| dieringe |
Hi,
I got 4 PCM1704 ridiculously cheap and put together the above components just following data sheets and evaluation board descriptions.
I use Nelson Pass' D1 output stage for the I/U-conversion.
The thing sounds incredible as it is, but as I just built it on experimenting board (yes I found one for the DF too!) and connected everything with wires I think there is much room for optimization. So my questions are:
1. What do I win by etching a PCB?
2. Does a quartz oscillator help and if so, which frequency at which part?
(I don't have one yet, the receiver makes the master clock by itself.)
The eval. board describes a 24.576MHz osc. for the DF1704, but if I understood it right, the DF has an oscillator built-in, so that wouldn't help much?
The elektor DAC 2000 (which uses same parts) has a 6.xxx MHz osc. at the CS8414.
Which is the better way?
(I want to play CD and DVD-Audio, so it should be compatible to all sampling frequencies.)
3. Maybe somebody experimented with the slow vs. fast rolloff-filter settings of the DF1704?
4. Power supply. I use a 2x15V transformer, one 7815 / 7915 pair, one
7805/7905 air for the DACs and one 7805 each for filter and receiver.
(output stage is completely separated)
I guess here is a lot of room for improvement.... Is there a schematic with a high-quality solution somewhere?
5. That's all I can think of for now, further suggestions are very welcome
thanks,
martin |
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| Hattori Hanzo |
| I would really encourage you to leave out the input receiver together with the digital filter i.e. to go the non-oversampling filterless route. Try to connect these PCM1704's to the EIAJ or I2S output of your player (decoder) directly. At least do yourself a favour and try it once before you start fiddling around with clocks, filter slopes, power supplies and complicated PCB designs. Please! |
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| Elso Kwak |
| quote: | Originally posted by Hattori Hanzo
I would really encourage you to leave out the input receiver together with the digital filter i.e. to go the non-oversampling filterless route. Try to connect these PCM1704's to the EIAJ or I2S output of your player (decoder) directly. At least do yourself a favour and try it once before you start fiddling around with clocks, filter slopes, power supplies and complicated PCB designs. Please! |
Connecting 24 bit PCM1704 DACs to a 16 bit format will result in a pretty low volume sound as the most significant bits are not used. You need some kind of shift register to get this right but why using a 24 bit DAC on a 16 bit format, NON-OS?
:bigeyes: |
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| dieringe |
yes I would probably like to try this although what I have now sounds better than anything I heard before.... :)
I would have to find the I²S connections in my cd/dvd-audio player and convert them to a format suitable for the PCM1704. I think these are 4 lines? Lots of work...
Is there a finished concept about this? How do I get left and right SDATA signals without the DF? |
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| borges |
Martin, you win a lot by puting it on a PCB. Most importantly, you put yourself in the driver seat when it comes to board partitioning, location of return currents (important if you're wireing together demo boards), grounding, and decoupling. Search for "Supply decoupling and layout of circuits with digital ICs" by Guido Tent. I got it from this forum.
I've used Eagle a lot with these chips. Beware that the footprint of the PCM1704 isn't straight out of the box. If you go for a board, remember to print it out and try to fit the chip! Are you using Eagle? If so I can mail you some of my stuff including the PCM1704.
What kind of analog filter are you using? I was never one of these non-OS guys, rather I go for custom analog filters and digital oversamplers born in Matlab and implemented in Xilinx.
I clock the DACs from the crystal oscillator in the CD drive itself. It is this clock (multiplied or divided) that is multiplexed into the SPDIF stream and extracted by the CS8414. On my current boards I can select between CD drvie clock and CS8414 derived clock. I don't let an oversampler handle clocking because the 384x or 256x clocks I get from CD drive or CS8414, respectively, make nice frequencies for a bitclock with 24bits at 8x oversampling. I actually use either one directly to the bitclk on the DAC chips and make the FPGA run asyncronously by polling it.
