John Curl's Blowtorch preamplifier part II

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Bill

Bells are based on power level. So when you are measuring voltage, As it goes up so does the current so for a fixed resistor load the power level goes up by voltage squared. That is why the formula for Bells is 2 x log(V1/V2).

Now we use decibels because the added factor of ten looks more impressive and helps to confuse the uninitiated.
 
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Exactly a Bell is power. dBV, whilst it still grates I can at least accept that the write understands its a relative gain rather than a relative power. And dont get me started on sound pressure vs sound power.

Of course its all because engineers are too lazy to use multiply and divide :)

(and yes I do still have a slide rule even if I've forgotten half the functions on it).

Aside, when I was still at school someone came from JET to give us a lecture on fusion. He mentioned that the plasma temperature was close to 200Million degrees. Some pedant piped up ' C or F', to which he replied 'doesn't matter you can't comprehend that sort of temperature'. I still can't.
 
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I know. Spend a couple of years as a fresh grad doing satellite concept design. A lot of link budgets and a lot of dBm and dBW. Which comes full circle. dBV, dBu etc mean something and no ambiguity. But dB on its own (in my curmudgeonly view of the world) should ONLY mean a power ratio. And I am happy if the rest of the work disagrees with me.

Life would be easier if audio was a 75Ohm system :)
 
Even more on jitter...

But rarely are these difference DESTRUCTIVE in nature, more like "voicing" in speakers and we pick our preferences.

But we are targeting something beyond "voicing" - something OBVIOUS to the ear as WRONG, something we have collectively coined... DIGITITUS !!!
My lastest audio "toy", the recently acquired laptop, is proving very useful, and convenient, in investigating this sort of stuff. Unless all the settings and environment are 'correct' then the sound is very much "digititus" - that is, it serves the purpose of allowing one to hear something on a video, play with audio material, etc, but is not pleasant to listen to, it rapidly becomes irritating on any extended listening. With all the tweaks in place the sound becomes "analogue" - what a word! - meaning that one can just relax and go with listening to the music, indefinitely.

Whether jitter is being addressed by the tweaks is impossible to say, but the end result is that the analogue circuitry of the laptop is being sufficiently left alone by digital muttering elsewhere in the machine, allowing "listenable" sound to be produced ...
 
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I would confine my tests to freqs in the most sensitive hearing range....3-5KHz.

Do you have a recommended jitter gen or CD jitter source/recording for me to add to this for comparison tests?


Anyone with real knowledge about the jitter on (used) commercial Cesium time standards (with a 10MHz output)?

THx-RNMarsh


The easiest would be to phase lock a good RF generator to the existing clock and then FM modulate it with an external generator. Swap its output for the internal clock going to the DAC. Any "synthetic" jitter will not really represent what happens in a specific architecture.

I don't think you will see much with the Yokogawa but I may be surprised. It was designed for HDD work where the data stream will have much higher modulations.

Here are some numbers for a current design Cesium standard:
http://www.ptsyst.com/Cesium-B.pdf

SSB Phase Noise 5 MHz :
1Hz - 95dBc
10 Hz - 130dBc
100Hz - 145dBc
1'000Hz - 155dBc

SSB Phase Noise 10 MHz :
1Hz - 90dBc
10 Hz - 125dBc
100Hz - 135dBc
1'000Hz - 145dBc

The 10 dB worse output at 10 MHz is normal even for stand alone oscillators. An old Cesium may need a new tube and won't be better. You can get a clean-up oscillator: 4145C Quartz Ultra Clean-up Oscillator | Quartz Frequency References | Time & Frequency References | Timing & Synchronization Systems | Products It offers a 10 to 20 dB improvement on phase noise. Its essentially PLL'ed to the Cesium's output (showing that the phase noise is not necessarily degraded by a PLL).
 
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You are probably right. (Those words on this thread should kill or at least stun half the folks here.) It has been 5 years since I last looked at external converters myself. There are more chips available and at a lower price so there should be more doing it at least reasonably well.

Of course the AES-3 standard is also old and holding up quite well for it's age. It was originally intended as a way for folks to upgrade to digital over existing audio networks. The BBC was behind a good part of it.

The pS vs. nS stuff has me very confused. Switching three orders of magnitude can lead to a lot of mistakes.

Getting 100 pS of jitter in a recovered clock with no crystal can be done for less than a few dollars in quantity: http://www.akm.com/akm/en/file/datasheet/AK4115VQ.pdf so why do it worse.

The ASRC (Asynchronous Sample Rate Converter) will use a different clock on each side and calculate the best fit between them. The jitter in the output is that of the output clock. However it can also imbed jitter and other errors on the input side and may have other artifacts from the conversion.
 
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Even more on jitter...
The contention is that very low rate clock centre frequency variance is relatively very high in amplitude (1/f noise characteristic) wrt audio band jitter, and is subjectively much more destructive.
This is why I have in the past mentioned jitter spectrum.
White noise clock instability causes a, well, white noise noise cloud over the reproduced audio.
VLF clock instability causes a 'vibrato' effect, and this is distinctly audible causing general lack of image clarity and stability, in particular centre placement and depth imaging, and sense of wrong pitch.

Allan Variance
AllanVariance/
Science Of Timekeeping

Dan.

This is a really marginal claim. First low frequency wander (called wow in analog reproducers) seems an odd source of "digital sound" since its orders of magnitude less than any analog playback system. Second, because of the way the modulation works the effects on the audio will be very low and so close to the fundamentals they are modulating they may not be detectable let alone audible.

A small philosophical point- if its easy to make something better you should. Even if there is not hard evidence that its audible. If its hard then its worth exploring the return on the effort. Getting 100 pS jitter on an SPDIF input is not hard (essentially free) so why not even if it may not be audible.
 
Philosophically, it also depends upon what one's goal is - if one is after the highest level of technical performance, and good subjective sound quality is an interesting by-product of that effort, that's one approach, obviously the centre of the efforts of many on this forum; another approach, which I pursue, is determining what easily altered factors directly affect the perceived sound quality, and then raising technical performance in appropriate areas so that there is no longer a negative audible impact. Both ways could lead to a good outcome, though I tend to feel my methods are somewhat more direct ... :)
 
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