Simple, good quality DAC

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TDA1549 internal opamp bypassed:

just to make it more clear
 

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Got the TDA1543 working. For the I2S I used two direct solderings to pcb's wires (jumpers) and one short (5cm) coaxial wire. Outputs and Vref have 1K metal oxide 0.5W resistors. 47k resistors to ground. And 4.7 uF regular electrolytics. In the psu wires I put a 22 uF and a 1 uF caps.

I just dont know where to get a decent 5V supply indide the player. For now I connected it to the uoput pin of the 7805 chip and the ground is directly to the case. But this is not a good solution...

The sound level is a bit low, even with the maximum level on the cdp (yes the volume control still works :) ). I tried to use my EF86 based pre but it didnt solve it (has not enough gain). As I use a opa549 gainclone, maybe I increase the feedback resistor to get more gain.

Miguel
 
Outputlevel of NON-OS TDA1543

miguel2 said:

The sound level is a bit low, even with the maximum level on the cdp (yes the volume control still works :) ). I tried to use my EF86 based pre but it didnt solve it (has not enough gain). As I use a opa549 gainclone, maybe I increase the feedback resistor to get more gain.

Miguel

Hi Miguel,
Also my experience: output is pretty soft, low with 1k resistors and passive conversion.
With 3k65 as Rconv as in Rudolfs schematic and a OPA604 opamp I got a lot more power and output.:bigeyes::idea:
(10 clicks on my volumecontrol or about 15 dB)
 
Hi Elso,

Can you point me a schematic of the Rudolfs dac? Yesterday I tried the OPA134s and could not get it working. The TDA1543 alone works well. And my implementation of the OPA134s seemed to work well too, with inverted input and unity gain (for starting). But when I connect the output cap of the TDA1543 to the input resistors of the opamp the sound becomes very degradated and with a lot of hum. I tested both the outputs of the dac and the opamp. I believe that I grounded well everything, so there must be other problem...

Miguel
 
IV-converter

miguel2 said:
Hi Elso,

Can you point me a schematic of the Rudolfs dac? Yesterday I tried the OPA134s and could not get it working. The TDA1543 alone works well. And my implementation of the OPA134s seemed to work well too, with inverted input and unity gain (for starting). But when I connect the output cap of the TDA1543 to the input resistors of the opamp the sound becomes very degradated and with a lot of hum. I tested both the outputs of the dac and the opamp. I believe that I grounded well everything, so there must be other problem...

Miguel

Hi Miguel, I connected the opamp just as in the datasheet of the TDA1543. The optional resistor at Vref is 3k3. For a start you can use the resistors as in the datasheet. The DC-blocking cap is after the opamp!.:cool:
 
Here is what I used. After the output cap I think I can connect any opamp there, like the opa549 from the gainclone that works well. So the opa134 should not give problems. I tried to use the Vref of the dac in the + pin but had no changes. time to think a bit more...
 

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Iv

Miguel,
This is what I use now:
R3= 3k65
C2= 3n3
???= GND
C4= 2n2 (barely readable)
Non-inverting input of the first opamp NOT connected to ground but to Vref of TDA1543.
I think you want to many things at once. First have IV-converter, one pole of low-pass filter possible, then have optional additional filtering. Currently first opamp= OPA604. This might change to AD8610. (with +/-12V supplies)
 

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Hi,

I am still confused that you throw such a big portion (>50% ) of the spectrum away

Looking at the frequency transfer of an audio system in from a musical cq. harmonic perspective: the top 10KHz to 20KHz is just the top octave from about 10 usefull octaves, so only 10% and probbably the least important 10%. Besides there is still usefull transfer at 20Khz through a 10KHz first order filter so the effect subjectively might not be so bad as the numbers suggest.

Regards,

Thijs

PS. miguel2, don't use ground as the non-inverting input for your I/V stage, use the ref pin of the TDA1543
 
Koinichiwa,

jean-paul said:
No, not forgotten. I am still confused that you throw such a big portion (>50% ) of the spectrum away. I wouldn't recommend this to others whatever explanation there is.

If cd has to be played this way I will buy a turntable again ( or go back to OS )...

Get a turntable anyway, But I will say that in this context the TDA1543 (or an even nastier chip) is an essential requirement.

