Extracting music from multi-channel discs

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baward

Well-known Member
Joined
Feb 18, 2012
Messages
109
Hi,

I use a Mac, and I want to rip the multi-channel content from my DVD-As and SACDs (Blurays and DVDs I can do more easily) so that I can play them from my NAS. What is the way to do this? Hopefully, there will be some Mac users here who can point me in the right direction. Thanks!
 
There is a dedicated thread somewhere, about how to rip the SACD's. (which, I'm sure someone here will post a link) I wouldn't think the fact that you have a MAC has any relevance in the equation. Also, if you can already rip Blurays & DVD's, it seems to me you should also use the same method to rip the DVD-Audios.
 
Last edited:
MrSmithers said:
Here it is, this excellent guide(s) will have everything you need. Basically for DVD-A/V use DVD Audio extractor. Blu Ray's MakeMKV first then DVDAE and for SACD you need a comparable player and the relevant software listed in the thread. All should be available for a Mac as well... :)

https://www.quadraphonicquad.com/forums/threads/converting-mch-discs-101-overview.22384/https://www.quadraphonicquad.com/forums/threads/sony-blu-ray-players-used-for-sacd-ripping.26078/
Click to expand...

Ah, yes. I guess I'm probably an outlier. I use DVD-Audio Extractor for DVD, DVD-Audio and Blu-ray, in conjunction with PassKey. :) SACD, I use my Oppo and Sonore software.
 
baward said:
Hi,

I use a Mac, and I want to rip the multi-channel content from my DVD-As and SACDs (Blurays and DVDs I can do more easily) so that I can play them from my NAS. What is the way to do this? Hopefully, there will be some Mac users here who can point me in the right direction. Thanks!
Click to expand...
You still need one of the stand alone disc machines to rip the SACD iso image. Even with a Mac. But you have full access to the required software and without any of the Windows workarounds.

MakeMKV - bluray images to .mkv
DVD Audio Extractor - DVDA to flac, DVDV to .dts (Don't use to convert dts to flac! Old codecs.)
ffmpeg - dts to flac, dts2496 to flac fully decoded (And a TON of other conversions, muxing, etc!)
XLD - Convert between all lossless formats. Includes the 32:1 decimation transcode for DSD to PCM
Sonar iso2dsd - convert dsd iso to dsd files (Then use XLD to convert to PCM)

If you do the dsd to pcm in two steps, you can get a virtually lossless transcode.
iso image to dsd files
dsd to 32 bit floating point 88.2k wav with 32:1 decimation in XLD (88.2k is the target sample rate)
Normalize the album as a whole to -0.01dbfs.
Render to 24 bit fixed and flac

88.2k is the target sample rate for the 32:1 decimation. This is critical to use and keep here! You can convert the final 24 bit renders to lower sample rate if you wish with minimal or no discernible damage as expected at the end.

Reason: The dsd to pcm transcode can result in peaks in the pcm file anywhere from -6db to +6db.
For the -6db case, you'll only lose one bit of precision (ie make a 23 bit file) and it will just be a little quieter. But for the other case, you'll clip those peaks if you go straight to 24 bit fixed! The floating point format preserves over zero data. You need a DAW app to work with audio files. (The normalize and render to 24 bit parts.) Audacity is free and covers simple basics. Reaper is the flagship DAW nowadays. It's actually affordable but it's also very comprehensive and might have a learning curve depending on your experience. Some people like Logic. I wouldn't recommend Protools anymore even to my worst enemy.
 
Thanks for all the fantastic responses, it gives me plenty to get my teeth into. With the more challenging older discs, I have the fallback of using my Oppo 203 and the separate channel inputs on my audio interface, and a multitrack DAW. But that all needs to be done in real-time, so I will avoid it if I can.
 
baward said:
Thanks for all the fantastic responses, it gives me plenty to get my teeth into. With the more challenging older discs, I have the fallback of using my Oppo 203 and the separate channel inputs on my audio interface, and a multitrack DAW. But that all needs to be done in real-time, so I will avoid it if I can.
Click to expand...
The oppo 103/203 has SACD ripping capability.
 
jimfisheye said:
You still need one of the stand alone disc machines to rip the SACD iso image. Even with a Mac. But you have full access to the required software and without any of the Windows workarounds.

