It doesn’t actually. It’s just an asynchronous error-corrected packet transport software. Once the UPnP or RAAT packets are “unwrapped” and re-assembled into PCM or DSD, the dCS Streaming board puts them out as identical I²S bitstreams to the rest of the system.
+google. . . …
I heard from dCS through my dealer; they are aware of the problem and will fix it. No timeline provided so a bit more patience is required.
In the meantime I will rework my playlists to contain a single sampling rate so no switching is required. unfortunately, my favorite song at the moment, Pneuma by Tool, is available on Quboz only as a hi-rez recording, so I’m gonna have to find the shortest hi-rez recording in the world to precede it in that playlist. Or, I could buy a Vivaldi transport and a CD of the album…
As an aside, I had a guitar lesson today (more like a jam session along with the stereo) and neither my instructor or I could get the correct rhythm going on KWS Lay it on Down since we lost the first 3 beats of the song. Sorta like having a drummer in a band who is always a couple beats behind…you’d go find a new drummer…
Here is hoping I don’t have to harp on this again.
Mitch
Mitch, maybe I’m missing something here 
This thread is specifically about the problem associated with missing initial bits when playing a DSD track immediately following a PCM one (ie. format change specifically from PCM → DSD). There was never an issue with just PCM tracks playing one after another regardless of the bit rates, or even after a DSD track.
But you seem to suggest having problems playing PCM tracks off Qobuz. That’s news to me. Can you describe the problem exactly? (Maybe this deserves its own thread actually).
Anupc,
Poorly worded summary on my point:
If I play a “normal” sample track (either from my NAS or Quboz), followed by a DSD or hi-rez track, the first second of the second track is truncated. BillK took the time to figure out it is actually the first .84 seconds.
The converse is true as well; if I play a DSD or hi-rez track, followed by a normally sampled track, the first .84 seconds of the second song is truncated.
This is a portion of the response I got from DCS Ltd:
“…this is a systemic problem. It is not a setup issue that is causing the beginning of DSD tracks not to be played. As you’ll see from the Forum thread, it does also impact Rossini, but the extra steps in the chain with Vivaldi (the Upsampler’s output rate for example) has added more complexity in getting the issue nailed down”.
I got another response from DCS America that also stated the DAC goes into mute mode while the Upsampler is switching sampling frequencies and as a result the beginning of a track that requires a new sampling rate is truncated. Once the Upsampler adjusts to a new frequency and sends the signal to the DAC, the DAC comes out of mute, and the first part of the track is lost. At least that is how I understand it, but I might be in error.
If I build a play list that is exclusively one sampling rate (all normal, or all hi rez or all DSD) there is no issue, unless the track played immediately prior is a different sampling frequency, in which case the first song of the playlist at the new frequency is truncated. All else that follows is fine.
I don’t know if you are able or willing, but build a 2 track playlist in Mosaic (bypass Roon). First song normal sampling and second DSD or hi-rez. You’ll hear the switching occurring in between tracks and lose the first part of the second track.
Is this the end of the world? Obviously not. I’m reworking all my playlists to be exclusively one sampling rate and that seems to avoid the issue. There are many songs however that are only available as hi-rez on Quboz, so I have to compromise and have those only on a “hi-rez playlist”.
BillK has a lot more experience with data transfer than I so he can chime in and correct anything above I’ve mis-stated.
Cheers,
Mitch
Anpuc,
I did start a new thread on my specific issue about a year ago:
I hate to beat a dead horse, but here’s a hypothetical:
Suppose you had around $80k to spend on the best digital streaming front end you could buy. A dealer sets up two rigs: a Vivaldi stack, and a Boulder 2120 with an Aurender W20SE. You have two songs you want to hear on both to assess which one is most resolving and faithful to the original recording. One of them is a normally sampled track, and one is available on Quboz only as hi-rez. You listen to the Boulder/Aurender combo and it is awesome. Next up ya listen to the Vivaldi set up. It sounds appreciably better, but the second song is truncated. The dealer explains this is a systemic fault due to switching of sampling frequencies (the Boulder/Aurender does not have this issue). He’s not sure when/if dCS is ever going to fix this. Which system do you buy?
dCS made a commitment to me to get this fixed, and I’m known for my patience so I’ll wait. But then again, I woulda been a doctor but I did not have the patients.
Cheers,
Mitch
That pretty much covers it.
