Converting SACD ISO to FLAC files

Tracks needing to be split at sector boundaries is a red book thing for CD audio. The SACD DSD layer is specified in the scarlet book. The scarlet layer is a different structure than the red book layer. Different concept of sectors and everything. The music ripped from the DSD layer isn't going to be constrained by ancient CD sector boundaries.

 

$7.50 includes stereo and surround layers. A bargain.

 

awwwww there goes the profit margin

 

Ok, now i understand. Thank you very much for your help! Just out of curiosity though: why the "audio data is not cut on a sector boundary" message then? Much obliged.

 

Redbook Audio CDs contain a spirally written data streams (files), sort of like an LP, except that these go from the inside out. CDs are not divided into sectors like HDs and more modern media such as DVDs (I believe) are.

I'm sure that if the Redbook standard were to have been designed today that they would have used sectors. But back then they were only thinking of CDs as being an audio format. It wasn't until a few years later that computer data started populating non-audio format CDs.

 

Thank you very much, but just so i understand: the message "audio data is not cut on a sector boundary" that i received from Traders' Little Helper was from downsampled files that i extracted from a single layer SACD ISO. I'm just curious about why i received that message. It's because Traders' Little Helper doesn't understand that a SACD works with a scarlet book layer and not with a red book one, so it checks the files by red book standards, resulting in this sector boundary error message? Cheers.

 

Not all people have the same needs, or the same solutions. So I tried to come up with a way to keep it all easily-accessible, easy to locate, and hopefully the various stages of the procedure separate, so people wouldn't get too confused with too much coming at them at one time. I have five different threads, titles worded the same, where you can contribute your experiences and help others. All you need do is do a search of "ripping workflow".

Take a look at this one, which is I'm afraid, one of the lesser-contributed ones:

Your Ripping Workflow #3 - multiple formats

As I said, I tried to organize them in five threads: Hard-drive or NAS structure strategy; issues about tagging; ripping for more than one format; dealing with multiple versions of the same recordings, tracks or songs; and a catch-all for stray questions.

For that matter, maybe I should start a Thread #6, just for surround or hi-res ripping. But these are the ones I want to help people keeping their plans focused without too much distraction for now.

It's my hope keeping the procedures separate, will keep people from distracting too much with good info that's nevertheless irrelevant to the task at hand, which should be placed in another of the threads. Learning minds need organization.

The "Vinyl workflow" thread is huge, and all over the place, redundant, conflicting, and redundant. I took my cue from that.

Support these threads, and help somebody else down the line!

 

The wikipedia on CD audio has a section that describes the frames and timecode and sector format of the CD: Compact Disc Digital Audio - Wikipedia

The gist is that the track boundaries have to happen at the sector boundaries. The sectors are 2,352 bytes of audio data. If you are burning files to an audio CD-R the files need to be multiples of 2,352 bytes in size so they end up on sector boundaries. Files that aren't multiples of 2,352 bytes will get padded with silence to make them the right size.

This is primarily an issue with tracks that are meant to have a gapless transition. Like for live recordings that Trader's Little Helper is designed to analyze. If the individual files aren't multiples of 2,352 bytes then you'll end up with some short brief silence between tracks that were supposed to have a gapless transition.

The warning from Trader's Little Helper is only an issue if you plan to burn the tracks back to an audio CD-R and the music has gapless track transitions.

The tracks you rip from the Red Book layer of the SACD will end up on sector boundaries just as the files ripped from a regular CD always will be.
The tracks you rip from the DSD layer and convert to high-res FLAC and then convert again down to 16/44.1 for CD will not end up on sector boundaries because they were never aligned to the Red Book sector boundaries to begin with.

 

It could be considered a bargain if one doesn't have the ability to rip SACDs. I have an Oppo 103 and 105 that both have the capability to rip SACDs. With the very generous help of an SH member I was able to figure out how to setup the programs for SACD ripping. Now I can sit down and rip 20-30 SACDs in a hour or so.

I have a good sized SACD library so to pay $5-$7.50 to have them ripped would prove to be very expensive. There are a number of players out there including those from Oppo that have SACD ripping capability. To me it would be worthwhile to check out the link below that lists those players and detailed instructions on how to rip SACDs. It would save those with a good sized SACD library quite a bit of money.

