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Old 10-22-2015, 08:46 AM   #81
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Originally Posted by karbomusic View Post
I'd like to see data to back that up since I have a very light touch of tinnitus and it absolutely interferes with real sounds of the same frequency.
Tinnitus is not imaginary in any way shape or form. It is not a psychological condition!
Actually there is even a form of tinnitus that is audible outside the sufferers head.

https://en.wikipedia.org/wiki/Tinnitus
"The diagnosis is usually based on the person's description. Occasionally, the sound may be heard by someone else using a stethoscope: in which case, it is known as objective tinnitus."
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Old 10-22-2015, 09:27 AM   #82
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I'm not sure who you have been hanging around to arrive at that number, Softsynth, but I've known many musicians who keep their PA system(s) in their house when not using them for live sound reinforcement. And even a mediocre PA system can handle this dynamic range in a typical living room or media room environment (the larger the room, the more power you'll need to attain higher volume levels, so smaller rooms in the home are not a particular challenge). No special acoustics or system required. And as technology continues to develop, it is likely that even the average Joe will have access to such systems in his home in the near future.
PA systems are not suitable systems for home listening, they do not offer a high level of fidelity without colouration, it simply isn't what they are designed for. It would be ironic to obsess about maximum dynamic range and then to pump it though a PA system!!!

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But this is missing the greater point that I intended to make, which is not that we should release at 24 bit, necessarily, but that we should track, mix and master at 24 bit (or greater) so that way we have the option of utilizing that bit depth for either 1) projects that require it, or 2) in the future as it is becoming an industry standard audio format. From there, you can always create lower resolution versions, for lack of a better term, but you can't increase the "resolution" if you start with a 16/44.1K file.

Regarding bit depth I make all my recordings at 24/48.
There has been plenty of information from respected sources that will explain that sample rates above this actually can cause problems back down in the audible spectrum, and should be avoided.

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My comment was in regards to recording orchestras, such as the London Symphony Orchestra or the Prague Philharmonic, so the worthy performance should be given in that context. But you do have a good point as it applies to the music that the average Joe makes in his bedroom.
No, I meant worthy performances from great Orchestras, not the majority of home production, the latter is a given. The majority of Pop and Rock from 1960s to now certainly would not quality in this regard either. Most are not remotely approaching the capability of 16bit 44.1khz, never mind 24/96.
Also most loudspeaker cabinets reduce the fidelity way below 16bits. Much audio resolution is effectively lost in loudspeaker cabinets.

Hard to find world class (Gramophone recommended performances etc.) performances that are genuinely benefiting from 24bit recording dynamic range and as good as the best performances that you can find on CD and vinyl.

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Tinnitus is not actually sound, it is imaginary, and as such, it does not interfere with our ability to hear real sounds.
This insults sufferers and is factually incorrect.


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Originally Posted by drtedtan View Post

As I said in the post you quoted, people may well prefer the reduced dynamic range of a compressed version of the performance, but that does not make it an accurate transcription of the performance as it originally occurred.

Which brings me back to my main point - if you record, mix, master at 24 bit (and 96K, which is becoming an industry standard) you can also create numerous other versions for different clients. That way you can deliver the 24/96K file to the director who wants to use your music in his movie and the compressed version to Mr. PC so he won't be scared if he forgets to turn the volume back down before the horns kick in.

But if you start with a lower bit rate version, you can't get those dynamics back (sure you can run it through an expander, but it won't be the same).
I record at 24bit. 48khz remains the industry standard, some may push for 96khz but no one loses sleep over that one.

Did you know that the vast majority of cinema systems are inferior to home systems and do not offer the DTS HDMA and Dolby HD of Blu ray OR EVEN DVD?
In fact the bitrate in cinemas is notably lower than lowly 2 channel CD for its multiple channels!

DTS HD and Dolby HD are audiophile home standards, and beyond the requirements or technical abilities of typical DTS and Dolby cinema systems.
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Old 10-22-2015, 09:30 AM   #83
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Originally Posted by Softsynth View Post
Tinnitus is not imaginary in any way shape or form. It is not a psychological condition!
Actually there is even a form of tinnitus that is audible outside the sufferers head.

https://en.wikipedia.org/wiki/Tinnitus
"The diagnosis is usually based on the person's description. Occasionally, the sound may be heard by someone else using a stethoscope: in which case, it is known as objective tinnitus."
Yea. To be 52 and a near lifetime of standing in front of Marshall stacks playing or in front of live PAs mixing, it technically could be far worse than it is. There is just enough there in the high end that from time to time, similar frequencies in what I am listening get interfered with. It sounds like that frequency is either louder than reality but most of the time it sounds like distortion due to them not being exactly the same and the subsequent interference between the two. This typically occurs when I'm making evaluations on a single source such as evaluating subtle differences in guitar distortion when tweaking stomp boxes etc. In that scenario it can sometimes be hard to tell which is which.

