New DirectStream software work

"New DirectStream software work
Do you have any pet new features or annoying bugs you'd like me to look at?"
Ted,hello... Can you add the total number of tracks with total time left on the full cd somewhere on the ds or pwt? I do like that option instead of guessing.
Mark
mark-d said
Can you add the total number of tracks with total time left on the full cd somewhere on the ds or pwt? I do like that option instead of guessing.
The DS doesn't know about CDs at all, it just sees a stream of bits. The bridge gets a little more meta-data, but I don't know if it has the info that you would like.

I don’t have anything to do with the PWT, but tho I’m sure that PS Audio engineering will read this, I doubt that they will be changing the PWT UI soon. I do agree that time left, total number of tracks and total time are often nice to see.

Shouldn’t that be in mconnect ? Btw the bit rate is there now.

mark-d said
"New DirectStream software work
Do you have any pet new features or annoying bugs you'd like me to look at?"
Ted,hello... Can you add the total number of tracks with total time left on the full cd somewhere on the ds or pwt? I do like that option instead of guessing.
Mark
That would be nice on the PWT, but currently it is not even on the radar.

Dennis

I assume this issue has something to do with the bridge or UI; however, I’ll pose it anyway. It strikes me as odd that the album art reloads when I remotely (JRemote/JRiver) change the volume.

thanks, Dale

amsco15 said I assume this issue has something to do with the bridge or UI; however, I'll pose it anyway. It strikes me as odd that the album art reloads when I remotely (JRemote/JRiver) change the volume.

thanks, Dale


It has to do with the UI, something is telling it to refresh the screen.

Dennis

Ted Smith said I don't have anything to do with the PWT, but tho I'm sure that PS Audio engineering will read this, I doubt that they will be changing the PWT UI soon. I do agree that time left, total number of tracks and total time are often nice to see.
Ok PS Audio engineering department,Ted and the rest of us want this feature on the PWT !!! What say you?

Cheers…Mark

Ted, any chances to experiment with pre-ringing? E.g. using different filters? I remember you mentioned, that the group delay appeared more important than pre-ringing, but I’m curious if it can improve the (micro)dynamics (add more “bite”, make it less “soft” or “polite”?).

How about color combinations for the display. Also a possible video output. As in my office the dac is very far away and as I do not like to use an iPad i cannot see the screen. Lastly a variation of what is important for each of us on the screen. As someone here wants a large display for bit rate how about the same for volume. .

Al

Is it possible to turn off everything (or as much as possible) that is not used? For example, if you turn the LCD off, the to be shown information is still being processed, right? And if you are using the Bridge, the rest of the input circuits are still on. The idea is to get the EMI and power draw to the minimum.

Ted Smith said Howdy Al

I thought I did explain what changes :slight_smile: Tho I expect that most people would describe the change in sound differently, for me it’s a slight loss of detail and pace. It just doesn’t make you want to move as much or draw you into the music as much. My hearing isn’t what it used to be but the top seems a little more muffled just after power up.

Like I said, perhaps 5 to 10 minutes seems to get back most of the involvement for me, so I suggest 10 to 30 minutes. But like burn-in every one probably has differing expectations/experiences.

-Ted

This is really interesting. After breaking-in the DS for a very long time (never turning it off from the back), the overall presentation became soft with a small soundstage. I tried a different DAC and the soundstage was big. Hmmm. As I do from time to time, I played the Ayre Acoustics short glide tone. As usual, this resulted in worse sound for quite a while (amazing since the short tone is 1 minute), but ultimately the soundstage re-emerged and the DS sounded better than ever (meaning amazing). Wish I had only run the tone through the DS rather than the whole system to test just the effect on the DS only.

Assuming this isn’t in my imagination, what might be the reason of the improvement? Thanks.

Alekz said Ted, any chances to experiment with pre-ringing? E.g. using different filters? I remember you mentioned, that the group delay appeared more important than pre-ringing, but I'm curious if it can improve the (micro)dynamics (add more "bite", make it less "soft" or "polite"?).
I do want to (re)try different filters now - but like I mentioned somewhere else on this thread I'm beginning to suspect that a lot of the differences that people hear on DACs with multiple filtering options are differences in the implementation of the filters rather than differences that are inherent in the filters in question. Anyway yes, less preringing is something that I want to listen to.
alrainbow said How about color combinations for the display. Also a possible video output. As in my office the dac is very far away and as I do not like to use an iPad i cannot see the screen. Lastly a variation of what is important for each of us on the screen. As someone here wants a large display for bit rate how about the same for volume. .
Display issues/changes like this are up to the PS Audio engineering people. As I mentioned in the first post I don't know how much time they will have to do new work on the UI. I do like being able to see larger versions of some of the info on the screen, but having it be customizable seems unlikely. Hardware changes like a video out aren't on the table now.
Alekz said Is it possible to turn off everything (or as much as possible) that is not used? For example, if you turn the LCD off, the to be shown information is still being processed, right? And if you are using the Bridge, the rest of the input circuits are still on. The idea is to get the EMI and power draw to the minimum.
I agree with the suggestion in general, I'm not sure how much there is that can be done now tho I am always looking for such oppotunaties - in general I took a lot of care in the hardware to minimize EMI, I use the slowest signals consistent with function everywhere, differential signals when possible, careful placement of traces and the shortest runs I could get away with, lots of shielding inside the boards. When we did FCC testing we passed easily (actually we removed excess filtering for better sound quality.)

