Converting DSD Files to FLAC

[LuvMyQuad]

Then what is all the fuss about? Many of you seem hell bent on proving me wrong!

I'm not trying to prove you wrong. I believe you probably hear what you say you hear. I'm trying to find out why what you hear is so different from what nearly everybody else hears.

For me it's about sound quality first, compatibility and convenience second. As for room correction time alignment ect. it's not that I'm against it just that I think it to be mostly unnecessary. I subscribe to the less is more theory as do a lot of the other "audiophile types" those that many of you seem to like to ridicule!

That is why I talk of a schism between the audiophile types (I referred to them purists before) mostly the stereo only crowd and those that promote digital DSP based systems and surround sound utilising a never ending number of channels. I would rather use four large matched speakers than a room full of dicky little surrounds and mismatched front and centre speakers! Then try to patch things up with room correction and time alignment!

For many, including me, room correction and "all that" is the easiest and surest way to better sound.

I think I said this already but to repeat, the difference made by room correction surpasses every other audio aspect save for mastering, including formats, conversions, DACs, amps, etc, etc. If those differences even exist, they are miles down on the subtlety scale compare to the sound the listening environment imparts on playback.

No one ridicules the "purist" for eschewing room correction. In my experience, most purists readily admit its importance, often in even stronger terms than I just stated. The difference is, the purist would tend to use a physical means to achieve the goal rather than place another component in the reproduction chain. I don't have that option. I cant place bass traps all over nor cover the walls with absorptive panels Nor do I have the knowledge and time necessary to experiment endlessly with that stuff. Digital EQ is the next best thing.

To say the need tor EQ is due to "a room full of dicky little surrounds and mismatched front and centre speakers! Then try to patch things up with room correction and time alignment!". Is way off base. It shows you are ignorant to what is possible with modern equipment.
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[GOS]

 

[par4ken]

Yesterday I did a few conversions using Foobar with the installable filters. I got very good results using the short filter and 176.4 Khz. The conversions have a sound character very similar to tracks downloaded from HDTracks. I haven't yet compared them to native DSD. I would suggest that if you must convert give these filters a try, they work in other applications besides Foobar as well. The filter is in the form of a text file, I guess that it is actually a script.

http://s-audio.systems/dsd-filter/?lang=en

 

[LuvMyQuad]

Yesterday I did a few conversions using Foobar with the installable filters. I got very good results using the short filter and 176.4 Khz. The conversions have a sound character very similar to tracks downloaded from HDTracks. I haven't yet compared them to native DSD. I would suggest that if you must convert give these filters a try, they work in other applications besides Foobar as well. The filter is in the form of a text file, I guess that it is actually a script.

I'll give it a try, at 88.2kHz. 176.4 kHz, that just seems like overkill.

What is the HDTracks sound character? Ideally, they should just sound like the master tape they came from. Are you saying HDtracks have a character all their own because they are HD Tracks?

What you need to compare is the disc based SACD track and the converted PCM file of the same track with matched levels.

 

[par4ken]

I'll give it a try, at 88.2kHz. 176.4 kHz, that just seems like overkill.

What is the HDTracks sound character? Ideally, they should just sound like the master tape they came from. Are you saying HDtracks have a character all their own because they are HD Tracks?

What you need to compare is the disc based SACD track and the converted PCM file of the same track with matched levels.

Yes try both. The reason I opted for 176.4 is that it permits the use of the "Small Filter" which affects the audible signal less. It's really only overkill if you are stuck for storage space. I would of tried even higher bit rates but my equipement can't handle it. I would agree that those higher rates are likely to do nothing more for the sound.

I always find that HDTracks are a bit bass heavy, I get the same type of sound from many of my pre-recorded reel tapes as well. That could be due to vinyl having a bit less bass, so other sources sound bassier. CD's at least the earlier ones sound very much like the vinyl. The Sony Japan SACDs which some complain about thin or weak bass sound just perfect to me.