Greetings,
Børge |
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| Hattori Hanzo |
| quote: | | I would have to find the I²S connections in my cd/dvd-audio player and convert them to a format suitable for the PCM1704. I think these are 4 lines? Lots of work... |
| quote: | | Is there a finished concept about this? How do I get left and right SDATA signals without the DF? |
Sorry, I wasn't aware that the PCM1704 is not stereo and you will definitely need some glue logic. It also depends on the format that is available in your player. Most probably there's no finished and proofed concept but there are similar examples. One is here: http://www.diyaudio.com/forums/show...&threadid=39993
Still, try to get around the input receiver and the DF once, maybe with another DAC and/or player... |
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| dieringe |
Borge, thanks for the paper hint, I will need this if I ever make a PCB...
I think I would use Eagle, it would be my first PCB, I did everything on experimenting board until now... So your Eagle stuff would probably help me.
I don't use any analog filter that I am aware of... the D1 output stage has 2 "1st order low pass filter" capacitors on in- and output and I even took them out because I felt they took away too much.
I think I can hear some aliasing but of course I don't know where it comes from. This is why I am thinking of optimizations.
You have a separate line for the clock from your CD drive?
I guess this is only possible with drive and DAC in the same case?
I would like to have the DAC independent of the drive/cd player.
Or would it be possible to use a computer DVD drive as DVD Audio player? I saw some people using computer CD drives but no DVD yet - ?
m. |
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| Fabian |
| quote: | Originally posted by Elso Kwak
Connecting 24 bit PCM1704 DACs to a 16 bit format will result in a pretty low volume sound as the most significant bits are not used. You need some kind of shift register to get this right but why using a 24 bit DAC on a 16 bit format, NON-OS?:bigeyes: |
It will have exactly the same volume as when it is fed with 24bit data. You have to align the MSB of the 16bit data stream with the MSB of the DAC input. A working basic circuit can be found by clicking the link provided by Hattori Hanzo. Modestly said, I think the PCM1704 is a good alternative to TDA 16bit DACs. :)
| quote: | Originally posted by dieringe
3. Maybe somebody experimented with the slow vs. fast rolloff-filter settings of the DF1704?
I don't use any analog filter that I am aware of... the D1 output stage has 2 "1st order low pass filter" capacitors on in- and output and I even took them out because I felt they took away too much.
|
Yes I've compared slow vs. fast roll of and I prefered slow filter. Switching the filter response is quite easy to implement. Just compare and see what's more apealing to your ears.
As you've already removed the filter from your D1 outputstage, I guess you will like the soft roll of setting. Removing the whole digital filter is like going one step further. |
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| Jocko Homo |
The DF1704 accepts I2S directly, without glue logic. It is all in the data sheet...................
So......you want to take out all of the filtering? Why? Ever hear of IMD and aliasing?
Jocko |
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| zinsula |
| quote: | Originally posted by Fabian
As you've already removed the filter from your D1 outputstage, I guess you will like the soft roll of setting. Removing the whole digital filter is like going one step further. |
Do you really advise to remove analog and digital filtering? At least one has to be careful about that, as it throws lots of garbage above 20kHz to the amp+speaker. The output of a D/A converter is stepped (settling time of 1704: 200ns), with resulting high frequency content. A 200ns step gives me 2.5MHz......
Ciao, Tino |
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| Hattori Hanzo |
| quote: | | So......you want to take out all of the filtering? |
Yes, as a proposal to answer some (all?) of dieringe's questions. Compared to his intention (receiver, DF, clock, PCB etc.) this would be an easy and worthwhile experiment. I cannot really comment on the DF1704 though.Maybe because it might sound good?| quote: | | Ever hear of IMD and aliasing? |
Yes, and also about the various debates regarding the filterless versus filtered approach. I didn't say the filterless way sounds best under all circumstances but I would at least give it a try. And it doesn't hurt... |
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| borges |
Without analog filtering, you get linear phase response. However, designing a linear-phase digital filter is actually hard to avoid, no matter the rollof.
Greetings,
Børge |
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| borges |
Martin,
mail me your email address and I'll send you my Eagle library featureing the 1704 and other oddities.
