It clicked with me that my speakers and room ended up that same way. Left to right moving soundstage images no longer start turning and heading back over my right shoulder after pulling right speaker forward. I figured it out about six months ago though but podcast made me connect with the advice
I was moving the balance control to the left towards the open side of the room. To correct that adjustment I pulled the left speaker out into the room 1/2" more making it 1/2" closer to me. That did the trick of centering the soundstage and locking it in. I think the right wall was adding some reinforcement to the loudness that could not happen on the left side. I tried changing the toe in on one side and then the other but ended up putting them back toed in equally. These speakers dont like much to in anyway. I have them each toed in roughly 5 degrees.
The left speaker is open into the room on the left side and is 43" from the front baffle to the wall behind the speakers. The right speaker is 42.5" from the front baffle to the wall behind the speakers and the right speaker is 44" from the right side wall to the center of the front baffle.
Almost to a T then⊠
I have my balance one or to clicks to the right (where it is not opening up to more house). Thank you two very much for sharing, and Iâll visit the podcast for more details and context.
Sorry to get us off topic, but have been working hard on speaker placement this Spring. Iâve used the manufacturerâs suggestions as a starting point, but still need to tune to perfection. Strange old home.
I think in a âperfectâ room with âperfectâ treatment and the speakers free to be positioned correctly it would be a lot easier to get the âperfectâ setup.
I dont think a lot of us have that option so then you are sitting in a âsound fieldâ for lack of a better term and all of the reflections and absorption from things in the sound field make it difficult to predict what is needed until you start fussing things into place.
I have learned a lot just hanging out here and it has made my system sound better than i thought it possibly could. Even in a room that is far less than perfect.
That is part of how John Hunter of REL balances speaker setup using the ânon-masterâ speaker during setup.
Iâve always been amazed that it works, but when done well one still senses a strong center image even when outside the speakers on each side.
Definitely a set up skill to aspire to.
Adrian,
start with the easiest and least time-consuming modifications. Install Nichicon UKZ capacitors on the analog board in the places I showed. You will do it within 10 minutes. Youâre only risking a few dollars. If you are dissatisfied, you will remove the capacitors without leaving any traces.
I notice that my colleagues do not believe that such a simple and cheap modification can be effective. I believe that it improves the sound more than replacing the output transformers.
I am gearing up to do some of these component level mods. Everything that you are doing makes sense to me and are simple to perform.
Since the DS is going to my office when the MKII arrives, I will have plenty of time to hot rod it. I like making one change at a time and listening for a while. Thereâs enough mods to keep me tweaking for a year. Fun!
I am using asr emitter 2 exclusive.
There is an option to adjust the input impedance.
What should I set for DS rca?
The photo is from the site
Pick which ever sounds the best to you, you canât hurt the DS with any setting. Lower impedances will lower the output level a little, but if you can still get the volume you want everything is fine.
Yes, Adriaan, the PS Audio stock transformer is a custom build, I think, to fit the allowed space. 2x120vac primary and 2 x 8vac secondary.
I do not know how to find a better transformer to replace this item, and a replacement will have to be external.
The sky is the limit, however, the stock power PCB can be left alone to supply the front display only and a Farad Super 3 external to supply 5vdc/3amps to the digital PCB.
I am a bit slow and I like to perform one stage at a time.
A question about the LPF design here. Maybe it has already been discussed? A friend dropped off an Abbas DAC to listen to and it has the gears turning again on some olde skule concepts Iâd sorta gotten away from lately.
For example Tantalum resistors. Iâve used through hole tantalum before and generally liked it alot (and have used Tantalum caps quite a bit as well). I happen to have a supply of SMD tantalum R in several values, with high hopes for those; was planning on using them in a preamp design but never got to it. Am thinking now about maybe trying some in the DS LPF circuit.
In the image I see that R3 and R4 are 1K1. I have 1K0 tant that could sub, but now the tuning is off. Hmm. Unless we could compensate (via the C8/C9)?
But think Iâd rather try a tantalum in the positions R1/R2 and R7/R8, if that is in fact the bottom of the feedback divider. Question- can we get the value of those R without having to remove and measure? Maybe weâll get lucky, so thought Iâd ask at least 
Looks to me like the LPF design is an MFB. Similar to the s.e. version below. Unfortunately Iâm not an active filter expert, and apologies if this has already been discussed.
Thanks, T
Image courtesy of TI:
anyone in OZ (Australia) looking for the transformer upgrade (supply your own) can report the Jim @ www.filatronicsavservice.com does a great job around the $120 mark - Transformers, RFI, NCF plug and Purple all running in âŠ
Those are the resistors that set VCOM, they arenât in your sketch of a filter. changing them from 10k to 1k raises the current needed from the power supply by 12.6mA. Iâm pretty sure that wonât crowbar the power supply. On the other hand, the caps with eight legs alternate plates between with each pair of adjacent legs. If you have even one set of adjacent legs shorted, you will be shorting the VCOM to ground and pulling the output of the opamps to ground (and out of valid range.)
R1 and R2 are 54.9 Ohms, R3 and R4 are 1.1k Ohms, R7 and R8 are 86.6 Ohms and R9 and R10 are 75 Ohms. They all are +/- 0.1%, 25 ppm / degree C, 1/8W high reliability thin film resistors.
Changing the values isnât recommended - they (along with the caps) all interact to provide the correct load to the reclocker upstream, the correct filter cutoff frequencies and the correct output impedance to drive the transformers. R9 and R9 values are less critical than the others, but lowering them will add (a little) more distortion to the outputs.
Ted, thank you- exactly what I needed 
And sorry, that was a recycled image with the letters A and B still in place. I assume those are the Vocm divider.
Thanks, T
I canât believe the number of mistakes I made reading your question. Iâm glad I just gave you the schematic
Yes going from 1.1k to 1k on R4 and R4 wonât make too much of a difference. It will lower the output level by about 1/3 of a dB. It will shift the filter cutoff up just a little which probably isnât a problem either.
Thanks for your help. I wish the R1,2 and R7,8 were standard values like 50R and 100R. Then I might try it.
Thanks, T
Hi @tedsmith,
What would cause DS to make a crackling sound similar to dragging a phono cartridge over a few tracks? It happens intermittently with varying loudness!
Assuming DS not modified:
Try bit perfect test to verify that the cable connection, etc. are good (or less likely, if you are playing DSD, that a volume setting or DSP setting is barely modifying the signal.) (How to run a bit perfect test with DirectStream â PS Audio)
If youâve changed any parameters on your source, e.g. buffering size, etc. check them. Perhaps see if the problem persists with a different source, or another DS input.
Does the display change at all during the noise? If so, knowing what it changes to would be very helpful.
On the hardware side, the obvious: loose cables or other intermittent connection.
With a little more information I might be able to be more precise.
Thank you for your prompt response Ted.
I use Bridge as the only input. The noise happens both while playing, while paused, and while muted. When I turn off the DS DAC, it does not happen.
I will start by replacing the analogue interconnects. Second, will inspect the soldering of the XLR connectors on the analogue board. I hope this resolves it.
Trying a spice model with a different OA so far, but as a sanity check- the circuit gain In to Out is about 19dB or so? The 3dB BW is about 60KHz (using a source impedance of 50R)?
And looking at Zin, thereâs a narrow spike around 100KHz or so where the input impedance triples?
Just wondering if my model is close to yours.
Thanks, T