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Old 05-02-2018, 04:34 AM   #81
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^yes.
maybe you do not see the current way is not the only way...the fast fourier transformation is only 1 type--there are lots of transforms.
nyquist/shannon had theory-it's not the only 1.
have you ever wondered why you see a signal go above and below a center line?
and how a speaker works the way they do?
time for a reathink m8--but people wont do that because it means a total change of systems--= too expensive in some minds perhaps.
the real expenses right now is how a computer crunches numbers..sampling is expensive,but could become cheaper -imo.

nobody bothered answering questions-but let's reapeat
"how many samples does it take to create a 1hz sinewave,@ srate of 44100hz?"

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Old 05-02-2018, 05:09 AM   #82
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"how many samples does it take to create a 1hz sinewave,@ srate of 44100hz?"
For practical purposes 44100 samples are going to be used. But 2 would be enough. (But then technically you wouldn't really be using a sample rate of 44100 Hz but rather 2 Hz.)
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Old 05-02-2018, 05:12 AM   #83
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DSD and Super Audio CD were just Sony's (failed) attempt to create a harder to copy audio format. I don't see any benefits for an audio engineer or the end consumer unless the 6 channel surround audio option in SACD is used.

DSP on DSD streams is extremely tricky and so in many cases the signal is just converted back into PCM for processing anyway.
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Old 05-02-2018, 05:27 AM   #84
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For practical purposes 44100 samples are going to be used. But 2 would be enough. (But then technically you wouldn't really be using a sample rate of 44100 Hz but rather 2 Hz.)
^heh-yep,hence a call for "variable sampling system"..
@96khz we waste lots right now eh..
getting rid of any filtering would be ideal to minimize signal distortion.
filtering is smearing/masking incomming and outgoing signals right now..eh?
VSS can work 'in theory' - but manufacturers need to adjust to new ways if they are even considered 'worth it'.
i'm not arguing-just looking of ways to optimize all of what we have. =)
seeing so much waste in our world-sampling is no different.
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Old 05-02-2018, 05:30 AM   #85
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just looking of ways to optimize all of what we have.
Why? Manufacturers should totally change a perfectly working system just because you want to save disk space? Or what exactly are you looking to "optimize"? (In practice it wouldn't even save space that much, most signals have some content up to 20khz, which one should probably be able to capture in order to cater for the human hearing range.)
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Old 05-02-2018, 05:36 AM   #86
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^heh-yep,hence a call for "variable sampling system"..
@96khz we waste lots right now eh..
Already exists. It's called compression. You can even have it in two flavours: lossless or lossy...
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Old 05-02-2018, 07:43 AM   #87
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That is a terrible article.

The author doesn't make a case for anything. The whole thing is more of a pointless rant. He doesn't present any of the real reasons to record and mix at higher sample rates.

Also, he makes some stupid, or at least poorly worded, statements:
"but we’ve read plenty of studies revealing that sounds above the audible band affect what’s within the audible band, whether it be imparting brightness or a sense of air and room ambience that’s difficult to measure."

This may be a promoting statement for recording and mixing at higher sample rates, but he doesn't account in this statement for the fact that most sound reproduction systems can't reproduce frequencies higher than 17 kHz at audible, accurate levels. This includes high-end, studio-grade monitors. Or, the fact that many microphones cannot capture these frequencies well.

Additionally, the ADC is always going to filter the signal above 20 kHz, regardless what samplerate you choose. Maybe the really high-end converters have separate LPF's for each samplerate, but I doubt it. It's just going to sample the filtered audio signal at a faster rate. Some of the cheaper interface ADC's don't even clock the higher rates, they just do a conversion of the file on the fly. Regardless, the ultrasonic frequencies are getting filtered out and are not mixing together in the DAW.

Using my MOTU interface at 48 kHz, the signal level at the Nyquist Frequency (24 kHz) is about -20 dB from the signal at 20 kHz. So, you might get a few extra frequencies recorded, but they will not be appreciable. The above measurement is on the drum bus after 10 tracks have mixed together and the dB values are -52 dBFS @20 kHz and -72 dBFS @24 kHz. I had to crank the volume of the channel into a +10 dBFS clip to get the signal high enough to measure in SPAN. So, we are talking about frequencies at levels so low that they will not be heard, even on the best monitors by 15 yr olds.

