SACD to ISO with Oppo & Pioneer BD players!

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ssully,

Did you try an elevated DOS command prompt?

If your files are in a folder off the C: try this:

C:\sacd_extract>sacd_extract -i 192.168.1.121:2002 -P -I

Where you put in your players IP instead of mine. I just did the Wired SACD and it worked fine!
 
notice I edited my post earlier....what you think??

I've read that DSD is typically 6dB quieter than PCM to prevent overloading the delta-sigma modulator. I've also read that the adjustment is not consistent from disk to disk or file to file and I've certainly found that to be true. You will probably be safe using +3dB, but the only way to get optimal results is to convert at 0dB and then measure the peaks for a second conversion.
 
SOLVED: I tried one more time with a different PC, connecting the BDP directly to the PC via ethernet (cat6). And when I opened the 'Network' window in the PC, there was nothing there, but a prompt appeared, telling me that device/file sharing was off by default, do I want to turn it on? Aha! Following the prompts, I turned it on, and voila, the BDP appeared as a networked media device, with IP 162.254.27.102 . Opened the BDP tray with the USB stick, loaded SACD, started iso2dvd_gui, specified the new IP, hit execute and it quickly started ripping the ISO.

In fact this works with the laptop PC too, because it, too, had device/file sharing off by default.

Couldn't be easier.;>
 
Use of 10.x.x.x IP addresses is pretty common - most home routers only use 192.168.x.x, but that limits you to 65k IP addresses. Not a big deal on a home network, but it matters on an enterprise private network. More explanation: https://en.wikipedia.org/wiki/Private_network
Thanks. Since my experience has been mostly with home networks, my statement reflects that experience.
 
SOLVED: I tried one more time with a different PC, connecting the BDP directly to the PC via ethernet (cat6). And when I opened the 'Network' window in the PC, there was nothing there, but a prompt appeared, telling me that device/file sharing was off by default, do I want to turn it on? Aha! Following the prompts, I turned it on, and voila, the BDP appeared as a networked media device, with IP 162.254.27.102 . Opened the BDP tray with the USB stick, loaded SACD, started iso2dvd_gui, specified the new IP, hit execute and it quickly started ripping the ISO.

In fact this works with the laptop PC too, because it, too, had device/file sharing off by default.

Couldn't be easier.;>

That's awesome. It's very possible that when I first set mine up, it was the same case. But honestly, if I had to do what you ended up doing...I don't remember doing it. Glad you got it......I rip a couple of SACD each evening...gonna take me a while.
 
Like others, maybe, I'm finding variation in how conversion-ready the ripped isos are. I have a quite old Sony Classical stereo-only SACD, Stravinsky conducting Le Sacre , and the Firebird Suite, with the Columbia Symphony orchestra -- it ripped just fine to iso, and it displayed correctly as multiple tracks in foobar, but unlike other rips (stereo and multi) I could not directly convert it to .wav tracks via foobar; the attempt returned 'unsupported format' errors. I succeeded by converting the iso to DSDIFF tracks with iso2dsd and then convert those to .wav with foobar....resulting in curious 88/16 format wavs. :mad:@:

(I already have most of my SACDs as 'laserdrops' so I'm pretty interested to see how closely my rips match my drops.)
 
Like others, maybe, I'm finding variation in how conversion-ready the ripped isos are. I have a quite old Sony Classical stereo-only SACD, Stravinsky conducting Le Sacre , and the Firebird Suite, with the Columbia Symphony orchestra -- it ripped just fine to iso, and it displayed correctly as multiple tracks in foobar, but unlike other rips (stereo and multi) I could not directly convert it to .wav tracks via foobar; the attempt returned 'unsupported format' errors. I succeeded by converting the iso to DSDIFF tracks with iso2dsd and then convert those to .wav with foobar....resulting in curious 88/16 format wavs. :mad:@:

(I already have most of my SACDs as 'laserdrops' so I'm pretty interested to see how closely my rips match my drops.)

Go to the settings or configuration preferences (forget the exact terms they use) in Foobar.

Set the output format to 32 bit floating point. 88.2k is correct and preferred for the sample rate.

Import the 32/88.2 wav files created into a DAW like Audacity or Reaper.
You'll see overs! (But not actually clipped as the 32 bit floating point format retains data that goes above zero.)
Lower the gains to a point where all peaks are below zero.
Render that to 24 bit 88.2k files.

If the DAW step is too complex, there is a setting in Foobar to auto lower the gain by a preset amount.
Set this to -6db. (A safe level. I've seen up to 4 or so db over zero after the conversion math.)
Then set the output format to 24 bit.


If you are outputting at 16 bit and also with no gain compensation between formats, you are stepping on the results pretty hard and it is very audible.
 
Go to the settings or configuration preferences (forget the exact terms they use) in Foobar.

Set the output format to 32 bit floating point. 88.2k is correct and preferred for the sample rate.


It's a function of the foobar sacd plugin. You can only set the sample rate.

All other sacd isos are outputting at 24 bits, so in this case in particular it's a function of the input dsdiff files.

It's not a big deal, really.


