Hi Coris
I picked the 4700uf 16v as its the highest value I have of a good capacitor.
I prefer Black gates in analogue and os cons in digital but the highest os con value I have is 1000uf,(I think that was the highest value they ever made anyway)
Will possibly look at other capacitors for a higher value.
Smiffy
I picked the 4700uf 16v as its the highest value I have of a good capacitor.
I prefer Black gates in analogue and os cons in digital but the highest os con value I have is 1000uf,(I think that was the highest value they ever made anyway)
Will possibly look at other capacitors for a higher value.
Smiffy
XLR output work as differential signal output. This it mean that two symmetrical but opposite phase audio signals, are send out on that output. Pin 2 and 3 are the active pins. One it send a so called negative phase signal, and another one it send out positive phase. Pin 1 is ground.
There are some converters XLR to RCA out there, but in my opinion it is not right to use such. It alter the quality of this output type.
If you have only RCA input, then it is not just right to use XLR and convert it to RCA.
105 model it have also native RCA outputs. Why not using this ones? So it is more reasonable, than converting or using XLR in a not appropriate way. XLR outputs are meant to be used only for an XLR (pre)amplifier input.
Oppo does not use transformers for audio analogue output signals, but only active circuits. All the outputs ar AC coupled in 105 model.
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Thanks Coris.
Was looking at 10,000 uf 40v dc but as the voltage off the diodes is 20v was thinking 22,000 uf 25v.Shoulb be enough of a safety margin.
Also replacing 3300uf digital cap with 10,000 uf Panasonic Low esr
John
Thanks Coris
30mm dia and 40mm height,the same as the Oppo ones
Mundorf M Lytic AG
And have found some 2 amp Schottky diodes
John
Well, I may say there is a little bit mistake about the instruction in this picture. I will use the opportunity now to correct it.
There is not necessary to remove the oscillator to use that clock inserting point. It is enough to remove the R147 and R92, which are actually 0 ohm components.
With R 92 in place the clock signal it is send to the multi-channel board too. R147 it cut the output signal from original oscillator. Removing both 0 ohm devices, the marked inserting point become fully isolated from the rest of the circuits, and it can be used for inserting the new clock signal.
Please keep in mind that doing like described here, the oscillator it still be powered and it output HF signal. To power the oscillator off it is enough to remove the L4 ferrite bead.
It is a clue to keep the original oscillator on board, for undo purposes, if something goes wrong while experimenting. For a definitive tweak in this area, one can safely remove the original oscillator. However, the R92 it have also be removed if the multi-channel stage is not used, and for a full isolation of the indicated inserting point, from the rest of the circuits.
Cutting the trace in between R92 and the inserting point (CN8), one can keep in function the original oscillator, to clock the multi-channel stage, while inserting into CN8 a new clock signal for stereo DAC...
There is not necessary to remove the oscillator to use that clock inserting point. It is enough to remove the R147 and R92, which are actually 0 ohm components.
With R 92 in place the clock signal it is send to the multi-channel board too. R147 it cut the output signal from original oscillator. Removing both 0 ohm devices, the marked inserting point become fully isolated from the rest of the circuits, and it can be used for inserting the new clock signal.
Please keep in mind that doing like described here, the oscillator it still be powered and it output HF signal. To power the oscillator off it is enough to remove the L4 ferrite bead.
It is a clue to keep the original oscillator on board, for undo purposes, if something goes wrong while experimenting. For a definitive tweak in this area, one can safely remove the original oscillator. However, the R92 it have also be removed if the multi-channel stage is not used, and for a full isolation of the indicated inserting point, from the rest of the circuits.
Cutting the trace in between R92 and the inserting point (CN8), one can keep in function the original oscillator, to clock the multi-channel stage, while inserting into CN8 a new clock signal for stereo DAC...
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I`m not sure about a better result... Both network interface and the driver interface it goes into the same processor for decoding. It are used almost the same algorithms, as the informations are (should be) the same (after decoding)...
Transferring the files to the hard disks (BD-R) it may introduce supplementary coding/decoding operations, lot of errors occur, which have to be corrected on the fly, mechanical systems are introduced too, into this alternative processing chain (writing/reading into the drivers)... Quite a lot processing.
I think there are not less "steps" into such process, but even more, than when using the chain as it is... However, you can give it a try, but my opinion is quite pessimistic about the expected improved result...
My conclusion is that the network interface it offer the best quality so far... A improved USB interface it come at the same quality level, or slightly more. I prefer to use streaming on network interface.
I will suggest you another way for improving the quality of the digital playback (as I do it). I copy bit perfect the CDs or whatever hard disk material, I process (64bit) and up sample the digital files, then playback such files through network or USB interfaces. This way of doing it increase the final quality quite well...
Transferring the files to the hard disks (BD-R) it may introduce supplementary coding/decoding operations, lot of errors occur, which have to be corrected on the fly, mechanical systems are introduced too, into this alternative processing chain (writing/reading into the drivers)... Quite a lot processing.
I think there are not less "steps" into such process, but even more, than when using the chain as it is... However, you can give it a try, but my opinion is quite pessimistic about the expected improved result...