"...just like any engineer would listen on an Auratone to make sure their mix will sound balanced on a car stereo."

Uhh, you check on a single Aurotone to see how it will sound in mono over something like a shopping store PA. Or, in stereo on single, small driver stereos. The car audio system is probably the highest fidelity system that most consumers are likely to listen to music on these days, especially the newer model cars. Junk car audio started going out of style in the late 1990's and most of today's models have very good systems in them, even the budget car models. The Aurotones may be excellent for determining how a mix translates to the average system, but I would not call car audio the "average system" today.

There are most definitely good & valid reasons for recording and mixing at higher samplerates, but the article did not make a case for any of them. I am surprised that MIX magazine published this article.
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Old 05-02-2018, 07:48 AM   #88
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For practical purposes 44100 samples are going to be used. But 2 would be enough. (But then technically you wouldn't really be using a sample rate of 44100 Hz but rather 2 Hz.)
Don't you mean 2 kHz?
You cannot reproduce any frequency less than half the samplerate, regardless how many points you want to use. I think 2 points @ 2 kHz samplerate would reproduce a 1 kHz sine wave, or maybe it's 3 points.

I don't know how many samples would be used @ 44.1 kHz to create a single cycle of a 1 kHz sine wave. Too much math for me.
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Old 05-02-2018, 07:49 AM   #89
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The author doesn't make a case for anything. The whole thing is more of a pointless rant. He doesn't present any of the real reasons to record and mix at higher sample rates.
He's probably talking about frequencies folding back into the audible band, that can happen with lower sample rates because the filter would need to be too steep to do it's job without artifacts - because there isn't enough wiggle room above 20k for it to do so. Do I think this manifests itself the final result of most music in such a way that we need to worry much, no I do not.

It's very easy to demonstrate with lab tests AFAIK but we don't run across it much in real world mixes - if we do, then I'm happy to listen to publicly released mixes anyone wants to present (instead of lab style demos).
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Old 05-02-2018, 07:55 AM   #90
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Don't you mean 2 kHz?
Bri1 did write 1 hz in his post, not 1 khz. Why such a low frequency, I can not know. Bri1's thought processes are pretty much impossible to understand.
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Old 05-02-2018, 07:59 AM   #91
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The strawman argument based on the claim of perception of sound above the range of hearing (and your loudspeakers too BTW) is just silly. Some of this may be innocent chasing of a red herring I suppose. Pretty awkward to publish though.

An argument against using HD formats might be that it's only chasing the last few percentage points to perfection and stuff like quality of mix is much more telling.

On the other hand, you can pretty much eliminate any variables in the recording/delivery format with this and not have to worry about outliers. In the same way that we don't compress images to make them viewable on a Commodore64, there's little reason to reduce anything down to SD with the storage device sizes we have now. For a master copy aimed at the home theater listener anyway. The phone listening crowd is kind of still in Commodore64 territory!

There ARE outliers. Surprisingly so with some instrument/synth plugins actually. And the bit where average DA/AD converters run cleaner at HD. Why not just eliminate any variable in the recording format (and any need to qualify the system from job to job) right? Who doesn't want that? The consumer delivery format has been 24/96 FLAC for a good long time now.

I think the only guy who doesn't want that is the guy who converted his CD collection to low bitrate mp3 (didn't know to check the default settings in iTunes) and sold the CDs. "I can't hear any difference! Lalalalalalalalalalala..." Sounds more like the sour grapes defense.

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Old 05-02-2018, 08:55 AM   #92
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the problem is-we have to catch sound at fixed rates which chops a continuous signal into tiny snippets or chunks of energy: this can result in signal distortions.
when we catch audio at 44100 snippets per sec- the result can trick the brain into decoding the vibrational image into what we can describe acceptably as 'sound':

simplifying stereo audio down to mono,then a single sided waveform {rectified signal} we can see a wave being modulated by amplitude over time (am) but not 'frequency modulation' over time (fm) ---reaper and other softwares work at 'fixed sample rates'- so long the programme can encode + decode the amount of samples per sec==all is well.
There are no "chops" or "snippets". There are only data points at fixed time intervals. When those data points are converted back to an electrical signal, an identical electrical signal will be created for all frequencies at and below the Nyquist Frequency. Unless, of course, faulty converters being used, ADC on the way in or DAC on the way out. But, that does not indicate a problem of the digital file, only the hardware used to convert the signal.

