What bit depth and sample rate should I use?

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Jarno
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Re: What bit depth and sample rate should I use?

Post by Jarno » Thu Jan 22, 2015 3:52 pm

jono not bono wrote:So, what are the actual pros and cons of recording at higher sample rates
Pros:
1. In case you have non-oversampling A/D and D/A coverters with poor anti-aliasing/anti-imaging filters the sound quality may improve slightly (flatter freq response near 20kHz and/or less aliasing distortion).
2. Some DSP algorithms may introduce aliasing distortion if not implemented correctly. So if you have ill-behaving plugins you can move artifacts they are creating to inaudible frequency range by using higher sample rates (and then removing them completely by downsampling with good quality sample rate converter).

#1 hould be non-issue nowadays (for last 10 - 20 years now depending on if you use pro or consumer grade converters). However #2 may still be valid point today.

Cons:
1. Storage space requirements
2. Processing power requirements
3. Some D/A converters may perform poorly with high sample rates
4. Introducing ultrasonic content into your analog audio chain (anything after D/A) may cause unwanted intermodulation distortion.
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Re: What bit depth and sample rate should I use?

Post by Paul Bryce » Thu Jan 22, 2015 4:42 pm

Woodcrest Studio wrote:15 IPS / 320nWb/m
That sounds very familiar :)
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Re: What bit depth and sample rate should I use?

Post by swamptone » Thu Jan 22, 2015 4:47 pm

Paul Bryce wrote:
Woodcrest Studio wrote:15 IPS / 320nWb/m
That sounds very familiar :)

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Re: What bit depth and sample rate should I use?

Post by jono not bono » Thu Jan 22, 2015 4:59 pm

Jarno wrote:
jono not bono wrote:So, what are the actual pros and cons of recording at higher sample rates
Pros:
1. In case you have non-oversampling A/D and D/A coverters with poor anti-aliasing/anti-imaging filters the sound quality may improve slightly (flatter freq response near 20kHz and/or less aliasing distortion).
2. Some DSP algorithms may introduce aliasing distortion if not implemented correctly. So if you have ill-behaving plugins you can move artifacts they are creating to inaudible frequency range by using higher sample rates (and then removing them completely by downsampling with good quality sample rate converter).

#1 hould be non-issue nowadays (for last 10 - 20 years now depending on if you use pro or consumer grade converters). However #2 may still be valid point today.

Cons:
1. Storage space requirements
2. Processing power requirements
3. Some D/A converters may perform poorly with high sample rates
4. Introducing ultrasonic content into your analog audio chain (anything after D/A) may cause unwanted intermodulation distortion.
Thank you for the advice. I use a MOTU HD192. Unfortunately I have not yet been fortunate enough to try anything out of this price bracket (yet). I'm just wondering how the HD 192 fares in quality and whether it would be a good idea to Record at higher sample rates? Also, I'm sorry to ask more questions but you seem very knowledgeable, could you explain what 32bit Float is and why I would use it over 24bit (I apologise if this is hijacking the thread but it's all important stuff to know about).

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Re: What bit depth and sample rate should I use?

Post by Piirka » Thu Jan 22, 2015 5:45 pm

Recording in 32-bit floating point is only benefitial if you are going to do a lot, actually a LOT, of offline processing.

Online processing (real time mixing) is always executed in the 32-bit floatig point realm, no matter the bit-lenght of your source material.

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Re: What bit depth and sample rate should I use?

Post by Jarno » Thu Jan 22, 2015 9:19 pm

jono not bono wrote:I'm just wondering how the HD 192 fares in quality and whether it would be a good idea to Record at higher sample rates?
Sorry, I can't tell for sure. I have no experience on HD 192. But it's modern pro/semi-pro unit and should be OK and not having problems of early cheap systems. According this article: http://www.soundonsound.com/sos/oct03/articles/motu.htm HD 192's converters are 128x oversampling. This should eliminate all aliasing/imaging problems in A/D/A conversion. It's probable you won't gain anything by using higher sample rates (when it comes to A/D/A conversion).
Piirka wrote:Recording in 32-bit floating point is only benefitial if you are going to do a lot, actually a LOT, of offline processing.
Or LOT of bouncing.
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Re: What bit depth and sample rate should I use?