Not that big a deal since I'm lucky to hear at all based on what they've been subjected to over the decades.
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Old 10-22-2015, 02:47 PM   #84
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Originally Posted by karbomusic View Post
Yea. To be 52 and a near lifetime of standing in front of Marshall stacks playing or in front of live PAs mixing, it technically could be far worse than it is. There is just enough there in the high end that from time to time, similar frequencies in what I am listening get interfered with. It sounds like that frequency is either louder than reality but most of the time it sounds like distortion due to them not being exactly the same and the subsequent interference between the two. This typically occurs when I'm making evaluations on a single source such as evaluating subtle differences in guitar distortion when tweaking stomp boxes etc. In that scenario it can sometimes be hard to tell which is which.

Not that big a deal since I'm lucky to hear at all based on what they've been subjected to over the decades.
I think if you had lived to 52 and you had not had any level of tinnitus at all you would have been quite lucky. I suspect a lot more suffer from it and do not admit it, especially when it is not severe.

The earbud generation will be familiar with it before their fathers ever were.
Mild tinnitus doesn't really impair anyone from evaluating audio quality, I feel sorry for people with severe forms of tinnitus.
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Old 10-22-2015, 02:54 PM   #85
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Mild tinnitus doesn't really impair anyone from evaluating audio quality, I feel sorry for people with severe forms of tinnitus.
Yea, pretty much the same here. It could be worse and I don't usually notice it unless it comes up in a conversation or the scenario I mentioned above. I have had two close friends have it temporarily - the kind that is the wooshing sound but luckily for both it went away. Mine sounds like head pressure or a light ringing like one would expect and is constant but as I said light enough that my brain usually blocks it out.
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Old 10-22-2015, 03:52 PM   #86
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Yea, pretty much the same here. It could be worse and I don't usually notice it unless it comes up in a conversation or the scenario I mentioned above. I have had two close friends have it temporarily - the kind that is the wooshing sound but luckily for both it went away. Mine sounds like head pressure or a light ringing like one would expect and is constant but as I said light enough that my brain usually blocks it out.
I'm in my early forties, mine is a gentle high pitched CRT like whine. It comes and goes.
I know people with much, MUCH worse.
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Old 10-22-2015, 04:25 PM   #87
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karbomusic: "To be 52 and a near lifetime of standing in front of Marshall stacks playing or in front of live PAs ..."

Yeah, I hear ya (or, well, let's pretend I can)!

The usual result? Commonly called "Mid-Range Burn".
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Old 10-23-2015, 07:47 AM   #88
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Why are you all arguing bit depth?

For multi-track recording we want a higher bit depth to lower the noise floor. Period.

Reproduction of finalized audio @16 bit has a dynamic range of 96dB. Who would want to experience a recording with all 96dB (or more) being utilized? 24 bit has a dynamic range of 144dB. The threshold of pain is around 120-130dBSPL. Seriously, reproducing real-world dynamics for entertainment is not necessary or desirable. I know I do not want to crank up a stereo to the level of pain just to be able to hear the softest parts of the recording. Hearing loss can occur within 30 sec of exposure to SPL greater than 115dBA.

Dithering can be used to increase the perceived dynamic range of 16 bit to about 120dB, and Oversampling can be used to increase the dynamic range as well. But, that is not on topic.

Question:
Can lower resolution recorded audio (e.g. 44.1kHz) gain an advantage by increasing the Project Sample Rate and converting the project items to the higher rate? Effectively, oversampling the entire project. Well, I cannot test this right now because my on-board soundcard is not capable of higher samplerates.
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Old 10-23-2015, 10:13 AM   #89
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In an effort to be understanding and reasonable, whether anyone thinks I ever am or not, I won't deny that, for example, Joaquins Void does make some valid points, particularly in his post #14.

But certain things strike me every time when this sample rate discussion rears its ugly head, namely throughout all the talk talk talk rarely, if ever, does anyone offer to help the inquirer by mentioning specific tools that could apply -- TDR's Ultrasonic Filter for instance:

https://vladgsound.wordpress.com/201...alpha-version/

Nor does anyone name names, such as which VST plugins to avoid or suggest using, those which would still work acceptably, those which should be kept well away from all this high sample rate business (I won't drop names here but a lot of 'usual suspects').