MakeMKV - bluray images to .mkv
DVD Audio Extractor - DVDA to flac, DVDV to .dts (Don't use to convert dts to flac! Old codecs.)
ffmpeg - dts to flac, dts2496 to flac fully decoded (And a TON of other conversions, muxing, etc!)
XLD - Convert between all lossless formats. Includes the 32:1 decimation transcode for DSD to PCM
Sonar iso2dsd - convert dsd iso to dsd files (Then use XLD to convert to PCM)
Click to expand...

dts (or ac3) to 'flac' can mean two things

dts-->decode to PCM(wav)-->flac (lossless compression + tagging)
dts--> put undecoded data in wav wrapper--> flac (this doesn't shrink the file size, it simply allows tagging)

The second option is doable using 'S/PDIF to WAV' in free Audiomuxer software. It's useful if you would rather keep the file as a bitstream, and let a downstream AVR decode it, while still allowing metadata (tags) to be added to it for display in player software.

Also foobar2000, with the right plugins, can do several of the things in the list up there, with a graphical interface, like converting between lossless formats, and DSD ISO to either DSD or PCM files (or both) , see below.




Reason: The dsd to pcm transcode can result in peaks in the pcm file anywhere from -6db to +6db.
For the -6db case, you'll only lose one bit of precision (ie make a 23 bit file) and it will just be a little quieter. But for the other case, you'll clip those peaks if you go straight to 24 bit fixed! The floating point format preserves over zero data. You need a DAW app to work with audio files. (The normalize and render to 24 bit parts.) Audacity is free and covers simple basics. Reaper is the flagship DAW nowadays. It's actually affordable but it's also very comprehensive and might have a learning curve depending on your experience. Some people like Logic. I wouldn't recommend Protools anymore even to my worst enemy.
Click to expand...

With foobar (plus DSD plugin) you can perform and monitor conversion of SACD ISO files to PCM. Below is the conversion setting window. The default 'PCM Volume' for conversion in 0dB (i.e., no change in level). This typically results in a peak level that is 'quieter' than the corresponding CD, by as much as -6dB. You can up the level in +1dB increments up to +6dB. As you see you can also adjust the LFE separately (this is an interesting option because SACD LFE mastering can be inconsistent from disc to disc, and the way an AVR treats PCM LFE is not necessarily the way it treats DSD LFE...it is a complex subject.) . Check 'Log overloads' to detect clipping during the conversion.
image.png


For example, suppose you start at the maximum amplification, +6 PCM Volume, on the hunch that a 0dB PCM Volume will peak at -6dB. Using foobar's View-->Console window to monitor the process you can see in real time (or afterwards) if there has been any clipping. If you see clipping in the Console log, reduce the PCM Volume of the conversion process by 1dB, and proceed again. Repeat as necessary until no clipping errors appear. (No need to check in Audacity, etc).

The DSD--PCM conversion is at 24bits . This is not selectable, though the sample rate is (44, 88,176, 352). The ISO can also be ripped as pure DSD (via the Output Mode selector).

'Preferable Area' btw lets you rip either the Stereo, or the MCH layer alone. Or both , with the default 'None' setting.
 
Doing this does seem to be a whole not more difficult on a Mac. I am sorely tempted to buy a Windows laptop and an external drive and do it with that. I am correct in assuming that a Windows machine would be easier? My family could make use of a laptop as well, which makes it all the more sensible.
 
Re DVD vs SACD LFE, in a nutshell:

DVD (and DVDA) LFE *expects* a +10dB boost before analog output to the subwoofer.
SACD LFE (*if* mastered according to SACD spec) does NOT. It 'expects' to be output 'as is'.

If your system doesn't 'know' this, you get wrong SACD LFE output level.

And if your system *does* know this, but the SACD LFE is not mastered to SACD spec..i.e., if it is mastered like a DVD -- again the LFE output level is wrong.

Handling this all in software (software converters, software players) adds another level of either confusion or opportunity to get it right, depending on your outlook.

Happy trails. ;)
 
ssully said:
dts (or ac3) to 'flac' can mean two things

dts-->decode to PCM(wav)-->flac (lossless compression + tagging)
dts--> put undecoded data in wav wrapper--> flac (this doesn't shrink the file size, it simply allows tagging)

The second option is doable using 'S/PDIF to WAV' in free Audiomuxer software. It's useful if you would rather keep the file as a bitstream, and let a downstream AVR decode it, while still allowing metadata (tags) to be added to it for display in player software.

Also foobar2000, with the right plugins, can do several of the things in the list up there, with a graphical interface, like converting between lossless formats, and DSD ISO to either DSD or PCM files (or both) , see below.