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^this may be good advice-- the less convertions-the better--- after all,we are just converting- light energy into sound energy__ which then gets transmitted via air or speakers back to light
more samples=more points of pleasure!!
we must consider speed of circuitry,speed of sound + speed of light + speed of consciousness.
it's all light-light is all that is. =)
WHat the hell are you talking about??? There is no light energy, AT ALL. Except for the Optical circuit of an analog Opto Compressor. Or, if you are transmitting you data via fiber-optic. Otherwise, it's audible air vibrations converted to electrical signals, converted to digital data, then in reverse.

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I strongly encourage you to acknowledge a straight line can not be a curved line..
1 is trying to reapresent the other-different.

closer the buildings-the less gaps there are to fall into-- only thing is- computers put huge gusts of wind to fight as well as 1 tries to hop from skyscraper to skyscraper!

just optimize audio for optimal amount of samples for recording + playback-- and lets all move on. i think 44100 is barely scraping the barrel for *professional users.*
As others stated. This doesn't hold any truth to audio samplerates, or how audio is digitally captured or reproduced.

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M8- your totally missing the point! a compressor does not give a shit what your hearing--it works on info per sec- if a compressor has only 1 dot per sec to work with--- well--get the point m8.? =hardwork.
Organically a sound can appear at any moment- if a recorder is not capturing at that precise 'point' that info is lost=simplez.
I have no idea what point you're trying to make here.

This was well said:
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Bri1, no offence, but it's very apparent that you don't understand the maths that's behind signal processing.

Did you even care to watch Monty's video?

Do you understand that any signal IS a sum of sinusoidal waves? It's not just a convenient representation.

Do you understand that if you have have an analog signal, properly band-limited below the Nyquist frequency, you can sample it and convert it back to analog again, and obtain the EXACT same signal? There's no approximation, no distortion, no "steps" or "straight lines". It's not "tricking our brain". You truly have the same signal again. The only caveat is that the filters can't be ideal filters, so you need some headroom and start filtering below the Nyquist frequency. That's why we don't use 40kHz but a bit more. (And, to be fair, a typical adult human can't hear anything above 16kHz already).

Do you understand that a higher sampling rate can only extend the frequency of the processed signal, and CAN'T improve anything at the lower frequencies? This means that the ONLY thing that you gain by going from 48kHz to 96kHz is encoding frequencies that only your cat or dog can hear. (And possibly introducing intermodulation distortion when you listen on cheap hardware.)
Bri1, it is obvious that you do not fully understand the concepts behind audio signal capturing. Instead of making strange and misleading statements about things you don't grasp fully, I recommend you ask questions for clarity instead. However, many people in this thread have recommended the fantastic video by Monty at Xiph.org which very clearly educates us all on sampling theory and Nyquist frequency. I too, recommend that you go watch that video several times before engaging in these sample rate conversations. What you're currently doing is spreading uninformed, misconceptions about audio samplerates. That is called MISINFORMATION, and is the root of continued ignorance for humanity.

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70khz seems a sensible setting as a new standard.
ditch the rest for hq audio.
the rest is for other types of communications... both above and below the nyquist rates...
most people not need to know what goes on both above and below at them rates other than military or medicals.
we work in a fixed range for 'audioists' =so lets optimize it! please. =)