I don’t know if processing for the display is done when the display is off, but there’s little processing that’s done now in the control processor that isn’t display related in the display code. I do think there is a some communication to the FPGA that isn’t necessary in some circumstances (I accidentally left some debug code in that reads more than it needs from the FPGA for example.)

The TOSLink input is just a single wire from the TOSLink receiver module to the FPGA, having switches in the hardware to turn off the input comparators for S/PDIF, etc. wouldn’t save nearly as much EMI as disconnecting the external cable. Not to mention that hardware changes are not on the table now.

Turning off processing inside the FPGA will (in most cases) cause more EMI - the FPGA does everything in parallel so you don’t gain anything timewise or free up resources if you “put an ‘if’ around a chunk of code”. Putting in an ‘if’ just doesn’t enable the clock on the last store of the answer into a register, all of the rest of the processing happens just the same. Having a lot more clock enables causes the fanout of the control nets to rise which (can) add more jitter…

PYP said
Ted Smith said {something about turning the power off and on causing temporary changes in the sound quality}
I played the Ayre Acoustics short glide tone ... what might be the reason of the improvement?
I can only speculate: like needing to degauss a tape recorder head there are asymmetries in audio that might be pulling various things in the audio path away from optimal: the simplest example is magnetizing the output transformers but also polar molecules in various dielectrics in wires or other insulation might be migrating and cause less optimal sound. Sweeping tones (and other break in signals) are often much louder than typical music and are often symmetrical (or, in the case of noise, have an average value of 0) - these may clean some of the asymmetries up... What ever the real reason, I've experienced improvements in virtually all components and cables in my system by running such tones thru my system over time.

Hi Ted,

I am new here at PS Audio, but I am following the DirectStream developments. A few years ago I was lured into the Linn camp for streaming and believed their view that hirez PCM was the best way to go. I built my database around PCM, translated my dsd material to hirez PCM and was happy - sort of-. (I have a Akurate DS).

Than for bluray I bought a Oppo-105 and started streaming DSD. Although I think that the Oppo brought a lot of extra detail, it lacks ‘musicality and fun’. So, which one could combine both? I am betting on PS audio right now!

Back to your question for this thread. What else in software? My suggestion: is it possible to program an option to translate all the inputs to 352 kHz/32 bits PCM (!) and feed this directly into your filtering cascade? This would give the definite comparison for PCM vs. DSD advocates.

In short, can you mimic a hirez PCM DAC? Or would this be the same as just lowering your sample rate a factor of 16?

berlin

The output of the DirectStream DAC is a passive filter of a 5.6448MHz stream of bits. That’s all the hardware can deal with.

The software inside the FPGA converts everything to 5.6448MHz in stages.

If the input is PCM at 44.1k or 48k it is converted to PCM at 88.2k or 96k

If the input is PCM at 88.2k or 96k (or was converted to 88.2k or 96k) it is converted to PCM at 176.4k or 192k

Then the result (or inputs that were 352.8kHz PCM or DSD) are converted to 28.224MHz and then back to 5.6448MHz and then to one bit for output.

That’s all software, but no matter what the software does it needs to end up with 5.6448MHz single bit noise shaped samples.

The rest of the FPGA software communicates with the control processor or decodes the digital inputs.

Ted, I too agree with Woot’s observation about the sound, in fact I ended up adding just a bit of eq in J River, though i blame my rolled-off hearing for the fact that I felt a need to do this rather than the DS. :) Mine is also nowhere near broken in yet, though I’m not sure if that’s directly relevant. In terms of software, I do miss the filter options though in truth I left everything at the default 98% of the time. One thing that I assume is in the pipe and which would indeed be useful to me sometimes is the ability to switch inputs via the new Bridge. (Unless Mconnect can do this but if so I haven’t figured out how.)

I’m much more concerned with getting the practicle problems fixed. Very happy with my DS. At the end of songs, I sometimes hear what sounds like the noise from an LP leadout groove. A small series of pops. I exclusively use the bridge with JRiver/JRemote. It doesn’t alwys happen but often enough…

amsco15 said At the end of songs, I sometimes hear what sounds like the noise from an LP leadout groove. A small series of pops.
That has to be the bridge (or JRiver) - but even so the next release of the DS software may well clean them up (it mutes the output more aggressively when the input signal has an invalid or rapidly changing clock.)
Bob said Ted, I too agree with Woot's observation about the sound
Could you point me to his post or try to put your observations into your own words - the only post I see from woot on this thread is a difference of preferences from al which doesn't help me to understand what you are saying.
One thing that I assume is in the pipe and which would indeed be useful to me sometimes is the ability to switch inputs via the new Bridge. (Unless Mconnect can do this but if so I haven't figured out how.)
I honestly don't know how such things are done for the PWD or DS - when I worked a little on the UI code I'm pretty sure I broke some of those kind of features and Dennis had to clean up my mistakes :) Perhaps he can chime in on whether newer bridge or control processor software is likely to support it.
Ted Smith said
amsco15 said At the end of songs, I sometimes hear what sounds like the noise from an LP leadout groove. A small series of pops.

That has to be the bridge (or JRiver) - but even so the next release of the DS software may well clean them up (it mutes the output more aggressively when the input signal has an invalid or rapidly changing clock.)


Are you using Wav files? If you are it could be metadata in the wav files. DBpoweramp and elyric put metadata in in wav file rips that the bridge can not handle.

Dennis

Thanks Ted and Dennis. I rip all my music using DBpoweramp into FLAC files. Also, the noise I’m talking about doesn’t happen when changing resolution (ex. 16/44.1 to 24/96). I’m noticing it between songs on the same red book FLAC rip or between any other red book songs.