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  1. #1
    Junior Member EuroChamp's Avatar
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    Best way to convert DSD to PCM

    Hi all,

    now I have ripped some SACDs and I am wondering, what whould be the best way to convert them to PCM? Which program do you use and which settings? and why?

    Bernhard

    P.S.: And no, I will not buy a DSD Dac at the moment ;-)

  2. #2
    Masters Level Member ted_b's Avatar
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    My guide (SACD Ripping) walks you through the settings I recommend in Korg Audiogate. The sample rate is up to you, based on the sweetspot of your DAC, etc. but basically when I used to convert DSD I used soft rolloff, Aqua dithering and I also used "normalize songlist" so Audiogate would make sure not to clip certain hot DSD titles.

    There is some code out there for Linux fans to use (DSD2PCM I think it's called). However, many folks simply let the player do it on the fly (thereby not having to permanently store additional PCM files). Players like A+, PM, JRiver and HQplayer all use very good DSD-to-PCM computations to do the conversion. This, of course, is a perfect solution if you someday plan to do a DSD-capable DAC.
    "We're all bozos on this bus"....F.T.

    My JRiver screencast tutorials : http://www.computeraudiophile.com/f1...tml#post272960
    My DSD database: https://docs.google.com/spreadsheets...ySk/edit#gid=0
    My SACD Ripping Guide (needs updating but still works): https://dl.dropboxusercontent.com/u/...r%20v4.0.1.pdf

    US Technical Advisor, NativeDSD.com: www.nativedsd.com

  3. #3
    I keep my SACD.iso files but also convert to uncompressed flac files using foobar2000 with the SACD plug-in.

  4. #4
    Sophomore Member gsquared's Avatar
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    mlknez, have you compared foobar2000 with the SACD plugin to Korg Audiogate? I've using Audiogate and the results are great, but would like to compare it to foobar2000.

    I'm not sure if I have the settings right. In the foobar2000/SACD plugin settings -> preferences, should I set the PCM Volume to +6dB or leave it at default 0? The first test I tried with PCM volume set to 0 was not right, but it could have been another setting I had wrong.

    Gary
    Mac Mini 2.3 GHz Intel Core i7, 16GB RAM -> HQPlayer, OS X El Capitan booted from SD Card -> Sonore microRendu -> LH Labs Pulse X Infinity -> Nord One UP Monoblocks -> Spendor LS3/5as | Controlled via Roon from sonicTransporter (Power Conditioning: Audience adeptResponse aR12). Twitter: @hirezaudio

  5. #5
    Junior Member EuroChamp's Avatar
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    Quote Originally Posted by ted_b View Post
    My guide (SACD Ripping) walks you through the settings I recommend in Korg Audiogate. The sample rate is up to you, based on the sweetspot of your DAC, etc. but basically when I used to convert DSD I used soft rolloff, Aqua dithering and I also used "normalize songlist" so Audiogate would make sure not to clip certain hot DSD titles.

    There is some code out there for Linux fans to use (DSD2PCM I think it's called). However, many folks simply let the player do it on the fly (thereby not having to permanently store additional PCM files). Players like A+, PM, JRiver and HQplayer all use very good DSD-to-PCM computations to do the conversion. This, of course, is a perfect solution if you someday plan to do a DSD-capable DAC.
    Dear Ted,
    thx for your answer. I have read your guide and saw your recommencation. But wanted some other opinions. At the moment I have foobar2000 + sacd-plugin.
    JRMC is not an option for me, my Atom330 has not enough power for realtime conversation.
    Did you (or anybody else) ever compare these two? Or is there a third option? - not the expensive saracon.

    Bernhard

  6. #6
    Junior Member EuroChamp's Avatar
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    Quote Originally Posted by gsquared View Post
    mlknez, have you compared foobar2000 with the SACD plugin to Korg Audiogate? I've using Audiogate and the results are great, but would like to compare it to foobar2000.

    I'm not sure if I have the settings right. In the foobar2000/SACD plugin settings -> preferences, should I set the PCM Volume to +6dB or leave it at default 0? The first test I tried with PCM volume set to 0 was not right, but it could have been another setting I had wrong.

    Gary
    Hi Gary,

    I think, you should set the gain as high as possible, but the keep in mind, the final PCM signal should never reach the 0dB limit. You can check it easy with the DR plugin in foobar - after dsd to pcm conversation. I try to loose not more than 1 dezibel.

    Bernhard

  7. #7
    Sophomore Member gsquared's Avatar
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    Quote Originally Posted by EuroChamp View Post
    Hi Gary,

    I think, you should set the gain as high as possible, but the keep in mind, the final PCM signal should never reach the 0dB limit. You can check it easy with the DR plugin in foobar - after dsd to pcm conversation. I try to loose not more than 1 dezibel.

