Video/TV/Cinema:
Overall, we can all state that we have gotten more picture quality, for the moving pictures, in the last few years.

Let's look at the old videos with VHS. It was still cassettes, then came the compact disc, DVD then the Blu-ray and replaced the VHS video cassettes. These old video players are no longer sold today.

Then the analog broadcasting method was changed to the digital broadcasting method. The old television receivers then had to be bought new or upgraded with a setup box. With 4K monitors, you can even make the writing a little sharper.

The result is as @Crinklebumps already noted more resolution, so more sharpness, more details and more colors.

Audio/Listen:
Let's remember the old analog tape cassettes where you could record music or speech with a tape recorder. Then came noise suppressors from the company Dolby. With the invention of the CD Rom, the analog cassettes became less and less until they disappeared completely. Some still have these old players at home and some digitized his cassette collection by means of the computer.

In the beginning there were 4 track cassette recorders available, some musicians bought the expensive hard disc recorders with DAT cassettes. Then there were tape recorders. The carrier material was plastic coated with a magnetizable layer of iron, iron oxide and/or chromium oxide crystals. These devices gradually disappeared with the new digital technology. Then came the computer, with digital audio workstations, new interfaces like VST from Steinberg and analog to digital converters in the form of audio interfaces.

Result of the digital age is, we hear more frequencies and hear more quality also the speech intelligibility has become much better. The technical progress has been implemented and benefits the users of these devices. The first CD ROM cost when it came out still 6 € and today by the mass production about 10 cents. Or you can use MP3 players with flash memory technology.

Spoiler: Advantages using the example of an audio CD

Advantages using the example of an audio CD

Using the CD as an example, some advantages of digital over analog signal processing can be seen: The information stored digitally on a CD does not change even after years, provided it can be recovered properly and there are no age-related defects. There is no "crosstalk" from one track to another, no high frequencies are lost. Even if the CD is played as often as desired, the data is not changed, as is the case with a record: there, the needle of the pickup "grinds away" a little material with each playback and smooths the edges - with the result that high frequency components in particular are attenuated.

 

I think you have accumulated a lot of BS inside your ears, perhaps try a QTip. I think over time when you get more experience you can come back to a thread like this with some humility, I was a newbie once too so I understand, but unlike you I was not afraid of being wrong, that's how you learn.

If you're mostly using in the box effects and instruments then 192 has an advantage, facts.

 

Guys, I know it's week end, pop corn time, but please, don't transform this into a personal fight.

 

I'm really not an expert in converter construction, so I may be wrong. But I wonder which capacitors you mean.
As far as I understand it, the sample rate plays a subordinate role in the DA conversion. What is interesting is the bit. The sample rate provides a series of bit values that must be converted to voltage. A network of resistors and switches decides which bit values are passed on as voltage - all in an integrated microcontroller. So what is important here is that you have enough resistors and switches that can cover the bit depth. And switches that can switch fast enough according to the sample rate.

Behind it sits an op-amp and a filter. So there you have your capacitors. But they have nothing to do with the samplerate.

 

Last edited: May 28, 2022

The quality improvement e.g. from 48 kHz to 96 kHz is, in my opinion, negligible, especially considering the necessary double storage and computing requirements. Even experts like to argue about how much, and whether this is even relevant - i.e. clearly audible - outside of a direct A/B comparison. In terms of bit rates, however, the quality leap from 16 to 24 bits is clear. It is a fact that a system, with correspondingly high sample rates, is loaded more and thus consumes considerably more performance. Many digital mixers, for example, offer only half the number of tracks at 96 kHz, or digital ADAT connections are then no longer available, or only in the special SMUX format, with half the number of channels.

Especially if you produce for the CD and MP3 format, you should think twice. Effectively, the quality improvement in the end falls into the realm of subtleties. Here it applies to try out whether it really comes to the own music to good. If the computer can handle 96 kHz projects without any problems, there is of course no reason to do without the "extra quality". I don't think it's necessary to run projects with 192 kHz in the context of home recording. If you think you hear drastic differences, you should check if the filter of your digital converter is not defective...

In any case, 44.1 and 48 kHz have established themselves as standard values. The former rate is used for audio CDs, the latter is mainly used for video and broadcasting. Even when creating MP3s and audio for the Internet, 48 kHz is recommended for the source material, since many conventional audio chips run "fixed" at 48 kHz. This means that even when playing an audio CD, conversion is necessary. Nowadays this is all possible with high quality algorithms, but the "high quality" of some consumer products has to be questioned. Professional converters, on the other hand, can handle different rates without loss. Often also with their multiples, i.e.: 88.2 kHz, 96 kHz, 192 kHz, etc.