 

[marcb]

Yes try both. The reason I opted for 176.4 is that it permits the use of the "Small Filter" which affects the audible signal less. It's really only overkill if you are stuck for storage space. I would of tried even higher bit rates but my equipement can't handle it. I would agree that those higher rates are likely to do nothing more for the sound.

I always find that HDTracks are a bit bass heavy, I get the same type of sound from many of my pre-recorded reel tapes as well. That could be due to vinyl having a bit less bass, so other sources sound bassier. CD's at least the earlier ones sound very much like the vinyl. The Sony Japan SACDs which some complain about thin or weak bass sound just perfect to me.

Or perhaps your DAC Is a little bottom end heavy? Vinyl cut using the same master as the digital version always sounds a bit weightier on my system - the opposite of what you describe. Many early CDs were cut from vinyl cutting masters, so that’s probably why you hear a similarity.

 

[par4ken]

Or perhaps your DAC Is a little bottom end heavy? Vinyl cut using the same master as the digital version always sounds a bit weightier on my system - the opposite of what you describe. Many early CDs were cut from vinyl cutting masters, so that’s probably why you hear a similarity.

It wouldn't be the DAC, vinyl rips sound the same as the vinyl does in person, ruling that out. Some phono pre-amps may add a bit more bass than mine does and my moving coil is a bit bright for sure. But still not a lot of difference between vinyl and CD (unless the CD is brickwalled). In many cases I actually prefer the sound of the vinyl to HDTracks. Don't get me wrong there are some very good sounding releases from HDTracks, but unfortunately many now seem to be brickwalled.

I guess that I should pull out one of my test records to check for deviation from flat response. I could then apply a bit of bass boost to my vinyl if required.

 

[jimfisheye]

HDTracks has some truly butchered brick wall masters. To the point where a 128k mp3 copy of an unmolested copy of the mix would run circles around them! They are not to be trusted. Vet everything. They DO have some of the real deal too sometimes so not to just be dismissed.

Rule of thumb, if you hear something gross it's probably the source. Garbage in, garbage out. DAC shootouts... Unless it's some grifter **** from Worst Purchase or Amazon (and thus more or less malfunctioning out of the box), DAC shootouts should be in the realm of treated rooms and the handful of beyond excellent recordings for test subjects. And frustratingly within perception bias range!

My vinyl system is kind of scavenger. (This gear is normally expensive!) So, not purpose built with careful choices. Cartridge is a Benz Micro MC-3, phono preamp is in a Mark Levinson #28 preamp. The bottom octave is a little heavy to my ear. I've read other comments like that about this cartridge too.

 

[paligap]

I was finally able to use my Oppo 103D to rip an SACD to my hard drive. I might save the DSF files, but I would still like to convert them to FLAC files for reasons not worth discussing. I used Foobar 2000 to do that, but then I noticed they were output at 44.1. Then I read through this thread and learned about changing some settings in the decoder. I changed the PCM sample rate, but what else do I need to change?

 

[par4ken]

Scroll back to my previous post. I would download and install one of the filters in that link. Please note that the "filter" is just a text file.

I would set the volume to +6dB otherwise most conversions will be a bit low in level. Alternatively you could normalize the level afterward.

My Lfe is set to 0, I'm not sure what's optimal I don't use the lfe channel anyway.

 

[ssully]

If you set the PCM volume to +6 you run the risk of digital overload, because not all SACDs are mastered at -6dB peak compared to PCM. Normalizing can't repair an overload. Better to leave at +0 and normalize *that* afterwards. Or monitor the process and find the right setting for the PCM volume adjustment that maximizes peak without exceeding digital full scale, which is what I do. Someone posted a script that does this automatically IIRC.

 

[dadregga]

(deleted)

 

[mkt]

+1 to all the info above. As I understand it, because DSD -> PCM (flac) is lossy, there is a difference (at least in theory idk about practice or audibility) in applying the 6 dB normalization during or after conversion.