    Bernhard
    Thanks Bernhard!
    Mac Mini 2.3 GHz Intel Core i7, 16GB RAM -> HQPlayer, OS X El Capitan booted from SD Card -> Sonore microRendu -> LH Labs Pulse X Infinity -> Nord One UP Monoblocks -> Spendor LS3/5as | Controlled via Roon from sonicTransporter (Power Conditioning: Audience adeptResponse aR12). Twitter: @hirezaudio

  8. #8
    Senior Member elcorso's Avatar
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    Quote Originally Posted by EuroChamp View Post
    Hi Gary,

    I think, you should set the gain as high as possible, but the keep in mind, the final PCM signal should never reach the 0dB limit. You can check it easy with the DR plugin in foobar - after dsd to pcm conversation. I try to loose not more than 1 dezibel.

    Bernhard
    From some (very good) recording engineer I learned: The 0db limit on PCM recordings it's not so important as in DSD recordings. Of course this don't mean to get clipping all the time in PCM.

    Roch

  9. #9
    Hey Guys well i perform with Sonoma DSD.I find that all the dithering settings in Weiss Saracon have a damaging impact when transforming DSD to PCM except which is truly amazing.Thanks!!

  10. #10
    Masters Level Member ted_b's Avatar
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    Quote Originally Posted by Lomew Bartho View Post
    Hey Guys well i perform with Sonoma DSD.I find that all the dithering settings in Weiss Saracon have a damaging impact when transforming DSD to PCM except which is truly amazing.Thanks!!
    Except what?
    "We're all bozos on this bus"....F.T.

    My JRiver screencast tutorials : http://www.computeraudiophile.com/f1...tml#post272960
    My DSD database: https://docs.google.com/spreadsheets...ySk/edit#gid=0
    My SACD Ripping Guide (needs updating but still works): https://dl.dropboxusercontent.com/u/...r%20v4.0.1.pdf

    US Technical Advisor, NativeDSD.com: www.nativedsd.com

  11. #11
    I don't really see the point in converting to PCM unless you need them for portable playback.

    Converting to PCM is a potentially lossy process, so you are best to use a player that can handle realtime DSD to PCM playback without altering the source files.

    JRiver Media Center seems to have some good options for this, though I used a custom 30kHz low-pass filter with a 48dB/octave slope instead of the 24dB/octave slope preset, as this filters out all the ultrasonic noise.

    Now that I have a DSD-capable DAC, I just bitstream the DSD files to it. (this is why I keep the source)

  12. #12
    Masters Level Member ted_b's Avatar
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    Skeptic,
    This thread is for the majority of CA folks, those without a DSD-capable DAC. Of course converting to PCM for no reason...is not recommended.
    "We're all bozos on this bus"....F.T.

    My JRiver screencast tutorials : http://www.computeraudiophile.com/f1...tml#post272960
    My DSD database: https://docs.google.com/spreadsheets...ySk/edit#gid=0
    My SACD Ripping Guide (needs updating but still works): https://dl.dropboxusercontent.com/u/...r%20v4.0.1.pdf

    US Technical Advisor, NativeDSD.com: www.nativedsd.com

  13. #13
    Quote Originally Posted by ted_b View Post
    Skeptic,
    This thread is for the majority of CA folks, those without a DSD-capable DAC. Of course converting to PCM for no reason...is not recommended.
    Up until a few weeks ago, I was without a DSD-capable DAC too - but that's why I recommend leaving it as native DSD and having the player convert to PCM in realtime, rather than destructively converting the file to PCM.

    Then if you ever do buy a device that supports native DSD, you can take advantage of it.

    That said, I do think the recent DSD hype is being overblown, and I wouldn't buy new hardware just to get native DSD support.

  14. #14
    Masters Level Member ted_b's Avatar
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    In both cases it is destructive; just one does not require you to store redundant PCM and DSD libraries. No one is recommending deleting the original DSD files.
    "We're all bozos on this bus"....F.T.

    My JRiver screencast tutorials : http://www.computeraudiophile.com/f1...tml#post272960
    My DSD database: https://docs.google.com/spreadsheets...ySk/edit#gid=0
    My SACD Ripping Guide (needs updating but still works): https://dl.dropboxusercontent.com/u/...r%20v4.0.1.pdf

    US Technical Advisor, NativeDSD.com: www.nativedsd.com

  15. #15
    Quote Originally Posted by gsquared View Post
    I'm not sure if I have the settings right. In the foobar2000/SACD plugin settings -> preferences, should I set the PCM Volume to +6dB or leave it at default 0? The first test I tried with PCM volume set to 0 was not right, but it could have been another setting I had wrong.
    I set the gain to the default 0 dB, load the ripped .iso, then run the DRM plug-in on it (it takes a while), then use the resulting peak numbers to set the gain accordingly before running the conversion. Note that you have to clear out the .iso and reload it for the new setting to take effect.

