a 20 KHz sinewave, and make it a 19 KHz sinewave, the application doing this re-sampling is not taking the original samples and moving them, it is interpolating a position between the original samples and synthesising a new sample, it will be as good as the algorithm the software uses - the sample rate of the source (44.1/48/96etc) is irrelevant, if the software is good, it will do a good job, if the software is poor, it will do a poor job. Luckily for us in 2023, this is a very solved problem and things like Reaper which still has a re-sampling mode on export, default to very good implementations for sample rate conversion whereas in the olden days, where the original Pentium processor was crazy expensive, it would take forever to export whilst re sampling. Any re-sampling algorithm, that is used today, does not simply draw a straight line between two samples and put a new dot the appropriate proportion along the line, The wave form represented by the original samples is effectively constructed, and the sample that is synthesised is placed on the reconstructed wave form, which is mathematically very precise relative to the original samples. This accuracy does not get better at high sample rates, these samples are temporal, not amplitudinal (ie - the inaccuracy is in the bit depth, not jitter in when the sample was made - unless the ADC was bad - in which case it’s bad at any rate!!) For those that are thinking about aliasing, again, the quality of the software you are using is far more profound than the sample rate you select, for example, a good piece of software may put a low resonance, brick wall filter at about 21 kHz to filter away higher frequencies, so they don’t cause aliasing. If your software does this, and many do, there is a good chance that the software developer has thought things through carefully. If you are dependent on sample rate to minimise aliasing, then there is a good chance that your software of choice has problems in many areas!

So what's your Opinion On Ableton Live 11

Can we talk about should EBU R128 and LUFS apply on RU-vid platform?

1:44 you are missing the compander part here.

For audio 24 bit 88.2 kHz is optimal. For video 96 kHz.

Naive question: listening to a band play live, don't the different instruments produce frequencies above the audible range of 20 kHz, which we cannot hear, but that interfere with similar frequencies from other instruments thus generating beat frequencies that we do hear? Not saying that happens, just asking. Now, if that would be the case, wouldn't we lose those beating sounds by recording instruments in separate tracks at let's say 44 kHz?