Earlier quoted context omitted.
If you're not on a US-based IP, you should check out https://src.infinitewave.ca/ It is an incredible resource to see the quality of the resampling algorithms used by the actual production software likely used in any digital audio workflow. You will see that while the best are indeed almost 100% transparent, many are not.
> If you're not on a US-based IP, you should check out https://src.infinitewave.ca/ There is also https://src.hydrogenaudio.org/ (with no IP based restrictions, AFAIK).
24-bit/192kHz music downloads and why they make no sense (2012)
201–210 of 274 posts
Re: 24-bit/192kHz music downloads and why they make no sense (2012)
#202As they say, most people listen to their music with equipment. Audiophiles listen to their equipment with music.
This is perfect, thank you this goes straight into my long-term memory bank. On a tangent, whenever someone mentions LP sounding warmer or whatever I like to point out that I prefer wax cylinders (a.k.a. phonograph cylinders).
Re: 24-bit/192kHz music downloads and why they make no sense (2012)
#203Earlier quoted context omitted.
Decades ago, I was treated to an ABX test in my brother's recording studio. I easily recognized and preferred a 24/192 master he played versus the 16/44.1 down-mix. I honestly don't know whether there was something wrong with the down-mix, but qualitatively it did feel like it was "muffled" and coming from speakers, while the master really felt like live performance. He was surprised that I could tell them apart. I a…
This is an extremely hard comparison to do well. I'll give a few examples as to why: Small differences in gain are ABX able much more readily than differences in noise at the 16 vs 24 bit level. So if the signal chain gives even a small difference in gain between the samples that's what you'll track. A reasonable conversion path to 16 bits for mastering will also apply dithering and some kind of brickwall limiting (y…
This was supposed to be running the DACs to match the source configuration, not resampling into some common format. I think that is an unavoidable part of the whole end-to-end ABX test concept.
Maybe it would be interesting to up-sample back into 24/192 and play both in that mode. But then people would argue about what type of up-sample to use.
I was in my mid 20s for this test. I understand my high-band hearing was better back then.
Re: 24-bit/192kHz music downloads and why they make no sense (2012)
#204Earlier quoted context omitted.
you have 22 bits for the typical audio voltage level, which you call 1.0 float but what if you "allow" double that voltage and call it 2.0 float? a strong pressure into the microphone generates a stronger voltage thermal noise limits you on the quiet signals, but not on the powerfull ones so 22 bit for typical -1.0 -> 1.0 range and you can add a few more bits on top of that for stronger audio pressures (voltages) whi…
Sorry, but this not how AD works. If your idea was valid, we'd have new generations of ADCs in our hands.
You have some low noise amplifier. There is a signal. You split it. The result on each side has >=1 bit worse noise floor, probably somewhat worse as we're not using superconductors :P-- as you expect: there is no free lunch.
Now: take one copy and attenuate it 48dB, further degrading its noise floor. Sample both. The attenuated copy is mostly useless, except when the input goes high enough that it would have hard clipped the other ADC.
So the tradeoff is that you lose a small amount of noise floor constantly-- out at the 20th bit, that you probably didn't care about (microphone self-noise is limiting you out there anyways at normal volume levels), in exchange for never clipping.
To turn this into a better ADC generally, you'd need the splitting stage to not hurt the noise floor, but it does.
The reason it's not the same as just lowering the gain so that you won't ever clip is that to get the same dynamic range you'd have to lower it by 48dB and now your ADC doesn't achieve its potential for typical signals. You could lower the gain by 3dB (or whatever the splitting cost you) and get the same results for the low gain signal and a little more headroom, but you would not get the massive headroom increase of this approach.
For this to work one must also have amplifiers with much wider dynamic range and SNR than ADCs, but we do.
The natural output for this approach is a float-- the most natural would be a weird float where instead of an exponent one bit tells you which ADC is in use and represents a factor of 256 or whatever, but in practice these recorders just output 32-bit floats. I haven't looked but I wouldn't be surprised if there were only two exponent values ever used in their output.
Re: 24-bit/192kHz music downloads and why they make no sense (2012)
#205Earlier quoted context omitted.
Decades ago, I was treated to an ABX test in my brother's recording studio. I easily recognized and preferred a 24/192 master he played versus the 16/44.1 down-mix. I honestly don't know whether there was something wrong with the down-mix, but qualitatively it did feel like it was "muffled" and coming from speakers, while the master really felt like live performance. He was surprised that I could tell them apart. I a…
Even for PC, I recommend some cheap studio monitors.
