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Literature/Professional Write ups on 192khz Sample Rate Dynamics Processors (HW)
Old 22nd December 2010
  #31
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Quote:
Originally Posted by SirElliot View Post
That depends on how low you are going. If you are stopping @ 20 Hz yes. However, moving further down, higher sample rates offer more dB gain since the frequency bandwidth is being rolled of under 20 Hz at a lower sample rate. One can look at the sensitivity below 20 Hz is lower @ 44.1 than 88.2 kHz.
I don't know what you mean by "dB gain." There is no inherent low frequency rolloff at lower rates, I don't know where this idea comes from.

Higher sampling rates offer increased high frequency bandwidth and arguably nothing but worse response at LF if processing is a consideration.


DC
Old 22nd December 2010
  #32
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Quote:
Originally Posted by dcollins View Post
I don't know what you mean by "dB gain." There is no inherent low frequency rolloff at lower rates, I don't know where this idea comes from.

Higher sampling rates offer increased high frequency bandwidth and arguably nothing but worse response at LF if processing is a consideration.


DC
I would imagine the confusion is on both sides. For I haven't experience any negative sides you mentioned.

Possibly you can explain what you mean by "worse response at LF if processing is a consideration" from a frequency standpoint.

Cheers!
Old 23rd December 2010
  #33
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Originally Posted by UnderTow View Post
Right up your alley then, Ethan! heh

( Ethan Winer likes to pretend his anecdotal evidence is science even after it has been debunked. When he runs out of arguments he calls people assholes: Hearing below the noise floor )

Alistair
Crossing topics to insult someone... I was kind of on the fence about you, but you just proved it.. you're a troll.. take a break.. make some music.. drink some alcohol.. chill the [email protected] out
Old 23rd December 2010
  #34
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Quote:
Originally Posted by SirElliot View Post
I would imagine the confusion is on both sides. For I haven't experience any negative sides you mentioned.
First, your claim that the bandwidth is 'rolled off under 20Hz at lower sample rates' is incorrect.

The Nyquist requirements are for highs, not lows. Any sampling rate extends to 0Hz. I don't know what you mean by 'sensitivity below 20Hz' or 'dB gain' at higher rates either, so we should probably start here and leave the bit about higher rates and greater difficultly of LF processing for another time.


DC
Old 23rd December 2010
  #35
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Quote:
Originally Posted by dcollins View Post
First, your claim that the bandwidth is 'rolled off under 20Hz at lower sample rates' is incorrect.

The Nyquist requirements are for highs, not lows. Any sampling rate extends to 0Hz. I don't know what you mean by 'sensitivity below 20Hz' or 'dB gain' at higher rates either, so we should probably start here are leave the bit about higher rates and greater difficultly of LF processing for another time.


DC
Can you please explain your meaning of "worse response at LF if processing is a consideration"?

Since Low Frequency can mean anywhere from 100 Hz downwards (some may say as high as 200) pending on who you ask, it would be helpful if you offered the frequencies you found offensive in your testing so, I can get an idea what you are trying to explain.


I would imagine the confusion may be based on field of audio we do.

Cheers!
Old 23rd December 2010
  #36
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Quote:
Originally Posted by SirElliot View Post
Can you please explain your meaning of "worse response at LF if processing is a consideration"?

Since Low Frequency can mean anywhere from 100 Hz downwards (some may say as high as 200) pending on who you ask, it would be helpful if you offered the frequencies you found offensive in your testing so, I can get an idea what you are trying to explain.
It doesn't matter which low frequency. There is no low frequency roll-off in sampling. You can represent 1 Hz with a 44.1Khz sampling rate. Or 0.1 Hz if you want.

The issue Dave is referring to is that the higher the sampling frequency, the less change occurs between each sample. As plugins process the audio in blocks of data, for low frequencies this means less precision at higher sample rates.

Alistair
Old 23rd December 2010
  #37
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Originally Posted by lionaudio View Post
Crossing topics to insult someone... I was kind of on the fence about you, but you just proved it.. you're a troll.. take a break.. make some music.. drink some alcohol.. chill the [email protected] out
I'm no troll and I wasn't the one doing the insulting. I just want that article adjusted or removed. As it stands, it is misinformation. There is already enough misinformation about audio on the net.

Alistair
Old 23rd December 2010
  #38
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Quote:
Originally Posted by UnderTow View Post
I'm no troll and I wasn't the one doing the insulting. I just want that article adjusted or removed. As it stands, it is misinformation. There is already enough misinformation about audio on the net.
I am truly worried about the potential development of your bloodpressure, mr Assange.


Happy Festivus!
Patrik
Old 23rd December 2010
  #39
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Quote:
Originally Posted by dcollins View Post
Is there a converter chip that performs better at 192k?


DC
What about the DAD stuff?
I find it interesting that their ppl and Lavry never go toe to toe as they seem to favour sample rates even over 192KHz.

