24 bits or 96 kHz? Which makes most difference?

Which is more important, bit depth or sampling rate? What sampling rate is the best? What sampling rate is best for audiophiles? What sampling rate do I use? Featuring Audio Phil.

UPDATE
I made a comment about a Wikipedia page in this video. One of my viewers has since improved it.

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[spectrelayer]

I love your sense of humor. You have a very clear understanding of the tech & you methodically dismantle the BS that so many pseudo-audio-experts spew. I record in 48k/24bits/chan, operate filters in 8x over-sample mode to minimize aliasing induced distortion. Final mix down to shaped-dither 48 or 44.1 in 16bits.

[clouds]

48k/24bits I looked at the options and those were the numbers that made most sense to me. Everything sounds great, I have flexibility.

[frogandspanner]

As audiophiles can hear what no mortal who have participated in scientific experiments can hear, perhaps they could donate their ears and brains to scientific research; preferably _ante mortem_ .

[edcataldo]

I only like listening to the highest level of audiophile technical cheekiness and this channel is cracking! Another well done episode mate!

[JesterMasque]

Finally another fantastic video about this topic besides Dan Warroll’s. We don’t think about needing to hear 96, 000Hz frequencies when recording at 192k. What we are thinking about (in terms of auditory effect) is the blending of digital audio samples as they combine together in a virtual space. This “smoothing” effect becomes much more apparent on things like cymbals in drum overheads & instruments with longer envelopes that ring out.

When combined with digital (or virtual) signal processing like high shelves, you are able to use more EQ with less apparent artifacting and aliasing when compared to lower resolutions. Some plugins compensate for this by upsampling inside the plugin for processing, then downsampling on the way out, but not all of them do, and I haven’t heard of a DAW doing this with their console strip (please correct if I’m mistaken).

The argument listed here about CPU power becomes a bit more null & void as time goes on. 192kHz is also much better for any kind of time or pitch correction, since higher resolution gives more samples to stretch & blend. The last benefit I’d mention for 192kHz is the lower latency times on system buffers. On certain systems, this is extremely desirable and beneficial since desktop computer systems these days can handle the load with much greater ease than ever before.

[fredfox]

I dithered around so long that I can only offer a truncated opinion that 24/48 is good enough for me.

[MYNAME_ABC]

Even CD quality is "only" 44.1 kHz (16 bit), enabling almost 22kHz bandwidth. That is what 10 year olds can rarely hear, even at higher volumes, and only when not masked by lower frequencies. If you are above 25 years of age - forget everything above 18 kHz.

[trevorbartram]

In terms mastering from tape to CD, what makes most difference is: (1) the quality of the source tape (2) subtle changes to eq etc during transfer to bring "the breath of life" to the presentation (3) the quality of analogue to digital conversion (ADC). The limitations of early ADCs was recognized by Tony Faulkner, who modified Sonys for better performance & the team at Pacific Microsonics who were developing HDCD.

[johnbrentford]

16 bit 44.1kHz is the limit on a CD no matter what you up sample to there is no additional information.

[DaftFader]

"I buy two copies of each CD so I have twice the resolution" ... that had me dying!

I genuinely used to be able to hear the difference between 48Khz and 96Khz when I was younger, but nothing more. Specifically when I would have a lot of high pitched distortion in a track with no LP cut off filter. But I'd be pushed to hear the difference now. My ears used to go up to around 22Khz, My left is down around 18-19k these days though and my right a bit lower than that maybe 17-18k (dam DJing headphones). Although sometimes I can tell if there's higher frequencies by how that top end range I can hear sounds, I can't actually hear them >20k frequencies anymore and there would have to be way too much of it for normal listening for it to even be noticeable to me in my audible range. (I've got speakers that go up to 25k, but I can no longer hear super sonic stuff at all sadly, dam age, I just have them so the cut-off is a bit further away from the range I actually can here currently).

[xanataph]

The reason why you never see bitstream used in DAWs is they can't handle the processing that way. They would effectively have to convert it to PCM on the fly, process and then convert back to bitstream.

[lucsolomusic]

I think the low noisefloor of 24 bit is convenient for recording but at the same time I think my music does require high sampling rates as well so 24/44.1 serves me well

[innovationsinm]

Was entertaining to listen to and glad that it’s pretty much factually accurate up till the SACD as really you missed how the supper high frequency low but rate encoding actually works and why the recording industry would want to use this for archival and data storage. Well worth listening and even subscribing to. 👍🏻

[ian-nz-]

As an electronics engineer, I really enjoy our down to earth technical videos, keep them coming! Regarding DSD & SACD, it's no coincidence that Mobile Fidelity Sound Lab use it for their digital releases... From a hardware engineering perspective, a bit stream D to A is much simpler than the alternative so it's much easier to do it right.

[mattlm]

If you record and mix in 44.1 or 48, then I suppose that's typically OK provided that the digital effects are oversampling to remove any aliasing from the application of those effects?

[stephenbaldassarre]

Virtually all ADCs of the 90s were 1-bit 64fs. In other words, DSD is just recording the first stage of converting to PCM and does away with the need for brick wall digital anti-aliasing filters, as well as reconstruction filters on play back. Nowadays, most quality converters are 2-bit or even 3-bit, at 128-256fs but the real world performance isn't that much greater than can be achieved with 1-bit 64fs ADCs. The main difference is you can get away with not having an analogue anti-aliasing filter (usually 5-pole) in front of the ADC.

[ac]

Audiophile William here, I've tried and tested many highend DAC's over the years, i never paid any attention to all the numbers just the sound. I borrowed a MSB Premier DAC that does 44.1kHz to 3, 072kHz PCM up to 32 bits and from 1 to 8xDSD, in the end i decided to keep my old modified Audio Note DAC from 1998 which sounded just as good if not better than the MSB Premier. My old Audio note dac manual says it has no over sampling, no jitter reduction, no noise shaping and no re-clocking and uses the highest grade AD1865, 18bit stereo converter chip what ever that all means?? Audio Phil tickles me every time 🙂

[Lif-]

Gotta love audiophile Phil.
I feel as if I've known him all my life!
I know he is just - as it were - your alter ego.
So every time I see him, In my head he's audiophile Shill.

[yasunakaikumi]

funny thing is some 1980s famous tracks were recorded at 32khz 16bit pcm lol

[PASHKULI]

24bit 48Khz for recording, mixing, mastering
(24bit 96khz only for raw recordings and source archive purposes;
for special applications, scientific, we might go for 32bit 96kHz but those are not really for music)

If we have 24b\48kHz, then DSP (digital sound processing) should include 'over-sampling' (×2 or ×4 the orig. freq. 48kHz in our case).
Record near 'hot' levels (test with low + percussive sound for worst case scenario of a small 'headroom VU' – eaten 'volume units' by the low freq. test sound, and louder Peaks – the percussive hit), if clipping occurs for a few samples DO NOT overthink it – they can be restored in post-recording\pre-mixing production!