SACD "ripping" over HDMI?

Stereodude

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Has anyone tried ripping SACDs using a PC based HDMI capture device and a HDMI "splitter"? My Oppo DVD player and Sony Blu-ray player can both output SACD over HDMI. I'm not sure if the Oppo can do DSD or just PCM output, but I believe the Sony does both. I realize it's not very elegant, limited to real time, and the HDMI capture device is expensive, but it should work... I can't really find any reports of people doing it though.
 

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I ripped my SACDs by buying a legacy Playstation 3 that had never been updated. It worked just fine. I understand that it's technically possible to de-flash a PS3 to the earlier firmware now, where it wasn't back when I did my discs, but I haven't really had a reason to keep up.

I sold the PS3 in question maybe nine months ago. It was taking up space and it was worth about four times what I paid for it.

As I understand it, a lot of cheap splitter devices will strip copy protection off the bitstream in normal operation, so just capturing PCM is at least viable.
 

Stereodude

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Actually as I read it, it looks like most of the non-professional HDMI capture devices will only grab stereo audio.
Which ones? The Black Magic Designs DeckLink Mini Recorder supposedly does 8 channel, but seems like it might be limited to 48kHz.

The Magewell Pro Capture HDMI looks to be more capable (and more expensive). I found this thread to be enlightening: http://shmups.system11.org/viewtopic.php?f=6&t=55592 It seems Magewell includes an executable that records HDMI audio to a file as part of their SDK. It's not clear if it can record DSD or not, though I'd want it as PCM ultimately anyhow.
 

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I was mostly looking at game capture systems.

If you can find one of the first-gen PS3s, one of the ones that supports PS2 and PS1 discs, there's apparently a fairly simple tool that simulates a bad firmware flash that causes the unit to default to its original shipping firmware. Once you're in that, you can find the instructions for straight-up ripping the discs. That might be a better overall plan than finding some voodoo of possibly compatible capture hardware.
 

Stereodude

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That might be a better overall plan than finding some voodoo of possibly compatible capture hardware.
But where's the fun in that? Plus being able to cap 1080p via HDMI might be a handy capability to have on reserve. Besides, it shouldn't really be voodoo, just a matter of the card supporting high sample rate LPCM.

Edit: I sent Magewell an e-mail to clarify the HDMI audio capabilities, specifically high sample rates 176.4kHz/192kHz and bitstreamed formats. I didn't specifically ask about DSD though.
 
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Stereodude

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So they replied...
Magewell said:
Pro Capture HDMI support LPCM 176.4KHz and 192 KHz for full 8 channels.
For AC3, DTS, E-AC3, Dolby TrueHD, DTS-HD, etc, you can capture those formats by SDK.
 

Stereodude

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I decided to get a Magewell card. Not just for SACD audio capture over HDMI, but also for HDMI video capture. It should be fun to play with.
 

Stereodude

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The Magewell card is supposed to arrive today. It should be interesting to play around with.

If it works for SACD capture, I can use SoX to throw away all the extraneous bits and sampling rate that's utterly wasted. I'm thinking 44.1kHz / 18-bit, maybe 20-bit in a 24-bit container should do the job since 176.4kHz / 24-bit is large file / bit wasting lunacy.
 

Stereodude

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That might be a better overall plan than finding some voodoo of possibly compatible capture hardware.
Maybe I should have listened to you... The Magewell card only enumerates itself as a Stereo recording device to Windows (at least per Audacity and the Windows Sound section). The application included with their SDK see's 5.1 LPCM audio from SACD from my Oppo (5.1/88.2kHz output), but has some issue with the file it records. There are audible glitches in the recording that aren't there when listening to the same 5.1 LPCM stream also split to my pre-pro. I will try my Sony BD player and see what it does with DSD or 176.4kHz audio.

FWIW, the 2 channel audio recording from Audacity sounds fine with no glitches.

Edit: It looks like both the 32-bit and 64-bit version of MultiAudioCapture.exe they include with the SDK have problems with sampling rates over 44.1kHz or 48kHz. It doesn't like 5.1 24/88.2kHz, 5.1 24/96kHz, or 5.1 16/176.4kHz audio.
 
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Mercutio

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Video capture was never 100% hassle free for me, no matter how hard I tried. I could get 98% perfect, but that last 2% just would not happen, which is why I gave up on vidcap. Now you're dealing with protected content streams and hardware that only has a barebone software suite and doesn't really have any competitors and isn't positioned for professional use and I can definitely see that same sort of issues happening. They'll probably tell you to try different cables or to cease all other activity on your 32-thread PC and 800MB/sec disk array because THAT's what's causing the problem. :thumbright:
 

Stereodude

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Video capture was never 100% hassle free for me, no matter how hard I tried. I could get 98% perfect, but that last 2% just would not happen, which is why I gave up on vidcap. Now you're dealing with protected content streams and hardware that only has a barebone software suite and doesn't really have any competitors and isn't positioned for professional use and I can definitely see that same sort of issues happening. They'll probably tell you to try different cables or to cease all other activity on your 32-thread PC and 800MB/sec disk array because THAT's what's causing the problem. :thumbright:
The splitter I bought takes care of the protected stream aspect just fine. I think it's a problem with their sample application. They did provide the source code for it (totally uncommented), but I don't really want to get into it that far. I'm not a developer and my experience is dabbling in embedded C, not writing a C++ application on Windows. The video side seems okay from the quick test I did. I'll play with that more in the future.