With foobar (plus DSD plugin) you can perform and monitor conversion of SACD ISO files to PCM. Below is the conversion setting window. The default 'PCM Volume' for conversion in 0dB (i.e., no change in level). This typically results in a peak level that is 'quieter' than the corresponding CD, by as much as -6dB. You can up the level in +1dB increments up to +6dB. As you see you can also adjust the LFE separately (this is an interesting option because SACD LFE mastering can be inconsistent from disc to disc, and the way an AVR treats PCM LFE is not necessarily the way it treats DSD LFE...it is a complex subject.) . Check 'Log overloads' to detect clipping during the conversion.
View attachment 58959

For example, suppose you start at the maximum amplification, +6 PCM Volume, on the hunch that a 0dB PCM Volume will peak at -6dB. Using foobar's View-->Console window to monitor the process you can see in real time (or afterwards) if there has been any clipping. If you see clipping in the Console log, reduce the PCM Volume of the conversion process by 1dB, and proceed again. Repeat as necessary until no clipping errors appear. (No need to check in Audacity, etc).

The DSD--PCM conversion is at 24bits . This is not selectable, though the sample rate is (44, 88,176, 352). The ISO can also be ripped as pure DSD (via the Output Mode selector).

'Preferable Area' btw lets you rip either the Stereo, or the MCH layer alone. Or both , with the default 'None' setting.
Click to expand...
Good stuff there @ssully - anyone know what the differences are for the "DSD2PCM Mode" selections?
And what is the "DoP for Converter" check box?

FOOBAR2K PREF..jpg
 
The are just different sampling filters. Basically DSD (megaherz sample rates) has to be downsampled for PCM. That requires filters to limit bandwidth. The likelihood that you will hear any difference between them is minimal. Though audiophiles will of course claim audible differences are 'bigly'.

DoP (DSD over PCM) is explained here


Might as well just post the user 'manual' (readme.txt) for the plugin (Super Audio CD Decoder):

Code: 
Install foo_input_sacd.fb2k-component file and restart foobar. Then open *.ISO image file for playback.
    When needed adjust output volume and samplerate at File->Preferences->Tools->SACD.
    To use editable tags check it at File->Preferences->Tools->SACD.
    To playback SACD-R/RW discs create the new playlist, insert disc into DVD drive and drag-n-drop
    DVD drive letter (root folder) on the created playlist. Or, if UDF file system exists on SACD-R/RW,
    open MASTER1.TOC file.

    If your DAC supports DoP through ASIO/WASAPI/DS driver you can set up DSD playback:
    1. Open File->Preferences->Tools->SACD page.
    2. Select "DSD" for "Output Mode".
    3. Open File->Preferences->Playback->Output page.
    4. Select "DSD : <Driver Type> : <Device Name>".
    5. If your DAC doesn't support multichannel playback use "Downmix channels to stereo" DSP.
    6. If your DAC doesn't support DSD source samplerate use DSP Processor plugin.
    If playback is in DSD mode you should get samplerates 2822400, 5644800, etc. and silence on VU Meter.
    In DSD+PCM mode output is DSD and all bells and whistles show up as expected.
 
    For PCM playback it is possible to use custom FIR filters. Some filter samples are put in filters subfolder.

    For WavPack playback in DSD mode move "Super Audio CD Decoder" upper than "foobar2000 WavPack Decoder" on
    File->Preferences->Playback->Decoding page.

    To produce DoPed PCMs check "DoP for Converter".
 
Last edited:
jimfisheye said:
. . . DVD Audio Extractor - DVDA to flac, DVDV to .dts (Don't use to convert dts to flac! Old codecs.) . . .
Click to expand...
I'm not sure what this means. When I use DVD AE to extract a DTS stream from a DVDV disc, I output the files as FLAC. Are you saying I shouldn't be doing that?
 
paligap said:
I'm not sure what this means. When I use DVD AE to extract a DTS stream from a DVDV disc, I output the files as FLAC. Are you saying I shouldn't be doing that?
Click to expand...
I think it's OK if the dts files are 24/48. If it's dts2496 though, DVD_AE only sees the core. So for those, export to dts and use ffmpeg to decode that to flac.
 
paligap said:
I'm not sure what this means. When I use DVD AE to extract a DTS stream from a DVDV disc, I output the files as FLAC. Are you saying I shouldn't be doing that?
Click to expand...
Its mostly a preference thing. FLAC is pretty much universal and can be played on nearly any setup. Leaving it as DTS requires a DTS decoder or plug-in somewhere in the chain, but as long as that is available, it works well. The conversion of DTS to FLAC takes a lossy codec (DTS) and converts it to a lossless codec (FLAC), which seems counter-intuitive to many. I convert everything to FLAC unless it contains video, in which case, I leave the audio stream as DTS. You can fully tag FLAC tracks, not so with DTS.
 
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