Experience with this DIY DAC ?

Got this DAC from ebay.

No sound when using quad speed setting!

Need help.

How do I source a192K digital stream ?
 

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If that DAC uses the CS4397 and CS8416 chips, you need to lift pin 2 of the CS8416 chip to ground and run a wire to pin 16 of the CS8416 chip.
There is much information about this DAC chip on message boards.
It will operate at 96kHz, but do not know if it is capable of 192k operation.
Good luck
 
Tks Hafp;

This DAC is based on CS4398 and CS8416. The CS4398 sampling rate is controlled by 4 dip switch to (manually) set M0, M1, M2 and M3 to ether HI or LO for sampling at 41K with de-emphasis off, 41K with de-emphasis on, 96K and 192K. It works with all setting except setting the control to 192K. When setting to 192K sampling, there is only hiss..

Wonder if anyone has experience with this DAC or is able to explain how to get the 192K sampling to work
 
Ahhhh the secret is out - this little DAC is fantastic...

I have this same DAC and am stunned at the sound quality for such a low price, especially with 16-bit lossless files. Unfortunately I am not able to try it at 192kHz setting as I don't have a source that runs this high alas.

Try replacing the op-amps for LM4562s - nice change ;)

For another tweak:

http://www.lampizator.eu/lampizator/LAMPUCERA/MAX/opamped_lampucera.html

Have you tired e-mailing the Ebay seller about the 192kHz setting? If it doesn't work then it's misleading information as it's sold as a 24/192 DAC!!!

cheers,

- John
 
Hi John;

I did email the seller. However, since this seller is in China/Hong Kong, I ran into comms problem with him.After a number of email exchanges, he insisted that the 192 sampling work.

I came across another forum (audio-karma) where another fellow is encountering identical problem with the unit.

I did some research and it appears that in order for the 192K sampling to work, the input to the DAC has to be at some higher rate than the norm 44.1/44K. There is mentioning that source to the DAC has to be something like DVD-A output. What I do not understand is that DVD-A is an encoded format and it will be required to decode the DVD-A data stream to PCM to feed into the DAC for it tow work. Do you know if that is the case ? If so, when DVD-A is decoded to PCM, what sampling rate the PCM signal is at ?
 
I think John has hit the nail on the head. The CS4398 can operate in 3 modes - single, double and quad, as mentioned. These modes control the oversampling function built into the device. If the appropriate mode is selected the CS4398 will oversample accordingly for the best performance. 128x in single speed, 64x in double speed and 32x in quad speed.

Single speed works with sample rates up to 50KHz
Double speed works from 50KHz up to 100KHz
Quad speed works from 100KHz up to 200KHz

So, for 44.1K input (which I assume is what you have?) the best mode is single speed because the DAC will internally oversample to the maximum extent possible. It appears that you have got it working in double speed at 44.1KHz but this is below the rated minimum for this mode. It's actually only if your source has greater than a 50KHz sample rate that you have to switch it into double or quad modes.

Now, if you have a 96KHz source you would need to manually switch into double speed mode. Automatic switching can be achieved by making a connection with pin 16 of the CS8416, as the 8416 detects incoming sample rates above 96KHz.

I agree that this is a great DAC. Have had mine up and running for a couple of weeks, not tweeked as yet, and am vvery pleased with it. Will have to try the LM4562 though. :cool:

Kevin
 
Glad to hear you like the sound of this DAC too Kevin. I have to say, with the LM op-amps and a few days running in and a hefty torroidal transformer supplying the power it bested my Audiosector DAC in a number of areas which I found startling. It doesn't have the rolled off treble of NOS designs either, and all the problems that go along with it.

This thing seems like a gem in terms of modding too. I intend to perform the mods I detailed in the earlier post sometime soon. Should sound even better just bypassing the second op-amp, and all the caps. I have the feeling it will be a giant beater of a DAC after these mods.

Add this to a Shigaclone and you have a top flight CD playback system for under £100 - stunning! Using mine with a PC and Foobar2000 and my 16-bit files have never sounded better :cool: Going to build a chassis for it next, but will keep it simple and probably just use a wooden chopping board. For some reason digital sounds better to me without a metal case!

- John
 
I am torn between doing the mod posted and modifying the output stage to the configuration used on the CS4398 eval board, which also does away with the coupling caps (bar one on the line output). I'm guessing Crystal will have built an eval board that will do their chip justice so it is probably worth a look. For now I'm just going to enjoy it though.

The CS4398 does upsample as part of the digital filter that's built into the chip, and the filter performs best if you run the chip in the mode that gives maximum upsampling of your source data. There's also a switched capacitor filter after the converter.