*think* optimizing the lower octaves is essential- the ranges of 38hz>1khz>5khz upto around 18khz being most sensitive to human ears.
a lot of eq's and plugs are not optimized for lower end of octaves eh..we mainly work below 7khz eh...?
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a variable sampling system would be different.modern. but if people do not feel a need to change or fix what seems to work,then all is good eh..
sampling could be seen as a fairytale where the porridge is either a little too hot,too cold--or just right.
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^yes.
maybe you do not see the current way is not the only way...the fast fourier transformation is only 1 type--there are lots of transforms.
nyquist/shannon had theory-it's not the only 1.
have you ever wondered why you see a signal go above and below a center line?
and how a speaker works the way they do?
time for a reathink m8--but people wont do that because it means a total change of systems--= too expensive in some minds perhaps.
the real expenses right now is how a computer crunches numbers..sampling is expensive,but could become cheaper -imo.
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^heh-yep,hence a call for "variable sampling system"..
@96khz we waste lots right now eh..
getting rid of any filtering would be ideal to minimize signal distortion.
filtering is smearing/masking incomming and outgoing signals right now..eh?
VSS can work 'in theory' - but manufacturers need to adjust to new ways if they are even considered 'worth it'.
i'm not arguing-just looking of ways to optimize all of what we have. =)
seeing so much waste in our world-sampling is no different.
In all of these you are enter fantasy land and derailing the conversation.
Variable sample rates are only used for audio FILE COMPRESSION. The CPU is not "decoding" a PCM WAV or AIFF file. They are not compressed files. Any form of file compression requires de-compression and increases the workload for the CPU. Recording using a lossless compression file type will save you some storage space, but there is no advantage to mixing with compressed file types.
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Old 05-02-2018, 08:58 AM   #93
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Bri1 did write 1 hz in his post, not 1 khz. Why such a low frequency, I can not know. Bri1's thought processes are pretty much impossible to understand.
My bad.
I assumed he said 1 kHz since many tests are done with that tone.
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Old 05-02-2018, 09:26 AM   #94
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fantasy--ok.
thing with science and music is both are looking at reapeatable measurable data points-- the realer aim of both is to adventure.. sometimes into the 'unknowns'.
people really get stuck into rigid thinking/ways and scoff at those that even look to tread new ground.
old ways are not always the better ways- we are using theory+tech+code that pre dates the birth of some actual users here...time to upgrade..no?yes!

all is light.... but as people we only see a tiny tiny fraction of the light--the band we call 'visible light'... the electromagnetic spectrum is much much greater definition than that!
personally,i think current sampling system is not true fm+wastefull of reasource-- but if that appears as fantasy--then so be it.
am not about to change the world-or want to. but if we can gain anything together >> i'm all ears for that.

btw- considering we look at sampling-- we must also look at the 'sources of distortion/noise' -
try this vid>

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Old 05-02-2018, 09:45 AM   #95
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fantasy--ok.
thing with science and music is both are looking at reapeatable measurable data points-- the realer aim of both is to adventure.. sometimes into the 'unknowns'.
people really get stuck into rigid thinking/ways and scoff at those that even look to tread new ground.
old ways are not always the better ways- we are using theory+tech+code that pre dates the birth of some actual users here...time to upgrade..no?yes!
In theory it sounds great, but there's one thing I'm having a hard time getting my head around, if this is actually describing a variable sample rate method of sampling, rather than an encoded file format:

How does the sampler know how many samples are needed to reconstruct a wave for any given time period? Wouldn't it need to look into the future to know what sample rate was needed?
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Old 05-02-2018, 10:32 AM   #96
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Wouldn't it need to look into the future to know what sample rate was needed?
not @ fixed rate of 60khz for eg:
because we deal we time-seconds--we use 'realtime' as 60bpm.
we do not need to reaproduce any hz below 20-we do not need (roughly) above 18khz - in that time period the samples would get shifted to positions according the where they are needed the most to reaconstruct waveforms of specific frequencies.

otherwise we could set to capture a trillion samples per sec say > to be sure!!

we do not make 20hz music all day-- but sometimes 18khz cycles are mixed into 20hz cycles--shifting of samples would help construct transitions of frequency rates much smoother for plugins to read as transient data.

when we create a pulse wave today @ 44100 samples-we get harmonics caused by just 1 single sample transition point... there is a huge waste of samples here constructing the pulse- extra samples at that single samples transition point may help to not produce higher harmonics >> that get folded back into our audio spectrums.

