Re: Schematics
rfbrw, my respect.
I thought you'd show something along the lines of http://users.verat.net/~pedjarogic/audio/tda1541a_dac/disdia_i-v.htm which is constant current loaded cap integrator whose source current is modulated by DAC output.
Attached is expected signal shape from your schematic.
rfbrw, my respect.
Elso, I'm confused here. Your schematic is pure good-old opamp integrator, subject to same RC signal shape as in my picture.Elso Kwak said:
I thought you'd show something along the lines of http://users.verat.net/~pedjarogic/audio/tda1541a_dac/disdia_i-v.htm which is constant current loaded cap integrator whose source current is modulated by DAC output.
Attached is expected signal shape from your schematic.
Attachments
Circlotron, there's not much use to try out on existing DAC. DAC would need to be modified to operate properly in delta mode. DAC itself becomes part of current-source integrator, thus IV Capacitor is needed instead of IV-stage. Normal IV and RC voltage integrator would misbehave there.
Elso's IV stage would work though if R3 was removed and C2 increased to limit the output swing.
Elso's IV stage would work though if R3 was removed and C2 increased to limit the output swing.
Re: Green Curve
Hi Wimms. I am sorry I must have been totally colour-deaf at the moment of my posting. Of course I meant my DAC is similar to the purple curve in your first picture on myscope.(post# 34) The funny thing is if you lower the frequency of the sine wave the steps get lost. Well your output curve shows that a lot of smoothing has taken place. On my scope it is slightly better but that may be due to additional filtering by the capacity of the cable and non-ideal opamp. What is the frequency of your sine wave in the pictures? (too lazy to calculate from the µs)
As a side note I am one of the few males who is not colour blind as the Moron for the military selection found out. Whe did the whole book of colour dot pages faster and faster.... I guess he thought I learned the book by head.
Again wimms I apologize for the confusion.
I have been thinking a long time about something you posted making the steps smaller in length by two staggered DACs or something like digital antidote by Taddeo. I gave up as even two balanced DACs did not sound better but worse especially in the highs.
Some top of the line Sony converter had two staggered DACs (TDA1541AS1) in the past.
I do not see a fundamental difference with Pedja's IV that I will build some time as the current to voltage converion takes place across the 1k resistors??
Elso Kwak said:
Hi wimms,
Interesting picture as the signal in my DAC after the IV looks exactly like the green curve. To get a smooth signal additional filtering is required.
Hi Wimms. I am sorry I must have been totally colour-deaf at the moment of my posting. Of course I meant my DAC is similar to the purple curve in your first picture on myscope.(post# 34) The funny thing is if you lower the frequency of the sine wave the steps get lost. Well your output curve shows that a lot of smoothing has taken place. On my scope it is slightly better but that may be due to additional filtering by the capacity of the cable and non-ideal opamp. What is the frequency of your sine wave in the pictures? (too lazy to calculate from the µs)
As a side note I am one of the few males who is not colour blind as the Moron for the military selection found out. Whe did the whole book of colour dot pages faster and faster.... I guess he thought I learned the book by head.
Again wimms I apologize for the confusion.
I have been thinking a long time about something you posted making the steps smaller in length by two staggered DACs or something like digital antidote by Taddeo. I gave up as even two balanced DACs did not sound better but worse especially in the highs.
Some top of the line Sony converter had two staggered DACs (TDA1541AS1) in the past.
I do not see a fundamental difference with Pedja's IV that I will build some time as the current to voltage converion takes place across the 1k resistors??
Re: Re: Green Curve
You know, your IV stage can be easily modded to allow testing of the idea discussed here. I think it would be possible to write a simple DSP plugin to foobar2000 that would generate required delta PCM on the fly, thus you could actually audition with real music and measure the results.
You'd need to remove R3 and set C2 to dac-current*TS to get 1V output swing. (TS is sampling time, 1/44.1K). If the DAC is outputting delta-current, then output of your integrator will have same signal shape as in my simulated interpolation.
Hi Elso. I see now. The original gif with purple curve I posted was created by simulating 1KHz sine, and then zoomed into the upper peak of the signal. The RC constant of the purple curve was 2.2usec iirc. The last gif simulating your IV was made with 3KHz signal iirc.Elso Kwak said:
You know, your IV stage can be easily modded to allow testing of the idea discussed here. I think it would be possible to write a simple DSP plugin to foobar2000 that would generate required delta PCM on the fly, thus you could actually audition with real music and measure the results.
You'd need to remove R3 and set C2 to dac-current*TS to get 1V output swing. (TS is sampling time, 1/44.1K). If the DAC is outputting delta-current, then output of your integrator will have same signal shape as in my simulated interpolation.