If you are outputting at 16 bit and also with no gain compensation between formats, you are stepping on the results pretty hard and it is very audible.

Nope, it's not. Not with the one SACD that is being output at 16. It looked (as waveform) and sounded fine when I listened. But I'll double check the level of the rest of teh tracks from that conversion, I only 'looked at' one..
 
Like others, maybe, I'm finding variation in how conversion-ready the ripped isos are. I have a quite old Sony Classical stereo-only SACD, Stravinsky conducting Le Sacre , and the Firebird Suite, with the Columbia Symphony orchestra -- it ripped just fine to iso, and it displayed correctly as multiple tracks in foobar, but unlike other rips (stereo and multi) I could not directly convert it to .wav tracks via foobar; the attempt returned 'unsupported format' errors. I succeeded by converting the iso to DSDIFF tracks with iso2dsd and then convert those to .wav with foobar....resulting in curious 88/16 format wavs. :mad:@:

(I already have most of my SACDs as 'laserdrops' so I'm pretty interested to see how closely my rips match my drops.)

What happens when you convert it with jRiver conversion mode? There are no settings and it outputs at a very high samplerate. Does it render as a 24 bit pcm file? That is what I use for my initial conversion from sacd iso to pcm, then I use Audiomuxer to downsample...
 
It's a function of the foobar sacd plugin. You can only set the sample rate.

All other sacd isos are outputting at 24 bits, so in this case in particular it's a function of the input dsdiff files.

It's not a big deal, really.




Nope, it's not. Not with the one SACD that is being output at 16. It looked (as waveform) and sounded fine when I listened. But I'll double check the level of the rest of teh tracks from that conversion, I only 'looked at' one..

I'm telling you from experience.

I used default settings the first time I ever tried it (which included 16 bit). Heard the results and said nope this doesn't work at all.
Noticed the 16 bit. Noticed the output difference business. Did what I said above.
Now I say the short answer is it's a HD to HD virtually lossless conversion.


When you load a file into Foobar, you can right-click (control-click) on the file - select 'convert' and then '...'
You'll see an output format tab to click on in the window that opens.
It's a clunky Windows app. You have to poke around for the controls. :)

PS. Being the DSD format, the SACD wouldn't know what "16 bit" even means. You're converting the different language of DSD to PCM. You need to use full quality PCM (and further, 32 bit fp) to capture the complete translation. Then frame it properly at 24 bit fixed.
 
I'm telling you from experience.

I used default settings the first time I ever tried it (which included 16 bit). Heard the results and said nope this doesn't work at all.
Noticed the 16 bit. Noticed the output difference business. Did what I said above.
Now I say the short answer is it's a HD to HD virtually lossless conversion.


I checked the 16bit waveform. There were three instances of digital clipping in the whole album. They are so short and far apart that they would not have been noticeable by ear.

But converting the Stravinsky iso again, still using foobar, this time it worked as 88/24. I don't really know why -- I was surprised. But such are the mysteries.. it may be due to the DSDIFF plugin I installed last time (seems unlikely), or to the actual location of the iso and .cue files output from iso2dsd -- I noticed when I moved the iso into a new folder, it no longer converted. Anyway, those clipping I saw before is gone because there's so much more headroom now, but there is actually still an instance , the loudest peak in the whole album, where it looks as if the original digital file is slightly overloaded, either because of how it the ADC was done, or something done during mastering. The peak is slightly 'flattened' even though it's well below digital full scale. Again, it would not likely be noticeable in normal listening.


When you load a file into Foobar, you can right-click (control-click) on the file - select 'convert' and then '...'
You'll see an output format tab to click on in the window that opens.


It's a clunky Windows app. You have to poke around for the controls. :)

I'm familiar with Foobar controls, thanks, it's been my player for many years now. If you're interested in the particulars: the standard foobar (v1.3.10) WAV output conversion bit depth parameter is set to 'auto' by default, which typically converts to 24 bit for a DSD source (as it did in all other cases except the Stravinsky disc). The SACD plugin has its own parameter settings for each mode of output (PCM, DSD, DSD+PCM). For PCM modes the settable parameters are sample rate (44100 is default, 88200 is my choice, though it's overkill), volume (i.e. 'gain compensation', +0 to +6dB increase, I vary this according to +0 peak level), and mode (Multistage (32fp) default is my choice, Multistage (64fp), Direct (32fp, 30 kHz lowpass), Direct (64fp, 30 kHz lowpass), Installable FIR (32p), Installable FIR (64fp)).

Anyway the issue is moot; as I wrote above, while 24 bit it output didn't work before for this disc, it does now, without me having changed the WAV conversion or SACD plugin output parameters at all.
 
Of all the Oppo and Pioneer models that are capable of doing SACD rips, which one is the cheapest? I'd be using it solely as a ripper, not a playback machine so it doesn't need any features in particular, and doesn't need to be a new model as I'm happy to buy used if necessary.

Already gone through two 'fat' PS3's that have the 'yellow light of death' and would rather put a bit of money in to a new(er) machine rather than try and have a 10 year old PS3 repaired.
 
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