even Justin realizes sometimes too many options are not too helpfull. even just agreements of 1 sampling rate suiting all scenerios would be a landmark decision--lol. forget fantasies.
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Old 05-02-2018, 10:52 AM   #97
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not @ fixed rate of 60khz for eg:
because we deal we time-seconds--we use 'realtime' as 60bpm.
we do not need to reaproduce any hz below 20-we do not need (roughly) above 18khz - in that time period the samples would get shifted to positions according the where they are needed the most to reaconstruct waveforms of specific frequencies.

otherwise we could set to capture a trillion samples per sec say > to be sure!!

we do not make 20hz music all day-- but sometimes 18khz cycles are mixed into 20hz cycles--shifting of samples would help construct transitions of frequency rates much smoother for plugins to read as transient data.

when we create a pulse wave today @ 44100 samples-we get harmonics caused by just 1 single sample transition point... there is a huge waste of samples here constructing the pulse- extra samples at that single samples transition point may help to not produce higher harmonics >> that get folded back into our audio spectrums.

even Justin realizes sometimes too many options are not too helpfull. even just agreements of 1 sampling rate suiting all scenerios would be a landmark decision--lol. forget fantasies.
So you're talking about a compression method for a file container, and not a new method of sampling?
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Old 05-02-2018, 10:54 AM   #98
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fantasy--ok.
thing with science and music is both are looking at reapeatable measurable data points-- the realer aim of both is to adventure.. sometimes into the 'unknowns'.
people really get stuck into rigid thinking/ways and scoff at those that even look to tread new ground.
old ways are not always the better ways- we are using theory+tech+code that pre dates the birth of some actual users here...time to upgrade..no?yes!

all is light.... but as people we only see a tiny tiny fraction of the light--the band we call 'visible light'... the electromagnetic spectrum is much much greater definition than that!
personally,i think current sampling system is not true fm+wastefull of reasource-- but if that appears as fantasy--then so be it.
am not about to change the world-or want to. but if we can gain anything together >> i'm all ears for that.

btw- considering we look at sampling-- we must also look at the 'sources of distortion/noise' -
try this vid>
Bwahhahahaha! LOL!

Man you really believe this dude? Why, because he's on YouTube?
He successfully demonstrates and explains how he doesn't understand how an AC motor works within the first 5 minutes.

AC motors do not run on Radio Frequency. How do I know this? Let's see... I spent 5 years going through an electrical apprenticeship learning about AC and DC theory along with electronics. Now I work for an electrical engineering company. I go to industrial facilities every week that utilize AC and DC motors. I'm not as knowledgeable as the EE's I work with every day, but I know enough to know that Gerard Morin is a quack. AC motors run at the electrical distribution frequency which is 60 Hz in the USA and generally 50 Hz in Europe and many other places. Those frequencies are WAYYYY below radio frequencies. Radio is from 3 kHz - 300 GHz. Maybe Mr Morin is confusing RPM with electrical AC frequency since the average AC motor runs at 1500 RPM, which would still be below radio frequency if confused for 1.5 kHz.

That video is from a delusional man who doesn't know what the hell he's talking about. 10 minutes of research would tell you that you should not believe anything that guy has to say.

Besides, we all have to work within the restraints of the technology that's available today.
PERIOD.
Sure, there may be some theoretical, future tech that will work way better than we have today. But that tech is not COMMERCIALLY AVAILABLE to the masses yet. Until then, IT IS FANTASY for all practical purposes.

I applaud your enthusiasm, Bri1! But, some of us have to live and work in the real world to accomplish things TODAY. Not tomorrow or 10 years from now.

We are all made of star-stuff, which is fine and well. But, we have to be able to define different frequency ranges so that we can discuss and calculate it in a way our tiny brains can comprehend. Being made of "star-stuff" doesn't mean that everything is light. It just means that every particle we can perceive and measure appears to have been created from within stars according to our current scientific models and theories. All of which are way outside of what information is necessary to decide which samplerate to record, mix or distribute digital audio files at.
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Old 05-02-2018, 11:04 AM   #99
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when we create a pulse wave today @ 44100 samples-we get harmonics caused by just 1 single sample transition point... there is a huge waste of samples here constructing the pulse- extra samples at that single samples transition point may help to not produce higher harmonics >> that get folded back into our audio spectrums.
Pulse and Square waves are not naturally occurring. A speaker cannot produce a pulse or square wave. Yes, sending that signal to a speaker makes a distinctive sound, but it is not a square wave that comes out of the speaker. There simply is no such thing. Only a computer generates a pulse or square wave.