My conclusion is that the network interface it offer the best quality so far... A improved USB interface it come at the same quality level, or slightly more. I prefer to use streaming on network interface.
I will suggest you another way for improving the quality of the digital playback (as I do it). I copy bit perfect the CDs or whatever hard disk material, I process (64bit) and up sample the digital files, then playback such files through network or USB interfaces. This way of doing it increase the final quality quite well...
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Well, this subject it may be quite OT, but I will answer you.
There is about software processing and wav/flac files converting, which it happen in 64 bit resolution. The processing under Audition CS6 is always 32 bit. The wav file is saved (upsampled) in 64 bit format, then converted from this bit length to FLAC.
The CD bit perfect file (44,1khz always in 32 bit format), it is first level corrected to never exceed -1db, then DC level corrected to be optimal -> upsampling (32bit) to 176,4khz -> wav saved in 64 bit length -> converted to FLAC from this 64 bit format.
The BD files (192khz) does not need sample converting if it is original recorded at 192khz (or more). If the BD file it originate from 44,1khz recordings, I prefer to put it on 176,4khz, and then to FLAC finally.
I`m use to use network or USB interfaces for playback, and the BD driver for DSD playback only. I always process the CDs as the improvement is substantial comparing to original, and much more convenient to archive it, store it, and playing it back this way from a computer, than through the driver inside the player... I maybe playback first time a CD directly from the driver, to know about its original quality, before the converting process.
I`m also use to record the streamed material, which mainly it originate from 44,1khz/16bit recordings. I record it in 64 bit length, processing accordingly as described and convert it to FLAC finally. The quality is also remarquable after such process, comparing to original. My assumption based on this experienced quality improvement for 44,1khz/16bit material, is that the original informations are kept inside the digital files in spite all the converting and processing from the recording to playback. It is interesting how very fine details it still be there inside the recorded informations and are revealed by an appropriate processing of these existent informations... But this is another story/subject.
There is about software processing and wav/flac files converting, which it happen in 64 bit resolution. The processing under Audition CS6 is always 32 bit. The wav file is saved (upsampled) in 64 bit format, then converted from this bit length to FLAC.
The CD bit perfect file (44,1khz always in 32 bit format), it is first level corrected to never exceed -1db, then DC level corrected to be optimal -> upsampling (32bit) to 176,4khz -> wav saved in 64 bit length -> converted to FLAC from this 64 bit format.
The BD files (192khz) does not need sample converting if it is original recorded at 192khz (or more). If the BD file it originate from 44,1khz recordings, I prefer to put it on 176,4khz, and then to FLAC finally.
I`m use to use network or USB interfaces for playback, and the BD driver for DSD playback only. I always process the CDs as the improvement is substantial comparing to original, and much more convenient to archive it, store it, and playing it back this way from a computer, than through the driver inside the player... I maybe playback first time a CD directly from the driver, to know about its original quality, before the converting process.
I`m also use to record the streamed material, which mainly it originate from 44,1khz/16bit recordings. I record it in 64 bit length, processing accordingly as described and convert it to FLAC finally. The quality is also remarquable after such process, comparing to original. My assumption based on this experienced quality improvement for 44,1khz/16bit material, is that the original informations are kept inside the digital files in spite all the converting and processing from the recording to playback. It is interesting how very fine details it still be there inside the recorded informations and are revealed by an appropriate processing of these existent informations... But this is another story/subject.
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As for CD vs. ethernet playback, they do not sound the same but there is no clear winner. The CD playback is more dynamic with a little more clarity in the upper mids but not fatiguing. Ethernet is more laid back with slightly softer top end and great mids, but some instruments in the upper treble get lost. I will experiment more. CD playback does sound great though (reminder I am using coax out to soekris dac).
Bits on the way
6800 uf audio power caps being replaced with 22,000 25v Mundorfs.Will measure output dc and if low enough link with wire
links or replace with Black Gate 100uf 6.3v bi polar
Also Power diodes to 21SQ100 then let the stuff burn in.
Post rectification digital side replace 3300uf with either a 4700 uf Black gate standard or a 10,000uf Panasonic FC
On digital side first 3 caps replace with 1000uf os cons. Would like to use 6.3v but feel that's a bit to close
to pre reg 6.1v.On the output side two 1.2v replaced with 2700uf os con and middle 3.3v replace with 1500uf
Will also take a peek at smps.
Smiffy
6800 uf audio power caps being replaced with 22,000 25v Mundorfs.Will measure output dc and if low enough link with wire
links or replace with Black Gate 100uf 6.3v bi polar
Also Power diodes to 21SQ100 then let the stuff burn in.
Post rectification digital side replace 3300uf with either a 4700 uf Black gate standard or a 10,000uf Panasonic FC
On digital side first 3 caps replace with 1000uf os cons. Would like to use 6.3v but feel that's a bit to close
to pre reg 6.1v.On the output side two 1.2v replaced with 2700uf os con and middle 3.3v replace with 1500uf
Will also take a peek at smps.
Smiffy