But I think the real elephant in the room in these threads is as follows: You give me practically any project, containing, let's say, a few tracks and instruments at least, and I will comb through it and find perhaps 100, perhaps 200 items that could be improved ... easily in most cases. Yet why all this worry or concern about sample rates when you already have scads of things technically wrong with your mixes already?

Why the constant fervour and (really) obsession with the sample rates so often to the exclusion of so many other features? Where is the balance here?

I've heard the untrained listener say, ooh, the song sounds a little too distorted, or too trebly, or too this or this, but I have never ever heard anyone say, Oh, you should have used a higher sample rate ... or, oh, your synth is aliasing!

Where has your perspective gone in regard to your total product?
Hey,

I don't know why I didn't respond to this earlier. I've been reading about it (couldn't find much beyond your link and the GearSlutz thread) and downloaded it.

So basically, I should put this between every single VST plugin, and it will filter the ultra-sonics. Will it still work in lower sample-rates? (for example, if I'm running 44.1, I'd have to set it to filter 22k or lower to have an affect, and that might interfere with the audible range. If I'm running 96k, then I could cut of ultra sonics at 48k, and so there'd be no chance of the upper frequencies being filtered. Right?

And will the TDR do anything to degrade the sound? If not, then I'll simple stick it between every VST on every track for the rest of my life.
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Old 10-23-2015, 10:30 AM   #90
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Hey,

I don't know why I didn't respond to this earlier. I've been reading about it (couldn't find much beyond your link and the GearSlutz thread) and downloaded it.

So basically, I should put this between every single VST plugin, and it will filter the ultra-sonics. Will it still work in lower sample-rates? (for example, if I'm running 44.1, I'd have to set it to filter 22k or lower to have an affect, and that might interfere with the audible range. If I'm running 96k, then I could cut of ultra sonics at 48k, and so there'd be no chance of the upper frequencies being filtered. Right?

And will the TDR do anything to degrade the sound? If not, then I'll simple stick it between every VST on every track for the rest of my life.
Check his response to me:

http://forum.cockos.com/showpost.php...8&postcount=28

The aliasing noise, if it exists on the output of the plugin, won't be helped by adding this plugin. The aliasing noise is part of the perceivable audio range.
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Old 10-23-2015, 10:51 AM   #91
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Mr. PC, I like your post here, a truly brave and valid effort to breathe new life into this thread. Uh, no just kidding I think.

Seriously, glad you brought this TDR Ultrasonic Filter question up again, as I was actually going to say more about it anyway but so much other has transpired on this thread.

I wanted to post a bit more because the filter, I felt, sort of got panned earlier as perhaps not of much good use. This is incorrect, though.

As I think was mentioned, this plugin won't necessarily save you from all unwanted ultra-highs, such as what some synth might generate at some stage of your work. It's technically possible but perhaps a little overboard to go placing this between every plugin or instrument, but you could I guess.

Specifically, what this plugin was designed for is to wipe out -- remove -- any or all ultra-high frequencies for whatever reason. Number 1 reason would most likely be that you don't want any stuff up there that will fold back, cross your Nyquist frequency to become audible aliasing when (or if) you reduce your sample rate to, for example, 44.1kHz perhaps at the end of your project. Obviously, if nothing is 'up there', there is nothing to fold back. A much lessor reason could be its use much like some use a high pass filter: If you have a lot of WAV/sound information well below hearing or, for some strange reason, way up high, then this stuff is simply using up energy in your file for no good reason. It can interfere with your attempt to get a desired peak value on a track if there's enough of it. Very low bass can really occupy a large energy space -- i.e., it takes a lot of energy to produce bass frequencies. With highs it is much less so but still can be measurable.

I mention that second reason above because I know some who are meticulous about removing any and all information that is not needed or not audible and not contributing to the quality of the product. Some are less interested in this sort of cleanup, and I won't enter that debate.

So, the filter is simply a type of pass/cut, just another (free) tool. If you have some strangeness going on at, say, 36kHz when your rate is set at 88.2k, that strangeness that is inaudible now can be quite audible when that file is later rendered to 44.1k, if not removed before render.
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Old 10-23-2015, 11:17 AM   #92
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You could also use Christian Budde's free VST, Rubber Filter, which is capable of 6-384dB/Oct.
http://www.kvraudio.com/product/rubb..._budde/details
http://www.pcjv.de/vst-plugins/eqs-filters/

