'Chibi' phase inverting mod. - diyAudio
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Old 19th May 2004, 06:19 PM   #1
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Default 'Chibi' phase inverting mod.

Hi,

Can anyone point me to where I can find the schematics for Chibi phase inverter for non-oversampling DAC? This mod is used by Scott and I saw it somewhere on the net too, but cannot find it now.
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Old 27th June 2004, 04:38 PM   #2
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Did you find it?
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Old 28th June 2004, 03:34 PM   #3
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No, I couldn't find it. I'm working on an opamp solution for i/v conversion, the Chibi mod is not an issue anymore.

If you know where to get, let me know, I'm still curious how it works.
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Old 28th June 2004, 05:50 PM   #4
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Judging from the TubeDac website, it is inversion done in the digital domain. There are a number of designs available on the web that include this option. It is just the audio datastream inverted prior to entering the dac and before nitpickers chime in I know one is supposed to be added after the inversion but very few bother.
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Old 28th June 2004, 09:31 PM   #5
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Quote:
Originally posted by rfbrw
Judging from the TubeDac website, it is inversion done in the digital domain. There are a number of designs available on the web that include this option. It is just the audio datastream inverted prior to entering the dac and before nitpickers chime in I know one is supposed to be added after the inversion but very few bother.
I would question the judgment and hearing of anyone who claimed the resulting 1-bit error was insignificant or inaudible.

Doing a simple bit-wise inversion of the data stream results in waveform that is shifted by one bit. All originally positive samples become negative but one bit larger than they should be. Likewise, all originally negative samples become positive but 1 bit smaller than they should be. That one bit offset equals 6 dB of distortion, across the board.
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Old 28th June 2004, 10:28 PM   #6
guido is offline guido  Netherlands
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You are always one bit of: see it as DC offset.
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Old 28th June 2004, 11:38 PM   #7
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Question away, but the general consensus would seem to be, albeit arrived at through observation, that it does not matter.
I have over time seen many schematics, some high-end and some for professional eqpt priced well into six figures and I have never seen phase inversion excuted to textbook specs. Here are two dacs, no longer in production, that invert phase 'incorrectly'.

http://www.audioasylum.com/images/DDE3.pdf

http://www.passlabs.com/pdf/d1-srv-man.pdf

Considering the serial nature of modern dacs it isn't really surprising that most don't bother with 1 bit addition.
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Old 29th June 2004, 01:23 AM   #8
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Originally posted by guido
You are always one bit of: see it as DC offset.
It’s not that simple: there is no such thing as a DC offset in the digital domain. In the analog domain, adding a DC offset is more of less innocuous because every point in the waveform, including zero, is shifted by the offset amount. In the digital domain, adding a bit to each sample will expand one half of the waveform and compress the other. The fact that the offset waveform is symmetrical about one instead of zero means nothing to the DAC: It still considers digital zero to be silence and the natural point of symmetry. To make ones-complement data stream inversion work correctly, you would have to bias the DAC so that an input of –1 resulted in 0 output.

Consider the simplified case of a 4-bit DAC. It has 16 permissible values ranging from –8 to +7, including zero. The left-hand column shows those values as a signed integer. The second column shows the binary representation of those values. The third column shows the ones-complement of these values. The fourth column shows the signed integer representation of the one-complemented values.

-8 1000 0111 +7
-7 1001 0110 +6
-6 1010 0101 +5
-5 1011 0100 +4
-4 1100 0011 +3
-3 1101 0010 +2
-2 1110 0001 +1
-1 1111 0000 +0
+0 0000 1111 -1
+1 0001 1110 -2
+2 0010 1101 -3
+3 0011 1100 -4
+4 0100 1011 -5
+5 0101 1010 -6
+6 0110 1001 -7
+7 0111 1000 -8

Suppose two successive samples have the values +2 and +4. That means the amplitude of the signal sampled doubled between one sample and the next. If we inverted the data stream, using the table above, the successive samples would have the values –3 and –5. The difference is less then double. The reconstructed waveform would be compressed and have a slower rise time than it should.

On the other hand, suppose two successive samples have the values –2 and –4. Again, doubling in amplitude each sample period. The inverted samples would have the values +1 and +3. The difference is more than double. The reconstructed waveform would be expanded and have a faster rise time than it should.

No matter how you slice it, the error amounts to 6dB.

It should come as no surprise to anyone here that the price of a DAC or its presumed "pro" status has no bearing on its quality or whether the designers got everything "right."
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Old 29th June 2004, 04:05 AM   #9
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Originally posted by jbokelman

It should come as no surprise to anyone here that the price of a DAC or its presumed "pro" status has no bearing on its quality or whether the designers got everything "right."
I am sure that the designers in question are well aware of the error of their ways but they just don't think it matters. If HDCD can sacrifice the LSB of the occasional sample and one particular design can simply truncate the output of the ASRC and still be well regarded, then they can probably be forgiven for thinking that way.
Consider the Tubedac, the dac with which the Chibi mod is associated. At its most basic the digital section consists of the CS8412 and the TDA1543. Add an inverter (or an XOR if you want to add a phase select switch) and you are done. Though the 1 bit addition is simple, the logic overhead is considerable especially if you think the possible gains are questionable. You might as well invert in the analogue domain.
However, I do think if one has gone to the bother of creating a balanced dac, e.g. as shown in the AD1852 datasheet, the 1 bit addition after inversion should be mandatory.
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Old 29th June 2004, 05:50 AM   #10
Werner is offline Werner  Europe
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Quote:
Originally posted by jbokelman

In the digital domain, adding a bit to each sample will expand one half of the waveform and compress the other.
No. You move the 'zero' point just as well one bit, and nothing gets distorted.

'Zero' is just a convention, in analogue as well as in digital.
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