Marco

Aha, I see I really have to upgrade to version 4, 3 can't do this. Does it matter (much) what signal is used for testing? I can imagine a few minutes of music is enough to generate a reliable weighing of the average speed and if the signal doesn't matter much it's even reliable to compare between different systems. Didn't expect this to be so easy :-) Thanks a lot Miska, excellent stuff!

Right. Certain things should be set static (Red Book, Modulator and Filter, choose or fix to highest demanding/most popular or generic one), the rest would be of help actually: having more or different memory yet getting a higher score in the same system means it's easier to see what various hardware and software settings do. And vice versa. I'm looking for general scores, so not interested in all possible filter and modulator combinations, just the real demanding popular one. A true benchmark. Just Red Book to X-speed DSD, EC & Cuda (Closed Form maybe?) enabled or not. This to keep the test program easiest to keep up to date and compact. Maybe it's just as easy to add all settings of HQPlayer, or that otherwise as you say, it is of no use at all. Could very well be. And yes, the proof is in the pudding, iow you need the real application (embedded/desktop/..) to know for sure. But what is really in the dark now, is how much a certain software setting or hardware difference changes calculation speed. We've seen back and forth discussions about greying out and (de)selecting certain settings in order to get at the optimum performance for a given system. In the end, Jussi is the only one with the insight to evaluate this as doable and of a test program's validity for the main programs . I would like to know if it can be done and if it's something to consider. Above all, Jussi, I can easily imagine you're busy enough as it is, but I also think it would get lots of customers on a much clearer upgrade path, be that for general GPU's, memory, CPU's or settings. You might even already have a similar environment at home, to measure what certain changes do. It could be build off of HQ Pro, as one has all the options, can write without selecting a real output. Generating a score depending on the (minimum?) achieved speed would be just the indication one needs to know how a system could be optimised, as well as what one shouldn't do. This of course, with a disclaimer that results doesn't make the author responsible. That's it in a nutshell. Of course, one can always install a trial version of HQPlayer and see if it works, but it really doesn't generate detailed of possible bottlenecks. I have no problem keeping an Excel sheet for this. I'll leave it at that.

Now what could come in handy is a single executable to run as a benchmark to test e.g. the EC filter/modulators (or just 1 for dsd64,128,256 and 512) and output a score. If this would be indicative of the performance of the actual load when playing music, it would be really usefull in exchanging scores cross platform. For instance dsd64 could score 480%, dsd128 220%, dsd256 101% (just above real speed) and dsd512 45%. One could easily test for best scores instead of black and white "it works"/ "doesn't work" as well as be able to exchange scores for comparison (be it with same hardware or not).

I'm taking you want to go the diy route? You can use these: https://www.minidsp.com/products/usb-audio-interface/mchstreamer And as for NAA you could use the Beaglebone Black, which supports 4 channels at up to dsd256. https://puredsd.ru/ Scroll down to the bottom for the firmware, but this image needs more editing for supporting 4 channels. Rather off topic, maybe better via DM.

Thanks Miska, those look like high quality, ready built options, but I`m looking mainly for a raw dsd datastream like provided by the Beaglebone/Botic combination. Kind of hoped somebody had been tinkering with e.g. the new NXP IMX8M, or some other Arm board that a capable GPIO programmer could get up and running in a day or two. I`m guessing it`s rather off topic and out of scope here on CA and will continue this question over at Diyaudio, but it is rather surprising to me that in 2019 there isn`t any capable hardware that can do more than 2 channels of dsd`s best format 🙂 Btw, any idea what makes that "chipless dac's" like the dsc1 put out loads less modulation noise/isi when fed with the AMSDM7 modulator? Enjoy your sunday 🙂

Hi all, Has anyone experience with NAA that supports 4 channel I2S dsd512? From what I can gather there`s never any confirmation about actual rates vs channels. Raspberry wouldn`t, Beaglebone hasn`t got the NIC to support this, Imx8m seems to but since I`m no build-your-own-driver-buff I wouldn`t know how to test it. What I`d like to do: 2 channel audio to stereo 2-way active out in dsd512. Anyone any experience in getting convolution on HQPlayer Desktop (or Linux/embedded) working to a NAA to 4 channels dsd512? Any direction will be greatly appreciated!