Greetings,
Børge |
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| Jocko Homo |
Too bad every signal encoded onto a CD is run through a brick wall on the A/D end.
It doesn't hurt?? Your tweeter may not agree.
Jocko |
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| dieringe |
| quote: | Originally posted by Jocko Homo
It doesn't hurt?? Your tweeter may not agree.
|
the tweeter shouldn't bother, it's a lowpass filter itself if it isn't a supersonic tweeter |
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| Jocko Homo |
Just make more HF thash.
Enjoy the sound!
Jocko |
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| dieringe |
| quote: | Originally posted by Fabian
Yes I've compared slow vs. fast roll of and I prefered slow filter. Switching the filter response is quite easy to implement. Just compare and see what's more apealing to your ears.
As you've already removed the filter from your D1 outputstage, I guess you will like the soft roll of setting. Removing the whole digital filter is like going one step further. |
well my perception is that the fast roll-off gives a little better resolution, the slow roll-off gives a more analog-like sound. can't really decide which I like better.
I have another question:
6. there is a bunch of capacitors at each PCM1704. As this is HF stuff I guess tantalum electrolytics are the preferred type?
m. |
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| vit |
I wouldn't use tantalum caps. It's better to use OsCon electrolytics bypassed with a ceramic or MKP film cap.
Recommended paper written by Guido Tent :yes:
Check out Olimex for cheap proto PCB.
//vit
edit: Olimex recommendation |
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| Peter Daniel |
| quote: | Originally posted by dieringe
there is a bunch of capacitors at each PCM1704. As this is HF stuff I guess tantalum electrolytics are the preferred type?
|
Here's what Madrigal's using in their $17K DAC . The chips under nude Vishays are PCM1704 http://www.marklevinson.com/image_l.../30_6DAC_lo.jpg |
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| rfbrw |
| quote: | Originally posted by Jocko Homo
Just make more HF thash.
Enjoy the sound!
Jocko |
If he can do this
| quote: | Originally posted by dieringe
<snip>
I think I can hear some aliasing
<snip>
|
with a 8x oversampling dac, he may well enjoy the HF hash. |
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| dieringe |
| quote: | Originally posted by rfbrw
If he can do this
with a 8x oversampling dac, he may well enjoy the HF hash. |
so what is it that I hear? jitter? |
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| rfbrw |
| quote: | Originally posted by dieringe
so what is it that I hear? jitter? |
Without actually hearing what you hear, I can't possibly comment but as the first image occurs at 352.8kHz when 8x oversampling is applied, I doubt its an alias thats to blame.
An error in your construction of the D1 output stage would seem to be a more likely candidate. |
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| Elso Kwak |
| quote: | Originally posted by Fabian
It will have exactly the same volume as when it is fed with 24bit data. You have to align the MSB of the 16bit data stream with the MSB of the DAC input. A working basic circuit can be found by clicking the link provided by Hattori Hanzo. Modestly said, I think the PCM1704 is a good alternative to TDA 16bit DACs. :)
|
I was referring to the case NOT using shifregisters.
The circuit by Hattori Hanzo will degrade the sound for sure and also contains too many parts.
:cool: |
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| Elso Kwak |
| quote: | Originally posted by borges
Without analog filtering, you get linear phase response. However, designing a linear-phase digital filter is actually hard to avoid, no matter the rollof.
Greetings,
Børge |
Hi Børge,
An analog Bessel low pass filter has linear phase or constant group delay in the pass band and is not hard to design.
The only problem is the Bessel does not have a sharp cut-off at the crossover frequency. This makes the case a difficult one if you go non-oversampling as you want to filter all high frequency products above 20kHz, yet want to preserve a flat response in the 20-20kHz band. In the very early days of the CD we had high order Butterworth brick-wall filters that inevitable had a lot of ringing on the pulse response. Common knowledge in DA-conversion is that the highest frequency of interest should be about 10 times lower than the sampling frequency in order to avoid aliasing distortion and filtering problems. In the case of NON-OS this simply cannot be accomplished as you would end up with a 4,4 kHz filtered audio signal. With 8x oversampling no problem. Unfortunately we cannot change the format of the redbook CD.
