a DAC has capacitors at the output to smooth out the digital signal. the question is are the capacitors in that DAC's output discharging as quickly as .005 of a millisecond?

 

(1.000ms ÷ 192.000smpls) × 32 smpl buf = 0,1666666667ms

0,166ms shouldn't be a problem for the coupling capacitors with the values typically used here.

 

I found the answer myself: recording at 192khz is a total different universe from recording at lower sample rates, if your system supports it obviously.maybe is not very useful when you apply affects and and when you edit, but recording at that rate gives you definitely better sound.It doesnt matter if eventually you ll have to master at 44 or 48 knz,because even ig fyou lower the sample rate to allow everyone to listen to your track, the original source is so definite that any loss during the conversion will leave a high quality final result compared to something that was originally recorded a 44 or 48.I think that for recording, 96 khz is the minimum to choose

 

 

Last edited: May 28, 2022

One thing you guys are forgetting.
If a DAC or ADC says 192k that doesn't mean you are supposed to do your project like that.
That means two things.

1. The device supports a Max Sample rate of 192 kHz and hence max bandwidth of 96kHz.
2. The reconstruction filter (DAC) and antialiasing filter (ADC) will work at 192kHz sample rate, hence better recording and playback AT ANY OTHER CHOSEN SAMPLE RATE.

Also, narrow band reconstruction filters cause issues with out own LPF implemented at our cochlea. As per research, atleast 4-5KHz bandwidth from fully blocked to fully passed is (so 20kHz is fully passed and Nyquist will be fully blocked) is best. So for reproduction etc, beyond 48Khz is fairly useless.
But when talking about reconstruction filter or Antialiasing filter (this is implemented at the ADC section), higher sample rate of the device will produce better sound at any chosen sample rate.

Wanna test it?
Have an external signal generator.
make sure you have an interface with 192 kHz support and then one with 96kHz support.
Now, set project SR to 44.1
Now generate 24kHz signal and send it to the interfaces. Now if the there were no Antialiasing filter, you would have got a 20.1 kHz signal.
But what do you get? Nothing. ( THEORITICALLY NOTHING BUT BASED ON THE FILTERS IMPLEMENTED YOU MAY GET SOME -90 OR -80 dB SIGNAL)

Now to test which device applies Antialiasing filter on the input at which sample rate, you can generate higher frequency signal. Say you generate 50kHz signal. So if you send it to the antialiasing filter of 96kHz , it will think, it is a 2kHz signal and even with a 44.1kHz SR project, you will see 2kHz signal with an interface that can go 96kHz at max, eg Audient Evo 8.
But keeping the Project SR to 44.1, if you send the same 50kHz signal to a good interface that supports 192khz, you will see nothing being recorded in your DAW project with 44.1k SR.

 

Please consider two things.

1. Consider talking to the AES panel and they will declare you as the FIRST EVER Goldern Ear who can differentiate 192kHz in blind test.
Once again, I am not joking here. Please do this.
2. If you are pretty sure that you didn't just encounter any Bias, do consider the point 1. Please.

And your whole judgement is based on vague data.
Firstly, even if you were an actually Golden ear god, there is no chance that you will be able to pick 100% of the time. 100% probability means it is THAT obvious and so everyone will have the same probability. So, that's a statistically vague data.
Read the "Law of Large Numbers".
What's your confidence data? That is related to the number of trials.
For your friend, 2 times in a row, in terms of Statistical analysis, Means just nothing..


So, if you have such tall claims, I would suggest you to contact AES ASAP.

Atleast Pass this test with 95% confidence.
https://www.audiocheck.net/blindtests_16vs8bit_NeilYoung.php



This is what an actual test would look like.

This means what?
I can identify 17kHz vs 20kHz pretty easily.
But still I couldn't get it correct 100% of the time.
First, the difference is fairly subtle. Plus there are other factors involved as well.
But the higher the confidence data, the lower the luck or guess factor.
Statistics is everywhere.

 

Last edited: May 27, 2022

Bob Katz did say this in his Book. You can't fool nature. In real world, ALL SOUNDS ARE Fixed Point, THERE'S NO FLOATING POINT (explaination follows). Out ears are almost 24 bit Fixed Point based on our Hearing threshold and Threshold of Pain. The loudest recorded sound is I guess 210 dBSPL which is like 35 bit fixed point. 32bit Floating point will give you a dynamic range of 1528 dB which is not possible in Reality. It is only possible inside PC for the internal calculations as more bits provide better calculation. Most DAWs, if not all, have internal 64 bit floating point So, You can use, say 100 utility plugins with +24 dB boost each and another 100 utility plugins with -24 dB boost each on a particular channel and still, you will get your original signal intact. That's where the internal 64bit floating point system shines. Even if you run your project at 32 floating point or 64, YOUR CONVERTORS CAN ONLY PLAY 24 bit. So, if you want to destroy hard-drive space, you are welcome.
Yes, sometimes, mixing engineers ask beat makers to use 32 bit floating point because those beat makers apparently knows nothing about gain staging and so, if they send the project at 32 bit floating point, it will not be in a destroyed state.

 

Last edited: May 29, 2022

Funny.