Question:
How steep does the filter need to be to go from 0dBFS @20kHz to -∞dBFS by 22.05kHz for finalizing in 44.1kHz?
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Old 10-23-2015, 11:22 AM   #93
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If you have a lot of WAV/sound information well below hearing or, for some strange reason, way up high, then this stuff is simply using up energy in your file for no good reason. It can interfere with your attempt to get a desired peak value on a track if there's enough of it. Very low bass can really occupy a large energy space -- i.e., it takes a lot of energy to produce bass frequencies. With highs it is much less so but still can be measurable.
that is wrong. in the digital domain there is no such thing as "energy", its 0s and 1s. digital domain doesnt work like an analog amplifier, for which is what you said true.

in the digital domain there is no energy eaten up to "produce" low frequencies, because in the digital domain frequncies are not "produced" but only described.

there is always thinking in analog concepts transported over into the digital domain. digital works very, very different to analog.
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Old 10-23-2015, 12:24 PM   #94
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that is wrong. in the digital domain there is no such thing as "energy", its 0s and 1s. digital domain doesnt work like an analog amplifier, for which is what you said true.

in the digital domain there is no energy eaten up to "produce" low frequencies, because in the digital domain frequncies are not "produced" but only described.

there is always thinking in analog concepts transported over into the digital domain. digital works very, very different to analog.
NO, you're way off. Playing a WAV file takes energy. Obviously. The BASS in this WAV file takes energy to produce. If you don't believe this, stand in front of the larger speakers of any PA system and feel the bass make your pantlegs wobble. Feel the bass virtually hit you in the stomach. Low enough bass signals, as in standing waves, if intense enough, actually can make you puke.

Either you didn't understand my post or else you responding before thinking it through. Think of digital files as stored or potential energy if you will. Too much low low bass will certainly effect your ability to get good final levels, even if all that bass is completely inaudible.
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Old 10-23-2015, 12:31 PM   #95
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in the digital domain there is no energy eaten up to "produce" low frequencies, because in the digital domain frequncies are not "produced" but only described.
To be fair that is true but sounds a little too pedantic based on what Tele is trying to get across. IOW it would only apply if we are analyzing the data with the pretense it *never* leaves the digital domain and that isn't what we are discussing here. All of this is completely concerned with the source-to-ears set of processes with digital somewhere in the middle.

In that respect, we can and often should treat it as if it were analog (as per his reply) because it is eventually going to be analog again. It would be sort of useless to remain in it's 1s and 0s form for eternity.
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Old 10-23-2015, 01:57 PM   #96
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It's okay -- he's confusing what it IS with what it DOES. That's all.
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Old 10-23-2015, 02:02 PM   #97
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Why are the high-end sample sets of orchestras, supposedly the best available, recorded in 96k, if, as someone mentioned, this is detrimental
to the fidelity?
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Old 10-23-2015, 03:27 PM   #98
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Why are the high-end sample sets of orchestras, supposedly the best available, recorded in 96k, if, as someone mentioned, this is detrimental
to the fidelity?


Not a strictly accurate statement. Some are delivered as 24/96, others 24/48 and 24/44.1k.

Don't assume 24/96 is automatically detrimental, however it can cause issues on playback (certain systems) and is challenging for a lot of recording equipment too, so greater care has to be taken when capturing the audio and in playback, for contentious benefits.

Vienna Symphonic Library - VSL.CO.AT the highest of the high end professional libraries deliver audio at 24/44.1k.

They record at 24/96 but deliver 24 44.1k. Most likely to maximise the number of professional customers that will not suffer technical issues with frequencies above the comfort zone of their electronics.

Further forum reading:
http://www.vsl.co.at/community/posts...96k#post209679

http://attachments.goldmund.com.s3.a...hite_paper.pdf
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Old 10-23-2015, 03:44 PM   #99
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that is wrong. in the digital domain there is no such thing as "energy", its 0s and 1s. digital domain doesnt work like an analog amplifier, for which is what you said true.

in the digital domain there is no energy eaten up to "produce" low frequencies, because in the digital domain frequncies are not "produced" but only described.

there is always thinking in analog concepts transported over into the digital domain. digital works very, very different to analog.
But as soon as you get to fixed-point digital audio, headroom is a concern. Filtering out frequencies that are inaudible or undesirable can increase perceived volume. That is working in a similar way, is it not?
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Old 10-24-2015, 02:35 AM   #100
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But as soon as you get to fixed-point digital audio, headroom is a concern. Filtering out frequencies that are inaudible or undesirable can increase perceived volume. That is working in a similar way, is it not?
Right. If we think in terms of 8bit sound to keep the numbers manageable we have the numbers -127 to 127 to work with. If you have a waveform at -6db it will peak at the number 64 leaving only another 64 steps up to the max. In principle anyway (numbers ex rectum).
And low frequency waves are naturally slower and so on, so having something going at the very bottom will have a direct effect on how many bits you have to work with per sample.
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Old 10-24-2015, 03:22 AM   #101
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So, now my main 2 questions,

- Could this filter in any way degrade the sound? If it doesn't 'touch' the audible spectrum, and only cuts ultra-sonics, there wouldn't be any downside, other than CPU usage, to putting it between every plugin, right? If there is, I might just put it after my synths, because those are what cause the only aliasing I can hear.

