SACD to FLAC Volume Conversion Issues and Fixes

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Yeah, but I don't listen to music that way typically. I have streaming options that center around .flac. For years I've just been maintaining a folder of .iso files I rarely do anything with. It just doesn't fit in with my scheme, and I'm realizing it makes no sense to keep maintaining a folder of files I do nothing with.
Please explain your 'scheme'? How do you access and play your audio files?
 
I'd rather not. I know you mean well, and I appreciate your desire to jump in and help. But I rather dislike when I post a question, and the response tries to get steered towards "well, what you really want to do is..." followed by what I don't want to do.

I really don't need to defend my scheme here - I simply wanted to see if anyone else had pursued and come up with a similar type of workflow to what I'm looking to do.
 
iso2dsd is exactly that. Iso to DSD. Which is only 1 step of the ISO to flac process. iso2dsd is basically a gui that utilizes sacd_extract.exe, which is what I'm using for my iso to dsd portion of my process I'm building.

I have a somewhat working process that then flattens the folders (sacd_extract.exe seems to have this folder struction creation hard coded with messes with things, so a script was built to flatten that down to the parent folders I created for stereo and multi), then a script that uses ffmpeg and sox to scan levels and come up with a recommended gain without having to do a full .wav conversion, then a script that uses ffmpeg to convert the .dsf files to flac with applied gain.

Problem is using ffmpeg to convert .dsf track by track results in the dreaded track transition pops issue - one of many reasons for why I now have decided I despise SACD and feel the world would've been a better place if it was never unleashed on us!

So - taking a step back, my angle is now to get full album dsf files with .cue, do a full album dsf to wav conversion with gain, then a wav to flac conversion with use of the .cue to split.

I'm finding that there doesn't seem to be a way to force sacd_extract.exe to do a full album extraction, but there seems to be an option to concatenate the files into a full album file. So now I'm working on that step.

Not sure why all the tools that exist seem to try so hard to drive away from full album extraction. Foobar can do it, but only by a convoluted process to "fool" it - you uninstall the SACD add-in to foobar, drag in the SACD iso which makes a single item playlist, reinstall the SACD add-in, and now you're in a state where your .iso is in a playlist as a single item, and foobar can play it - which means you can convert it to a full album file. Problem is, it seems to only do the stereo if it's hybrid. Not sure if I ever came up with a workaround for that, I abandoned that workaround once I found foobar can get around the pops if you set it to do one file at a time.
 
The fear of audibly wasting bits if your PCM files peak at -6dB -- which all of this seems to be based on -- is mainly audio nervosa , especially if the source is from analog tape, and there's an argument that peaking a few dB down from 0dBFS is good insurance against intersample overs at playback(....which are also probably inaudible, unless there's a serious mass of them)

NB the current default output level for foo_sacd_input (v1.5.11) is +6dB. A 6dB boost. That absolutely WILL result in 'visible' overloads in the PCM waveform, from some SACDs*.

To maximize the peak level without overload, I check the 'Log Overloads' option in foo_input_sacd and monitorthe process in the foobar Console, which shows such errors in real time (the log is also saved as a text file if you don't feel like monitoring). Start at the default (+6) and check for overloads. If they appear, try +5. Rinse and repeat until no overloads. Then rip the disc at that setting (checking the log file for overloads just in case).

Tedious at first , when I had to convert my few dozen existing SACDs, but now my rate of buying converting SACDs is like once or twice a year, so not a big deal.

(*MJ's Thriller has a few overloads even when ripped +0dB ...but they are inaudible. They are errors printed into the SACD.)
 
replaygain keeps saying the levels are in the range I want already, but all conversions give me .flac files at -5. Sure, I can manually boost, but I need to automate. Yet to find a way to get a reliable read and setting set. It seems the only way is to convert, check each file, manually set - this is way too many steps and way too time consuming. We can land a man on the moon, but we can't read an accurate level of an audio file and set a gain automatically to rip a decent flac?

To maximize the peak level without overload, I check the 'Log Overloads' option in foo_input_sacd and monitorthe process in the foobar Console, which shows such errors in real time (the log is also saved as a text file if you don't feel like monitoring). Start at the default (+6) and check for overloads. If they appear, try +5. Rinse and repeat until no overloads. Then rip the disc at that setting (checking the log file for overloads just in case).

This is literally what I do when I convert my SACDs to FLAC using foobar (manually boost, check levels, re-convert as needed). It is time consuming but I am obsessive compulsive and try to maintain the 0.9+ peak per album, so it works for me.

(I also understand that it's audio nervosa--a term I've never heard before but makes sense--but I just like doing it anyway)
 
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