volume control 2k lin.pot?
Volume Control ADC
--- firmware supports volume control via the ADC header.
For such firmware simply wire a 10K pot to the ADC header---
---------------------------------------------------------------------
Could I use a 2k linear pot instead?
If it requires a 10k never lower,
I could boost it with an 8k resistor.
The reason I'd like to use the 2k pot
is I have a ten turn pot and thought
I might have finer control at the top end.
Volume Control ADC
--- firmware supports volume control via the ADC header.
For such firmware simply wire a 10K pot to the ADC header---
---------------------------------------------------------------------
Could I use a 2k linear pot instead?
If it requires a 10k never lower,
I could boost it with an 8k resistor.
The reason I'd like to use the 2k pot
is I have a ten turn pot and thought
I might have finer control at the top end.
Russ: I am experiencing some volume control glitches with my BuffPRO 9038. Sometimes, at the start of a new playlist (album, etc) but not between tracks, there will be a louder than usual POP, and the volume will go way up (but apparently not to full scale, thank goodness). Then with further input to the volume knob (pot) it will drop to normal and then operate normally.
The 9018 never had a single instance of anything like this. Same source (USB).
Setup: BuffPRO/Amanero-Hermes-Cronus. DAC is AC grounded, to chassis, XLR pin one to chassis (AES recommended approach).
Any ideas?
The 9018 never had a single instance of anything like this. Same source (USB).
Setup: BuffPRO/Amanero-Hermes-Cronus. DAC is AC grounded, to chassis, XLR pin one to chassis (AES recommended approach).
Any ideas?
Sounds like something is happening to the PS to cause a hard reset. I have never seen anything like that.
The only condition I know of that could possibly cause that would be that the DAC was in a reset condition without the controller resetting (and thus setting the correct volume). That should never occur unless the power supply situation is iffy.
You might want to break out the scope. It could be a sample rate change (or even mute/un-mute condition) is causing a current demand situation that the supply is not handling.
The only condition I know of that could possibly cause that would be that the DAC was in a reset condition without the controller resetting (and thus setting the correct volume). That should never occur unless the power supply situation is iffy.
You might want to break out the scope.
It could be a sample rate change (or even mute/un-mute condition) is causing a current demand situation that the supply is not handling.
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Ah yeah - I suppose it could be a bad pot. But I am not sure that matches the described conditions where the problem occurs. I would literally put a scope on the supply and watch what happens during the problematic transitions
Or alternatively - just adjust/replace the power supply to one that is known to be able to handle the large current swings/demands.
Or alternatively - just adjust/replace the power supply to one that is known to be able to handle the large current swings/demands.
Definitely not a power supply issue, 30VA tranny, 22,000µF of C, and Belleson 3A capable regulator. It is weird though.
Note, i am using only one XO (45.1584, I convert everything to DSD in software) in the Cronus, so maybe something weird is happening RE sample rate changing, sequencing with Amanero/Cronus?
At first I suspected a bad pot, but then I noticed it happening with no pot movement. Is it worth mounting up the other oscillator to see what happens (I have it)?
I am planning to mount a bit better pot later (just to have better bushings smoother action)...
Note, i am using only one XO (45.1584, I convert everything to DSD in software) in the Cronus, so maybe something weird is happening RE sample rate changing, sequencing with Amanero/Cronus?
At first I suspected a bad pot, but then I noticed it happening with no pot movement. Is it worth mounting up the other oscillator to see what happens (I have it)?
I am planning to mount a bit better pot later (just to have better bushings smoother action)...
OK...
Just add a jumper from the XO output pin of the 45.1584 to the Cronus PCB for the other XO? Simple as that...
First I'll go ahead and mount the other XO, to make sure this could be the issue.
Just add a jumper from the XO output pin of the 45.1584 to the Cronus PCB for the other XO? Simple as that...
First I'll go ahead and mount the other XO, to make sure this could be the issue.
Would be interested in your impressions concerning the difference in SQ between DSD and PCM on the new chip vs the 9018. Of course, all the downstream analog gear also matters with each format. But the 9018 differences between DSD and PCM are fairly distinct (in my opinion). Haven’t been able to play with my new chips yet...
Yep...
Thanks Russ & Brian. First I mounted the other XO, and then tested and it appeared to fix the problem, so then I jumped the two XO positions via a u.fl cable per Brian's suggestion. Did a bunch of start, stop, empty playlist, start stop cycles and no sign of the issue again. I cannot be sure until a couple of days go by, but it seems good now.
franco: I have not even bothered listening to PCM on the 9038, as i found DSD (converted via Audirvana +) to be better on the 9018, I just assumed the same would be the case with 9038. A+ has some very nice flexibility in filter parameters as well, so one can experiment. It oversamples to DSD rates at 32 bits first, and then modulates to single bit from there. Right now I am feeding the 9038 DSD 128, as that is all Amanero can do (DoP limitation)-the Amanero firmware for native DSD under Linux is still beta (at best). I am hoping Russ will offer his XMOS solution and make sure it runs native DSD with Linux (we at Sonore are happy to help him if he needs it, wink, wink) so I can try DSD 256.
