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[alcarp]

Would appreciate advice on the correct volume setting in JRiver.
I notice that PCM tracks generally sound warmer and richer in my system than DSD tracks which sound a bit clinical.
I've noticed that  the volume bar is at 100% for PCM and swings to 0 for DSD tracks.
Sound is not muted.
I have my DAC volume at 100 since I'm using a preamp for controlling volume.
My JRiver settings are:
Audio Device: PS Audio USB Driver [ASIO]
Device Settings > Volume > JRiver MC 20
Tools > DSD Bitstream in DOP Format
Under Settings:
Bitstreaming: Yes [DSD]
Under Volume mode the choices are: Application Volume/Internal Volume/System Volume/Disabled volume
I have Internal Volume checked.
The 3 vertical bars at the top right light up in blue for both PCM and DSD so I guess that there is no DSP being done?
Is it normal for the volume bar to go to 0 for DSD tracks?

[blgentry]

Since you have bitstreaming set to ON for DSD files, that means JRiver doesn't touch the data when sending it to your DAC.  It just sends it with no modifications.  So, that means JRiver can't change the volume of those files.  It's bitstreaming them!  That's why the volume control shows 0.  It's disabled because of the bitstreaming.

I have no idea if what you're hearing is a consequence of the recording, your DAC, or something else.  But you *might* try turning bistreaming off.  Then set up the Output Format section in DSP Studio to convert DSD to PCM.  To do so, just set everything from 352,800 Hz and up to a rate that your DAC can handle.  For example, on my DAC, it's maximum supported PCM rate is 192,000 Hz.  So I set 352,800 and up to be converted to 192,000.  That's how I listen to DSD because my DAC doesn't support DSD.

Why would I suggest this if your DAC *does* support DSD?  Because some people say that nice some nice sounding DACs that happen to support DSD, do a better job with PCM and you get better sound quality by converting to PCM first.  I don't know if it's true, but it's probably worth testing.  It's just a few clicks of the mouse.

Good luck,

Brian.

[Spike1000]

Why would I suggest this if your DAC *does* support DSD?  Because some people say that nice some nice sounding DACs that happen to support DSD, do a better job with PCM and you get better sound quality by converting to PCM first. . . .

Damn good call Brian, I was was thinking of dabbling with a DSD capable DAC myself. There may be absolutely no need as PCM may well be more than good enough, possibly better. . . 

So 'more' may not indeed be better so I shouldn't just rush in but carefully assess the benefits rather than just assuming that 'more' is indeed 'more'. 

Spike

[ferday]

here's some fun reading...it's applicable cause it involves MC and DSD

http://archimago.blogspot.ca/2013/09/measurements-pcm-to-dsd-upsampling.html

[dtc]

The DSD standard is typically below PCM levels (usually 6 Db), so when doing  comparisons make sure the dB levels you are listening to for a given track are the same. A lot of comparisons of DSD versus PCM are dong at different volume levels, which skews the results.

[alcarp]

Good call .. this 6 db may account for the difference I'm hearing.

Re volume control I've now set the JRiver volume control to Disabled since I'm using the preamp to control voume. Now the volume bar in JRiver is at 0 for both PCM and DSD tracks.

[blgentry]

The DSD standard is typically below PCM levels (usually 6 Db),

I did 5 WHOLE minutes of research on this and I couldn't find any reference to a volume level difference.    Can you provide more details?  Link or explanation, either is fine.

Brian.

[Matt]

Just a note that you can configure MC to add +6dB to the PCM by doing Options > Audio > Configure Input Plug-in... > DSD input plug-in... > Increase volume of DSD to PCM by +6dB.

There's kind of some disagreement about whether using that is right or not.  We have it off by default because we have some DSD files that would clip with it on.  But other people on the inside say it absolutely should be on.  So take your pick!

[dtc]

I did 5 WHOLE minutes of research on this and I couldn't find any reference to a volume level difference.    Can you provide more details?  Link or explanation, either is fine.

Brian.

Sony set the SACD standard level at - 6dB to avoid clipping, as Matt said.  Therefore DSD can sound softer.The -6dB standard was adhered to by SACDs but it is hit and miss with DSD digital files. Some SACD players even had a 6dB boost built into them, if I remember correctly.  I am traveling (National Parks in Utah, currently at Escalante Grand  Staircase) with limited Internet access and do not have a quick reference, unfortunately,

[blgentry]

Sony set the SACD standard level at - 6dB to avoid clipping, as Matt said.

Thanks for the reply.  It doesn't make sense to me though.  This is digital we're talking about here.  You can measure the levels.  So, if you can set it to -6 dB for maximum level, it won't go above -6 dB.  You could just as easily set it to -3 dB, or 0 dB.

The whole concept of setting levels low to "prevent clipping" is from the analog world.  Or, more precisely, it's a practice that's used when you don't know what your peak levels might be.  So you set them lower to try to prevent overload.  With a digital file, that you recorded, and mastered, you know the highest level because it's something you can analyze.

I'm not saying either of you are wrong about -6 dB being some sort of SACD standard.  I'm saying there has to be some other explanation for why they do it.  Overload (clipping) is a non-issue for non-real-time digital audio.

Brian.

[dtc]

Thanks for the reply.  It doesn't make sense to me though.  This is digital we're talking about here.  You can measure the levels.  So, if you can set it to -6 dB for maximum level, it won't go above -6 dB.  You could just as easily set it to -3 dB, or 0 dB.

The whole concept of setting levels low to "prevent clipping" is from the analog world.  Or, more precisely, it's a practice that's used when you don't know what your peak levels might be.  So you set them lower to try to prevent overload.  With a digital file, that you recorded, and mastered, you know the highest level because it's something you can analyze.

I'm not saying either of you are wrong about -6 dB being some sort of SACD standard.  I'm saying there has to be some other explanation for why they do it.  Overload (clipping) is a non-issue for non-real-time digital audio.

Brian.

Pure DSD file are recorded to a -6dB reference level and never changed. To change the level, you need to convert to PCM. You cannot change the level in DSD format. I do not believe there is any software that does that. Certainly the main editing programs like Merging Pyramix and Weiss Saracon have to change to PCM to change the overall volume. Hence, pure DSD is recorded at a given level and left at that level. Sony and Philips set the standard of -6dB from the beginning. So, when recording to DSD or when converting from PCM to DSD, many people just use the -6dB standard.