Hearing the difference between Aliasing and Antialiased data is not the same as "Hearing the difference between 192k and 48k".

 

Thank you.

 

Although, I know, you will copy the answer from Google anyway..
But still..

Define the exact differences you will hear when listening to mp3 or any perceptual encoded daata reduced format vs PCM format.
I am being trained on particularly that so I might still hear small differences at only certain areas, sadly but I know the differences one should expect for.

 

i didn t copy the answers from anywhere, unlike you probably did, but with your last statement about the difference in the quality sound of mp3 formats , you ve won everything, congratulations

 

About my hearing..
Well, You heard Martin Garrix, Zedd - Follow on YouTube?
(At least this is one of the songs where I have found this discrepancies of YouTube compression)
So, can you hear the weird noise?
Sharp narrow band?

Let me spoon feed you some more.
It happens from 00:06-00:08 and then 00:11 to almost end.
Can you hear it?

I can sadly.
Not because I am some superhuman or superior shit.
But because due to extremely absorptive conditions in my listening area, I have way too much HF sensitivity, even when B&K curve is applied because usually, ER (Early reflections) causes diffusion in the HF and when there's no ER at all, your HF sensitivity gets heightenes extremely.
I tested the same thing.
With Flatish response, the HF became unbearable for all songs, from R&B, to Reggae to even Orchestral or Blues stuff.
So after talking to some people like kyle Mann, Bob Katz etc, I implemented B&K curve.
Even after that, I can hear more details with my Loudspeakers than any Headphones I have (R70x, M50x)

Honestly, it is worthless to reply to people like you, who are reluctant to go by hard evidences and data that can be reproduced with redundancy.
But still, you were questioning my hearing, I thought of answering this.

Can you hear the Noise?
It is only there in the YouTube version and not on spotify or iTunes.
So can you?

And if you can't, either you are old and have presbycusis which is okay, or you have lost HF sensitivity.
Do check.

See, we have a common ground for Testing.
@Mud Jones @duskwings

 

Last edited: May 29, 2022

Well actually, when producing it’s possible to hear a (f)actual difference between high (88.2/96) & regular (44.1/48) samplerates. As HIGHER samplerates actually INTRODUCE / INCREASE a lot of intermodulation distortion across the entire frequency range.

In practice, more often than not, HIGH samplerates actually end up sounding, and being INFERIOR compared to regular (44.1/48) samplerates.

Make sure to watch the first 8 minutes specifically:

 

I asked you to asnwer.
You passed the question carefully.
Good job.

 

Dan Worrall is just great.
All of his videos are one of the most informative videos on YouTube on Audio.

 

what answer do you want? do i really have to t ell you that the sound quality of a compressed format like mpt lacks in dynamics, and the more you compress the file the worse the sound is, do I really have to explain you what the ears of every normal person (sorry to exclude you) already know and can tell, or do you want me to explain you why this happens? google the answer, so yu ll have the explanation to why you cast a bunch of crap when you say you can tell e relevant differenceonly on certain areas, too bad that the difference you talk abot refers to cuts and compressions and conversions that contribute to the variation of a format to another.Obviously if you convert a wav file to mp3 without modificatios you can t hear a difference, but at that point there s no need to convert, but you already know that because you r so smart and we are all stupid and deaf because we think that a 10 mb song sounds the same when it s reduced to 3 mb, you must be the idol of the kindergarten

 

This is the BS jargon I was exactly looking for, lol.
Thanks for providing the answer.

You didn't answer.
Here's what to look for when judging mp3 from wav.
Or for that matter, most Perceptually encoded format from PCM or Lossless Encoding.

Perceptual encoding means a set of filters, that reduces spectral comtenta that was supposed to be Masled for most casual listeners.
How was the value for the filter were found then?
Based on several tests on subjects.

You may feel loss of Transients and High percussion which, if exaggerated, should sound like putting a cover in front of the tweeter.
Check the Reverb of the main lead instrument and vocal and try to figure out how the reverb of then sounds. It will be way harder if you try to perceive reverb as a whole. Try to find one particular instrument and its Reverb. Listen to it in mp3 vs Wav.
With a lot of practice, you will be able to feel that the Wav file Reverb has less spatial width and depth (which is subjective, I know) but you will feel a gurgling or whirling sound.
A good example of whirling sound can be the first few seconds of "Linkin Park - Crawling" (They have that sound as an effect in the music video).
Also, you will hear a chattering (an easy artifact of bad Gating).
You will hear a Tinning quality around sharp transients. That's a post ringing.