Post by UltimateOutsider » Thu Jan 22, 2015 9:21 pm

Jeff Hayat wrote:
UltimateOutsider wrote:The problem is that this also means that you're getting less protection from those 256 samples than you were getting at 44.1K.
Huh?!?!?
Don't know if you were disagreeing or just not understanding what I was trying to say.

Sample buffers provide protection against audio glitches while your computer's processing audio. A 256-sample buffer provides less protection from these glitches at a 48kHz sample rate than it does at 44.1kHz, both because it represents a shorter amount of audio time at the increased rate, and because your computer has to work harder to deliver audio at that higher rate. So even though you may experience less latency while using a sample buffer of a given size at a higher sample rate, you're also running a greater risk of getting clicks, pops, and dropouts in your audio if you don't increase the size of the buffer to scale with the increased sample rate.

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Re: What bit depth and sample rate should I use?

Post by Jeff Hayat » Thu Jan 22, 2015 11:27 pm

UltimateOutsider wrote: Don't know if you were disagreeing or just not understanding what I was trying to say.
Neither. I was basically saying, "WTF are you talking about?!?!"
UltimateOutsider wrote: Sample buffers provide protection against audio glitches while your computer's processing audio.
WTF are you talking about?!?! Protection? You mean like Cosa Nostra? That kind of protection?

Maybe you know what you are talking about, but you just don't know how to say it.

First off, I assume when you say sample buffers, you mean audio buffers. Do you know what an audio buffer is? I'll explain it. An audio buffer is a reserved segment of memory. It is used to hold audio data to compensate for processing delays, aka latency. When the sample rate is higher, the audio data will pass through the buffer faster, and then the latency will be lower. The smaller the buffer, the less time it takes for audio data to pass through it, and the more processing power/CPU work is needed. So if your processor is not up to the task, you may have to raise your buffer/increase latency, or work at a lower sample rate in order to keep your buffer and latency down. There is no 'protection' going on.

Cheers.

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Re: What bit depth and sample rate should I use?

Post by Early21 » Fri Jan 23, 2015 1:10 am

I think preserving more detail that we can't hear (above a sample rate of 44.1K or 44K) is equivalent to recording more ultraviolet and infrared on our videos. Let's not forget that we can't hear digital. We can only hear actual sound pressure waves coming out of speakers, which are the result of conversion of digital to analog. They do not come out with stair steps. I'm using 44.1, which takes less space.

I do see an advantage to using 24-bit depth, as it gives you more headroom than you need. It takes a third more disk space, but I can't imagine that's a problem for anyone. You can't make a mistake in input levels (too low I mean) and you can't record a signal that's noisy (internally, that is - you can certainly record a signal that has noise in the chain). You just don't have to think about it. At 16-bit, you have to give it a little thought. At least in my experience. I noticed a difference when I switched from 16 to 24.

So -- 44.1K or 48K and 24-bit. There you go. Use dither on mixdown to 16-bit. Or don't. In truth, nobody will hear the difference.
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Re: What bit depth and sample rate should I use?

Post by Elektrobolt » Fri Jan 23, 2015 4:55 am

Digital audio macho guys chest it out.

LOL, too funny! :D
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Re: What bit depth and sample rate should I use?

Post by fretthefret » Fri Jan 23, 2015 8:55 pm

jono not bono wrote:
Wow. I really enjoyed that video. Clears up a lot of opinionated Dogs Brown.

So, what are the actual pros and cons of recording at higher sample rates (bearing in mind I can't hear any difference above 44.1khz)? With Bitrate, I'm more than happy with 24bit but could someone explain what the benefits of using 32bit Float are? I've never understood it!

Thanks

Jono
First - I didn't author the article and have no connection to it.
Second - It's not wrong. Nor garbage to be wholly dismissed as others claimed. It is what I said, a good place to START.

As for the 24bit 32bit float difference:
To sum it up in one line without the pages of technical engineering data:
32 bit float allows for processing or storage of a higher dB range (Dynamic Range) before being subject to waveform truncation ('clip').
With 32bit float, using the 6dB per bit rule, you end up with about 1530dB dynamic range.
24 bit is just over 144 dB.