There has been a few very well respected mastering engineers on this board that have said they have some of the best convertors to their ears. IMHO experience (not scientific tested) I have preferred DSD over PCM, and 96KHz over 44 on a good setup. Can't vouch for 192 though.

Of course DSD and DXD drops bits over standard LPCM (correct?)
Old 23rd December 2010
  #40
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Quote:
Originally Posted by UnderTow View Post
It doesn't matter which low frequency. There is no low frequency roll-off in sampling. You can represent 1 Hz with a 44.1Khz sampling rate. Or 0.1 Hz if you want.

The issue Dave is referring to is that the higher the sampling frequency, the less change occurs between each sample. As plugins process the audio in blocks of data, for low frequencies this means less precision at higher sample rates.

Alistair

Cheers UnderTow

I am actually wondering where did he draw to the conclusion on the following:

Quote:
Originally Posted by dcollins View Post

Higher sampling rates offer increased high frequency bandwidth and arguably nothing but worse response at LF if processing is a consideration.


DC
Since I have not experienced a worse response at LF at higher sample rates, I am curious what frequencies were creating this problem.

I understand the usable frequency range can scale down to 0.5 Hz. However,
my question to DC Collins is what low frequencies did he find offensive in his testing to draw such a conclusion.
Old 23rd December 2010
  #41
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Quote:
Originally Posted by SirElliot View Post
Since I have not experienced a worse response at LF at higher sample rates, I am curious what frequencies were creating this problem.

I understand the usable frequency range can scale down to 0.5 Hz. However,
my question to DC Collins is what low frequencies did he find offensive in his testing to draw such a conclusion.
It's a well-known phenomena called "Finite Wordlength Effects." All filters are affected by it. Generally speaking, it's caused by quantization (finite resolution) of the coefficients that generate limit-cycles or frequency response errors at the the low end and near Nyquist.

As a DSP developer, you should be able to simulate in it your code by reducing the resolution of the coefficients.

I guess the first place we heard it was in the old Sonic Solutions eq There were ripples in the response at LF even at at 44.1. This was in IIR.


DC
Old 24th December 2010
  #42
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I'm not certain, but I dont think it's a stretch to conclude knowing how to make plugins in synthmaker doesnt guarantee a thorough knowledge of digital audio.

Sean
Old 24th December 2010
  #43
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Originally Posted by diggo View Post
I'm not certain, but I dont think it's a stretch to conclude knowing how to make plugins in synthmaker doesnt guarantee a thorough knowledge of digital audio.

Sean
My knowledge in audio is based on analogue hardware design. However, from a design standpoint, the two intertwine in terms of obstacles in which you must overcome. Since the subject is frequencies, the co-relation amongst the two is not two far apart.

From my experience chatting with those who focus on one over the other, the terminology in wording can bring forth confusion at times.

In the case of Synthmaker, one still needs to have good understanding on what they are aiming for to accomplish the end result. Synthmaker only compiles the code so one still needs to have his/her math correct in order for it work based on the designer’s concept.

What DC Collins mentioned is easily understood and, I have devised ways to overcome the objective. I will leave it at that for I don’t want to moderator to assume I am spamming the board.

Happy Holidays!
Old 24th December 2010
  #44
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Quote:
Originally Posted by dcollins View Post
It's a well-known phenomena called "Finite Wordlength Effects." All filters are affected by it. Generally speaking, it's caused by quantization (finite resolution) of the coefficients that generate limit-cycles or frequency response errors at the the low end and near Nyquist.

As a DSP developer, you should be able to simulate in it your code by reducing the resolution of the coefficients.

I guess the first place we heard it was in the old Sonic Solutions eq There were ripples in the response at LF even at at 44.1. This was in IIR.


DC
Right!

I now see what you are referring to. I've conducted many tests with other brands and noticed how the frequencies can shift by a few cycles or oscillate once they are called to process a higher sample rate on some designs.

I have implemented your suggestion in the past on a few of my designs to prevent such errors. Thank you for bringing it to my attention nonetheless.

Happy Holidays!
Old 24th December 2010
  #45
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Quote:
Originally Posted by SirElliot View Post
In the case of Synthmaker, one still needs to have good understanding on what they are aiming for to accomplish the end result. Synthmaker only compiles the code so one still needs to have his/her math correct in order for it work based on the designer’s concept.
I'd never seen this program before, but it looks interesting. From what I could tell, you can just go to the comp.dsp site, grab the bi-quad library and drag it into the window and it would run. Without any tears. I guess it's one reason why most digital EQ's give identical results.

Quote:
What DC Collins mentioned is easily understood and, I have devised ways to overcome the objective. I will leave it at that for I don’t want to moderator to assume I am spamming the board.
Far from it. I bet they would welcome a discussion of what is unique in your implementations.


DC
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