Yes and Jocko is right about the brick-wall filter at the recording end before the AD converter. Here obviously we can do nothing about it.
:eek: |
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| Guido Tent |
they still don't get it
The PCM1704 has a clear digital side (pins 1 to 10) and an analog side (11 to 20) hence should be laid out 90 degrees rotated compared to what Levinson does
some things are so simple...... |
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| Elso Kwak |
Hi Peter,
Funny seeing OPA627AP's in a US$ 17,000 DAC and not OPA627BP's.
Highend, esoterica........?? :rolleyes: Maybe be in the next, even more higher priced version?:D |
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| dieringe |
ok, but what's the type of these caps?
what are Vishays, why are they nude and why are they on the chips? :xeye:
poor design, only 2 PCMs altogether I guess?
edit: no thats wrong :cannotbe: |
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| Elso Kwak |
Yeah, and what is that 743 thingy. Oh, I see a LT1027 5V reference.......:D
Where do we need that for in this DAC?:confused: :confused: |
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| Guido Tent |
it looks like the vishays' legs are shorted at the PCB, or am I wrong ?
cheers |
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| dieringe |
| quote: | Originally posted by Guido Tent
The PCM1704 has a clear digital side (pins 1 to 10) and an analog side (11 to 20) hence should be laid out 90 degrees rotated compared to what Levinson does
|
but what are AD826 doing and why are they on the digital side? |
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| dieringe |
| quote: | Originally posted by Elso Kwak
Yeah, and what is that 743 thingy. Oh, I see a LT1027 5V reference.......:D
Where do we need that for in this DAC?:confused: :confused: |
ML is a power supply fetishist |
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| Guido Tent |
| quote: | Originally posted by dieringe
but what are AD826 doing and why are they on the digital side? |
good question. Maybe they serve as clock buffers, but these should be at the senders' side......... |
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| Guido Tent |
| quote: | Originally posted by dieringe
but what are AD826 doing and why are they on the digital side? |
good question. Maybe they serve as clock buffers, but these should be at the senders' side......... |
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| Peter Daniel |
| quote: | Originally posted by dieringe
poor design, only 2 PCMs altogether I guess?
|
Two per channel |
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| Peter Daniel |
Here's what they say about Vishays:| quote: | | These are genuine R-2R ladder DACs, compared with the delta-sigma (1-bit) DACs used by many other 24-bit processors, and are hand-adjusted in manufacture, using a precision Vishay trim resistor, a microscope, and a computer test setup. |
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| dieringe |
| I'd like to have additional single ended outputs. If I just connect the positive output to an RCA socket, I will lose the error-correction function of the symmetrical setup. Is there a simple (and well-sounding) way to combine + and - to make single-ended output? |
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| dieringe |
| quote: | Originally posted by Peter Daniel
...are hand-adjusted in manufacture, using a precision Vishay trim resistor, a microscope, and a computer test setup.
|
A MICROscope! :confused: :confused: |
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| Peter Daniel |
| I wouldn't mind if you translate this, as I'm curious about the microscope technique myself;) |
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| analog_sa |
| I guess the laser is used for a couple more cuts to trim the nude resistors. And this is done under a microscope. |
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| dieringe |
| quote: | Originally posted by Peter Daniel
I wouldn't mind if you translate this, as I'm curious about the microscope technique myself;) |
"Another highlight is the internal upsampling of the 360S. Madrigal
prefers to generate integer multiples of the received data rate
instead of "inventing" intermediate steps. The 44.1kHz of a cd for
instance are calculated to 352.8kHz, 48 or 96 to 386kHz. And as we
deal with a Mark Levison, maximum smoothnees in calculation and sonic
result are practically self-evident."
....
no you don't want to read the rest of it. the microscope is not really explained. only something like trimming the resistors to 0.0002% for over-all minimum distortion of the whole converter circuit
"the ernormous list of technical delicacies could be continued for pages..."