If you set to +6 dB during conversion, there could be clipping.
If you set to +6 dB after conversion, you waste bits.

Is that right? I convert twice, once to find the right gain (in theory +6 dB) and once for real.

 

[paligap]

If you set the PCM volume to +6 you run the risk of digital overload, because not all SACDs are mastered at -6dB peak compared to PCM. Normalizing can't repair an overload. Better to leave at +0 and normalize *that* afterwards. Or monitor the process and find the right setting for the PCM volume adjustment that maximizes peak without exceeding digital full scale, which is what I do. Someone posted a script that does this automatically IIRC.

What about splitting the difference at +3? Does the type of music make a difference?

If all that happens if you leave it at zero is that you might have to turn it up a bit, maybe that's the easiest approach.

 

[himey]

I add +3db on Sacron and I haven't had any issues. If you aren't doing a ton of rips you could listen first, and if it's not very loud, add 6db. If it is louder than usual, only add 3db.

 

[ssully]

Adding +3dB should be safer since -3 and -6 dB are the most common 'recommended' DSD peaks with reference to straight conversion to PCM. Adding +6 will of course cause overs in any SACD mastered at -3dB peak.

The type of music makes no difference IME. And just listening for 'if it's not very loud' seems imprecise. You basically have to listen to the whole record, or skip to the parts you know are loud, and then guess if the loudest part will trip up the conversion or not.

Yes, leaving it as zero basically means you turn it up a bit if you want it to be analogous to PCM releases on your playlist. Or use Replaygain.

I don't use replaygain because it wrecks bit-perfectness for streaming DD and DTS files, and I'd rather just set playback level once and forget it, so I take the time to get it right during the foobar convert process using monitoring with the Console.

Using +0 and 'manually' normalizing after could work, but I like a track to be normalized to the *album's* peak, not the track's (replaygain calls this 'album gain'). This maintains the mastering choice of track-to-track peak variation. And that's a bit tedious to do manually.

 

[par4ken]

If you set the PCM volume to +6 you run the risk of digital overload, because not all SACDs are mastered at -6dB peak compared to PCM. Normalizing can't repair an overload. Better to leave at +0 and normalize *that* afterwards. Or monitor the process and find the right setting for the PCM volume adjustment that maximizes peak without exceeding digital full scale, which is what I do. Someone posted a script that does this automatically IIRC.

True of coarse, that's why they made the setting adjustable. I've yet to have a problem with using the +6 dB setting however. If clipping does occur you can simply reconvert at the lower setting. It would be useful if there was a listing of those "hot" releases.

 

[LuvMyQuad]

I don't use replaygain because it wrecks bit-perfectness for streaming DD and DTS files, and I'd rather just set playback level once and forget it, so I take the time to get it right during the foobar convert process using monitoring with the Console.

I didn't even know you could apply replay gain tags to DD and DTS files. Good to know.

 

[Cheezmo]

I don't think it has been mentioned on this thread but if you have a Mac, I highly recommend DSDMaster (in the App Store) for converting DSD files to PCM. It has a feature to automatically normalize over file groups which take the guesswork out of the whole +3/+6 gain part of the equation. It will analyze all the tracks on an album and normalize them all the same amount without any going over.

https://apps.apple.com/us/app/dsd-master/id829431988?mt=12

 

[konoyaro]

I don't think it has been mentioned on this thread but if you have a Mac, I highly recommend DSDMaster (in the App Store) for converting DSD files to PCM. It has a feature to automatically normalize over file groups which take the guesswork out of the whole +3/+6 gain part of the equation. It will analyze all the tracks on an album and normalize them all the same amount without any going over.

https://apps.apple.com/us/app/dsd-master/id829431988?mt=12

While I'm a fan and user of Bitperfect software on Mac, it's worth mentioning that the same developer has pretty much gone AWOL. Note that neither of the apps have been updated in years.

I'm fine with "if it ain't broke don't fix it" but where he used to be responsive to bugs it's now just radio silence.