SAA7210 to PCM56, a Dutch-American connection

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tubee said:
Bernard, for temperature stability, is a simple 78L05 not good enough?
I have been fiddling to shift range of pot allready with different R values.
Will try further.

I don't know.
My CD players never have been stable.
While the pots get adjusted there is no cover on the CD player and everything gets less warm.
Room temperature also is not always the same.
So IMHO everything should have as small drift as possible.

My DAC now seems to be stable but I have the pot only on one side because I fiddle canceling MSB error with combinations of chips.

But for the moment I try to figure what is the best sounding filter:

Passive LC, active GIC or transformer + RC.
 
tubee said:

yep, i've got a few of them (and seen the lot).
And to anyone reading this: all gone, none availalble.

Plan is to use it with four pcm63's and the pass i/v stage i have lying around. Got all the stuff, only a powersupply, clock and gluelogic required. And a nice pcb. Had enough of the 1541 for now, going to be busy with this for a while.
 
http://www.diyaudio.com/forums/showthread.php?threadid=69663

Post 17 & 21 has a bessel filter. No Gic btw

Bernard, you are experimenting with a 300//600 ohm transformer, where did you get it from?


And i must unfortunately admit here: I had TWO Philips line stepup transformers, sitting in a Mu metal housing with tube-like pins from a tube amp. I have demolisched 1 of them as study object, the other i still have. But what to do with one when i want stereo.....:bawling:
 
QSerraTico_Tico said:

The bandwith limiting will occur in the low-pass filter every AD converter uses! If this is a brickwall filter it has much ringing. The point being will another brickwall filter after the DA converter null this ringing or make it worse?
I will ask a friend of mine to burn the CD in redbook.
My friend did record a squarewave into a redbook CD on CD-ROM.
Much to my surprise the sqaurewave after playing on NON-OS DAC came out without any ringing.
So I don't understand how's it possible as one need a low-pass filter before the ADC.
He used an Onyx400 mixer and firewire cable to the PC.
 
Originally posted by Bruno Putzeys on the Prosoundweb forums
The reason why I was at the vacuum tube gathering was to give a talk on DACs. I took special time out to address the nonoversampling question.

For one, proponents' suggestion that NOS dacs have better impulse response is readily disproven using a square wave signal that is not critically and synchronously sampled. Test CD's tend to contain a square wave signal that reads -1,-1,...,-1,-1,1,1,1,...,1,1,-1,-1,-1,...
When looking at this signal, NOS dacs seem to have an edge, but as anyone who understands sampling theory will see, this is a very peculiar situation.
I made a CD containing synchronous (ie at a submultiple of fs) square wave signals sampled at various delays and one at a non-synchronous frequency. The outputs from two DACs (one normal, one unfiltered NOS dac) were fed to an oscilloscope and I hope the message hit home.
 
I've placed the pcm56's next to the 8 logic chips. This time the R channel is completely silent and easy to adjust in MSB alignment on ear with headphones, with crickets noise on the first track of Roger waters, Amused to Death. Used a simple 100K pot, it is set around middle now, the previous 15 turn pot was at the end. So there was a small error in the PS of the R dac in previous circuit. Now the soundstage can be fully tested at last.
Wil swap some clocks now.
 
Bernhard said:
Please try:

Iout to gnd: 200 ohm
33 ohm
To gnd: 33n
1.8m
To gnd: 68n
0.9m
To gnd: 3.3m in series with 200 ohm

This provides +3dB @ 20 kHz and -18 dB @ 44 kHz

Load >= 47k ( amplifier )


Thanks Bernard for filter values.

I have listened to the bunch of pcb's yesterday all evening.
I am sure the better dampened cabinet is sounding some better as the plastic CD660 base, but little bit dull sometimes. (leaded MDF and acryllic sheet base, and leaded cdm2)

The first used AD844 stage had a lift in treble, wich annoyed me after some time. Now the treble has a rather obvious roll-off, and i miss some treble. Soundstage is not deep but wide. Each instrument can completely followed as isolated instrument, bass is some coloured but deep. Sound is some mellow or something, never sharp or harsch.

Treble @20k is about 3dB down i guess. I want to compensate the treble roll-off, have mailed to Pedja Rogic (thnx Pedja :) )He advised me to re align my used filter to get a notch response. I will try your suggested steeper filter Bernard it sounds good to me.
I have bought a simple C/L meter, nice tool for selecting caps and winding L's.
And yes Tico the ripple is coming back with steep filtering, see also:

http://www.pedjarogic.com/non_os/lpf.htm

http://www.pedjarogic.com/1541a/L-sinc.htm
 
tubee said:

Now the treble has a rather obvious roll-off, and i miss some treble. Soundstage is not deep but wide. Each instrument can completely followed as isolated instrument, bass is some coloured but deep. Sound is some mellow or something, never sharp or harsch.

Treble @20k is about 3dB down i guess. I want to compensate the treble roll-off

Those values are not what I use ( my filter is much steeper ), just a quick and dirty modification of your schematic that compensates the treble roll-off and has the corner frequency @ 20 kHz.

The two 200 ohm resistors give the same 100 ohm for the I/V resistor as in your schematic.
Their values can not be changed as they are part of the filter.

You may use a switch to bypass the 3.3 mH to compare between compensated and uncompensated treble rolloff.

The 33 ohm need to be adjusted.
I will tell you how when you have built the filter.

audiofilter.jpg
 
You use a trannie for i/v? Tell me about inductors..;)

I need more amplification to get to 2V out. I/V is now allready about 200ohms (not 100 as stated in filter)

Amp is a CC E85CC with 2k7 on anode, about 180 ohm on cathode, no bypass, 200V unregulated PS. Hums slightly too.
 
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