I'm guessing Sandyk is talking about inline playback and you're talking about offline upsampling/playback?

AFAIK there`s no other combination possible, as for it to work it needs a Botic driver: http://bbb.ieero.com/

It does do 512, but you need to populate the clocks with 45 & 49 MHz versions. Any higher would maybe be possible if it weren`t for the 100mbit NIC reaching its limits, I assume. It runs rock solid, sounds very organic and transparent and it serves as an NAA. I did replace the smt electrolytics with tantalums. Really nice!

4est, many thanks! I couldn't find this flutter related problem on the Internet, so I reckoned it had something to do with my hardware. I'll see if 1096 works also. Btw if you're interested in a, imho, better sounding source (and/or in a network connection), you might want to check out the Twisted Pears' Cronus/Rhea etc. No fluttering there whatsoever and easy to connect to the DSC1. I'm not sure if Mute is operational though..

Ah, the dreaded mute-problems. I am running the output of the shift registers at different speeds than the datastream from Amanero. I only get pops when turning the dac on and off and when shutting down the pc. For the rest no problems when starting, changing filters or starting/exiting HQPlayer. This is without a mute circuit populated. I think it works because the storage registers still get their clock independent of what amanero does, but I haven`t given it much thought yet. The problems i had were different: fluttering/ noise related problems in one channel. It is amanero based, because with a different CPLD (CPLD 1080 channels swapped) the noise problems also swapped channels. Maybe it is my version of the board, but I have 2 and have the same problems with both at dsd512. 256 and lower worked fine. You`re saying don`t have those problems at those speeds?

Eclectico, DSD streams basically work by means of averaging the electrical values by a low pass filter. In your example it could work by averaging a stream of 0,2 volt, next clock cycle 0,0 volt, next clock cycle 0,2 volt again, next 0,0 etc etc. On average this gives 0,1 volts. As Jussi pointed out, the values are not changed every 22 microseconds as with PCM 44,1 KHz coding, but with DSD512 around every 42 nanoseconds/24 MHz. Averaging all these high data rate changes is easier done as well as less jitter sensitive when done with a FIR-filter.

The newer firmware versions also don`t work for me (on Windows OS). After lots of reflashing I settled for "CPLD_1080_XC12"and "firmware_1097c". Lots of hassle to get dsd512 working. What version did you end up with?

Audio note almost exclusively uses the AD1865 dac chip in their products. It's a 18 bit ladder dac. Afaik they don't use an "analog fir filter", so it might not be applicable to their products. Transformers are funny things: "Slightly over damped" means something like a slightly lower load than the 1700 ohm output impedance they state in the datasheet of the ad1865, and also keeping the energy transfer maximum seems to point at this. Many dacs they manufactured incorporated an enclosed filter as well as a transformer, with a ratio of 1:1 or 1:2. The filter used resistors to load the current output of the chip. In later years they completely abandoned filtering of the deliberate kind whatsoever, don't know what they do for the last 5 years or so. It's hard to manufacture a high bandwidth transformer at high impedance sources like a -sort of- current source. Though 1700 ohms arent that bad, the dsc1 has an output impedance of about 470 ohm, many times lower, and it would probably also be more uniform in its impedance. Ladder dacs tend to change output impedance depending on the output level, so distortion products are more erratic. Also, I'm not sure if capacitive coupling in the transformer is sufficiently mitigated to state the rf demons of dsd and ladder dacs are sufficiently filtered out without added filters behind one, after going through a transformer. I would love to see some wide bandwidth (normal and common mode, above e.g. 100KHz) measurements of the dac output to prove that is the case and/or capacitive input measurements of the transformers' input and coupling to the secondary windings.

Just to be clear: it's my copy and paste fiddling that makes it look as if ted_b is suggesting we should all listen to another ones DSC1. I wanted one post and quote 2 persons. Sorry for that ted_b!