There are two issues here as mentioned before:
-
The DAC takes time to determine the data it is receiving is DoP DSD and switch to DSD mode from PCM
-
The dCS units discard all data received when switching into DSD mode, which causes the loss of the first 0.84 second of the DSD track.
What makes this complicated is the unit makes different assumptions about whether the datastream will remain in the same format under different circumstances.
I mentioned earlier in this thread, when playing DSD data:
Via USB:
Played continuously starting at Track 1:
Track 1: First part cut off
Track 2: Plays normally
Jump to Track 2:
First part cut off
Jump back to the start of Track 2:
Plays normally
Via Network:
Played continuously starting at Track 1:
Track 1: First part cut off
Track 2: First part cut off
Jump to Track 2:
First part cut off
Jump back to the start of Track 2:
First part cut off
This is because:
-
When playing via USB, the DAC remains in DSD mode when playing to the next track, so there’s no PCM/DSD switch. When jumping to the next track it does reset to PCM mode, causing the drop, but when jumping back to the beginning of the same track, it stays in DSD mode.
-
Via network the dCS hardware closes the connection at the end of the track and resets to expecting PCM when the datastream stops as it awaits a new connection for the next track - whether it’s due to a track ending due to completion of the track or jumping to the next track or to the beginning of the current track.
There is something else occurring as well, as the DoP spec says a DAC should take no more than about 180 µsec to determine a data stream is DoP rather than PCM, so it apparently also takes the dCS hardware around .75 second or so to switch to DSD mode and unmute.
Now the conditions above only occur when DSD data is being pushed to the DAC; if the tracks are instead pulled from the DAC using Mosaic, the issue does not occur because Mosaic knows it is requesting a DSD format track and changes DAC modes ahead of time.
It’s not just a bug fix; it’s a design choice that needs to be made as well, as if the DAC buffers the data then properly starts DSD playback from the buffer after changing modes it will be forever .87 seconds behind, though that’s not really an issue as it’s not much different than PCM playback when the PCM buffer is enabled. It can’t cause lipsync issues as the buffer option does as no video sources emit DoP except perhaps when using a device like the GeerFab D. BOB to provide DoP from an SACD/BD combi player to a dCS DAC.
I have to admit that I haven’t tried to see whether there is a DSD to PCM delay, but it may be different on the Rossini due to the lack of a separate upsampler as is present in the Vivaldi stack or Vivaldi One.
Were I dCS I would probably make the buffer option control both PCM and DoP buffering, if they wanted an option to shut it off.
As before I still wonder if they can fix it, as the buffer memory used for PCM may be wired directly to the PCM data bus and the DSD decoding engine may not be able to “see” it. Or, the decision to just drop DoP data on the floor when switching the DAC to DSD mode may have just been a poor technical decision or one made because they needed to get the code released before they could solve the issue.
Regardless, as it’s been eighteen months since a fix was promised to me, as stated I have to wonder if one is actually coming.
I do not expect one to come with the APEX mod, as it was promised that the fix would come in the next software release, and the Rossini APEX is labeled “2.0,” which is the same software revision as the current Rossini, so I expect the new hardware to be installed but the software will remain the same, pending the “future” update.
Bill,
Thanks, I learn something new from every one of your posts…I’m an engineer, but a civil one (so I’ve been told). You wanna know anything about moving dirt - I’m your man…
Cheers,
Mitch
Thanks; as you may be able to tell I am an engineer as well but I move bits and sometimes electrical charges for a living. 
Careful with that stuff. I took Intro to Electrical Engineering twice in my college days and it wasn’t because I liked it so much the first time…
1 Like
Yes, I’m familiar with the problem as far as DSD is concerned. However, Mitch says he’s got problems even when it’s completely PCM, but with varying bit rates;
That, I’ve never seen, and cannot duplicate any such problem on my Vivaldi stack.
I probably mis-stated my problem…happens when I play a DSD or hi-rez from Quboz after a normal track and vice versa…
I’m at the limit of my knowledge on sampling rates/DSD etc. time for some learning I guess. This is a good place to get that, lots of Einstein s out there…
I think the confusion is that DSD and “Hi Res” are two different things; the latter is simply a high resolution PCM file so there shouldn’t be an issue playing multiple PCM files back to back, just with a switch to/from DSD.
If you are experiencing a mute when switching between two PCM files, that’s a new variation of the bug I haven’t seen.
Bill,
I’m away from home for the weekend and will double check this on Monday - have an awesome weekend!
Cheers,
Mitch
1 Like
Sadly, 48 hours later, I think that is our answer.