Wimms,
In your graph you are comparing apples with pigtails. The exponential RC time constant curve is for an RC integrator, the straight line approximation is for a constant current integrator like Elso's. What your idea does is changing the steps of the staircase, but the results will be the same as the unmodified stair case: straight line with a constant current integrator, exponential with an RC integrator. I am pretty sure you are fully aware of this, and I can only guess why you chose to use two different integration methods in this 'comparison'!
Your idea may change the step size, but so would oversampling. It does nothing on the basic operation of the way of integrating. If you stay with the same sampling rate, and you lower the step size, you will get a lower recovered analog level, with a lower 'noise' level, but still with the same signal-to-noise-ratio. This is a no-win situation.
It's your decision to follow your idea, but don't expect a large reward for your efforts.
Jan Didden
In your graph you are comparing apples with pigtails. The exponential RC time constant curve is for an RC integrator, the straight line approximation is for a constant current integrator like Elso's. What your idea does is changing the steps of the staircase, but the results will be the same as the unmodified stair case: straight line with a constant current integrator, exponential with an RC integrator. I am pretty sure you are fully aware of this, and I can only guess why you chose to use two different integration methods in this 'comparison'!
Your idea may change the step size, but so would oversampling. It does nothing on the basic operation of the way of integrating. If you stay with the same sampling rate, and you lower the step size, you will get a lower recovered analog level, with a lower 'noise' level, but still with the same signal-to-noise-ratio. This is a no-win situation.
It's your decision to follow your idea, but don't expect a large reward for your efforts.
Jan Didden
Let me ask the obvious stupid question.......
If this was a way of making a good DAC- I/V setup.....and cheaply, just like MASH/Bitstream/one-bit/etc...........wouldn't those clever Japanese companies that develop all this mass market crap tried it already????
Seems that maybe they would have. I tend to agree with rfbrw. I don't seem to get it either.
Jocko........getting a sunburn on the roof of his mouth.
If this was a way of making a good DAC- I/V setup.....and cheaply, just like MASH/Bitstream/one-bit/etc...........wouldn't those clever Japanese companies that develop all this mass market crap tried it already????
Seems that maybe they would have. I tend to agree with rfbrw. I don't seem to get it either.
Jocko........getting a sunburn on the roof of his mouth.
Confused?
Hi Jan,
Now I am the one to be confused.
In my DAC I see exponential discharge curves of a capacitor and wimms is proposing a straight line between steps. Right?
As I wrote earlier on this forum the whole problem originates from a far too low chosen sampling frequency for Redbook CD.
At first the industry came with a brickwall filter, then they invented 4 or 8 times oversampling.
And now I am among those guys favouring NON-Oversampling.......
janneman said:
Hi Jan,
Now I am the one to be confused.
In my DAC I see exponential discharge curves of a capacitor and wimms is proposing a straight line between steps. Right?
As I wrote earlier on this forum the whole problem originates from a far too low chosen sampling frequency for Redbook CD.
At first the industry came with a brickwall filter, then they invented 4 or 8 times oversampling.
And now I am among those guys favouring NON-Oversampling.......
Re: Re: Schematics
Wimms, you cannot come to an open forum, try and sell me a lemon when I know it is a lime and get upset when I contradict you.
There are two issues here. One is the value of your proposal, something best left to the likes of circlotron et al.The other is what your proposal actually IS and that is the issue that concerns me. All the maths and insults in the world will not alter the fact that it is NOT linear interpolation, analogue or otherwise. Linear interpolation as a mathematical concept is as simple as it gets and its hardware implementation is just as simple as shown by Beard, Cambridge, Wadia and numerous diy implementations.
wimms said:
Wimms, you cannot come to an open forum, try and sell me a lemon when I know it is a lime and get upset when I contradict you.
There are two issues here. One is the value of your proposal, something best left to the likes of circlotron et al.The other is what your proposal actually IS and that is the issue that concerns me. All the maths and insults in the world will not alter the fact that it is NOT linear interpolation, analogue or otherwise. Linear interpolation as a mathematical concept is as simple as it gets and its hardware implementation is just as simple as shown by Beard, Cambridge, Wadia and numerous diy implementations.
janneman,janneman said:
You are attacking me for no reason.
May I suggest you to make few simulations at least? Elso's integrator is no more but RC integrator and IV resistor, with all its RC exponential issues. You can not apply pure current integrator directly onto staircase current - you would get garbage out. Simulate! And not just with single sinewave, but complex signal.
I was making no 'competition' with graphs, but only used them to convey my point, noting exagerations. Why do you accuse me? Elso's integrator can not output straight line approximation, as cannot anything else that is not based on memory of previous sample. Obviously, oversampling is. Many are preferring as little OS as possible. Not because of digital processing btw. As low order filtering as possible.
And I don't understand what you say about step size or basic integration operations. This has no relation to what I'm saying.
