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#21 | ||
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diyAudio Member
Join Date: Oct 2009
Location: Brunei
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Quote:
Quote:
__________________
-------- TDA1541a-S2 / 6072a / MediaplayerDac -- Michell Gyrodec -- Aikido Preamp -- Pass F5 -- Audio Physic Tempo 2SE -- Yamamura Cabling |
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#22 | |
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diyAudio Member
Join Date: Jun 2011
Location: SCOTLAND
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Quote:
As one dac is used per channel, each pair of dac outputs should have (theoretically) the same amplitude and polarity of noise which will therefore cancel in the transformer. |
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#23 | |
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diyAudio Member
Join Date: May 2010
Location: Santo, TX
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Consider the additional noise added by unnecessary gates switching at the same time. When the source of the sample clock changes state, a spike of noise and ground bounce is injected into the circuit. Moments later, before the noise has had a chance to dissipate, the first inverter receives the changing clock edge and inverts it. The Schmidt trigger references Vcc and ground to determine the trigger points but the reference is polluted with noise. Then, as it is changing state, the first inverter injects additional noise and ground bounce into the circuit, which, in turn, degrades the operation of the second inverter. Your buffer has degraded the clock signal and increased jitter. A single pico-gate is sufficient to invert one data line with minimal impact on the rest of the circuit. Also, inverting the data to one dac is insufficient to create a proper differential output. You must negate each sample value otherwise you will add a DC offset. Transformers, like the 9545, work best when there is no DC offset. |
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#24 |
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diyAudio Member
Join Date: Mar 2002
Location: diepe zuiden
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I'm getting flashbacks from this thread. DC yes, but not a lot (lsb). Allthough it goes wrong at one of the end values iirc. All discussed (a lot) before..
As the currents return to +5, it makes sense to use a diff setup. In theory the current going through +5 is then 0 (assuming a perfect dac), the changing audio currents that is. The offset current of the 1541 doubles, but that's only DC. If you want to do the inverting properly, the best way is a dig. filter and use the 1541 in another mode (not i2s). In that way you can correctly invert the feed to one channel. or a lot of logic in a cpld or fgpa.
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GuidoB |
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#25 |
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diyAudio Member
Join Date: Oct 2009
Location: Brunei
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#26 |
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diyAudio Member
Join Date: Jun 2007
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In case of the tda1541 one LSB would result in an offset current of 61nA I would think most any signal transformer could tolerate that.
2's complement inversion falls apart for the most negative number so the logic would also have to handle that case. Besides if the data is encoded with "real" or absolute zero at 0.5 lsb then its actually correct to only invert the data. In any case I'm not losing any sleep over this. |
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#27 | |
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diyAudio Member
Join Date: Jun 2011
Location: SCOTLAND
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Quote:
From post 3409 I tested this dual-mono mode recently, using a simple I2S encoder that converts L/R to L/L for chip #1 and R/R for chip #2. Both chips receive same BCK and WS signals, only the DATA signal for each chip differs. I didn't use balanced mode so possible "issues" with inverting DATA wouldn't occur. From post 3409.. I tested this dual-mono mode recently, using a simple I2S encoder that converts L/R to L/L for chip #1 and R/R for chip #2. Both chips receive same BCK and WS signals, only the DATA signal for each chip differs. I didn't use balanced mode so possible "issues" with inverting DATA wouldn't occur. Interesting experiments ahead to see just what happens (audibly) with inverting the data. Last edited by batteryman; 8th July 2011 at 08:08 PM. |
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#28 |
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diyAudio Member
Join Date: Jun 2011
Location: SCOTLAND
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I couldn't wait to finish the dual differential design so decided to experiemtn with the I/V transformers.
I disconnected the existing opamp & filter components from an old Philips Cd471 and soldered an LC Tee* filter loaded by 150R to the 1541 outputs. The Sowter transformer primaries were paralled and connected across the 150r load and the secondaries series connected and fed directly to the amps aux input. *2x 1mh in series and 100n to ground from the midpoint (online calculators with F-3 at 16khz, Z = 100r produced 2 x 1mh and 200n which I did try) Without the filter there were audible artifacts as expected, but vastly reduced by the filter without major loss of hf (to my old ears!) and certainly listenable to and not a fet, tube, opamp or transistor in sight! |
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#29 | |
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diyAudio Member
Join Date: Jun 2007
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Quote:
Here is the same measurement again only this time its measured differentially. |
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#30 |
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diyAudio Member
Join Date: Oct 2004
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