luxury54 makes a very good point. At this level it's virtually impossible to make a real comparison without an A/B switch that you can throw while listening. Wander off to the soldering iron for a quarter of an hour and you've forgotten what you heard, but the placebo effect will tell you that you've improved on it later. ;-)
With a digital source, I would say splitting the SPDIF signal into two DACs, trimming their outputs to be the same level (very important - at a primitive level the brain thinks "louder=better") and switching between them in real time is the best way.
I should have bought two DACs and left one a step behind the other in terms of modifications.
Kevin
With a digital source, I would say splitting the SPDIF signal into two DACs, trimming their outputs to be the same level (very important - at a primitive level the brain thinks "louder=better") and switching between them in real time is the best way.
I should have bought two DACs and left one a step behind the other in terms of modifications.
Kevin
This thread is a gold mine for us that have "discovered" thís little diy gem...
To you Kevin, as you mention recalculating the filter to achieve a lower impedance, and to luxury54 as you found the roll-off to be 13kHz, I found this link at Cirrus: http://www.cirrus.com/en/pubs/appNote/AN048Rev2.pdf
I haven't simulated the existing filter yet to compare it with the values advocated by Cirrus, nor have I tested "the real thing" with RightMark yet, but by just comparing component values it seems that the existing filter is a bit off, and may very well have a roll-off point way below the 50kHz Cirrus recommends.
I will try this out as soon as I have more Solder Wick and get the components needed.
I also found this att Cirrus web site: http://www.cirrus.com/en/pubs/errata/ER578D1.pdf
giving new recomended values for the PLL filter. I haven't got a clue how this would influence the sound or general performance, but someone here can perhaps explain?
To you Kevin, as you mention recalculating the filter to achieve a lower impedance, and to luxury54 as you found the roll-off to be 13kHz, I found this link at Cirrus: http://www.cirrus.com/en/pubs/appNote/AN048Rev2.pdf
I haven't simulated the existing filter yet to compare it with the values advocated by Cirrus, nor have I tested "the real thing" with RightMark yet, but by just comparing component values it seems that the existing filter is a bit off, and may very well have a roll-off point way below the 50kHz Cirrus recommends.
I will try this out as soon as I have more Solder Wick and get the components needed.
I also found this att Cirrus web site: http://www.cirrus.com/en/pubs/errata/ER578D1.pdf
giving new recomended values for the PLL filter. I haven't got a clue how this would influence the sound or general performance, but someone here can perhaps explain?
The Pin 20 mod is to put the CS8416 clock recovery PLL into a different mode. Basically, it has two modes of the PLL. One aims to reduce jitter in the wide band and the other in-band. Depending on the type of DAC connected to the device, the sensitivity to jitter is different and it's optimal for delta-sigma converters to have this signal pulled high to select low in-band jitter.
Hey Kevin,
So the pin 20 connect of the CS8416 should be done when this chip is used with the CS4397 DAC chip; to optimize jitter correction.
Please confirm.
By the way, really appreciate the technical gurus like you and luxury54 taking the time to post detailed information on the mods; pictures are a big plus!
I'm basically a parts swapping DIY guy, so the I would not be able to figure this stuff out without your techincal instructions.
Cheers
Hey Kevin,
So the pin 20 connect of the CS8416 should be done when this chip is used with the CS4397 DAC chip; to optimize jitter correction.
Please confirm.
By the way, really appreciate the technical gurus like you and luxury54 taking the time to post detailed information on the mods; pictures are a big plus!
I'm basically a parts swapping DIY guy, so the I would not be able to figure this stuff out without your techincal instructions.
Cheers
Yes, that's a very handy reference for designing this type of filter.
This site Here is also very useful, I find. If you select the filter tool then multiple feedback LPF you can plug in the values for the filters in the DAC and see what happens. There is only a single input leg, so you have to ignore the network attached to the non-inverting input of the amp, but the results are valid.
Yes, that's correct. - for a Delta-Sigma converter. If you were feeding a R-2R DAC you would want the other mode. It should be pulled up to digital VCC with a 47K resistor. The resistor is important because it is reconfigured as an output after the chip has reset!