It doesn't allow you to use it when running at 48k, btw, as of course it wouldn't do anything; the high frequencies would have already folded back. I think the default setting (21khz) is best? There's a 20k setting but I suppose it might affect the audible frequencies. I don't understand the point of "steep and reduced". Why cut any less than 100%?
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Old 10-24-2015, 04:43 AM   #102
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Mr. PC,
If it genuinely ONLY cuts inaudible frequencies then it should only be a benefit, so long as it does not gobble up your CPU cycles too. This is unlikely to be the case.
Usually the digital filters are trade offs and compromises on each other, for instance more accurate frequency response, less or no aliasing but with more ringing, versus poorer frequency response and no ringing. Best to look at this subjectively, see which you prefer.
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Old 10-24-2015, 07:30 AM   #103
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inaudible frequencies
This phrase really bothers me.
We think we know what this means.

I believe humans are capable of much higher "frequency" perception than scientific tools can yet accurately measure.

So, my feeling is that this phrase can be limiting, and ultimately restrictive to progress.

Just sayin'


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Old 10-24-2015, 07:41 AM   #104
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I believe humans are capable of much higher "frequency" perception than scientific tools can yet accurately measure.
The tool to measure it is the ears so all you need to do is ABX it and prove it. So even though I was just countering ABX testing in another thread, this is an ideal reason to use it and what it was made to prove/dispel.
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Old 10-24-2015, 08:16 AM   #105
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This phrase really bothers me.
We think we know what this means.

I believe humans are capable of much higher "frequency" perception than scientific tools can yet accurately measure.

So, my feeling is that this phrase can be limiting, and ultimately restrictive to progress.

Just sayin'
Hi Studio Tan,
Say we could actually hear (or even somehow register inaudibly) tones higher than 20khz, what could they possibly have to do with music and sound appreciation?
Are all frequencies that your ears can register pleasurable?
I know that in my forties I can still hear unpleasant high frequencies, well out of the range of music. Ask yourself this, even if you can hear it, is it relevant?

My hearing (and hopefully yours) is still capable of registering sounds well above those AUDIBLE in music programme. The accepted human hearing range of around 20-20khz covers a broad spectrum far beyond what is useful for desirable music and sound reproduction. This includes the audible harmonics, which may or may not be desirable.

The useful range is about 30hz-12khz (30hz produced cleanly -not contaminated with higher frequency output - is a really deep sound), anything else is just a nice bonus. Listen to pure tones of 12khz and 15khz these frequencies have nothing to with music appreciation, 18khz is downright unpleasant. If your tweeters could only reach 10khz with considerable roll off thereafter you would still hear all of the music.
The real difference between a quality treble unit and a poor unit is in the way it handles lower treble/upper midrange and mid treble. Extreme frequency reproduction is a red herring.

Further reading:
http://www.listenhear.co.uk/general_acoustics.htm

http://www.cco.caltech.edu/~boyk/spectra/spectra.htm
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Old 10-24-2015, 08:22 AM   #106
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Here's a view from the other side.

Note, this is posted without endorsement. Potential marketing bias in the piece, and the fact that correlation is not causation are duly noted

http://recordinghacks.com/articles/t...-beyond-20khz/
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Old 10-24-2015, 08:43 AM   #107
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Here's a view from the other side.

Note, this is posted without endorsement. Potential marketing bias in the piece, and the fact that correlation is not causation are duly noted

http://recordinghacks.com/articles/t...-beyond-20khz/
This is quite interesting.

I suppose when I made my vinyl example in another thread, I should have said...I can FEEL the difference, not HEAR!



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Old 10-24-2015, 08:55 AM   #108
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Here's another interesting bit from the "other side": https://en.wikipedia.org/wiki/Hypersonic_effect

Anyone who's heard gamelan live knows that speakers do not reproduce the same effect. I don't pretend to know why, or if it has any relevance to audio recording as we know it
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Old 10-24-2015, 10:10 AM   #109
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Originally Posted by Judders View Post
Here's a view from the other side.

Note, this is posted without endorsement. Potential marketing bias in the piece, and the fact that correlation is not causation are duly noted

http://recordinghacks.com/articles/t...-beyond-20khz/
While I applaud and agree with Mr Blackmer's vision of the future of audio he clearly states in his paper why we should not be concerned with the ultrasonics at this time.

