zonka

I probably should have described my setup in case that is causing the problem. I have one instance of JRiver on my main computer in another room running as the server. I have another running as a client via ethernet at my stereo setup connected to my DAC (I'm using a mini pc with the i7 processor). This is where I'm having the trouble. So I'm wondering if there are any specific settings I may need on my server side version, feeding my client? Thanks:) I gave tried all the DSD options with the same message.

Auto-correct got me - obviously, I meant native dsd:) Anyone else have some tips or settings I should try? Thanks

Trying native dad. I forget the exact message, but the generic error that the chosen output isnt possible and to choose another. Thanks for taking the time to reply:)

I may be missing some settings but I keep getting the error that the file can't be played back with the current settings. I know my DAC can handle up and through DSD 512 (Gustard X22). I am using an i7 Broadwell cpu (16 gb ram) so I figured it had enough power for the conversion. I have JRiver set to Bitstream "Yes" but no PCM to DSD playback options work. I am using XMOS ASIO player that should support it (as mentioned prior). Are there other settings I need to adjust? Thanks for any help someone might be able to offer.

Unfortunately, I think the Nuc is the limiting factor. When I go into control panel to choose a sound device, both usb and optical are missing 88.2 in the supported formats. It has to be a driver issue - I would think it could handle 88.2, I just need the right driver.

I've got an interesting problem that I can't figure out for the life of me. I just got a Parasound P5 preamp. I want to run my NUC 6CAY with Windows 10 running JRiver. By either USB or mini toslink I can't get the ability to play 88.2 or 176.4 files (all others through 192 work for toslink and the USB tops out at 96). It has to be a driver issue, I'm sure but I can't find it. I have the most current drivers but I'm wondering if there is an old driver that would work that would "unlock" those sample rates. Does anyone have any ideas or experience with this? I'm kind of new with this and have tried googling everything I can think of and haven't made any headway. This NUC also had an issue with 24 bit from HDMI (would only play 16 bit). Very long story/history, but apparently Intel stopped supporting it and didn't tell potential buyers. It turns out a 3 year old driver had the 24 bit functionality so I found that and it worked - problem is I've now upgraded to the P5 preamp an it doesn't take HDMI (only USB or optical). I'm hoping there is a similar work around. Thanks for any help you can offer.

Here's my setup: main windows 10 pc has all my music on it. It runs JRiver MC23 and I have it running as a server. I then have a Nuc, also running windows 10, connected to my receiver. The Nuc is also running MC23. I have the Nuc set up to play from the server library on the other computer and a separate setup where I just point the Nuc to my music folders on my pc. Is there a preference as to which is better? Is it better to stream from the server or play the files directly from the pc (when pointing directly to the music files)? Hope this even makes sense - I'm a networking novice. Thanks for your help.

Great reply - and thank you. This makes sense to me. The foobar DR meter came back with 3 different tracks with a peak of 0.0 with rest of tracks under the ceiling. I don't hear any distortion - I just see very slight clipping, in red, in audacity (very little). I could probably just leave it alone or drop it a few dbs and everything will be fine. Thanks again for your detailed post. Drew

Sorry for a silly question. If a flac track clips does that mean the literal file is cut off at the top or does that point in the file extend above the threshold (whatever that is)? In other words, if I lower the gain will I get the "whole" track now without clipping or are those peaks cut off for good? I don't even know if I'm phrasing this correctly - I apologize if it is hard to understand. Thanks for any light someone can shed. Drew

So here's another question. Since I play my flacs from a usb drive directly to my oppo, would I benefit from a usb jitterbug filter or is that more designed for use with playing files from a computer? Thanks

I just read that thread but i think it is saying just the opposite of my situation. My dsf files are noticeably louder than a flac pcm file made from the dsf file. My receiver recognizes the file's as dsd and pcm, respectively, and the oppo isn't adding anything as I play tracks through usb. So.....If the dl meter is telling me peak of -7db for both files then it is because the dsf file is first converted to pcm by foobar making both files identical. Therefore, if I raise the flac file by +7db I've brought the volume back up to the dsf file when played by my oppo. Am I finally starting to get this through my thick head?

Good info - so maybe +3db is safe? Is there software that can tell me exactly how much to add (that would require software that can decipher the volume, if that is even the right word, of a dsf file)? Otherwise, it seems like guesswork - which isn't a huge deal as long as it is close. I just don't want to go over the ceiling. Thanks for your insights and education of a novice.

Would I "normalize" or "apply replaygain"? If I apply it to an entire album I probably want replaygain as it maintains the volume differences between album tracks, right? I want it to be a permanent alteration of the file (I have backups). So +6db, right? Thanks

Thanks for your reply. Yes, I used dbpoweramp as well. I just don't know why the dl meter would return identical values for the two files when one is obviously not as loud (unless I don't understand what the dl meter values mean - a distinct possibility!). I'm just trying to find an exact value I need to boost the gain on the new file so it is the same as the dsf file.

I have a track in dsf format - sounds great. I converted it to flac for comparison purposes (to 24/88.2). The track also sounds nice but the volume is much reduces. When I ran them both in Foobar's Dynamic Range Meter they both showed peak volumes of -7.12. How is that possible when one track is clearly not as loud. Do I need different software to find the actual volume differences between the tracks? Sorry if this is a silly question - I'm pretty new to this stuff and I am trying to see if I hear in degradation converting dsf to flac (24/88.2) for storage purposes and to have all tracks in flac. However, I do want them to be the same loudness so I'm trying to understand how the DR Meter works and what the values are telling me. Thanks for any help you give:)