For me this is not an option! :dead:
which paper is this from? |
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| Peter Daniel |
| Image HIFI (2001) |
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| Guido Tent |
| quote: | Originally posted by Peter Daniel
I wouldn't mind if you translate this, as I'm curious about the microscope technique myself;) |
wow, 0.0002%
do they know about tempco of resistors ?
putting one off silicon destroys the behaviour of an R-2R.......
cheers |
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| A 8 |
What exactly do they adjust with a resistor?
From what I can read of the datasheet there are no adjustment options on the PCM1704.
/Michael |
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| Elso Kwak |
| quote: | Originally posted by A 8
What exactly do they adjust with a resistor?
From what I can read of the datasheet there are no adjustment options on the PCM1704.
/Michael |
Hi I guess it is the IV-resistor form output of the opamp (OPA627AP?) to the inverting input. Non-inverting input connected to ground.
As the picture is of one channel you can trim the inverted and non-inverted signal to be as equal as possible. Been there, done that, with AD1865 and TDA1543 in balanced mode, not with a microscope, just with a trimmer pot.
:cool: |
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| Peter Daniel |
| quote: | Originally posted by Guido Tent
wow, 0.0002%
do they know about tempco of resistors ?
putting one off silicon destroys the behaviour of an R-2R.......
|
Could this be a reason they put them directly over the plastic casing of a chip? |
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| Elso Kwak |
| quote: | Originally posted by Peter Daniel
Could this be a reason they put them directly over the plastic casing of a chip? |
I have read somewhere that the idea was to get thermal tracking of the DAC and the resistor. If the DAC heats up the resistor also does.......I don't know squat if this makes sense or how the PCM1704 changes when heated up. Maybe it is all made up to be "interesting" to audiophiles.......This type of Vishay resistors have a very low Tempco. (about 1ppm) :rolleyes: :rolleyes: :rolleyes: |
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| A 8 |
| Ok, as you guy say they probably put it face down on the chip for temp drift reasons but would the not get RFI issues as its partially over the digital section of the chip? |
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| Guido Tent |
| quote: | Originally posted by A 8
Ok, as you guy say they probably put it face down on the chip for temp drift reasons but would the not get RFI issues as its partially over the digital section of the chip? |
Hi
For RFI it doesn't work
Temperature balancing only works for slow changes, the time constants are small.
cheers |
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| rfbrw |
| quote: | Originally posted by Elso Kwak
I was referring to the case NOT using shifregisters.
|
I think you will find shift registers in some quantity or another hard to avoid.| quote: |
The circuit by Hattori Hanzo will degrade the sound for sure and also contains too many parts.
:cool:
|
And what is the basis for the assumption? The size of a circuit stems from the function the circuit is supposed to perform not from some notion of audio 'goodness' based on the number of parts. |
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| Guido Tent |
| quote: | Originally posted by rfbrw
And what is the basis for the assumption? The size of a circuit stems from the function the circuit is supposed to perform not from some notion of audio 'goodness' based on the number of parts. |
Hi
The only thing I can think of is increased jitter due to more noisy gates, but if you implement it correctly, it shouldn't be a problem
so I agree, more is not always worse
regards |
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| borges |
| quote: | Originally posted by Elso Kwak
I have read somewhere that the idea was to get thermal tracking of the DAC and the resistor. If the DAC heats up the resistor also does.......I don't know squat if this makes sense or how the PCM1704 changes when heated up. Maybe it is all made up to be "interesting" to audiophiles.......This type of Vishay resistors have a very low Tempco. (about 1ppm) :rolleyes: :rolleyes: :rolleyes: |
So what can you do with two PCM1704s? Obviously, you can give one inverted data and go differential all the way to the output. But do you _really_ need perfect matching of gain and/or offset for that? With true differential signalling, a small signal-correlated common-mode signal (i.e. different gain in the + and - branches) should be accepted.
Or do you think they try in some way to add the output of two 24-bit converters to somehow gain a total resolution of 25 bits or more? Right now (on the second day of a cold and approaching bedtime) that's one of the few applications I can think of where you really need extreme matching.
