Balanced PCM1702 question

[Elso Kwak]

Hi All,
Guido (R2D2), sorry for confusing you with Guido Tent. It happened before!
For balanced operation I made a scheme with three discrete opamps. Two opamps are for IV-conversion and one is for output buffering and balanced to single ended conversion. I also incorporated a third order Bessel low-pass filter. -3dB point at 10kHz. The first pole is in the IV-converter. I can post the general schematic but will not post details of the discrete opamp.

Thijs, There was a famous Japanese DAC using 16(?) parallel Burr-Brown PCM56. I don't remember the name or brand. A long time ago I had the PCM56 in a balanced configuration in my Sony player. So I had four PCM56 in the working. The PCM56 had very good bass but poorly defined highs. Later I changed to AD1851 that had much better highs but weaker bass.

[Elso Kwak]

Hi Guido B,
Attached the schematic of a balanced IV-converter and outputbuffer. I got the idea from a Crystal datasheet. The whole filter is calculated with Burr-Browns FilterPro(R). A nice side effect is that any offset caused by the DACs is cancelled as long as the offset is equal for both DACs. This eliminates the ouput cap.
I similated both inputs with Microcap. It simulated good and measurements confirm that. No overshoot at a 1kHz sqaurewave.
As a experiment I changed the filterresponse to Butterworth and a 15kHz -3dB point. The squarewave showed the slight overshoot exactly as in the FilterPro papers. Sound was worse more "digital" so I quickly changed it back to Bessel filter response. This result withheld me from trying higher order Butterworth filters. Kusunoki states that the bad sound is not due to oversampling per se but due to the digital filter. I don't agree as you will understand.
http://www.tnt-audio.com/intervis/kusunoki_e.html

[Elso Kwak]

Oops, attachment lost in transit. Trying again!

[tbla]

the dac you talk about, is the HIBARI from audio note japan.......

[jean-paul]

Hi Elso, I don't see the use of the -3 dB point of 10 kHz. You will throw more than half the spectrum ! To me it seems like listening cd's through a telephone since it has almost the same bandwidth.

This can not be called HiFi.

[hifi]

.. my TDA1543 by the way is not dead-quiet .. it has significant background noise (it is the simple DAC kit with passive I/V) ... I'll let you know how thing progress, thanks for all the help..

I find that rather strange, I find my TDA1543 dac to be very quiet! I use a ground plane and TL431 regulators, maybe that makes a difference?

Maybe your pream/amp dont like the hf? or what IV do you use?



/ micke

[Elso Kwak]

Originally posted by jean-paul
Hi Elso, I don't see the use of the -3 dB point of 10 kHz. You will throw more than half the spectrum ! To me it seems like listening cd's through a telephone since it has almost the same bandwidth.

This can not be called HiFi.

Hi Jean-Paul,
It is all about choices!
I want a smooth 3150 sine wave.
I can put the -3dB point at 20kHz. This will not remove the staircase on the sinewave.
It does not sound dull as I feared.
NON-OS sounds very bright to my ears. Maybe this explains a bit.
I have been critisized a lot for the low crossover point. That's why I tried the 15kHz filter. It is a dilemma we will never solve in a NON-OS DAC. If you want a steep, brickwall filter you get ringing and overshoot and bad sound. If you choose a filtertype without ringing like the Bessel characteristic you don't get a steep filter. Especially the soft knee or shoulder of the Bessel filter can not be avoided. A higher order Bessel filter at a higher crossover will not solve the problem. I can not bend the laws of physics. Steep filtering is not possible without ringing. If you have a scheme that accomplishes steep filtering without ringing I would be most interested. As we all know the bandwith of the redbook CD is choosen too low. This was due to the constricts at the time of introduction of the CD.
I have also experimented with notch filters like in the Zanden DAC. Notch filters give even more ringing and overshoot. The result is a very weird square wave.
Feel free to use a higher crossoverpoint f.a. at 20kHz. It will not remove the staircase signal, only round it off a little.

[Elso Kwak]

Originally posted by hifi


I find that rather strange, I find my TDA1543 dac to be very quiet! I use a ground plane and TL431 regulators, maybe that makes a difference?

Maybe your pream/amp dont like the hf? or what IV do you use?



/ micke

The HF garbage can excite your preamp and poweramp. Intermodulation products can fold back into the audio band making the sound worse and the noise more audible!
My NON-OS DAC is producing all kinds of weird noises when not using the lowpass filter I just posted.

[Elso Kwak]

Originally posted by tbla
the dac you talk about, is the HIBARI from audio note japan.......

Hi t,
You mean this one?
They did not even bother to mention the type of the DAC chip.
http://www.audionote.co.jp/digital/content.htm

[guido]

Thijs,

If the CS8412 gives jitter and you buffer it with one buffer, you will feed your 32 dacs with the same signal. Then they all suffer from the jitter. Maybe if you use separate buffers to feed the dacs you might have different timing at the dac input. But if it gives an improvement?

There are three options to reduce jitter i have seen in projects on the net:

one box solution: use an old player to tap i2s. You need an old player, newer models have the oversampling done inside the decoder. I took a 1994 philips apart the other day, this already has this (SAA7345&TDA1545, no i2s anyway). If you go for this, you need to upgrade the cdp's clock to reduce jitter.

async overclocking, like Elso uses.

reclocking in dac with fs and feedback of the clock signal to the cd player. The way i am building my dac. It is not ready yet, so no impressions yet on sound. Not really an universal solution i must admit.

use a pll to create a new stable clock, which follows the incoming clock.

Greetings,

GuidoB