This means that the analogue output is free from crud close in to the audio passband so you can use a gentle low pass filter in the analogue output stage (-3db @ 38KHz on the eval board) that doesn't roll-off the treble. This is where it wins over NOS designs, of course.

I mainly feed mine from a Squeezebox 3 playing FLAC files ripped using EAC (this DAC makes a huge improvement over the SB3's analogue output!). Interestingly my Alpha One CD player didn't sound too hot initially. When I looked at the RF waveform from the laser it was obvious why so that now has new optics and is sounding good again, otherwise a Shigaclone would have been the next project for me!

Kevin
 
To answer Puffin's question, the oversampling is as follow:

Single speed mode: I/P sampling rate up to 50K - 128x oversampling
Double speed mode: I/P sampling rate up to 100K - 64x oversampling
Quad speed mode: I/P sampling rate up to 200K - 32x oversampling

I think Kevin is right on in saying that the best configuration is the single speed mode and I am not sure why it works when I configure the DIP switches with double speed mode when the I/P sampling is at 44.1K though?

Currently I am using the OPPO970 as the 'transport' to feed into this DAC. I've done some 'capacitor upgrade' to the OPPO970 in both the analog O/P circuit and also, the PS. I have also added bypass capacitors at various 'points' of digital chip of the OPPO.

I remain presently surprised that this DAC sounds 'subjectively better ;-) than the mod OPPO970 analog out. Currently, I am using this DAC to feed into a Yain tube buffer (also modded using input from other threads in this forum), than to a tube amp.

I will try out the LM4562 chip when I receive them.

BTW, the OPPO970 appears to be capable of outputting at 192K with its HDMI connector. I am not sure yet if it is straight forward to tap of the digital signal from the HDMI connector to feed into this DAC?



Kevin - Do appreciate if you will post your experience with this DAC when you start 'modding' it.

Do encourage others who have experience with modding this DAC to support this thread.

One thing for sure though - it does sound good as is.

One other note - I did change the capacitor values for the PLL for this DAC. It appears that the stock capacitor values are the ones recommended for CS4397. If you look at the spec for the CS4398, the PLL values recommended are different from those that are mounted in the stock board.
 
I don't know a lot about HDMI. My feeling is that, if the appropriate mode is selected, the DAC is operating at or near its' highest sample rate and the oversampling is carried out internally in a digital filter whose characteristics are carefully matched with the switched capacitor filter that follows the DAC.

Upping the rate elsewhere in the chain is not likely to be advantageous, IMHO, but that's purely a gut feeling that I can't back up with any listening. I've not fed anything other than 44.1 into it yet.

The CS8416 outputs a "96K" signal that indicates if the incoming sample rate is above an arbitrary threshold that's somewhere between 48k and 88k but to determine the input rate exactly would require some additional hardware as far as I can see. This would be required to switch the DAC into Quad speed mode for inputs above 100K.

Sounds like a microcontroller driving a display on the front panel, counting the sample rate via one of its' timer channels and configuring the CS8416 and CS4398 according to what it sees. ;)

Kevin
 
Well, a tube output stage is something else that was in the back of my mind to try. Very interesting.

Interesting also that you changed the PLL mode of the 8416 to minimise outband jitter. I noticed this setting appeared to be non-ideal. Looks like I'm going to be busy with this DAC but given that I'm already very happy with the sound I have high hopes for what can be achieved with a little work.

Kevin
 
Kevin, have you decided which mod to implement yet? I can't help feeling that the op-amp mod, with no capacitors in the way to muddy the sound will be very impressive indeed.

What other options are there... can the output from the first op-amp be taken directly from the op-amp leg and just put through a 2.2uF film cap - for example - and directly to phono sockets? Would that work? Probably easier than trying to squeeze a 14K7 resistor in there as per the first mod I posted.

- John
 
It should do. The 14K7 resistor is only an attempt to cure the offset anyway, at the expense of injecting power supply noise.

I haven't obtained / figured out the whole circuit yet so I'm not 100% sure what the purpose of the 2nd opamp is. I suspect it's just a buffer stage and totally unnecessary.

It might also be interesting to take a look at what Cirrus do on their Eval board Here (Page 11). Pretty much what you describe. DC coupling from the DAC to the first OpAmp and just a single decoupling cap on the output. If you really get keen you could use an integrator to feed back the DC error and cancel the offset.

When I can prise myself away from listening to it I think I'll try both the valve route and the Op Amp. See what sounds best.

Kevin