 

Exactly, that's the point: it is obvious that the 192khz feature is dedicated to professional recordings, we are talking about showbusiness where quality is the key. There is a constant battle in innovation up there, due to the strong competition to place your own product on top of the list. Even a relatively small added value given by a higher sampling rate can make a difference
On the contrary, I don't think the 192khz recording feature is dedicated to bedroom producers: there is no need of it, plus the cost of a new workstation will impact your budget (yes: you will need a more performing computer and much more space on the HDD).
I believe that, for home recording, 48k/24bit is more than enough, if you're asking yourself "do I really need 192khz?" means that actually you don't.

no panic and focus on the mixing stage

 

Last edited by a moderator: May 28, 2022

Oh god, it s true that 44 and 48 are the standards when it s about listening,but recordinbg is another thing,If you can t tell the differnce in terms of quality from 48 to 96 or from 96 to 192 you better have your ears checked, because there s an abyss, the fact that the final product is then converted to lower sample rates doens t mean anything, or better ,it means that you convert something with a superior quality compared to something that was recorded at lower sample rates, unless you think that since you record at 44 khz, eventually your master will have the same quality because the sample rate is the same. All your measurementes dont mean anything, because listeners listen with their ears, and if you can t perceive the difference,it s not because the sample rate makes no difference,but because you are deaf.
By the way, my computer has an i eight intel i5 3450, 16gb ram, only one ssd for the system, and for mechanical drives for storasge and other stuff,no graphic card, and guess what, i can record at 192khz

 

DDMF Meta Plugin to oversample only plugins that acutally need it > oversampling everything to hurt poor cpu.

 

Yes @recycle, I agree with you on that: What can be done technically will be done and if you can make money with it, these technical possibilities will flow into products that you can then monetize.

There is in Germany a research institute with the name Fraunhofer Institute, which are the ones who have developed the MP3 standard, by the way with an American company. MP3 costs thus no cent for the consumer. That is times a positive message. What you can earn money in the future is the AI and voice assistants in the car, for example.

MP3 Link: www.iis.fraunhofer.de/en/ff/amm/consumer-electronics/mp3.html

 

Oversampling Waves Audio Plugins using DDMF Metaplugin

 

People here like to attack because most have very little experience. Go to gearspace if you want real feedback.

 

While reading some comments I came across some strange statements about 32bit. Without wanting to address anyone specifically, why do you think there are 32bit interfaces? To fill your HDD faster? Certainly not.

I have a 32bit interface. Actually I have two. And you know what? They are awesome!
In studio - where I have enough control and overview - I record in 24bit. But when I'm not in studio, I always work in 32bit. It's pretty much impossible to clip recordings that way. 3dB over, no problem. 20dB over, no problem. Even if I had a peak with 50dB above zero - which is almost impossible in practice - it would not be a problem for me. I simply turn down the corresponding part in post and the peak is absolutely perfect. No clipping.

But you need a converter that works natively in 32bit float to do that. Storing a 24bit recording as 32bit, on the other hand, makes no sense. Because your source is just 24bit.

 

a man in australia wiped his ass with a gimpie gimpie leaf, and the pain was so strong and persistent that he eventually shot himself.I felt a little excluded so I decided to add something that had nothing to do with the thread too.Nonetheless I still haven t figured out what it all has to do with the undeniable better quality of recording at 192khz. 44,1 and 48 khz are the standard values for listening (standards that were set long ago when thery were the best option btw), nobody said they are the standard for recording, and the thread is about recording, I don t care about how demnading recording at such sample rates is.And agin, rather than casting numbers that nobody cares about, i insist that you record a guitar at different sample rates and check if it makes a difference

 

the output of a dac chip will be a series of pulses and when they bleed together and amplified they become analog. the pulses coming out will naturally have a duration, no matter how small they can't be instantaneous. my question is how long are the durations of those small impulses and if those impulses are happening as fast as the digital side of the process. my guess is no. as you said, there needs to be a filter after the DAC processor, in the amplifier circuit. but even before the signal reaches that amplifier those impulses need to be smoothed somehow, and the only way I know how that could be accomplished is by some sort of very fast 'buckets' (capacitors) to momentarily hold the charge that is sent out. I wonder if every manufacturer builds this to a standard that matches the 192khz internal processing.

 

Ah now I see what you mean. Fast and impulse resistant, probably PP metal film capacitors... But I have to be honest, electrical engineering was never my greatest strength and I dropped it after 2 semesters. So it's not really my area of expertise.

 

For recording - the more, the better. Here if you can go into MHz territory, if it's possible. After all in this context it means frequency of "reading" the analog signal and turning it into digital. That's why the more - the better.

