OK I finished wiring up the pre-amp. See picture below. As well as wiring up the selector switches, I have put in a few filter caps as suggested by Emilio. There are caps on the regulators, and I have also put caps right on the input pin of the JLSounds board. I only wish I had bought a wider enclosure - it got a bit tight soldering the switch that is next to the no-DAC components/wiring.
I'm now listening to the no-DAC using Ubuntu OS, HQPlayer 3.12 upsampling to DSD256. Observations so far.
1. There is NO noise. Or no noise that I can hear from my couch. The prototype no-DAC, that was sitting out in free air, suffered from various hf noise and whooshing type sounds. I think that it was picking up lots of interference. Now I have enclosed it completely in a box, and put bypass caps on all the power inputs - no noise.
2. Only pops I get are when I change sampling fequency (I upsample to 11.3 MHz for 44.1kHz family including SACD, I upsample to 12.3 MHz for 48kHz family). Even then the pop is innocuous. There are NO pops when tracks start and stop, or when I hit the next track button or stop button
NOTE - I haven't tested with Windows10 just yet, but last time I used windows there were lots of pops as described by Ray. But Ubuntu seems fine - note that I am using Jussi's kernel 3.33.0.55 (available via the signalyst.com website). Ubuntu still doesn't support native DSD (only DSD as DOP) but Jussi's modified kernel will support native DSD up to DSD256.
3. As requested by Emilio, I have measured the output of my no-DAC. I created a 440Hz, 0dB test signal using Audacity as a 16 bit wav file. Upsampling to DSD256 via HQPlayer, I measured 1.55V at the output of the transformer.
4. And the sound is - absolutely lovely. PCM files, from 44.1k to 192k, and DSD files are handled with aplomb. I'm still a vinyl man at heart, but most vinyl these days is cut from hi res digital files. I reckon that those files will sound better if I buy them from HDTracks and play them via the no-DAC, rather than buying them encoded onto vinyl, and decoding with my Lenco. I will still listen to my older (analog) LPs on the Lenco, but for now I am pretty happy.
Regards, Hazard
Hi Hazard,
Good work ! So, with an output of 1,55V with balanced, I believe that we can use only buffer, without complicate our life
Thanks Ray, excuse my bad English, no problem if you implement the board design with ne555, but before doing so, simply try to bypass the tlp185, as did Audiodan of nexthardware forums. To keep the galvanic isolation you can feed the circuit or with a dedicated power supply or with the power supply side of the usb jlsounds. Using basically the design of the filter LME49710 board that you can download from jlsounds site, You can review your circuit. One important thing is that my circuit that I used on my nodac with output stage FET in dsd128 in windows environment and naa hqplayer without problems, used by a friend of mine that unlike me, uses linux and has a stadium output tube, it have the bump in stop. I would not want that the problem has also influences the output stage ... Sorry but my electronic knowledge is less than a goat. Bye Fabio.
Thanks Fabio and, yes, I have been thinking about eliminating the tlp185 device just as you suggest.
Ray
I have a dam, tried most of the filters, the no dac is much better , buy a long margin imo.
I think that difference is all in output stage, the dac can to influence but a true comparing it need the same output stage. Sure that Dsd is future, but now there are only a few native files
Before we go much further, my inclination is to modify the flip-flop and SE boards with the modification suggested by Fabio, which should mean the possibility of eliminating the pop at start, should this be an issue in your context. That still leaves the potential stop pop but is a step in the right direction.
That is unless anyone has another solution?
I would appreciate some feedback from you chaps on whether to modify the boards.
Ray
That is unless anyone has another solution?
I would appreciate some feedback from you chaps on whether to modify the boards.
Ray
1 possibility is to have the pin as a dedicated DSD mute function only from the JLsound board...
unmute after the start of the DSD signal and immediate mute when a stop is detected...this should require no change to the current design...
Thanks Flowerpot. My understanding is that what you describe is what pin 7 (Codec change) pin on the JLSounds board is supposed to do, hence the current board design.
That said, I was waiting to see if Jussi had anything to offer first but he doesn't seem to have picked up on the question so I'll drop Lubyen a line.
Ray
Chaps, just to be clear, even with Fabio's mod there can be no guarantees that a pop-free solution will result; as you may have picked up from the thread the actual results will depend on the context in which you operate the system.
I'm happy to include the mod but don't want people to think everything will be hunky dory as a result. This is an experimental thread so if you're not prepared for that side of things this may not be for you. Of course, if you can put up with this inconvenience you will be blessed with wonderful sounds.
I've started to look at modifying the boards and maybe providing options for configuring it.
Ray
Another useful piece of learning to share; it seems that we will not be able to use the HQPlayer volume control with the simple low-pass filter approach we're experimenting with here (unless you're prepared to hear white noise).
HQ Player - Page 190
Looks like we'll need to look to use analogue volume controls. It's a shame, but there it is.
Thinking now required for a volume control for my flip-flop/BCF project.
Ray
HQ Player - Page 190
Looks like we'll need to look to use analogue volume controls. It's a shame, but there it is.
Thinking now required for a volume control for my flip-flop/BCF project.
Ray