ANY flat-top wave is distortion, by definition, can only be created in the audible spectrum through harmonics. I guess you could say that some of the ultrasonic harmonics would need to be present to recreate a perfect square/pulse wave, but the fact remains that neither of those waves exist in the physical world of sound waves traveling through air to get to our ears.
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Old 05-02-2018, 11:08 AM   #100
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Originally Posted by insub View Post
ANY flat-top wave is distortion, by definition, can only be created in the audible spectrum through harmonics. I guess you could say that some of the ultrasonic harmonics would need to be present to recreate a perfect square/pulse wave, but the fact remains that neither of those waves exist in the physical world of sound waves traveling through air to get to our ears.
OT: Some will disagree with me but the waveform creates the harmonics instead of the harmonics creating the waveform. The reason a "true" square wave doesn't exist is that it's impossible to go from 0 to "top" instantaneously without time passing among other reasons I don't fully understand.

That's the cool thing about an oscilloscope and an FFT, we can see distortion in the form of the harmonics that are inevitably created by "distortion" of the waveform even before we can visibly see the change in the waveform itself.
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Old 05-02-2018, 11:08 AM   #101
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@ insub - bruv,i'm just throwing shyte out there into the ether-sueme =)
oh? 3khz > 300ghz fits our audio range?
as suggested,change would be really expensive for some and totally not welcomed-- but change is the only constant i believe--lets go with any flow that may come along eh if it floats us..
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Old 05-02-2018, 11:15 AM   #102
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Pulse and Square waves are not naturally occurring.


but the fact remains that neither of those waves exist in the physical world of sound waves traveling through air to get to our ears.
oh-lol so here you admit none of this is natural--this is my point bruv!!
fake-faked and faking it more seems to be the order of daw+sampling reality today!! lolz.
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Old 05-02-2018, 11:17 AM   #103
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Sometimes I have a hard time telling the difference between the malicious humor videos and true delusion or Dunning-Kruger effect at play.

Malicious humor like the "microwave your iPhone to kill germs" or "enter rm -rf / to clear your browser history" delivered deadpan.

I suppose, to play devils advocate: Motors can produce RF noise. Or so I've been told in explanation to how the heck a leslie was jamming my wifi. Maybe the guy heard something along those lines and his lack of background led into fantasy from there?
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Old 05-02-2018, 12:32 PM   #104
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Originally Posted by serr View Post
Sometimes I have a hard time telling the difference between the malicious humor videos and true delusion or Dunning-Kruger effect at play.
Definitely, I am guilty of the latter in this statement.

The more I learn about things I thought I really knew a lot about. The more I realize how ignorant I am about everything.

[I know you were commenting on the video. I think some people have gathered a lot of information (& misinformation) and compiled it in their minds in a way that only makes sense to them. Therefore, they come to ridiculous conclusions which they honestly believe are 100% correct. It is unfortunate for the uneducated person they encounter, because the delusional one knows enough facts to sound convincing. And, thus, we spin into idiocy. The internet is such a great source for information & misinformation. The more that's available, the harder it is to separate the wheat from the chaff.]
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Old 05-02-2018, 12:39 PM   #105
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Originally Posted by Bri1 View Post
oh-lol so here you admit none of this is natural--this is my point bruv!!
fake-faked and faking it more seems to be the order of daw+sampling reality today!! lolz.
Bri1, I get that you are trying to be humorous. But, some people will not be able to tell the difference between you humor-isms and truth. They will become confused and leave this thread less educated than when they entered.

The reality is that we have higher fidelity in capturing and reproducing audio than ever in recorded history at 44.1 kHz sample or higher, and all that fidelity is within reach of the average person living in greater than a third-world area. So, regardless which samplerate, ≥ 44.1 kHz, you choose to use, the modern DAW is capable of more accurate recordings of audio than ever before.
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Old 05-02-2018, 01:46 PM   #106
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Tesla woo - check
Free Energy conspiritardation - check
Perpetual Motion - Check!