Re: 24-bit/192kHz music downloads and why they make no sense (2012)
#206Earlier quoted context omitted.
Running tape at half speed has almost nothing to do with digital resampling, which is what playing digital audio at half speed is generally all about. My use of DAC was a thinko, I've edited at least post to correct it since in the current context we're always talking about ADC. Apologies for that.
Wrong again. As a sound designer I can choose to import a 192kHz file into a 48kHz PT session in two ways, one as resampled audio which means pitch & duration stay the same, OR I can choose to import it without SR conversion, in which case the audio plays at quarter speed & pitch is 2 octaves lower. We use both techniques ALL the time, every day. It's a common technique every sound designer uses. You are arguing abou…
Re: 24-bit/192kHz music downloads and why they make no sense (2012)
#207Earlier quoted context omitted.
Ok dude, you obviously never recorded anything. Twelve mics on a drum kit, 60 tracks of rhythm guitars, several bass guitar layers, vocals, backing vocals, electric organ, percussions, saxophone solo. Do you think recording them at 44.1 somehow creates a shared "cloud-based" aliasing artifact that I store in S3?
> Ok dude, you obviously never recorded anything. https://ardour.org/ is my website.
Firstly, it's an amazing experience to randomly interact with people like you - I love and use your software. Hats off and thanks for what you offered to the industry!
But secondly, your statement makes even less sense to me: obviously artifacts do add up. Yes, not linearly, like any complex audio in general. But the more tracks with artifacts I have, the more artifacts I have overall. It's not like they cancel each other (outside of normal frequency cancellation).
Re: 24-bit/192kHz music downloads and why they make no sense (2012)
#208Earlier quoted context omitted.
The central point is that AD conversion can and will introduce artifacts. DA process wil intrduce more artifacts. The "imperfect" is a huge range and AD/DA converters play a role in that. We are not talking about "golden cables" bs here, conversion does introduce measurable artifacts in the audio path. The more tracks you record the more artifacts you have. Can everyone hear them? Definitely no. Can they be heard - y…
No, the central point is that the analog signal handling before AD introduces vastly more "artifacts" than the AD or DA does. In addition, nobody cares about "measurable" artifacts (or rather, they should not). What matters are "audible" artifacts. We have measuring equipment that is vastly more sensitive than human ears (e.g. your recording equipment that can pick up signals far above 22kHz). What's measurable is no…
And absolutely - I blind tested coverters extensively. Mbox2, Black Lion Audio upgraded converters, UA, Prism.
Re: 24-bit/192kHz music downloads and why they make no sense (2012)
#209Earlier quoted context omitted.
Sorry, but this not how AD works. If your idea was valid, we'd have new generations of ADCs in our hands.
> In a 32-bit float recorder, you have two ADCs working in tandem to create a single audio file. One “low gain” ADC is optimized for high-level audio, and the other “high gain” ADC is optimized for low-level audio. If the high gain ADC clips due to loud sounds, the low gain ADC does not. And if sounds are too quiet for the low gain ADC to capture clearly above its noise floor, the high gain ADC still has plenty of he…
Re: 24-bit/192kHz music downloads and why they make no sense (2012)
#210Earlier quoted context omitted.
Sorry, but this not how AD works. If your idea was valid, we'd have new generations of ADCs in our hands.
It's discontinuous. You have some low noise amplifier. There is a signal. You split it. The result on each side has >=1 bit worse noise floor, probably somewhat worse as we're not using superconductors :P-- as you expect: there is no free lunch. Now: take one copy and attenuate it 48dB, further degrading its noise floor. Sample both. The attenuated copy is mostly useless, except when the input goes high enough that i…
So, basically, no better than the best AD converters we already have?
My understanding of the fundamental limit to AD performance is that the brownian noise level is around the 22nd bit level. So even if you come up with techniques to successfully measure down to that level, you're basically picking up .. inevitable, irremovable, irrelevant noise.
Possibly there are gains to be made by not worrying about the noise floor and caring more about the lack of clipping, but I'm not seeing people screaming about that. The "noise" seems to be "N bits of dynamic range", not "slightly less dynamic range but it will never clip!"