
Home  Forums  Rules  Articles  The diyAudio Store  Gallery  Blogs  Register  Donations  FAQ  Calendar  Search  Today's Posts  Mark Forums Read  Search 
Digital Source Digital Players and Recorders: CD , SACD , Tape, Memory Card, etc. 

Please consider donating to help us continue to serve you.
Ads on/off / Custom Title / More PMs / More album space / Advanced printing & mass image saving 

Thread Tools  Search this Thread 
28th December 2012, 07:35 AM  #11 
diyAudio Member
Join Date: Mar 2003
Location: Haarlem, the Netherlands

Presumably each DAC's output produces a constant current until it receives its next sample. Therefore, you will have two DACs producing current per channel, which basically doubles the current compared to a singleDAC circuit.
Interleaving two four times oversampled DACs like this does give you eight times oversampling, but you get some extra lowpass filtering for free. That is, as each DAC keeps its output constant for 1/(4 fs) instead of 1/(8 fs), the zeroorder hold filter transfer goes to zero at all multiples of 4 fs rather than 8 fs. This makes life easier for the analogue post filter and probably reduces sensitivity to jitter somewhat, but it needs to be compensated for in the digital oversampling filter. 
28th December 2012, 07:41 AM  #12 
diyAudio Member
Join Date: Sep 2007
Location: Hangzhou  Marco Polo's 'most beautiful city'. 700yrs is a long time though...
Blog Entries: 120

Its that 'free low pass filtering' that gives the clue as to this not really being 8X OS. The point of 8X OS is to move the image frequencies out further from the wanted frequency band  so they would be centred around 384k, but they aren't in this case as you point out. So to me its unclear what the technical benefits might be.
__________________
“A new scientific truth does not triumph by convincing its opponents ... but rather because its opponents eventually die, and a new generation grows up that is familiar with it.”  Max Planck 
28th December 2012, 08:14 AM  #13 
diyAudio Member
Join Date: Mar 2003
Location: Haarlem, the Netherlands

Hi Abraxalito,
Why do you think the images are not centred around 8 fs? I would expect that the images are centred around 8 fs, but the zero order hold suppresses both 4 fs and 8 fs. This is all assuming perfectly matched DACs. If you have mismatch between the DACs, you will get some imaging around 4 fs back, but much less than with just a single DAC running on 4 fs (depending on how well the conversion gains of the DACs match). Regards, Marcel 
28th December 2012, 09:22 AM  #14 
diyAudio Member
Join Date: Sep 2007
Location: Hangzhou  Marco Polo's 'most beautiful city'. 700yrs is a long time though...
Blog Entries: 120

Hi Marcel,
It strikes me that to get frequencies centred around 8Xfs you do need to update the DACs with new data every 8X clock, not have them holding the same data for two of these clocks. Otherwise seems too much of a free lunch. But if you're right then this would allow subunity OS ratios, as follows... Suppose I wanted to do a NOS DAC but instead I ran the two DACs at 22kHz, but interleaved  it would be a cool solution if it worked, but I can't see it. You reckon this would work? If so I plan to design the first subNOS DAC How far could we extend it? Run 4 interleaved DACs at 11kHz? As I said, seems too good to be true. <edit> Here's my handwaving argument for why it can't work. An 8X OS DAC reproduce frequencies up to 8 * 22kHz = 176kHz. To create a 176kHz sinewave at full scale it needs to swing from positive full scale to negative full scale on each cycle. This is the worst case. But if only one of two DACs (whose outputs are summed) is updating at this rate then the combined output can only swing down to 0V because the first DAC is still holding the previous positive fullscale value. The output can only swing to ve full scale after both DACs have been updated, which of course takes 2 clocks. So no longer can we create 176kHz at full scale. Perhaps we could create 176kHz at half scale (6dB) though if I thought carefully about it
__________________
“A new scientific truth does not triumph by convincing its opponents ... but rather because its opponents eventually die, and a new generation grows up that is familiar with it.”  Max Planck Last edited by abraxalito; 28th December 2012 at 09:34 AM. 
28th December 2012, 10:34 AM  #15 
diyAudio Member
Join Date: Mar 2003
Location: Haarlem, the Netherlands