Now how is this useful in processing/Mixing?
SOUND is the word we use for the translation of a waveform (or more appropriately to the topic of mixing MANY INTERACTING waveforms of different frequency)

to illustrate:
Take 10 complex frequency sources at 0dB and sum them to your 2-channel Buss.
At your 2-channel Buss you will be well over the 0dBFS limitation and your 24bit project will be a clipped, aliased and distorted mess.
But this is not the case with 32 bit float!
32 bit float allows for the dynamic range to be extended just beyond 1500dB - more than enough to handle the processing of the sources.

Now you can't directly translate that 32bit float out through a 24bit D/A process but you can process it and store it digitally at 32bit float.

So if you saved the processing you are doing in a 'Pulse-Code Modulated' (xPCM) format like a WAV file at 24 bit you would have a clipped, aliased and distorted mess.

if you saved the processing you are doing in a WAV file at 32 bit float you would have the APPEARANCE of a clipped, aliased and distorted mess! Examination of the visual representation of the wave would look like a big fat sausage or block.
A simple "Normalize to 0dB" would restore the entire stored encoded waveform back to listenable without it being a clipped, aliased and distorted mess.

So where it matters internally is that you can drive the gain of a source , or summed sources, well into the RED (above 0dB) and enjoy all the many benefits and effects it has on your other sources while being summed or processed.

Does that make the concept a little less muddy?
(probably not! but the more you learn about it and experiment with it the clearer it will become!)
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Re: What bit depth and sample rate should I use?

Post by peakae » Fri Jan 23, 2015 10:12 pm

Thats actually why I am using 32bit float, it is to easy to accidentally destroy a bounce or mixdown.
It's a safety net, that IMHO won't tax your computer that much more than 24bit.
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Re: What bit depth and sample rate should I use?

Post by fretthefret » Fri Jan 23, 2015 11:13 pm

peakae wrote:Thats actually why I am using 32bit float, it is to easy to accidentally destroy a bounce or mixdown.
It's a safety net, that IMHO won't tax your computer that much more than 24bit.
It's actually more than a safety net!

It is an indisputable fact that 32bit float can more accurately represent a source signal than can 24 bit or 16bit or 8bit.
In fact, 32bit (according to the math) is about 65,536 times better at accurately representing a signal than is 16 bit and 256 times more accurate than 24 bit!

If we were only recording a single source signal and then outputting that unprocessed single recorded signal, we could probably get away with 16 bit or lower and still get a halfway decent representation. Though outside the test lab nobody is recording a 1khz sine wave, exporting it to CD and calling it music!
When we are processing and manipulating the multiple complex frequency signals (as happens during mixing) you want to start with the best possible sources and give it the best possible processing in order to get the best possible end result.
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Re: What bit depth and sample rate should I use?

Post by Jarno » Sat Jan 24, 2015 12:08 am

fretthefret wrote:First - I didn't author the article and have no connection to it.
Nobody said you did.
fretthefret wrote:Second - It's not wrong. Nor garbage to be wholly dismissed as others claimed. It is what I said, a good place to START.
Yes it is. It's garbage. First of all it plays "connect-the-dots" game, which is fundamentally flawed approach to digital audio. I don't have to go any further into details. This alone makes it wrong, garbage, flawed, utter cow excrement! It is not good place to start. It's horrible place to start, because it gives you wrong impression on how digital audio works. It's as good place to start as ancient greek mythology when you try to understand basic laws of quantum physics.
fretthefret wrote:256 times more accurate than 24 bit
You are comparing apples to oranges: fixed-point vs floating point. 32-bit float is in matter of fact just as "accurate" as 24-bit fixed, but in much wider dynamic range.
fretthefret wrote:If we were only recording a single source signal and then outputting that unprocessed single recorded signal, we could probably get away with 16 bit
Are you talking about The Good Old "Stacking" Myth? That quatisation errors of multiple tracks will magically accumulate into final mix? Please tell me it isn't so.
fretthefret wrote:Though outside the test lab nobody is recording a 1khz sine wave, exporting it to CD and calling it music!
Of course not. But still 44.1kHz/16-bit digital audio can exactly reproduce any incoming music if
1. It's bandwith limited to less than 1/2 of sampling freq
2. You don't count quantisation distortion/dither noise below -96dBFS.
That's scientific fact. Proved decades before anyone even thought about digital audio.
fretthefret wrote:you want to start with the best possible sources
But what if your "source" (audio moderately isolated room recorded with microphone through pre-amp and converter) already has about 80dB dynamic rage (something you will get in home/project studio if you are lucky)? Would 24-bit recording help you? Of course not, the last 8 to 10 bits are just noise.

