John Curl's Blowtorch preamplifier part III

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The 26uS only applies when you get to +-1 bit sine stream, if it is =+3 bits you get about 7uS or so, that means 1 -90db sine is difficult to locate in space, well if it is burried down in the backround noise I think you can expect it to be difficult to locate.

The rise time limit (means the shortest possible rise time) for a signal in a system bandlimited to 20 kHz (or slightly above) is ~18 us.
 
Sad most of the argument here had been over simply that if you want to record an instrument that goes out to 50kHz you need to sample at 100kHz. This was always just plain obvious.
I just want to record the content that can be heard. 20k as a frequency limit, but also a bit more bandwidth to assure capturing the envelope information we are sensitive to.

Jn
 
Can you say what hardware - adc etc - and recording software was used?

Cymbal was Zildjian K Custom Hybrid Limited Edition 14" Reversible Hi-Hat top.
Mic was Schoeps MK41 capsule with CMC 6 body (aka Schoeps CMC 641).
Mic and premap-to-ADC cables were custom manufactured Jam designed XLR cable wire.
Preamp was Grace Designs M101.
ADC and master clock source was RME ADI-2.
SPDIF from ADC into Focusrite Scarlett 6i6.
Recording software was Reaper 64-bit with project sample rate set to 192kHz (to avoid any unwanted resampling).

Hope that's enough info for you... :)
 
~0.115V/µs

You calculated slew rate which is not important. Rise time, by definition, is the time needed for the step response to get from 10% of its final amplitude to 90% of the final amplitude (after settling). Rise time would be same for the 100mV step or 10V step, if the system is not slew rate limited. It represents linear parameter, invariant with amplitude.
 
I just want to record the content that can be heard. 20k as a frequency limit, but also a bit more bandwidth to assure capturing the envelope information we are sensitive to.

I ran the 2,4,8 NRZ scenarios, there are expected differences that might help the filtering when you need every dB you can get. I think we are finally on the same channel, I could do some windowed bursts of 20kHz and look at phase and timing differences with the ideal brickwall filter.
 

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@cdbd,

following the same line of reasoning, we have to assume that the loudness war brought the best possible sound, because nearly everyone contributed to it.

It is a question of how to achieve the best possible quality; we know that the vast majority of consumers does not care about this quality, but should that be an argument?
John Curl was talking about the best effort approach back then and surely the direct to disc (like Crystal Clear or Sheffield Lab) were aiming at the small proportion that was interested in it.

@syn08,
Which exactly applies to you as well, as much as to everybody else. So stop playing the content ueber censor and moral headlight role, and mind your own business of promoting the High End Audio FUD elsewhere. Based on Oohashi, of course.

I somehow ( :) ) thought that the self-reflection would not be your cup of tea.
Of course you're right, the same applies to me, could it be the reason why in my posts there is little to find about why the others are all incompetent, the discussion ridiculous and whatever else.

Wrt to Oohashi et al., you are constantly fooling yourself; look at your last assertion that nobody could reproduce their findings.
Exactly the opposite is true, if other experimenters used the same/similar approach - including imaging - along with psychoacoustical tests as well, they found corrobation for the so-called hypersonic brain effect.

Btw, the link to the debunking of Oohashie et al.'s work by SY, that you've promised is still missing.
 
I ran the 2,4,8 NRZ scenarios, there are expected differences that might help the filtering when you need every dB you can get. I think we are finally on the same channel, I could do some windowed bursts of 20kHz and look at phase and timing differences with the ideal brickwall filter.
I would do bursts at various frequencies, say 2k,5k, 10k, and 20, but only for 8 NRZ.
Sample the analog bursts at 44.1.

It will be interesting to see how long it takes the output to go steady state, and what amplitude they get to. My thinking is that the farther from nyquist at sampling, the faster the settling.

Another thought is, will the settling be sample length dependent.

Jn
 
Btw, the link to the debunking of Oohashie et al.'s work by SY, that you've promised is still missing.

My memory must falter, could you please remind me where I promised you anything? Link please.

Otherwise, pleased don't play dumb, you know very well that the Oohashi paper is a total (fortunately, isolated) failure, quoted only by those with a vested interest in propagating audio grade FUD. The shortcomings were discussed in each and every audio related forums you mentioned it, and you ended up in the trash bin without exception, only to start over again the next chance you got. Do you need quotes/links? Athough you were quoted this analysis ad nauseum, here's again for others to go through: Audio Myths | PS Audio How do you qualify using IM marred speakers to play ultrasonic sounds, knowing that the IM products would fall straight in the audio band? And that later a corrected experiment by Ashihara completely debunked the Oohashi results? I still want to believe this was an honest mistake by Oohashi, due the lack of knowledge regarding sound reproduction.

And yes, it's the picky (when it comes to sensory testing) you which is supporting this failed study and conclusions by Oohashi. I'm afraid this does not qualify as a honest mistake.
 
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Going back to tape, it hit me that all the traditional tape systems have low pass filters along with the native filtering of the tape process AND a HF bias ranging froo 30 KHz to 250 KHz (and even 4 MHz) so there is a lot of similar issues with tape to digital recording.

Has anyone ever looked at transient response of a tape system for ringing etc? Aliasing is a well known phenomenon with tape. And what is the effect on transient response of the high pass filter at the bass?

Its interesting that digital shares a lot of these issues but has much higher linearity, has been the focus where I have seen little discussion of it in tape recording. Some pro recorders incorporated phase equalizers but not all and even with those corrections tape is not really phase perfect.
 
I would do bursts at various frequencies, say 2k,5k, 10k, and 20, but only for 8 NRZ.
Sample the analog bursts at 44.1.

It will be interesting to see how long it takes the output to go steady state, and what amplitude they get to. My thinking is that the farther from nyquist at sampling, the faster the settling.

Another thought is, will the settling be sample length dependent.


Settling time will be that of the filters used, and independent of sampling per se. That's the point I find most lacking in this discussion. It's a newer, subtler form of Nyquist denial.


Much thanks, as always,
Chris
 
I take this timing thing with a grain of salt. When editing sound effects one of he tricks to make a gun shot, explo sound louder but stay within the limits, both peak (-8dbfs) and RMS (-18dbfs) we would take 2,3 sometimes 4 different gunshots and pile them up a frame apart (30ms ). No one could tell it wasnt one sound. And when the CGI guys put the mussel flashes in the picture less than 4 frames apart the shots sounded like one so we would cheat them and seperate them. And this is listening to just the gunshots track solo.
 
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