For producing and mixing - 96 is more than enough, except for very rare cases(when a plug-in includes emulation of resonators in any form - higher sample rates create completely different results(example - 2caudio Kaleidoscope)).

For finishing\release - 44.1 with HQ downsampling. Cases when you need more in this case are so rare that you'll know about them before even begin the production process.

 

Because 32 floating point bit in DAC or zoom devices are pretty misleading as well.
What they don't tell you is, whether the recording is done at 32 bit or the internal processing.

Here is a data for Zoom F6.

As you can see, the dynamic range possible on this device is even a slight lower than theoretical 16 bit FIXED POINT.
Yet this device is a 32 bit flaoting point device.

Statistics and Data, provided by the selling companies are like Bikini, most of the time.
They reveal a lot, but, HIDE THE MOST IMPORTANT PARTS.
So, to understand what is what, you will have to keep an open mind, read AES journals and papers and may be test things youself.
All this tests can be easily done using REW software in someway as well and REW is free.

So if the interface or recording inherently has 16 bit Fixed point dynamic range and every DAW uses 64 bit floating point internal math, then by recording something with F6 in 32 bit floating point, you are technically wasting HDD space nonetheless.
What you get is 32 bit flaoting point for internal calculations.

 

Last edited: May 29, 2022

Not being a jerk.
Whenever you have some time,
Check this.

https://www.audiocheck.net/blindtests_16vs8bit_NeilYoung.php

Audiocheck.net has a lot of blind tests and lot of informative practical and theoretical stuffs as well.
Also, you can get the "Golden Ears" DVD set and practice there as well.

Trust me, you will be amazed to understand or feel, how difficult it is to find 8 bit dithered audio from 16 bit audio. Yes, it is understandable and perceivable but the differences are very very subtle. Plus to hear the differences, pretty clearly, you will need an excellent listening environment.
Off course you can try Closed back cans but there are certain things in cans that people don't like, eg In head Localization due to no HRTF, No Interaural Amplitude or Time difference, no Pinnae Transformation, No phantom center, exaggerated 2k-4k region (an artifact of all the things mentioned above that are missing in headphones and there are good headphones that actually do compensate for this) etc.

 

Hi @Anubhav Ukil, thanks for your contribution, I really appreciate you as a dedicated sound engineer.
I also think it is very important to find out the truth. Keep it up, get the best out of the recordings.

 

Last edited: May 29, 2022

if starting data are correct, 3uF will be enough, see below.

Well, I guess that if a consumer or monitor or whatsoever speaker limit is, let's say, 22kHz, it really doesn't matter, mech inertia won't stand such high frequency follow.
BUT if your concern is if you get chopped waveform out the DAC, see the clip @Fowly posted in page 1, the fear of the "Zero order hold" is not to be feared, values are interpolated, see the analog oscilloscope output on the clip.
Then there is you ear limit, of course.

Cannot say if you can strip it down to a classic condenser circuit for a DAC (think not), I guess it's much more complicated, but in case, condenser time constant is T=RC, and 5xT is total discharge time, so C(F)=5T(s)/R(ohm) for the requested total discharge time.
What is R? Well in this case we can't think the speaker as pure resistive load (is inductive also), but for the sake of simplicity, will think is.
So for 192kHz is about 3.2 uF condenser and we're speaking of peak charge vs 0, that is not the case with contiguous samples, adding L to the equation will give intermediate T values.
Does it serve all this psychobabble? Duh, I guess not. DAC designers are gently requested.

A VERY interesting lecture from Mark Waldrep, Ph.D., spanning from 44kHz to DXD:

Spoiler: Who is Mark Waldrep, Ph.D?

Mark Waldrep, Ph.D., AIX Records & iTrax, President Mark Waldrep, Ph.D., aka Dr. AIX, has been playing, engineering and producing music for over 40 years. He learned electronics as a teenager from his HAM radio father while studying music. Mark received the first doctorate given by UCLA in 1986 for an electronic music composition. Other advanced degrees include an MS in computer science, an MFA/MA in music, BM in music and a BA in art. As an audio engineer and producer, Mark has worked on projects for the Rolling Stones, 311, Tool, KISS, Blink 182, Blues Traveler, Britney Spears, the San Francisco Symphony, The Dover Quartet, Willie Nelson, Paul Williams, The Allman Brothers, Bad Company and hundreds more.Dr. AIX writes a daily blog at RealHD-Audio.com and runs HRAPlanet.com, information resources for those interested in high-resolution music.

See above.

I used to work also in 3 sigma (99%) and for scarseful data enabling Student factor.
But NOT in audio equipment.