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Old 05-02-2018, 02:45 PM   #107
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Wow! The loonies are out today.
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Old 05-02-2018, 03:45 PM   #108
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Originally Posted by insub View Post
The reality is that we have higher fidelity in capturing and reproducing audio than ever in recorded history at 44.1 kHz sample or higher, and all that fidelity is within reach of the average person living in greater than a third-world area. So, regardless which samplerate, ≥ 44.1 kHz, you choose to use, the modern DAW is capable of more accurate recordings of audio than ever before.
Exactly. We live in a golden age of audio now.

That's a good way you stated it and it kind of makes a great argument for sticking with 44.1k. But since HD came along came along and cleaned up a couple outliers and made converter circuits easier to build, it still seems wrong to consider going back!

Maybe someone will find a way to hide some inaudible data for an easter egg type thing or something.
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Old 05-02-2018, 04:07 PM   #109
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The strawman argument based on the claim of perception of sound above the range of hearing (and your loudspeakers too BTW) is just silly. Some of this may be innocent chasing of a red herring I suppose. Pretty awkward to publish though.

An argument against using HD formats might be that it's only chasing the last few percentage points to perfection and stuff like quality of mix is much more telling.

On the other hand, you can pretty much eliminate any variables in the recording/delivery format with this and not have to worry about outliers. In the same way that we don't compress images to make them viewable on a Commodore64, there's little reason to reduce anything down to SD with the storage device sizes we have now. For a master copy aimed at the home theater listener anyway. The phone listening crowd is kind of still in Commodore64 territory!

There ARE outliers. Surprisingly so with some instrument/synth plugins actually. And the bit where average DA/AD converters run cleaner at HD. Why not just eliminate any variable in the recording format (and any need to qualify the system from job to job) right? Who doesn't want that? The consumer delivery format has been 24/96 FLAC for a good long time now.

I think the only guy who doesn't want that is the guy who converted his CD collection to low bitrate mp3 (didn't know to check the default settings in iTunes) and sold the CDs. "I can't hear any difference! Lalalalalalalalalalala..." Sounds more like the sour grapes defense.
I get what you are saying but this is worth a look.
https://www.youtube.com/watch?v=YgEjI5PZa78
I don't think many would doubt this guy's credentials and there is an interesting practical experiment too.
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Old 05-02-2018, 04:09 PM   #110
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Maybe someone will find a way to hide some inaudible data for an easter egg type thing or something.
Surely that can be done. Image steganography is very well established.
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Old 05-02-2018, 04:13 PM   #111
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Originally Posted by martifingers View Post
I get what you are saying but this is worth a look.
https://www.youtube.com/watch?v=YgEjI5PZa78
I don't think many would doubt this guy's credentials and there is an interesting practical experiment too.
The next day he released his album in WAV format

The test was interesting too. As I remember, I got the same ones right that his assistant did, which made me wonder if the source material has a lot to do with whether one can hear any difference or not, so a % score of how many you guess correct on the test may not be what to look for, but rather what % of people guess correctly for the same sample. Had a nice little chat with him in the comments section about it.
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Old 05-02-2018, 04:42 PM   #112
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which made me wonder if the source material has a lot to do with whether one can hear any difference or not, so a % score of how many you guess correct on the test may not be what to look for
Couple things...

If the order of the correct choice was randomized and you always chose those then yes, and.... source will *always* play some part in the fray. That's actually one reason to use the 'better' format because it's far easier to just play it safe than try to understand just when and where it may show up in which context of which mix/source (I'm talking 320 kbs vs 44.1k not 96k vs 44.1k). One doesn't need to explicitly hear it in some random test for that to be potentially beneficial but it goes south once people start bloating the significance into night and day - I suggest they actually observe the difference between night and day as a reminder.

Of course as far as the statistical relevance goes, I'll be impressed when someone gets it 100% of the time every time or 98 out of 100 instead of 4 out of 5 or a lucky one off 5 out of 5, until then, they are truly on the fringe of reliably hearing it, and statistically irrelevant by the very virtue of the fact they missed one or more.