Here's an even simpler test for you, Load up a few synth plugs (not sample based) use something like Diva and similar synths, then add some drums and something like Waves IR-1 convolution reverb and exaggerate the wetness a bit, maybe some distortion on one of the tracks like guitar rig or something, hit play and change it from 44.1 to 96, I can do this without pressing stop just switch from 44.1 to 96 as its playing if you can, if you can't hear the difference, I would suggest you consider having your ears checked or cleaning the wax out of them. But please don't try to spread false information, perhaps your bad ears are not a good judge. Yes I know I'm talking 96 vs 44 but if you cant hear that difference between that then 192 definitely wont matter for your ears.
An improvement in sound will carry over even when dwindled down to an mp3.

 

Last edited: May 27, 2022

This is what happens in real world. False info spreader asks others to stop spreading false information. Kudos. But seriously, please inform AES and they will really be fucked up completely if they found out that there's a man who is actually golden ear. please contact AES immediately.

I have never seen the mixture of Dunning Kruger effect and Bias (either Conformational or confirmation or Authority or egocentric Bias or a mixture of all Biases) in one thread. It would seem pointless to argue with you any way. But jokes apart, do email AES panel members.

 

Last edited: May 27, 2022

Ok you caught me, in truth I was hired by Mackie to spread fake news. You're right 192 is bad and 44.1 is awesome.

Have a nice day.

 

Did you poeple actually read the thread?who gives a shit about what you hear on a a cd or a dvd of a mastered product?i was talking about recording, as i said before i found the answer on my own; the higher the sample rate, the better in terms of performance and sampling.Obviously those who don t even know how a real instrument is made and just toy around with virtual instruments and loops won't notice a big differnce, but try to record an electric guitar or an acoustic guitar with two mics, and let's see if recording at lower sample rates gives the same results, but from people who insisted on another thread that different daws sound the same I expect everything

 

Last edited: May 27, 2022

This argument is very similar to 8K TVs vs 4K - because of the distance we sit from the TV most of us probably can't tell the difference. But if I was producing a video and 8K was available I would certainly go with that because eventually those 8K TVs will be big enough for us to see the difference. I would hope our hearing will similarly improve with time via technology or biology to at least restore/keep it to how it was when we were children.

 

Since the thread is turning into an ideological topic I would like to participate too.

It all started from "The Luddites", an English radical movement of the 19th century. A movement against industrialization, automation, computerization, or new technologies in general. They feared that technology could replace the less skilled workers, it was a very clear political stance.
Throughout the industrial and post-industrial revolution, the anti-technology topic has been raised several times and in various forms
Lately, fighting against technological progress appears more like a silly hipster attitude than a real political stance, in desperate cases, I would call it more simply Neophobia (yes, it is a mental desorder)

Neophobia
https://www.medicinenet.com/neophobia/definition.htm

The Luddites
https://www.history.com/news/who-were-the-luddites

Amish (a religious community against modern world)
https://www.theguardian.com/society...if-adopt-technology-amish-resist-modern-world

 

,166ms is a sample rate of 6000. that's a far cry from 192,000 times per second. what im trying to find out is what kind of capacitors DACs use in them that can handle a 1/192000th of a second impulse without decay that will bleed into the next sample's converted analog signal.

the analog side of DAC is designed to provide enough energy to move a speaker, and i wonder if there's a measurable difference in the electronic decay of one sample impulse in a 192k DAC and say, a 96k DAC. i need to see it on a scope, and i'd need to see every DAC on the market play back a one sample impulse over time and see the decay in order to know whether it is snake oil or not. probably some manufacturers are better than others.

I think 192k ought to be enough to drive an ultrasonic heterodyne speaker system for example, so i do believe in high frequency sound devices.

 

I believe it was Bill Gates who said '44.1khz ought to be enough for anyone'.

 

Last edited: May 29, 2022

Some good literature :

http://lavryengineering.com/pdfs/lavry-sampling-theory.pdf
https://www.lavryengineering.com/pd...the_optimal_sample_rate_for_quality_audio.pdf

 

Last edited: May 28, 2022

4K, 8K etc are Resolution, number of total pixels.
Similarly, Bit depth is the resolution of audio.

Video and audio are entirly different perceptions.
Again, I am not against 192k.
No one should be.
I am against Bullshiting.
And everyone should be.
A few threads above, I explained my understanding about 192k and if you are normal humanbeings, you can Google and there are numerous Research works published on AES and independent researchers about this topic.

Audio is a strictly dynamic process. Dynamic as in, ever changing. There's no static part of audio. Audio has to evolve over time for us to perceive.
Video has a static component. Video is literally a bunch of still images. We can perceive each still image if you want to.
We cannot perceive anything static in audio, because it doesn't have any.

But since this thread has derailed from its original purpose entirly due to some tall claims and biased bs, there's no point replying anyway.

 

My point wasn't that audio and visual are similar but that the arguments are.

 

I work in 96khz and only recently had to do a 44.1khz mix. There is definitely more struggle in the high frequencies, broader cuts in the highs have to be made and filters utilized in a broader way, where in 96khz all of this is more forgiving, you can hear the aliasing clearly in 44.1 after working in high resolution.
Working in 48khz should fix a lot of these problems and is probably the sweet spot. 96khz is very cpu demanding.
It does not mean you can not make a good mix in 44.1khz, not at all, it just means you need to work a bit more.
The reward is in the cpu, if you have big projects you can really use a lot of plugins even heavier ones like acustica.