First of all, of course some people will be able to perceive frequencies outside of the average range. When talking about biological entities you must have some reference point and generalization. Just as perfect vision is referred to as 20/20, this is based on the average individual and most certainly does not infer that there are not individuals with better than 20/20 vision. 20Hz-20kHz is the same description of human hearing. But reproduction and reality are not the same. We experience sub frequencies with our entire bodies, and perhaps the same can be said for all frequencies including ultrasonics.

But, the real kicker here is exactly what Mr Blackmer already stated. That reproduction of sound begins at capture (typically using microphones) and ends at the speaker striking the air. Both of which are rarely accurate in the sub and ultra frequency range even at the most expensive units on the market today.

Even professional systems cannot afford entirely pristine chains of equipment. Also, there are other things to consider in the real world such as durability and interference. So, you can complain that the metal grill is interfering with the frequency response of your uber microphone, but how much will you complain when the membrane is broken because you removed its protective grill???

Why concern yourself with ultrasonic frequencies when the average to even the most expensive speaker system cannot reproduce them accurately, and the majority of the human race will not be capable of hearing it in the first place?

We only really concern ourselves with higher sample rates because of the processes necessary during mixing. A single audio file at 44.1kHz will accurately capture its given input in the commonly accepted human range of hearing and likely exceed any playback system that a person will experience it on.

Perfect reproduction is not necessarily desirable either. A recording is not like experiencing the real thing. Compare how you feel listening to real musicians playing acoustic instruments in the same room with you to listening to a CD of the same. There is more to attending a live event than just how well it was recorded or mixed or mastered. You can see the expressions of the musicians as they become engulfed in their craft and you can hear and feel the reactions of the other spectators around you among a myriad of other stimuli such as smell, lighting, decor, etc.

In a recording however, an exact reproduction is not really what people want in most cases. They want the mix to be something that is not achievable in reality. The clarity of each instrument should be greater than what you experience live. If this were not true then there would be no such thing as multi-track recording. Also, there are expectations of sound such as compression. We've been listening to compressed audio for our entire lives. It makes that sound not only expected but desirable. There are other expectations based on genre. But, none of these expectations include ultrasonic frequencies.

Concerning using an ultrasonic filter between plugins: Quality plugins should internally address aliasing problems and filter out ultrasonics prior to output. I just compared Klanghelm's IVGI against Softube's Saturation Knob. While IVGI is not completely alias free the aliased frequency amplitudes generally remain below -45dBFS at any point that I could discern. Saturation Knob, OTOH, was more prone to aliased frequencies reaching nearly -20dBFS! According to an email I received from Mr Frenzel at Klanghelm, "SDRR (& IVGI) has a lot of trickery inbuilt to avoid aliasing." This was readily apparent in my test. Sat Knob also began noticable aliasing at much lower frequency than IVGI (about 3500Hz compared to 6800Hz). This was tested in a project with the samplerate set to 44.1kHz. The aliasing had already occurred prior to the plugin's output, so using an Ultrasonic filter afterward will not help in this test.
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Old 10-24-2015, 10:26 AM   #110
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While I applaud and agree with Mr Blackmer's vision of the future of audio he clearly states in his paper why we should not be concerned with the ultrasonics at this time...
I don't disagree with what you're saying, but many people from the "all you ever need is 16bit/44.1kHz" tribe seem to be saying that any capture or playback device that has a flattish frequency response from 20Hz to 20kHz should be indistinguishable from the real thing, because that is all we can hear or sense in any context. Which is nonsense, because people can distinguish recordings being played back through speakers from other sounds most of the time.
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Old 10-24-2015, 10:46 AM   #111
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Agreed. I know you didn't say I was, but for the record, I am not in the "16/44.1 is all you need" camp.

I mix at 44.1kHz. Why?
Because I have equipment and situation confines that preclude higher sample rates. My MOTU 828 mkII can only accept 8 channel ADAT expansion (via 8PRE) at 44.1/48kHz. I know newer units can use two fiber ADAT connections for higher sample rates but my legacy model cannot. So, right there our multi-tracked drum recordings are samplerate restricted. I don't use 48kHz because my end goal is not video, and I don't perceive the increased 3.9kHz to be worth the down-conversion problems that may occur at output worthwhile.

Besides, I usually only have time to work on mixing while I'm on the road for work (I travel a lot). And, I don't want to pack an interface around with me, so I use headphones and the laptop's onboard soundcard which is not capable of more than 44.1/48kHz either.