I have a DIYINHK XMOS interface that can do native and high rates, but it has no isolation and I find isolation (with re-clocking) of the USB interface to be a good thing.
Thanks Russ & Brian. First I mounted the other XO, and then tested and it appeared to fix the problem, so then I jumped the two XO positions via a u.fl cable per Brian's suggestion. Did a bunch of start, stop, empty playlist, start stop cycles and no sign of the issue again. I cannot be sure until a couple of days go by, but it seems good now.
franco: I have not even bothered listening to PCM on the 9038, as i found DSD (converted via Audirvana +) to be better on the 9018, I just assumed the same would be the case with 9038. A+ has some very nice flexibility in filter parameters as well, so one can experiment. It oversamples to DSD rates at 32 bits first, and then modulates to single bit from there. Right now I am feeding the 9038 DSD 128, as that is all Amanero can do (DoP limitation)-the Amanero firmware for native DSD under Linux is still beta (at best). I am hoping Russ will offer his XMOS solution and make sure it runs native DSD with Linux (we at Sonore are happy to help him if he needs it, wink, wink) so I can try DSD 256.
I have a DIYINHK XMOS interface that can do native and high rates, but it has no isolation and I find isolation (with re-clocking) of the USB interface to be a good thing.
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There was a thread active a while back...
The Best DAC is no DAC
It describes a particular XMOS board that could be tapped for a raw DSD signal, that signal filtered for HF noise, and the result sent merrily off to a preamp. I messed with it but found too many limitations for my own use case. But the suggested XMOS board seems pretty nice. It does DSD 256, DoP 128, plus PCM. The output is isolated and reclocked. For the price, perhaps it would answer some of your questions? Mine is sitting unused so I wouldn't mind sending it to you. I have way too many projects stacked up, and I imagine Russ and Brian are in the same boat!
PM me if I can help.Frank
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There was a thread active a while back...
The Best DAC is no DAC
It describes a particular XMOS board that could be tapped for a raw DSD signal, that signal filtered for HF noise, and the result sent merrily off to a preamp. I messed with it but found too many limitations for my own use case. But the suggested XMOS board seems pretty nice. It does DSD 256, DoP 128, plus PCM. The output is isolated and reclocked. For the price, perhaps it would answer some of your questions? Mine is sitting unused so I wouldn't mind sending it to you. I have way too many projects stacked up, and I imagine Russ and Brian are in the same boat!
Frank
I have been using the I2SoverUSB board for several years with my 9018 chip DAC and now with my new 9028 chip DAC to play DSD256 (not DoP) using HQ Player on Linux server coupled with a mRendu NAA. The USB board has two power supplies - input & output. Works great. No idea how it compares to the other USB boards for SQ.
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Yeah, the JLsounds would probably work, but I find it to be a bit primitive, with older XMOS (external PHY) and relatively unsophisticated power supplies/regulators. A good deal for the price though and perhaps worth trying.
I like using Cronus because it is compatible with the Pulsar clock, which I want to try... Problem is Amanero reverses DSD channels, and Linux native DSD compatibility is very sketchy right now (beta), so limited to DSD 128 with DoP.
A well sorted XMOS solution from TPA/Russ which plugs into Cronus would be ideal.
I like using Cronus because it is compatible with the Pulsar clock, which I want to try... Problem is Amanero reverses DSD channels, and Linux native DSD compatibility is very sketchy right now (beta), so limited to DSD 128 with DoP.
A well sorted XMOS solution from TPA/Russ which plugs into Cronus would be ideal.
Some of this information is out there so I am free to talk about it.
The DAC always uses internal 32-bit data pipeline for PCM and DSD. The special part for DSD is that it also expanded to this same internal format, and importantly it does this without any sample rate conversion! (in the conventional sense - decimation)
For PCM inputs the final sample rate is the rate after the oversampling filter, while for DSD it is the native DSD rate. But each "0" or "1" is scaled to 32-bits - so that it can have volume control applied. Everything then goes to the HyperStream-II modulator.
While downsampling to DSD to PCM is indeed a harmful process, for example: one that goes from a 2.8MHz DSD signal to a 352.8kHz PCM signal. Inside the ESS DAC what is happening that the DSD data is basically converted to a 32-bit stream but while retaining the original 2.8MHz sample rate!!! This is really important - because it means that the data is the same - it just can now be scaled.
The ESS DSD process is not a downconversion where the sample rate is decimated - instead you should think of it as an upward conversion where each 1 bit value because a 32bit value
It's actually a particularly awesome way to handle DSD. I can't say I have heard anything better than the ES9028/38 at handling DSD with volume control.
I hope that helps!
Cheers!
Russ
Russ -
Is this true for all of the ESS chips, or only for the new PRO versions? This is appealing to the "we don't need no stinking processing" folks among us.