  16. #16
    I've come across a case where there is some weird clipping going on, apparently with DSD->PCM conversion. The track in question is track 2 of the new Audio Fidelity America - America release... Sandman. The left waveform is and SACD rip converted to PCM. It looks the same with AudioGate or the foobar plugin. The right waveform, is a rip of the CD layer.

    As one can see the peak is clipped off on the DSD->PCM conversion. I tried various gain settings in AudioGate with no change. Anyone have any theories as to what is going on here?

    I recorded the analog output of both the SACD and CD from an Oppo BDP-93 and didn't see any differences there so it appears to be something going on in DSD->PCM conversion.
    Attached Thumbnails Attached Thumbnails screen-shot-2013-10-21-12.31.08-pm.png  

  17. #17
    Masters Level Member ted_b's Avatar
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    You tried Audiogate's "normalize" function (for playlist)..which takes the average DSD peaks and reduces it enough to remove clipping, such as in this case? DSD can be hot, yes.
    "We're all bozos on this bus"....F.T.

    My JRiver screencast tutorials : http://www.computeraudiophile.com/f1...tml#post272960
    My DSD database: https://docs.google.com/spreadsheets...ySk/edit#gid=0
    My SACD Ripping Guide (needs updating but still works): https://dl.dropboxusercontent.com/u/...r%20v4.0.1.pdf

    US Technical Advisor, NativeDSD.com: www.nativedsd.com

  18. #18
    Normalize made the whole track louder, but the clip is still there...

    screen-shot-2013-10-21-2.00.24-pm.png

  19. #19
    Is there any software out there that will render the original DSD as a waveform?

  20. #20
    There's some disagreement about DSD vs PCM levels. 50% modulation is the traditional peak level for DSD, but the SACD specification allows exceeding this. It's also possible the DSD encoding baked the clip in (range handling to avoid overflow of the noise shaper).

    JRiver supports up to 100% modulation from a DSD file before it would clip in the integer domain. And since JRiver uses all 64-bit floating point math, you can attenuate to avoid clips by using R128 Volume Leveling (including dealing with intersample overs).

    You might try JRiver and use the 'Analyze Audio' tool to determine the peak level.
    Matt Ashland, JRiver

  21. #21
    Here is JRiver's analysis. If anything it is mastered at a pretty low level, right? I used JRiver to convert to AIFF and the waveform has the same clipped peak.

    screen-shot-2013-10-21-2.54.55-pm.png

  22. #22
    Quote Originally Posted by cheezmo View Post
    Here is JRiver's analysis. If anything it is mastered at a pretty low level, right? I used JRiver to convert to AIFF and the waveform has the same clipped peak.
    Click image for larger version. 

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    I wouldn't expect any software or DSD hardware to clip until the R128 True Peak exceeded -6dB for DSD (for JRiver it would be 0dB, but some hardware or software assumes -6dB is the max).

    So with a R128 True Peak of -8.7dB on the track, I think the most likely answer is that the flat-line is baked into the DSD signal.
    Matt Ashland, JRiver

  23. #23
    Masters Level Member
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    Quote Originally Posted by Matt Ashland View Post
    There's some disagreement about DSD vs PCM levels. 50% modulation is the traditional peak level for DSD, but the SACD specification allows exceeding this.
    SACD spec allows max +3.1 dB level and the max is quite clearly defined, so it is kind of easy to determine.
    Signalyst - http://www.signalyst.com
    Developer of HQPlayer

  24. #24
    Quote Originally Posted by Miska View Post
    SACD spec allows max +3.1 dB level and the max is quite clearly defined, so it is kind of easy to determine.
    It's not me you need to convince :P

    Some well known DAC chips clip on the analog output after 50% modulation when playing DSD.

    And some DSD files in the wild reach almost 100% modulation.
    Matt Ashland, JRiver

  25. #25
    Masters Level Member
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    Quote Originally Posted by Matt Ashland View Post
    Some well known DAC chips clip on the analog output after 50% modulation when playing DSD.
    I would guess these are the ones that do digital processing for DSD... Some DAC chips also clip on inter-sample overs too...

    And some DSD files in the wild reach almost 100% modulation.
    Those shouldn't be coming from SACD masters at least, because SACD pressing plants should refuse masters exceeding the +3.1 dB level. For example HQPlayer keeps watch and prevents output exceeding the spec level. Maybe some modulators are not keeping eye on the output.

    Not that it would be problem for HQPlayer's PCM conversion either, because there's practically no max sample value for the processing pipeline.
    Signalyst - http://www.signalyst.com
    Developer of HQPlayer

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