Kevin
Luxury54: I've constantly been comparing this DAC to my DPA PDM One - seies 2 and also my Audiosector (Peter Daniel) NOS DAC. And I am definately not imagining the bass I am getting. However when I first started using the LM4562 (in a Shanling CD player) they took about a week to really start sounding their best. Before that they were very sterile and unemotional.
The bass I am getting at the moment is very 'tactile' - you can 'feel' the size of the instrument... whether it is a double bass or a kettle drum - each feels realistic and with plenty of 'body. If there was little bass believe me I would throw it away as bass is very important to me
I am currently at home every day as I am suffering from M.E. so I listen to music constantly and am very critical - to me the LM4562 most definately does not appear to lack bass.
- John
p.s. I am currently using Foobar2000 with the ASIO plug-in. This is running into a Trends Audio UD-10.1 USB to SPDIF converter. I am mostly using 16 bit files.
I will try 24/96 files using an Edirol UA-1EX --> optical --> 'BIG' DAC too and see if I still have this bass.
The bass I am getting at the moment is very 'tactile' - you can 'feel' the size of the instrument... whether it is a double bass or a kettle drum - each feels realistic and with plenty of 'body. If there was little bass believe me I would throw it away as bass is very important to me
I am currently at home every day as I am suffering from M.E. so I listen to music constantly and am very critical - to me the LM4562 most definately does not appear to lack bass.
- John
p.s. I am currently using Foobar2000 with the ASIO plug-in. This is running into a Trends Audio UD-10.1 USB to SPDIF converter. I am mostly using 16 bit files.
I will try 24/96 files using an Edirol UA-1EX --> optical --> 'BIG' DAC too and see if I still have this bass.
I have simulated the LP filter with component values as described above. Althought I can't verify a roll-off at 13kHz, but there's a strange 3dB dip around 10 kHz, while the -3dV roll-off is at 43 kHz. The dip is worse when changing R8/R9 from 36k to 22k.
The values described by the Cirrus AN gives a very nice straight line from 5Hz to 48kHz and a steep roll-off at 50 kHz. This is the way I will go, rather than removing the filter all together (by removing all the 1nF cap's as I have it now).
Just a reminder to those who decide to keep the second OP in place - Don't remove C1/C4. If you do, the bass is gone.
The values described by the Cirrus AN gives a very nice straight line from 5Hz to 48kHz and a steep roll-off at 50 kHz. This is the way I will go, rather than removing the filter all together (by removing all the 1nF cap's as I have it now).
Just a reminder to those who decide to keep the second OP in place - Don't remove C1/C4. If you do, the bass is gone.
SoNic_real_one said:
yes they are made from vinyl,i just wanted to show the AD's crisp highs and very low bass reproduction due to it's high slew rate ,compared to an NE5532 that i have changed in a Cambridge Audio Azur phono preamp
i wouldn't advise changing the 22k resistors to other values because higher values would give an artificial gain of the circuit that would no longer be in the DAC'S 2Vrms output specifications
also it's the best way of adapting the impedance of dac's output (22ohms) with the following circuit;
i have also compared the sound of the tube output stage that i have made with the opamp sound since they can work in paralel and i'm able to A/B test it,and the best highs are obtained with no filtering at all that's why i thrown away all the caps leaving just that 56pF cap in the opamp's feedback not to oscillate (in reality 100pF) and also having NO capacitor at all in the sound path - i have 0.02v DC offset on RCAs (this is because i wasn't able to get 100% perfect match between the four 22k resistors,having on one side of the circuit a 20k resistor in series with a 5k trimmer i could have obtained 0.00V DC offset, but i don't like trimmers since they are not perfectly stable over shocks
luxury54,
Awsome posts on your trails and results. Could you please post some pictures of your work when you have the time.
Did the +5v connect to the pin 20 mod on the CS4397 chip yesterday.
I tapped off pin 28 of the CS8416 chip pin, which measured +5.02vdc and ran a wire to the 47k resistor connected to pin 20 of the CS4397.
Do not hear anything negative from this test hook up.
Would you guys recommend a better +5vdc tap I should use to supply the the vcc for pin 20.
Cheers.
Awsome posts on your trails and results. Could you please post some pictures of your work when you have the time.