Also, the most obvious thing, any random chaotic sound, eg Applause, will sound disturbing.

You can't hear this right?
At first chance no one can.

But you are capable of hearing them clearly.
Engage M-S matrix and listen to S signal only.
Once you can hear this in the S signal clearly, switch back to normal mode.

Still, it is pretty hard to distinguish mp3 320 kbps with 44.1/16.
Yes, the data has been reduced but remember, the spectral filters applied are decided based on a large sample of trained listeners.

This is the answer to the question.
What you wrote is some tautological mumbo jumbo BS.

Also, thanks for proving the point.

 

I heard this like 7 years ago in a very interesting discussion in a forum. It was no less than the VoS (co-TDR) developer who was explaining it. At least he tried, because other "top level DSP devs" ... . I've had to wait 7 pages until someone mentioned it
Props

 

once again you prove that you are deaf and base your experience (what??) on things that you read but you can t trust your ears at all

 

What?

Are you a comedian?

It is utterly butterly clear that YOU ARE THE ONE, who creates opinion based on authority Bias and literally have No ear experience at all and clearly want to continue speaking jargon based on any random BS, tries to talk in subjective BS and blaim others.
We all have had the guy, who himself was a small Weiner yet called everybody else a small Weiner.
You are like that guy, based on your replies.

P.S. I do trust my ears and sound reproduction system, probably more than you trust in your Biases and Mumbo jumbo BS

 

Last edited: May 29, 2022

is mumbo jumbo your imaginary friend?i bet in all the time you spent here writing crap nobody gives a fuck about, you din t find a second to plug an instrument into your soundcard ( if uyou know what an instrument is) and tried to record at different sample rates to check if you could find a difference.It would have been a waste of time for you, obviously, since you are deaf, among other things

 

The real problem here isn't that people can't hear the difference, it's because they're using cheap monitor cables. Try these...

https://www.analogueseduction.net/t...-emperor-double-crown-loudspeaker-cables.html

 

I'm 51 and already little deaf of my left ear .Do you think that I care about the bitrate of the melody that I listen on my phone ?
In my opinion 44,1/48 Hz is the standard ... Humans can detect sounds in a frequency range from about 20 Hz to 20 kHz. (Human infants can actually hear frequencies slightly higher than 20 kHz, but lose some high-frequency sensitivity as they mature; the upper limit in average adults is often closer to 15–17 kHz.)
192 Khz only if you are an alien ...

 

they seem very affordable

 

so since an adult can hear a frequency renge from about 20hz to 18khz, and this value decreases with age,you think that even recording at 48khz is too much?why not recording at 20 khz , just to be sure that even children can hear everything?

 

Everyone who wants to sell a product tries to emphasize the best - the selling point - and to conceal the less good points. And most customers don't want to deal with complex details, so why complicate things? But you definitely have an argument here.

Ok, I got you. I would have to look closer in detail, since I have not worked with the F6. BTW this leads the whole topic ad absurdum. No converter I know of achieves a fully usable 24bit, because microphone, power supply and preamp add their own noise to the signal. Especially with older devices. What about tape? Should we switch back to 16bit conversion? The noise floor is not the point of 32bit ADC at all!

If the company has developed anything of its own at all, anyone is free to look at the relevant patents.

I can't say anything about the F6. I do not have such problems with my Mixpres. The limiting factors are always either the microphone or the internal preamp. The whole point of multi-stage ADC in a 32bit vector is not that you have 192db SNR or you can enjoy preamp noise at more than 1500dB below digital 0, but that you have virtually infinite headroom in the AD conversion. And THAT is absolutely worth the extra storage space when I'm working outside on location!

 

Last edited: May 29, 2022

Enjoy the blessing of 8x the file size of every project and 8x more CPU load for each plugin for no audible benefit compared to oversampling where it makes a difference. Unless you're selling sample libraries there's little to be gained by going over 48/24, and even less so for 96/24.

192kHz has 4x smaller latency (if your interface actually keeps up and scales perfectly which is not always the case), but that doesn't help you when you have to set your buffer higher anyway because your CPU can't keep up in a session that's any bigger than 8 tracks.