Hi Abraxalito,
The problem with your subNOS DAC is that the first zero of the response of the zeroethorder hold gets rather close to the band of interest, even in the band of interest when you have more than two DACs. You could solve that using returntozero DACs, but on top of that, the requirements on DAC matching will be impossibly tough. My reasoning is as follows: Start with a digital signal with sample rate fs. According to discretetime signal processing theory, its spectrum is repetitive, having spectral copies around all multiples of fs. When you put this into a theoretical perfect impulse DAC, its output will be a series of infinitely high and infinitely narrow (Dirac) impulses having a charge or flux that corresponds to the digital number and, of course, having a sample rate of fs. Now comes the digital interpolating lowpass filter with sample rate 8 fs. If the filter works properly, its output signal only has spectral copies around multiples of 8 fs. When you put this into a theoretical perfect impulse DAC, its output will be a series of Dirac impulses with a sample rate of 8 fs. When you demultiplex the signal across 2 perfectly matched perfect impulse DACs running at half the clock rate, whose clocks are shifted by 1/(8 fs), and add the outputs, you get the exact same series of impulses as you would have got with the single impulse DAC. Hence, the spectrum is also the same, and spectral copies only occur around multiples of 8 fs. For a number of practical reasons, a reallife DAC does not output infinitely high and narrow impulses. Instead, it may output a pulse with finite height and width T. Mathematically, such a signal could have been obtained by passing the impulse signal through a zeroethorder hold filter, that is, a filter having an impulse response that is constant during a time T and then drops to zero. Being linear and time invariant, such a filter cannot create any frequencies that weren't there before, it can only filter the frequencies that are already there. In fact it acts as a lowpass having a sin(pi*f*T)/(pi*f*T) magnitude response. Assuming perfect DAC matching, the only difference between a conventional system with a single DAC running at 8 fs and the Sony system with two interleaved DACs running at 4 fs is the different value of T. Conventionally, T=1/(8 fs), but for Sony, T=(1/4 fs). Hence, the Sony zeroethorder hold filters a bit more. Now with mismatch. For simplicity, start with total mismatch: one DAC works and the other doesn't. The working DAC sees an input signal with sample rate 4 fs. Hence, its input signal spectrum has spectral copies at all multiples of 4 fs. The whole system then degrades to a fourtimesoversampling system. With partial mismatch, the output signal spectrum will be something in between the cases with no and with complete mismatch. For example, suppose the first DAC has 1 % more conversion gain than intended while the second DAC has exactly the right conversion gain. You could get the same signal by taking two DACs with the intended gain, and adding one DAC running at 4 fs with 1 % of the desired DAC conversion gain. The spectral copies at odd multiples of 4 fs will then reoccur, but at only 1 % of the level you would have had in a singleDAC system. As the desired signal is twice as strong as in a singleDAC system, the ratio of desired signal to 4 fs spectral copy is still 200 times (46.02... dB) larger than with a single DAC running at 4 fs. By the way, it is true that you can't make 176.4 kHz anymore with the Sony system. That is the consequence of the time T of the zeroethorder hold increasing to 1/(4 fs): the first zero of the filter response will be at 176.4 kHz when fs=44.1 kHz. For CD that doesn't matter because the desired signal is below 22.05 kHz anyway. Regards, Marcel Last edited by MarcelvdG; 28th December 2012 at 10:38 AM. 
28th December 2012, 12:24 PM  #16  
diyAudio Member
Join Date: Mar 2012
Location: Bradford

Quote:
Perhaps I misread the symbols...the pin I took to be inverted WS is marked WS with a bar. The filter is CXD1144. I'll look at the datasheet more closely when I've got some thinking time, and look with a 'scope. For now, I note that the 337ESD manual illustrates the waveforms at the two WS DAC pins with a single picture. I came to praise Sony... 

28th December 2012, 01:31 PM  #17 
diyAudio Member
Join Date: Jul 2008

sony's first staggering DAC circuit was employed in DASR1.( it was a DAC with transport CDPR1)
According to an article of a japanese audio magazine about them, it is suggested that the CXD1144 have poor attenuation around 4fs, so it will be some help the attenuation of 0th order hold of each staggered 4fs DAC. sony developed CXD1244 next year ,which have sufficient stopband attenuation. so they do not need to employ staggering in next model, haha. Last edited by Shinja; 28th December 2012 at 01:37 PM. 
28th December 2012, 04:39 PM  #18 
diyAudio Member
Join Date: Oct 2006

Looking more closely at the 507ESD schematic.
The WS pins are marked LE/WS and APT/WS. Not very revealing, but certainly seems to indicate that one is not the inverse of the other. The other thing that I'd not noticed before is that, while each dac chip's AOL pins are summed to feed left out, and each chip's AOR pins are directly summed to go to right out circuit, i.e., each 1541 feeding it's r ch to r ch, left ch to left ch, as I'd said before, the CXD1144 data L is fed to only one dac, and the data R is fed only to the other. That is a brain teaser. Also, the dac chips, which are labeled TDA1541R1 in the physical reality, are designated TDA1541R5 on the diagram, which suggests to me that the chips are custom matched. Darned interesting setup, which I would not have paid attention to until it came time to mod my 507, had the OP not brought it up. 
28th December 2012, 07:50 PM  #19 
diyAudio Member
Join Date: Mar 2012
Location: Bradford

LE and APT are Sony, nonI2S options. The chip is set to I2S. One of your WS should be notWS, like this

28th December 2012, 08:41 PM  #20  
diyAudio Member
Join Date: Mar 2012
Location: Bradford

Quote:
When I've finished dinner, I'll get the 'scope out. Dual trace but not dual beam. What's the best way of investigating the relationship between the two WS signals? 

Thread Tools  Search this Thread 


Similar Threads  
Thread  Thread Starter  Forum  Replies  Last Post 
Need advice for a 93 Fender "The Twin"  Topper Harley  Tubes / Valves  23  7th January 2011 07:57 PM 
6.5" and 8" Sony Subwoofer Drivers from Sony All in one HT Systems  form109  Subwoofers  7  1st March 2010 03:14 PM 
Fender "Evil Twin" Hi/Low output = same level  jl_lensman  Tubes / Valves  4  24th April 2009 10:49 PM 
Fender "Twin Reverb Amp" has no output.  Pioneer621  Tubes / Valves  8  21st January 2009 05:20 PM 
New To Site?  Need Help? 