Additionally, this always tends to center around high frequencies in the top 10% of the scale which makes some sense but I now find odd because there are as many places nuance lives anywhere in the content. What I mean is we have people arguing for days about who might be able to hear some change at 19k when they couldn't tell the difference between small but larger compressor changes at any frequency to save their life.

People waste so much time on this for no reason other than fear of the insignificant, the older I get the more foolish I feel for the times I myself fell for that mind trick.
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Old 05-02-2018, 05:01 PM   #113
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Couple things...

If the order of the correct choice was randomized and you always chose those then yes, and.... source will *always* play some part in the fray. That's actually one reason to use the 'better' format because it's far easier to just play it safe than try to understand just when and where it may show up in which context of which mix/source (I'm talking 320 kbs vs 44.1k not 96k vs 44.1k). One doesn't need to explicitly hear it in some random test for that to be potentially beneficial but it goes south once people start bloating the significance into night and day - I suggest they actually observe the difference between night and day as a reminder.

Of course as far as the statistical relevance goes, I'll be impressed when someone gets it 100% of the time every time or 98 out of 100 instead of 4 out of 5 or a lucky one off 5 out of 5, until then, they are truly on the fringe of reliably hearing it, and statistically irrelevant by the very virtue of the fact they missed one or more.

Additionally, this always tends to center around high frequencies in the top 10% of the scale which makes some sense but I now find odd because there are as many places nuance lives anywhere in the content. What I mean is we have people arguing for days about who might be able to hear some change at 19k when they couldn't tell the difference between small but larger compressor changes at any frequency to save their life.

People waste so much time on this for no reason other than fear of the insignificant, the older I get the more foolish I feel for the times I myself fell for that mind trick.
Yeah, I agree, but the game of diminishing returns is part of art, I think.

Try telling an oil painter that they should just use cheaper paint because that 2% gain in quality will go unnoticed by the great unwashed masses.
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Old 05-02-2018, 05:07 PM   #114
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Try telling an oil painter that they should just use cheaper paint because that 2% gain in quality will go unnoticed by the great unwashed masses.
I get it... that's the crux of it, inspiration vs reality vs detail 99% will never see/hear. I wish I could remember my sig from a few years ago that basically stated...

"It truly doesn't matter if the 'mojo' is real, a figment of your imagination or both, use it in any way you can because worrying about where it lives is destructive, and using it is rewarding."

However, when we take the creator hats off and put the lab coats on, we sort of need to adhere to the lab coat rules because anything professed in that conversation needs to be confirmable, so that our gear actually works LOL. Nail how to behave and what to expect when wearing either, everything gets simpler and easier.

To be fair, I went on and on about my last project and spending lots of time adding elements most will never hear until they've heard the song at least 20 times, maybe 100 times so I'm all for nuance, just trying to frame the proper contexts/importance.

/ramble
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Old 05-02-2018, 05:15 PM   #115
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karbo:

I've come to the conclusion that music is ultimately neither an art nor a science, but a craft.
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Old 05-02-2018, 05:22 PM   #116
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karbo:

I've come to the conclusion that music is ultimately neither an art nor a science, but a craft.
Writing, or improvising, music is art by almost any definition.

Mixing could certainly be argued to be a craft though.
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Old 05-02-2018, 05:28 PM   #117
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Writing, or improvising, music is art by almost any definition.

Mixing could certainly be argued to be a craft though.
I mostly write and improvise - inspiration certainly (hopefully at least some of the time), but that's nothing without craft.
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Old 05-02-2018, 05:32 PM   #118
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karbo:

I've come to the conclusion that music is ultimately neither an art nor a science, but a craft.
I'm completely on board with that.

https://youtu.be/Xm-izoJZ-Bw?t=1430
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Old 05-02-2018, 05:34 PM   #119
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I get it... that's the crux of it, inspiration vs reality vs detail 99% will never see/hear.
I think the point is that the artist doesn't care what the unwashed masses will notice. They aren't making art for them, and that extra 1% is for the satisfaction of the artist.
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Old 05-02-2018, 05:36 PM   #120
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I mostly write and improvise - inspiration certainly (hopefully at least some of the time), but that's nothing without craft.
Of course, but all art involves craft, whether painting, sculpting, writing or whatever. That doesn't mean that none of those things may be classed as art.
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