SACD Ripping using an Oppo or Pioneer? Yes, it's true!

 

hush now I almost have my website done!!
$4/$6.50!

 

I was using ISO2DSD to convert my SACD ISO files to Sony DSF files and than converting them to Hi-Res FLAC files using dBpoweramp. But if i wish to convert those same SACD ISO files directly to Hi-Res FLAC files, can i just use foobar2000 with no problems and good results? Can i just select the SACD ISO tracks on foobar2000, choose "convert" and that's it? Cheers.

 

Yep!

 

Thanks. Every SACD has 24 bit and 88,2 KHz?

 

Not quite. The SACD layers are usually encoded in DSD64. If you have an .ISO you can extract that to DSD files in .DSF format (not compatible with all hardware/software) or just convert to 24/88 FLAC. 24/88 is a common way to convert, but that's not what's on the actual disc.

 

I think you also need the SOX resampler component for foobar. Not mandatory but better than the built in one.

 

Can any SACD have less than 24/88 then, like 24/44.1? Those "Mercury Living Presence" releases for example. Cheers.

 

Those old MLP albums were originally recorded on analog tape I assume. They would've had to be transferred to digital before an SACD could be made. And if any editing or mastering work was needed, it probably took place in in PCM, since DSD has always been fairly limited in terms of what can be done in that format. So basically your SACD could, in theory, be upsampled/converted PCM. The only pure DSD recordings with no PCM conversion are usually classical releases where there are only 1 or 2 instruments and everything is recorded "live".

 

I was wondering if there were cases of SACDs created by mastering in analogue chain and recording directly to DSD?

 

I'm not saying it's impossible but these would be very few in number. It's very difficult to do anything with DSD files in a DAW program. Stuff is commonly converted back and forth from PCM to DSD. The Native DSD site often has notes about this, though they play games and use the terminology "DXD" which is simply high bitrate PCM.

In general the problems with editing, playback, copy protection, etc. are the reason why DSD failed to gain the wide sort of long-term adoption that industry types thought it would. It's basically on life support now.

 

What he meant - I guess - is that DSD and PCM are different formats and work in a different way. Therefore there's no "PCM 88/24" in DSD; you need to convert it to have that.
As for PCM -> DSD conversion, I think digital mastering nowadays is frequently done at 192/32bit or 96.
Sometimes 88 was preferred to 96 because you can downsample for CD at 44 without roundings. 96 was chosen to downsample to 48.
Even in case of PCM to DSD conversion, there wouldn't be any sense in using a source that is less than 88.
But then again, you have digital shops selling "HI REZ" files which are upsampled from CD, so who am I to tell?

 

Is this the SOX plugin you're talking about?

Resampler-V DSP plugin for Foobar

 

I use this one:

A new resampler DSP for foobar2000

 

Geez, I thought there was only one. How naive of me

The one I use is called foo_dsp_resampler. I can't find the source webpage now.
Anyway, if it has SOX in it, anyone will do. They do the same thing in the end.

 

Sorry about this next question, but i'm not very familiar with this subject: when converting a 3-channel stereo SACD to DSF through ISO2DSD, should i just choose the "multi" channel mode option and the software will automatically recognize that the multichannel SACD has only 3 channels and not 5 as usual? Cheers.

PS 1: And why did the end DSF multichannel files got so much bigger than the original SACD ISO? The ISO is like 3GB while the DSF is 5GB!

PS 2: on another subject, i've been using dBpoweramp for some time now to downsample a lot of Hi-Res audio files (i keep the hi-res originals in external HDs). In the DSP effects, i put the resampling first in order, followed by the bit depth setter with triangular dithering. My question: what's wrong with dBpoweramp for resampling? Why choose SoX over it?

 

These questions are beyond my expertise, I'm sorry.

All I can say is that I never used dBpoweramp because I'm lazy and I had enough software installed already. I did a research and came upon a website that compared different resamplers and the distortion/artifacts they introduced. Among my options, SOX was the winner so that's what I chose. Quite surprisingly, the built-in resampling tool in Audacity fared very well too, but Foobar+Sox combo is faster to use on multiple files.

As for the size, I have no idea. Maybe some file space is used to encode the 2 ghost channels you mention. Just a lot of information to say "there's nothing here". Just guessing.
Once you mixdown to PCM stereo you should come out with a normal sized end file.