John Curl's Blowtorch preamplifier part II

[john curl]

Wavebourn, resistors have capacitance too. Earling Fredrikson told me himself that this was the reason when I visited B&K in 1974.

[PMA]

They have physical dimensions, so they must have some capacitance.

[fas42]

Originally Posted by RNMarsh

OK Keep going... we're listening and watching.

Thanks, Richard. OK, results are in and it worked as well as I expected it to. This resampling exercise is particularly stressful because the natural fit for downsampling 352.8 is to use 88.2kHz as the go between, rather than 96 as here.

So, did the resampling, down and up, of only the right channel, which appeared to be slightly more dynamic, making it mono on the way. Then, did a diff of the original, 2L waveform and the resultant, 2 pass 352.8 file. This was what came out:

2L50SACD_tr1_DXD_stereo_2012-12-29_14-09-58-04.jpg

Looks pretty terrible, but then have a look at the spectrum:

2L50SACD_tr1_DXD_stereo_2012-12-29_14-09-58-04b.jpg

This is very suspicious, a typical noise shaping curve after 40kHz, suggests there is no real information beyond that frequency, it's purely processing "rubbish". So, apply equalisation to brickwall attenuate everything after 40kHz, and get this:

2L50SACD_tr1_DXD_stereo_2012-12-29_14-09-58-05.jpg

This looks much, much better, but there is still some apparent noise there. Can't zoom any more, so use the amplify function to boost it, by 50dB:

2L50SACD_tr1_DXD_stereo_2012-12-29_14-09-58-06.jpg

Lo and behold, we have our original waveform back, inverted, as demonstrated by the copy of the result of the 2 pass resampling which follows the amplified difference "noise". In other words, a slight attenuation crept in at some point, which is why the difference process didn't perfectly null -- easy to compensate for ...

Frank

[Telstar]

Originally Posted by RNMarsh

OK Keep going... we're listening and watching.

I'm not at home so i cant check any of those files.
But sure you can write to 2L and ask Information on the process used and eventual editing.

Thanks for the good work, Frank also from me.

[Telstar]

Frank, so from what I understand they have recorded armonics up to 40khz.
If I'm correct, this is still VERY good. I personally believe that we should aim for 50khz, and that should sound MUCH better than brick walled at 20-21k. With most modern tweeters reaching 40k this should be clearly audible.

I used their piano sonata and while that shouldn't have any harmonic above 20k, the DXD version was do much closer to the real thing...

[fas42]

Originally Posted by Telstar

Frank, so from what I understand they have recorded armonics up to 40khz.
If I'm correct, this is still VERY good. I personally believe that we should aim for 50khz, and that should sound MUCH better than brick walled at 20-21k. With most modern tweeters reaching 40k this should be clearly audible.

I used their piano sonata and while that shouldn't have any harmonic above 20k, the DXD version was do much closer to the real thing...

The point of the exercise I did is that sampling at 96kHz, or even 88.2 was sufficient to encode all the musical, or event information of the DXD recordings that I tested from 2L. The DXD recording process theoretically can capture real audio information up to 176kHz but I haven't seen any evidence so far that the 2L crowd have done that for frequencies over 40kHz.

Which means, why do you need 352, 176, 192, etc? More space, download time, etc.

Yes, playback can sound better at the higher sampling rates, but my experiments indicate that this is a by-product of the way the playback systems work -- not because you're hearing "more information" ...

Frank

[elektroj]

Originally Posted by john curl

Here is some additional stuff on 30ips analog recorders. Please note date.

Thanks, John! I will study it with great interest.

[elektroj]

Originally Posted by Esperado

.....
At the very beginning of the digital recording, while so much sound engineers were talking of the DACs quality, i pretended that most of the audible differences were due to the analog stages after the DACs, not the DACs themselves.
Used a cheap Philips cd player with a Delta sigma 14 bits DAC followed by a modified analog stage to demonstrate-it in my studio.
.......

1. First Philips CD players weren't cheap.
2. They were 14 Bit Multi-Bit, not Delta-Sigma.

[Telstar]

Originally Posted by fas42

The point of the exercise I did is that sampling at 96kHz, or even 88.2 was sufficient to encode all the musical, or event information of the DXD recordings that I tested from 2L. The DXD recording process theoretically can capture real audio information up to 176kHz but I haven't seen any evidence so far that the 2L crowd have done that for frequencies over 40kHz.

Which means, why do you need 352, 176, 192, etc? More space, download time, etc.

Yes, playback can sound better at the higher sampling rates, but my experiments indicate that this is a by-product of the way the playback systems work -- not because you're hearing "more information" ...

Frank

Yes, I agree on your conclusions.
Now, why higher (than 88.2 which is min to preserve what current mics can acquire, at least those from 2L) sampling rate sounds better if the HF content is the same?
Somebody said earlier in this thread "noise", and I agree - pushing the hf noise further makes it less audible (by whichever mechanism the ear/brain use).
DAD (digital audio Denmark) affirms that 352k are required to get away with filtering. Considering the damage that analog filtering does to the music signal I agree with them. (Sorry no DBT done).
Chord believes that 1,5mhz are needed instead. I can't verify that because no multibit dac exist that can support such sampling rates.

The most sound way to reproduce digital would be using 88,2k material of good quality (ie non manipulated more than redbook) and upsample it on the fly to at least 352k, and get rid of analog filtering, or at least push it very high (>300k) where it won't destroy phase coherence up to 40khz.

If someone wants to organize blind tests, I'm all in favour of them.

[elektroj]

Originally Posted by fas42

Which means, why do you need 352, 176, 192, etc? More space, download time, etc.

Recommended reading:
http://www.lavryengineering.com/pdfs...lity_audio.pdf