All a matter of taste.

Sayonara.
 
Low-Pass Filter

jean-paul said:
No, not forgotten. I am still confused that you throw such a big portion (>50% ) of the spectrum away. I wouldn't recommend this to others whatever explanation there is.

If cd has to be played this way I will buy a turntable again ( or go back to OS )...


Hi Jean-Paul,
Agter a night sleep I decided to come back on the subject.
First your verdict was upgraded from telephone sound to LP-sound. That's quite a improvement and if the sound is as the LP I have reached my goal and I am quite happy.

May I ask you what crossoverpoint (-3dB) filterorder, and filtertype (Bessel, Butterworth, Chebysheff, etc. ) you suggest?. With FilterPro it is quite easy to calculate the values and modify the circuit.

I had the opportunity to compare my DAC with the most expensive JK-Acoustics DAC in 't Harde.
http://www.jkacoustics.nl/DACReference.htm
Johan Ketelaar was quite perplexed that I had such a good sound from a homebrew affair. Last Friday I had on audition in my home a Arcam FMJ CD33 and a Marantz SA-17S1. The Arcam was beter on CD's than the Marantz:att'n: The Arcam is Euro 2000, the Marantz Euro 2500.
My DAC had more bass but less highs than the Arcam. The Marantz had a anomaly in the bass, just did not have it under control. I decided not to buy either of them, as I preferred the sound of my own DAC/Philips CD-650. To my ears it sounds more musical more like the real thing. A week ago I attended harpsichord and clavichord concerts here in town. I don't hear the exaggerated highs at the live concert I hear from upsamplers and Hifi-ish sets. Also the volume is not that high as most Hifienthusiasts like to play, in fact volume is quite low. I concur with Kuei. Y.W. that it is all a matter of taste but I like to gauge my ears every now and then, before I loose track on the real sound.

I concur with you if you devide 20 K by two you get 10k or 50% of the audioband. I prefer to use a logaritmic scale and then 10k-20k is a very small portion of the spectrum. The filtering I use is choosen after extensive listening tests.:cool: As Thijs points out 10k-20k is only 10% of the usable octaves...
 
just a thought on non-os filtering...

I've been looking at a non-os DAC for a while now. Looking around on the net has shown:
- There must be something special about non-os given the DIY interest and the great reviews.
- >10K measurements for a non-os DAC are poor, although it does not seem to affect the sound from what people are reporting.
- The out of audio band artifacts are high and difficult to filter given the 44.1Khz sampling frequency.
- Common concensus seems to be that for os solutions its the digital filter that affects the sound quality.

What about applying a simple sample rate doubler that ups the effective sample rate to 88.2Khz and use an analog filter? The analog filter will be much easier to implement a cut off above 20Khz if the sampling rate is doubled.

Either a chip with 2x upsampling (althugh no longer a non-os solution!!) or a simple DSP to modify the i2s bitstream would work?? I'm sure someone with more experience than me has thought of this, but I'm thinking that this approach might be a good compromise to the problems outlined with non-os.

Feedback?
 
Re: just a thought on non-os filtering...

deandob said:
I've been looking at a non-os DAC for a while now. Looking around on the net has shown:
- There must be something special about non-os given the DIY interest and the great reviews.
- >10K measurements for a non-os DAC are poor, although it does not seem to affect the sound from what people are reporting.
- The out of audio band artifacts are high and difficult to filter given the 44.1Khz sampling frequency.
- Common concensus seems to be that for os solutions its the digital filter that affects the sound quality.

What about applying a simple sample rate doubler that ups the effective sample rate to 88.2Khz and use an analog filter? The analog filter will be much easier to implement a cut off above 20Khz if the sampling rate is doubled.

Either a chip with 2x upsampling (althugh no longer a non-os solution!!) or a simple DSP to modify the i2s bitstream would work?? I'm sure someone with more experience than me has thought of this, but I'm thinking that this approach might be a good compromise to the problems outlined with non-os.

Feedback?

Hi Deandob,
I have thought about the upsampler approach but gave up when I realised the upsamplers like CS8420 and AD1890 - 1893 have a digital filter "on board".
:bawling:
 
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