Once and for all ... just watch and listen to what people who really know has to say...
Monty Montgomery about sampling: https://www.youtube.com/watch?v=cIQ9IXSUzuM
Ethan Winer about audio myths (including bit depth around 46 mins): https://www.youtube.com/watch?v=BYTlN6wjcvQ
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Re: What bit depth and sample rate should I use?

Post by fretthefret » Sat Jan 24, 2015 1:11 am

Jarno wrote:
fretthefret wrote:First - I didn't author the article and have no connection to it.
Nobody said you did.
fretthefret wrote:Second - It's not wrong. Nor garbage to be wholly dismissed as others claimed. It is what I said, a good place to START.
Yes it is. It's garbage. First of all it plays "connect-the-dots" game, which is fundamentally flawed approach to digital audio. I don't have to go any further into details. This alone makes it wrong, garbage, flawed, utter cow excrement! It is not good place to start. It's horrible place to start, because it gives you wrong impression on how digital audio works. It's as good place to start as ancient greek mythology when you try to understand basic laws of quantum physics.
fretthefret wrote:256 times more accurate than 24 bit
You are comparing apples to oranges: fixed-point vs floating point. 32-bit float is in matter of fact just as "accurate" as 24-bit fixed, but in much wider dynamic range.
fretthefret wrote:If we were only recording a single source signal and then outputting that unprocessed single recorded signal, we could probably get away with 16 bit
Are you talking about The Good Old "Stacking" Myth? That quatisation errors of multiple tracks will magically accumulate into final mix? Please tell me it isn't so.
fretthefret wrote:Though outside the test lab nobody is recording a 1khz sine wave, exporting it to CD and calling it music!
Of course not. But still 44.1kHz/16-bit digital audio can exactly reproduce any incoming music if
1. It's bandwith limited to less than 1/2 of sampling freq
2. You don't count quantisation distortion/dither noise below -96dBFS.
That's scientific fact. Proved decades before anyone even thought about digital audio.
fretthefret wrote:you want to start with the best possible sources
But what if your "source" (audio moderately isolated room recorded with microphone through pre-amp and converter) already has about 80dB dynamic rage (something you will get in home/project studio if you are lucky)? Would 24-bit recording help you? Of course not, the last 8 to 10 bits are just noise.

Once and for all ... just watch and listen to what people who really know has to say...
Monty Montgomery about sampling: https://www.youtube.com/watch?v=cIQ9IXSUzuM
Ethan Winer about audio myths (including bit depth around 46 mins): https://www.youtube.com/watch?v=BYTlN6wjcvQ
Oh boy....

I've already watched the video by Monty.

You spoke up. But aren't really listening (nor reading apparently)!

And until you actually "learn it" and prove it to yourself rather than repeat someone else's idea (and bad info too) after watching a fixed test on youtube it's pointless to go further - it's like arguing with a religious zealot.

I happen to record acoustic sources at the high sample rate and increased bit depth because there is a definite quality difference!!
Whether Monty, Ethan or Jarno say different ...
Not only does Physics back it but It is something easily noticeable to my ears doing blind comparison tests!

I would also go with the actual tested results at DOLBY Labs long before Monty or Ethan!
You know them, they're the real scientists who have spent billions over the years to determine that 96kHz 24bit is the most appropriate for accurate 8 channel reproduction in the world's movie theaters (and 192kHz 24 bit for 6)!

But I guess despite all the evidence and science you must be right, since you watched a video on youtube.
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Re: What bit depth and sample rate should I use?

Post by Piirka » Sat Jan 24, 2015 1:34 am

Please listen to Jarno on this one.

So many out there with no clue about digital audio and/or the human hearing's inner mechanic.
Whatever you hear has to be within the human hearing range.

If you hear a difference between 44.1 and 96K with your converters, it has some poorly made filtering.
Anyhow, it will sound WORSE at 44.1, NOT BETTER at 96k.

Those videos are absolutely NOT a good place to start. Why learn it wrong from the beginning?

PS. There are nothing in digital audio named Resolution. Does not excist.