Did the +5v connect to the pin 20 mod on the CS4397 chip yesterday.
I tapped off pin 28 of the CS8416 chip pin, which measured +5.02vdc and ran a wire to the 47k resistor connected to pin 20 of the CS4397.
Do not hear anything negative from this test hook up.
Would you guys recommend a better +5vdc tap I should use to supply the the vcc for pin 20.
Cheers.
luxury54,
Thanks for the clarificaion; my bad on the last post, which should have said:
Did the +5v connect to the pin 20 mod on the C8416 chip; tapped off the lead going to pin 28 of the CS4397, which measured +5.02vdc; ran a wire from there to a 47k resistor connected to pin 20 of the CS8416.
With that being corrected, is the +5vdc supply line for this mod critical. Would it be better to take it directly from the +5vdc power supply regulator rather than taking it from a tap on the board, like I did above.
Cheers.
Thanks for the clarificaion; my bad on the last post, which should have said:
Did the +5v connect to the pin 20 mod on the C8416 chip; tapped off the lead going to pin 28 of the CS4397, which measured +5.02vdc; ran a wire from there to a 47k resistor connected to pin 20 of the CS8416.
With that being corrected, is the +5vdc supply line for this mod critical. Would it be better to take it directly from the +5vdc power supply regulator rather than taking it from a tap on the board, like I did above.
Cheers.
I've just had another go at simulating it and I agree. Something goes wrong with the shape of the filter with the component values we have found to be fitted. Not sure I can explain why but it doesn't work at all.
Simulated this too and I agree again! Perfect filter shape. 0.1db down at 20KHz, 3db down around 50KHz. I will be replacing everything with these values next time I get the soldering iron out.
To clarify, that is:
R8, R14, R15, R16 = 13k7
R10, R11 = 3k32
C36,C37 = 1nF
C32, C34 = 220pF
With all the caps except C32 removed it's 0.5db down at 20KHz with a 3db point at 63KHz. The common mode rejection starts to suffer at high frequencies (probably too high to matter) because the filtering that remains is unbalanced.
Kevin
hafp said:
yes,i think is a good solution to take that 5v directly from one of the regulators...i will try that myself since on my board cs8416 is fed also with 3,3v
i think would be a better solution t change that small 3,3v regulator to a 5v regulator because cs8416's datasheet says that it can accept voltages up to 5v
I'm still using the 5532 currently but I plan to upgrade that. Had a rummage this morning and I haven't got any 220pF caps to try the new filter.
Will have to put an order in for some bits, including some better Op Amps.
Beware that on my board the CS8416 is fed from two +3.3v supplies so it might not take kindly to being pulled up to 5v. It would be worth checking what it is powered by first, although I doubt it would do any harm through a 47k resistor. In addition, one of the 5v supplies (i believe there are two) drives the DAC so be careful that you don't connect the 8416 to that. I would imagine the 8416 is quite a noisy device and not something you want polluting the DACs power rail.
I'm not sure if there's anything to be gained by running the 8416 at 5v. It could well be that it's less noisy on the lower voltage. Only one way to findout I guess. Try it and see if it sounds any different.
Kevin
Great information from luxury54 and Kevin.
Yeah, the voltage to the CS4316 has me a little bit confused too; but the +5dvc to pin 20 via 47k resistor works pretty well.
One DAC schematic I have of the CS8416/CS4397 has pin 21 of the 8416 being supplied with a +5vdc; another schematic has that same pin supplied with 3.3v, along with pin 23.
Thus my asking the question of where would be the best place to tap the 5v power.
What Kevin notes about keeping the DAC power separate to prevent noise from getting back to the CS4397.
Will change the power tap and let you know how it goes.
Cheers.
Yeah, the voltage to the CS4316 has me a little bit confused too; but the +5dvc to pin 20 via 47k resistor works pretty well.
One DAC schematic I have of the CS8416/CS4397 has pin 21 of the 8416 being supplied with a +5vdc; another schematic has that same pin supplied with 3.3v, along with pin 23.
Thus my asking the question of where would be the best place to tap the 5v power.
What Kevin notes about keeping the DAC power separate to prevent noise from getting back to the CS4397.
Will change the power tap and let you know how it goes.
Cheers.