So, I am relying on the VST plugin creators to know when and how to deal with ultrasonics from within their plugins! Thus, only processes that require oversampling will be oversampled in my projects.

I'm a hobbyist anyway. NO ONE has ever said, "Oh man, that aliasing has f'd up your mix dude!" Most people recognize that my work is pro-sumer and are generally surprised at the quality. But, what I really want them to be excited about is the song and the performance! Lay people will comment on a horrible mix because it's distracting. But, they will not notice a good mix. They will just pay attention to the song and determine if they like it or not.

So, for me, the end all-be all is: Can I hear the aliasing or not?
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Old 10-24-2015, 10:56 AM   #112
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So, for me, the end all-be all is: Can I hear the aliasing or not?
I think that's a very sensible criteria
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Old 10-24-2015, 11:23 AM   #113
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Even though a lot of this thread has been a fairly interesting discussion, I can't help but juxtapose all of this concern with the fact that cassette tapes are currently making a real comeback. Apparently, the cassette factories are operating at full speed, getting them out as fast as they can. We're not talking about any metal tape or anything fancy here, just cheap cassette tape.

Why do I bring this interesting bit of news into this? Well, if you try to measure the quality of these tapes and talk in digital terms, you are getting 6 to 8 bits worth out of those tapes. The frequency range isn't much better -- again this is no kind of expensive tape. And the standard playback speed? Hahaha!

But ... you hear anyone complaining? Yeah, I didn't hear any serious complaints back in their heyday, either. Funny how it goes with music formats. Meanwhile, obsolescence of formats is moving right along at a pretty good pace.
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Old 10-24-2015, 11:32 AM   #114
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I think that's a very sensible criteria
Also: do you plan to do significant amounts of time stretching.

But yeah. Those two things. That pretty much solves this mystery to me.
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Old 10-24-2015, 11:33 AM   #115
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Even though a lot of this thread has been a fairly interesting discussion, I can't help but juxtapose all of this concern with the fact that cassette tapes are currently making a real comeback. Apparently, the cassette factories are operating at full speed, getting them out as fast as they can. We're not talking about any metal tape or anything fancy here, just cheap cassette tape.

Why do I bring this interesting bit of news into this? Well, if you try to measure the quality of these tapes and talk in digital terms, you are getting 6 to 8 bits worth out of those tapes. The frequency range isn't much better -- again this is no kind of expensive tape. And the standard playback speed? Hahaha!

But ... you hear anyone complaining? Yeah, I didn't hear any serious complaints back in their heyday, either. Funny how it goes with music formats. Meanwhile, obsolescence of formats is moving right along at a pretty good pace.
Yeah man, it's the new nostalgia kick for hipster bands.

In 20 years time it will be 96kbit/s MP3 encoded with "vintage" 90's software
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Old 10-25-2015, 02:46 AM   #116
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Hi Studio Tan,
Say we could actually hear (or even somehow register inaudibly) tones higher than 20khz, what could they possibly have to do with music and sound appreciation?
Are all frequencies that your ears can register pleasurable?
I know that in my forties I can still hear unpleasant high frequencies, well out of the range of music. Ask yourself this, even if you can hear it, is it relevant?

My hearing (and hopefully yours) is still capable of registering sounds well above those AUDIBLE in music programme. The accepted human hearing range of around 20-20khz covers a broad spectrum far beyond what is useful for desirable music and sound reproduction. This includes the audible harmonics, which may or may not be desirable.

The useful range is about 30hz-12khz (30hz produced cleanly -not contaminated with higher frequency output - is a really deep sound), anything else is just a nice bonus. Listen to pure tones of 12khz and 15khz these frequencies have nothing to with music appreciation, 18khz is downright unpleasant. If your tweeters could only reach 10khz with considerable roll off thereafter you would still hear all of the music.
The real difference between a quality treble unit and a poor unit is in the way it handles lower treble/upper midrange and mid treble. Extreme frequency reproduction is a red herring.

Further reading:
http://www.listenhear.co.uk/general_acoustics.htm

http://www.cco.caltech.edu/~boyk/spectra/spectra.htm
Are you saying we should just low-pass everything above 12k?

I'm actually trying that now, a low-pass filter at 15k with a 1.8 bandwidth. I don't know if it sounds better or not.

Also, I just realized GlissEQ can filter up to 96k, so I could use that instead of the TDR? And really if 12k+ is undesireable, I'll just slap a low-pass at 20k every time I EQ.