More bits = Larger Dynamic Range. Not "better resolution" or smaller/finer details of audio reccorded.
One bit is the same whether it's the first or last. One bit has only 2 values 1 or 0 (on or off).

Do the math: 2x2x2.... as many times (bits) you want. All you gain is........... Larger Dynamic Range (from top, 0dBFS, and downwards), and no finer "anything".

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Re: What bit depth and sample rate should I use?

Post by fretthefret » Sat Jan 24, 2015 1:38 am

I guess all these scientists are also wrong because Ethan and Monty and Jarno say so....

http://www.analog.com/en/content/relati ... e/fca.html

So to be clear, just because Monty and Ethan and Jarno believe different, all the cream-of-the-crop top scientific minds at both the world leader in audio research and the world leader in Digital Signal Processing are WRONG?!
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Re: What bit depth and sample rate should I use?

Post by jono not bono » Sat Jan 24, 2015 2:08 am

I'm so confused now. What videos aren't a good start? The Monty and Ethan Videos seem like a good start to me.

I'm gonna stick with 24bit 44.1khz.

Shame really, that no matter how high quality anyone records at, you still have to destroy your masterpiece by converting it to a little MP3 at the end of production, so someone can store it on an Ipod. Probably listening to it (if you're lucky) on a pair of $10 headphones. Haha.
Please make me stop buying stuff.

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Re: What bit depth and sample rate should I use?

Post by Jarno » Sat Jan 24, 2015 10:57 am

fretthefret wrote:I guess all these scientists are also wrong because Ethan and Monty and Jarno say so....
No! They are talking about signal processing, NOT signal sampling. Two completely different things.

If all processing inside the DAW was done with same bit-depth/sample rate as audio is recorded with, I would recommend using 192kHz/32bit recording without hesitation and would call anyone not doing so an idiot. But that's not the case: Signal path inside your DAW is 32-bit floating-point and most hight-quality plugins do oversampling internally (whenever it matters).
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Re: What bit depth and sample rate should I use?

Post by fretthefret » Sat Jan 24, 2015 4:40 pm

Jarno wrote:
fretthefret wrote:I guess all these scientists are also wrong because Ethan and Monty and Jarno say so....
No! They are talking about signal processing, NOT signal sampling. Two completely different things.

If all processing inside the DAW was done with same bit-depth/sample rate as audio is recorded with, I would recommend using 192kHz/32bit recording without hesitation and would call anyone not doing so an idiot. But that's not the case: Signal path inside your DAW is 32-bit floating-point and most hight-quality plugins do oversampling internally (whenever it matters).

Wow....

Again, you obviously didn't really read the scientific paper in the link.
Nor even understand the application of the content.

FYI, just so you know who you've hitched your opinion wagon to,
Ethan Winer has been banned from a few of the top professional public forums on the subject.
His tests are fixed and most of his claims false!

Ethan's claims of no difference in audio quality are based on a very poor monitoring chain running through 2 DJ club mixers with low end soundblaster cards - such than none would even expect to hear a difference through such a poor chain!
He's also outright refused public invitations by several studio professionals over the years to defend his claims of not being able to hear the differences - as most of the audio professionals could INDEED hear the differences!

You're promoting him as the authoritative basis for your argument...
He's NO scientist (not even close!), nor ethical, nor authoritative on the subject - by any stretch of the imagination!

While there are indeed MANY instances of misleading claims in the Pro Audio World, is that really the particular charlatan you want to promote?

All you have to do to protect yourself is to learn the actual science yourself and quit repeating false claims drawn from Internet tricksters like Ethan!
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Re: What bit depth and sample rate should I use?

Post by Jarno » Sat Jan 24, 2015 6:38 pm

fretthefret wrote:Again, you obviously didn't really read the scientific paper in the link.
Not throughout, but according to pieces I read it talked about processig and why it must be done with higher bit-depth. For example, from summary section:
John Tomarakos wrote:When using 16-bit A-D and D-A Converters in an audio system that will process `CD-quality' signals having a dynamic range of 90 - 96 dB, a 16-bit data path may not be adequate as a result of truncation and rounding errors accumulating during execution of the DSP algorithm
Exatly the same thing I'm talking about.
fretthefret wrote:Ethan Winer has been banned from a few of the top professional public forums on the subject.
I'm not surprised. The best way to get you banned from audiophile forum is to go and say any good quality wire is as good speaker cable as $5000 gold-soaked cable which is manufactured specially for you taking ultimate care that cable's and your star signs are compatible. Ethan is in the same boat.
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Re: What bit depth and sample rate should I use?