I'm suggesting a way to use pure current integrator in place of RC integrator with IV resistor. If you think that you already use pure current integrator, then you are confused.
wimms said:
Wimms, I am not attacking you or accusing you. I addressed the merits, if any, of your idea. I think if you read my post less emotionally, you will agree. I apologise if I gave the impression to accuse you of something less then honesty. But it is well known that people - unconsciensly - often present data in a favourable way.
You could be right that I misread Elso's I/V converter. I'll look at it again.
But that doesn't change the basic fact, and please let me know if I am right, that your idea changes the step size, but NOT the 'shape' of the step. So, whatever the integration method, it can only differ in level or amplitude, NOT in basic shape. I think that part of rfbrw's frustration came from the fact that this obvious fact is being glossed over.
So, I think you have received advise from different people looking at your idea from different angles but with the same conclusion. Does that count for something?
Jan Didden
wimms said:
There is so much information out there defining linear interpolation that it beggars belief that you cannot find it. One can only conclude you are not looking for it.
So once more.
For a series of samples A, B, C, D you have an output of A, (A+B)/2, B, (B+C)/2, C, (C+D)/2, D. That is linear interpolation. It assumes the points are joined by a straight line.
RFBRW's post explaining linear interpolation is correct... at least as far as he goes with it, a series A,B,C,D does indeed lead to a series A, (A+B)/2, B, (B+C)/2... but what is the point? it also leads to a series A, (A+((A+B)/2)/2, (A+B)/2... and so on... looks like 'upsampling' to me... but WIMMS is suggesting a method of 'integrating' the difference.... essentially 'upsampling' to a 'infinite' level of samples... Seems like a pretty neat idea to me.
Steve
Steve
steve said:
It is indeed a crude form of oversampling, and is found in bitstream dacs operating in conjunction with a FIR based oversampler. You'll find a 4x variant in Wadia dacs. It is, in theory at any rate, infinitely expandable though in practice it is limited by the serial clock rate.
BTW steve, the next step practical step in terms of ease of implentation would be 4x,though there is, I think, a dac with 3x linear interpolation. 4x would yield [4A],[3A+B],[2A+2B],[A+3B],[4B].I have a schematic from the Japanese magazine MJ that follows the SAA7220 with 64x linear interpolation.
I have read this topic with some interest. Can someone answer me un-emotionally:
1) How are you going to accurately, in the analog domain, represent both n, and n-1 samples to 16+ bits precision? There is a difference between paralleling DACS and running them completely independantly (you don't get the averaging effect).
2) I know have this current source integrator. My current source would need to output a current that is exactly correct for the step size that will change for every sample. How will this be implemented? Will this not be prone to errors as well?
3) The comments was made that this would be like "infinite oversampling". Mathematically, that does not seem true to me at all. Going back into the depths of the brain, this just seems like a trapezoidal approximation which is just a second order approximation of the analog domain. That is far from infinite. In fact, I don't think mathematically it will be a whole lot more accurate than a 2* oversample with a first order anti-alias filter. As opposed to square waves, now you have triangular waves. Same harmonics, they just roll off faster at higher frequencies.
1) How are you going to accurately, in the analog domain, represent both n, and n-1 samples to 16+ bits precision? There is a difference between paralleling DACS and running them completely independantly (you don't get the averaging effect).
2) I know have this current source integrator. My current source would need to output a current that is exactly correct for the step size that will change for every sample. How will this be implemented? Will this not be prone to errors as well?
3) The comments was made that this would be like "infinite oversampling". Mathematically, that does not seem true to me at all. Going back into the depths of the brain, this just seems like a trapezoidal approximation which is just a second order approximation of the analog domain. That is far from infinite. In fact, I don't think mathematically it will be a whole lot more accurate than a 2* oversample with a first order anti-alias filter. As opposed to square waves, now you have triangular waves. Same harmonics, they just roll off faster at higher frequencies.
Just a quick 'clarification' I used the term 'infinite upsampling' as an analogy. Perhaps it was a poor one, but the previous poster explained the idea of 'linear interpolation' by describing a series ... and showing the calculation for the 'mid-point' of that series... I was trying to point out that the technique in question provides an 'analog function' that does not occur in the 'digital domain', hense the only 'digital analogy' would be to 'upsample'.. meaning that you would need to make more than just that 'midpoint' calculation to approximate that 'analog function' but would infact have to break the sample into ever smaller 'time units' .... same 'analogy' used in Calc 101 to explain the integral...
I don't think anyone here has any illusions about somehow 'creating' new data out of thin air... just looking to find the most logical interpretation of the data that is present.
Steve
I don't think anyone here has any illusions about somehow 'creating' new data out of thin air... just looking to find the most logical interpretation of the data that is present.
Steve
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