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Concerning using an ultrasonic filter between plugins: Quality plugins should internally address aliasing problems and filter out ultrasonics prior to output. I just compared Klanghelm's IVGI against Softube's Saturation Knob. While IVGI is not completely alias free the aliased frequency amplitudes generally remain below -45dBFS at any point that I could discern. Saturation Knob, OTOH, was more prone to aliased frequencies reaching nearly -20dBFS! According to an email I received from Mr Frenzel at Klanghelm, "SDRR (& IVGI) has a lot of trickery inbuilt to avoid aliasing." This was readily apparent in my test. Sat Knob also began noticable aliasing at much lower frequency than IVGI (about 3500Hz compared to 6800Hz). This was tested in a project with the samplerate set to 44.1kHz. The aliasing had already occurred prior to the plugin's output, so using an Ultrasonic filter afterward will not help in this test.
An Ultrasonic filter will only work at higher project sample-rates, because mixing at 44.1k means the aliasing will be baked in at the output of every plugin. Are you saying that Quality plugins will all include a built-in low-pass? But if that's not without at least 2x oversampling, it will be affecting the audible range.
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Old 10-25-2015, 03:27 AM   #117
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Are you saying we should just low-pass everything above 12k?

I'm actually trying that now, a low-pass filter at 15k with a 1.8 bandwidth. I don't know if it sounds better or not.
No, I feel you should try to capture as much AUDIBLE range as possible.

I was referring to the hearing process, how relevant ultra high frequencies are above a certain range of frequency. I am not making any kind of blanket suggestion of chopping off high frequency data!
Lots of good quality loudspeakers treble output starts to roll off gently after 12khz, significantly down at 15khz, though may extend above 20khz, albeit with little meaningful output.


I doubt you would hear a great deal of difference rolling off after 15khz. It could be interesting experimenting though.

A good quality FM signal can have lovely sound (not commercial heavily dynamically compressed pop channels though).
Most FM radio stations have poor (dynamically compressed) sound but if you have the option to listen to a quality bandwidth rich radio station like BBC Radio 3 on FM on a quality tuner (maybe not an option where you are) that gives you bandwidth of 30hz to 15khz and never sounds lacking at either extreme, especially on live concerts. I don't remember ever hearing fatiguing sound from Radio 3 at any point.
Most FM Radio is well below CD quality though.
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Old 10-25-2015, 04:08 AM   #118
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A good quality FM signal can have lovely sound (not commercial heavily dynamically compressed pop channels though).
Most FM radio stations have poor (dynamically compressed) sound but if you have the option to listen to a quality bandwidth rich radio station like BBC Radio 3 on FM on a quality tuner (maybe not an option where you are) that gives you bandwidth of 30hz to 15khz and never sounds lacking at either extreme, especially on live concerts. I don't remember ever hearing fatiguing sound from Radio 3 at any point.
That's funny, I never knew that, but I have listened to Radio 3 a lot, including on good systems, and I've always found the top-end lacking compared to CD. Like there is a slight dullness to the sound - I think it doesn't detract from the music they play because you get a similar effect from many concert halls.

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Old 10-25-2015, 04:25 AM   #119
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That's funny, I never knew that, but I have listened to Radio 3 a lot, including on good systems, and I've always found the top-end lacking compared to CD. Like there is a slight dullness to the sound - I think it doesn't detract from the music they play because you get a similar effect from many concert halls.

Interesting, I find CD can often sound fatiguing in a unique way that live sound does not, Radio 3 FM live concerts (for me) mimic live sound better than most CDs. This is probably down to production rather than format though.
Of course it will depend on the tuner and signal too.

Most sample instruments are not fatiguing to my ears either, and they do not have those bandwidth limitations.
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Old 10-25-2015, 08:12 AM   #120
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Aha! I just found my problem (of having Aliasing, even running at 192k and with the filter. I had to go to File-Project Settings and check "set project sample rate" to 192. For some reason the "request sample rate" in the audio preferences wasn't doing it on export.

Here are two samples, one with aliasing (that annoying low sound. I could have put an automated HPF on it, but this is the better solution because I like the sound of the other gritty stuff happening, just not the 'gru gru gru' of aliasing).

I was starting to think it was something other than aliasing, but I think this proves it. Much better than using some 3rd party wrapper, I think. Here are samples mixed in 48, 192, and 384

https://soundcloud.com/albert-mckay/...n-test/s-51VcF
https://soundcloud.com/albert-mckay/...-test2/s-boMVn
https://soundcloud.com/albert-mckay/...-test3/s-r34Je

Here it is in context (without the higher sample-rate).
https://soundcloud.com/albert-mckay/song-66/s-Yy3bH

So, I was running the project at 192, and hence heard reduced aliasing in the project file, but on export I guess reaper was mixing at the export sample rate, and hence the increased aliasing.
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