Post by fretthefret » Sat Jan 24, 2015 7:06 pm

Jarno wrote:
fretthefret wrote:Again, you obviously didn't really read the scientific paper in the link.
Not throughout, but according to pieces I read it talked about processig and why it must be done with higher bit-depth. For example, from summary section:
John Tomarakos wrote:When using 16-bit A-D and D-A Converters in an audio system that will process `CD-quality' signals having a dynamic range of 90 - 96 dB, a 16-bit data path may not be adequate as a result of truncation and rounding errors accumulating during execution of the DSP algorithm
Exatly the same thing I'm talking about.
fretthefret wrote:Ethan Winer has been banned from a few of the top professional public forums on the subject.
I'm not surprised. The best way to get you banned from audiophile forum is to go and say any good quality wire is as good speaker cable as $5000 gold-soaked cable which is manufactured specially for you taking ultimate care that cable's and your star signs are compatible. Ethan is in the same boat.

Jarno,

I appreciate what you are saying! I'm not faulting you for being mislead by the false claims of Ethan.
Just please educate yourself before arguing something as fact.

No Ethan is not in the same boat. (unless that boat is the Banana Boat)
He wasn't banned for making statements on $5000 cables but for presenting false claims and rigged tests as hard science on bit depth, sample rate and audio quality.

You felt it necessary to shoot down and break apart what I commented on previously in this thread and give poorly backed claims from tricksters and yet the evidence and science from all the top-in-their-field researchers at the world's leading audio research corporations completely backs what I've said!
I'm not here to waste my time pounding you over the head with evidence and science as you are an individual and can make up your own mind (for right or wrong).
But when your mixes fall short when compared to others in the marketplace you'll know why - hopefully listening to those tricksters doesn't mislead you enough to compromise quality and cost you opportunity!

And Ethan is still a charlatan - albeit a charismatic one in a Jim Jones drink the magic kool-aid kind of way!
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Re: What bit depth and sample rate should I use?

Post by Jarno » Sat Jan 24, 2015 7:22 pm

fretthefret wrote:Just please educate yourself
Done that for last 40+ years of my life.
fretthefret wrote:But when your mixes fall short when compared to others in the marketplace you'll know why
I've always known. Reason have always been found between my mixing console and my chair.
fretthefret wrote:And Ethan is still a charlatan
Quite a serious accusation. I would never say anything like that about any person in public forum, unless this person were convicted criminal. I don't know about laws of your home state/country, but in many places you could became convicted criminal for making that kind of statements.
Cubase 8 Pro/7/4/SX1/VST3.7 | Waves Gold | Melodyne | PC i7-4770/8G/2xSSD/Win7 64 | MacMini | Frontier Tranzport
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Alesis 3630 | AudioLogic MT66 | Joemeek VC1Q | dbx386 | Focusrite VoiceMaster | Line6 PodXt/BassPodXt | Boss GT-3
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Re: What bit depth and sample rate should I use?

Post by fretthefret » Sat Jan 24, 2015 7:25 pm

Jarno wrote:
fretthefret wrote:Just please educate yourself
Done that for last 40+ years of my life.
fretthefret wrote:But when your mixes fall short when compared to others in the marketplace you'll know why
I've always known. Reason have always been found between my mixing console and my chair.
fretthefret wrote:And Ethan is still a charlatan
Quite a serious accusation. I would never say anything like that about any person in public forum, unless this person were convicted criminal. I don't know about laws of your home state/country, but in many places you could became convicted criminal for making that kind of statements.

Not worried! FACT is FACT!
"A charlatan (also called swindler or mountebank) is a person practicing quackery or some similar confidence trick in order to obtain money, fame or other advantages via some form of pretense or deception."

"A person who makes elaborate, fraudulent, and often voluble claims to skill or knowledge; a quack or fraud."
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Re: What bit depth and sample rate should I use?

Post by Qbass-007 » Sat Jan 24, 2015 7:35 pm

It's just too early in the morning for popcorn :lol:

I'd use NMB non-metalic sheathed Romex cable for $75.00 per 250 roll before spending THOUSANDS on gold plated cable :D

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