Good night,
Børge |
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| rfbrw |
| quote: | Originally posted by Guido Tent
Hi
The only thing I can think of is increased jitter due to more noisy gates, but if you implement it correctly, it shouldn't be a problem
so I agree, more is not always worse
regards |
That's not my bone of contention. What I am questioning is the notion that one decides how many chips to use based on some scale of audio 'goodness' and then sets about shoehorning the design into them.The logical conclusion of this ridiculous notion is that ASICs and FPGAs ought to be avoided like the plague. I often wonder how these chaps think digital audio devices format or demux data in the first place. By leprechaun, perhaps? |
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| banana |
| quote: | Originally posted by Elso Kwak
I have read somewhere that the idea was to get thermal tracking of the DAC and the resistor. If the DAC heats up the resistor also does.......I don't know squat if this makes sense or how the PCM1704 changes when heated up. Maybe it is all made up to be "interesting" to audiophiles.......This type of Vishay resistors have a very low Tempco. (about 1ppm) :rolleyes: :rolleyes: :rolleyes: |
It reminds me of the build in I/V resistor in PCM63.
However, the real benefit of this build in I/V resistor approach was not mentioned in the datasheet.... |
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| dieringe |
| quote: | Originally posted by Hattori Hanzo
Sorry, I wasn't aware that the PCM1704 is not stereo and you will definitely need some glue logic. It also depends on the format that is available in your player. Most probably there's no finished and proofed concept but there are similar examples. One is here: http://www.diyaudio.com/forums/show...&threadid=39993
Still, try to get around the input receiver and the DF once, maybe with another DAC and/or player... |
He only says that the PCM1704 sounds better in non-os config than his tda-xyz converter. This is no miracle. He doesn't state non-os is better than os.
So I'm not very confident about this. In addition the shift register circuit is incomplete, yields phase shift etc.
He couldn't get stopped-clock operation running which is the only worth trying IMO, so he has no evaluation of it.
Did anyone really compare a stopped-clock non-os pcm1704 to os with a df11704?
Anyone has ideas how to implement something like the integer-oversampling ML does according to the article?
m. |
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| rfbrw |
| quote: | Originally posted by dieringe
He only says that the PCM1704 sounds better in non-os config than his tda-xyz converter. This is no miracle. He doesn't state non-os is better than os.
So I'm not very confident about this. In addition the shift register circuit is incomplete, yields phase shift etc.
He couldn't get stopped-clock operation running which is the only worth trying IMO, so he has no evaluation of it.
|
Not sure why you think it is incomplete and definitely don't know where the phase shift idea comes from. Having tried both methods with the PCM63/1702 I could discern no difference between the two methods. They both varied wildly with the music played. FWIW the better NPC digital filters use the stopped clock method so perhaps there is something in the idea that there is an advantage in having the data static for a short while prior to conversion.| quote: |
Did anyone really compare a stopped-clock non-os pcm1704 to os with a df11704?
|
Herr Altmann, perhaps?
| quote: |
Anyone has ideas how to implement something like the integer-oversampling ML does according to the article?
m.
|
What article?
Integer oversampling is exactly what the DF1704 does. All it means is you oversample by whole numbers. If you are prepared to decimate you can apply the same techniques to non-integer oversampling.
Oi, werner, any objections to the above ? |
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| borges |
| quote: | Originally posted by dieringe
Did anyone really compare a stopped-clock non-os pcm1704 to os with a df11704?
Anyone has ideas how to implement something like the integer-oversampling ML does according to the article?
m. |
What is stopped-clock non-os pcm1704? Does it mean that it receives indata at 44.1kHz and holds its output constant at 1/44100 seconds?