Were you ever able to check for muting going between two different sample rate PCM files?
Bill,
I just listened to Roxette Wish I Could Fly (16 bit 44.1 hz) followed by Tool Pneuma (24 bit 96 hz) and I lost the first fractional second of the second song.
Both came from Quboz.
If ya have the time, setup a 2 song playlist with these and let me know if it happens to you too…
I’ll admit up front I don’t fully understand sampling and bit rates so I might be headed down the wrong path on this test…
Agreed, no response after 48 hrs is not encouraging. Sorta reminds me of my experience a couple years ago when I was trying to get help with the Puccini UClock compatibility with newer Apple OS. Lots of electrons spent here back then trying to get some attention and help.
Nearly $80k invested in dCS’s flagship equipment and can’t get an answer to a simple question:
“You guys gonna fix this?”
Slainte,
Mitch
Bill,
Just sent an email directly to dCS requesting information and help at this address: [email protected].
Drop ‘em a line and we’ll see who gets a response first, if any at all…
Slainte,
Mitch
Here is the email I just sent them.
Rather than give the entire background and litany of issues I’m having with a Vivaldi DAC/Clock/Upsampler truncating the first second of a song that requires a sampling rate different from the song that preceded it on a playlist, the attached threads, on your community website, that have gone unanswered are provided.
The first thread was started by user BillK who is having the same issue with his Rossini gear. That thread was started in Jul 2020. dCS has yet to respond.
And, here is a thread I started myself to specifically ask for assistance with the problem I’m having with my Vivaldi gear. That thread was started in Feb 2021. dCS has yet to respond.
I did get a response recently thru my dealer that dCS is aware of the issue and will figure out a fix. It’s been 2 weeks since that commitment was made, and I’d like to ensure that there is a fix possible, and for dCS to tell me when it’ll be available.
I’ve posted lots of info on both the above threads, and included some analogies and comparisons that should underscore the importance of fixing this. I’ll leave with just couple of them:
Suppose you had around $80k to spend on the best digital streaming front end you could buy. A dealer sets up two rigs: a Vivaldi stack, and a Boulder 2120 with an Aurender W20SE. You have two songs you want to hear on both to assess which one is most resolving and faithful to the original recording. One of them is a normally sampled track, and one is available on Quboz only as hi-rez. You listen to the Boulder/Aurender combo and it is awesome. Next up ya listen to the Vivaldi set up. It sounds appreciably better, but the second song is truncated. The dealer explains this is a systemic fault due to switching of sampling frequencies (the Boulder/Aurender does not have this issue). He’s not sure when/if dCS is ever going to fix this. Which system do you buy?
———————
I really enjoy having friends over and showing off my system. The song I like to play for them to really experience the system’s clarity, and power is Pneuma by Tool, available on Quboz only as a hi-rez recording. So, before they come over I have to have the system running, and before they arrive, make sure the last song played is also in hi-rez, otherwise I have to explain why the first second of Pneuma is missing.
“Mighty fine system ya got there Mitch, too bad we missed the beginning of the song…”
———————
I’ve been on the audio merry go round for around the last 30 years, grabbing for the brass ring every time I passed it. I’ve spent way more than I can remember on audio components. About 2 years ago, as I was about to retire I decided to get off the merry go round, bite the bullet, and just buy the best of everything I wanted/needed to enjoy my system and focus on other things.
One of those things is playing guitar. One of my goals was to build a system in front of which I can sit, and play along as if the artists are in the room with me. The system I have allows me to do that, other than the front end of songs being truncated. If you were at a concert, and the mixing board had this kinda glitch you’d probably not be too happy and might even leave.
———————
A final analogy. Suppose you bought your dream car, a Mercedes SL63. It was the best car you ever owned, other than in second gear between 2,000 and 2,500 rpm the engine stalled. And then suppose this was a systemic problem and others were having the same issue. Do ya think MB would just shrug it off, or tell their customers a third party might be able to tinker with the software and make it better?
The favor of a response is sincerely requested.
Dave Mitchell
Yorktown, VA
I confess this situation has me flummoxed. Not for technical reasons, but for point of pride. I don’t experience this issue, whether because I use Roon or due to sheer dumb luck. But that doesn’t matter. Others do, and dCS have acknowledged the issue, have promised to fix it, but have failed to fix it. As David notes above, how can someone with a world class system be expected to be an amnassafor for that system when ir demonstrably fails to perform the most elemental of modern playback tasks?