The way it's usually done is to take your 44.1/16 signal and insert 7 zeros between each original sample. Then you convolve that zero-inserted signal with a finite-duration, windowed sinc with 7 non-zero values between each zero (and the central one). The result of this is that each 8th sample is an original 16-bit number from your starting pcm signal and that between those, 7 derived (interpolated) samples are inserted. (You may choose to round those down to 24-bit number or any resolution you choose.) If your windowed sinc was infinitely long, the resulting digital signal contains the same information as the original signal. The benefit of oversampling is that input data is fed to your DAC at an 8 times higher rate. A PCM DAC like '63, '1702 and '1704 holds its analog output constant for one sampling period. The shorter the sampling period, the easier the rounding-off job required by the analog filter (because the output from the DAC is already greatly rounded off). Also, because intermediate signals are inserted, the differences between consecutive analog outputs are small.
In practicality, an 8x oversampler is built as three 2x oversamplers in series. That vastly simplifies the implementation. Because of this and because it is easy to divide a clock by 2^n, oversampling rates usually come as powers of two.
Oversampling involves no phase shift. It delays the data, but because the windowed sinc is symmetrical and of finite duration, all frequencies have the same delay.
Greetings,
Børge |
|
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| dieringe |
| quote: | Originally posted by borges
What is stopped-clock non-os pcm1704? Does it mean that it receives indata at 44.1kHz and holds its output constant at 1/44100 seconds?
|
forget it, I have to read more datasheets.
What I meant was holding the left channel while the right channel is loaded, but this is probably done anyway by the DAC?
But then I wouldn't need a shift register either - ?
I have no experience with this stuff, so I have lots of misunderstandings and better shut up.
I hoped there existed a concept I could try to understand...
m. |
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| dieringe |
Ok here is my stopped-clock idea:
For the left channel I OR the BCK with LRCK, feed the result into left PCM1704 BCK (this means BCK is stopped when right channel data is transferred), and OR the right channel with negative LRCK, feed into right PCM1704-BCK (or vice versa). Both channel DACs see all the SDATA but would ignore them when there BCK is stopped. Could something like this work?
m.
edit: ok, something must be done to WCLK also in a similar fashion... |
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| borges |
Martin,
you should do like I did with these matters:
1) Get a Xilinx or Altera FPGA demoboard
2) Learn Verilog or VHDL
3) Design any digital glue logic you can imagine
If you go for Verilog, I've got working code to read I2S and write to PCM1704 (and friends). My code is for polled bckl, which I get from the I2S of the CD drive or from the CS8414. (Should probably work with synchronous clocks too, but my SpartanIIe demoboard is a little low on clock resources.)
Maybe you can achieve non-os operation with simple glue logic. If you do, at least make sure you're using the upper 16 bits of the DAC. To do that, wouldn't you have to shift in 8 more zeroed lsbs after the 16 bits of data from the receiver?
Greetings,
Børge
| quote: | Originally posted by dieringe
Ok here is my stopped-clock idea:
For the left channel I OR the BCK with LRCK, feed the result into left PCM1704 BCK (this means BCK is stopped when right channel data is transferred), and OR the right channel with negative LRCK, feed into right PCM1704-BCK (or vice versa). Both channel DACs see all the SDATA but would ignore them when there BCK is stopped. Could something like this work?
m.
edit: ok, something must be done to WCLK also in a similar fashion... |
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| rfbrw |
| quote: | Originally posted by dieringe
Ok here is my stopped-clock idea:
For the left channel I OR the BCK with LRCK, feed the result into left PCM1704 BCK (this means BCK is stopped when right channel data is transferred), and OR the right channel with negative LRCK, feed into right PCM1704-BCK (or vice versa). Both channel DACs see all the SDATA but would ignore them when there BCK is stopped. Could something like this work?
m.
edit: ok, something must be done to WCLK also in a similar fashion... |
The logic you need depends on the CS8414 format you choose. Some formats make things easier than others.
Re demoboards and HDLs.
By all means get a development board, I have 4 and am designing a 5th but be aware the logic for stopped clock operation will fit in a XC9536 cpld that costs around 5 Euros. It is available in 44pin PLCC, a format relatively easy to work with whether SMD or socketed and schematic entry tools are freely available. A full blown FPGA development board on the other hand will cost anything from around 100Euros to the price of a small car.
As for HDLs,they are hardware DESCRIPTION languages. There is little point bothering with them until you fully understand the hardware you are describing. |
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