I will probably send them a friendly note, though I don't have high hopes.
 

Stereodude

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So this is how the audio device appears to Windows. A stereo device, but with an 8 channel balance control.

magewell_hdmi_audio.png
 

Stereodude

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So I sent Magewell a link to a zip file with 95MB of audio samples containing 15-20 second recordings of:

2.0-16-176kHz.wav
2.0-16-44.1kHz.wav
2.0-24-192kHz.wav
2.0-24-44.1kHz.wav
2.0-24-48kHz.wav
2.0-24-88.2kHz.wav
2.0-24-96kHz.wav
5.1-16-176.4kHz.wav
5.1-24-48kHz.wav
5.1-24-88.2kHz.wav
5.1-24-96kHz.wav

They confirmed a bug in the SDK and have sent me a link to a new version that's supposedly fixed. I will test it later today after work.

I'm not sure if I'm better off starting out with 16-bit/176.4kHz audio or 24-bit/88.2kHz audio. I realize there's technically more information in 16-bit/176.4kHz, but more of it is likely useless IMHO. Since I don't plan to keep raw PCM capture, I'd tend to think starting with 24-bit/88.2kHz is the better option given that I plan to convert it to 20-bit/44.1kHz. I guess I can try it both ways and see if the output appreciably differs.
 

Stereodude

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I downloaded and tried their updated SDK. It got rid of the major glitching. It looks like there's still some sort of static pop / glitch that occurs in the audio every 4-5 minutes, but I'm not 100% sure about that. It happened at about the same amount of time from the start of the recordings with both SACD players. However it was at a different spot in the track so I know it's not on the disc. I'm guessing it's a slight clock sync issue. The capture card is likely deriving it's own clock from the PC, not synchronizing to clock from the HDMI receiver, but I need to do a little more troubleshooting here. I need to split the audio off to my receiver and listen and see if the static / pop is audio through it. It was audible listening to the live mix on the PC through headphones, so it's not just in the recording.

Then, with the first disc I was messing around with I found the CD layer did not match the SACD mix. I had hoped to use a .cue sheet from the CD layer to cut up the recorded SACD content after lining up the start of each since the SACD recording will just be one big .wav file. I lined up the start of the first track to the ms, but by the end they were out of sync with each other by over 1 second. I need to do more testing since the CD layer was ripped, not recorded over HDMI so I can't conclusively rule out the HDMI interface as the cause of the track length mismatch, but it seems extremely unlikely.

I was trying to see whether the Oppo 24/88.2 or Sony 16/176 would be the better source. I converted down to 20/44.1kHz (I think from memory) using the same command line in SoX for both. Then I discovered the Oppo and the Sony don't output SACD at the same volume over HDMI. It seemed to be about 3dB different. I tried to normalize them to the same level, and then convert to .FLAC but they didn't quite match. The final file sourced from the Sony gave me a larger FLAC file. I don't know if that means it has more detail in the file or if bits are being wasted on artifacts or what. The amount of noise in SACD (DSD) -> PCM conversion above 20kHz is disheartening. You can see the frequency response roll off similar to a CD, and then there's this huge blob of high frequency noise inherent to SACD. Obviously there's a lot more of it from the Sony courtesy of it's 176.4kHz sampling rate vs the Oppo's 88.2kHz, but SoX uses a low pass filter when resampling so all that should get rolled off and averaged out courtesy of Nyquist in either case so I don't think it should cause the file size differences, but I could be wrong.

Anyhow, these various things sort of doused my enthusiasm for the project since it looks like I would have to make two recordings slightly offset from each other (like a 5 second delay) and use one to kill the pops / glitches from the other (assuming they occur on a regular periodic basis) which will be tedious to do. Then, I'd have to manually chop up the files (a pain). Added to that I'm still not sure which player I should use as the source.
 

Mercutio

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That definitely sounds like too damned much work. I was hoping you had hit on something that might work universally for new recordings and formats but it doesn't sound like that's the case.
 

Stereodude

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I came upon this thread some months ago. I finally got around to buying a cheap'ish used compatible blu-ray player (BDP-S5100) from eBay for ~$35 shipped. I plan to rip all my SACDs with it and convert them to PCM and compress them to FLAC. I haven't tried it yet.
 

Stereodude

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:cool: So it works. The first SACD ripped both the stereo and multichannel layer fine. Working on a second one now.
 

Stereodude

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Ripping to a .ISO is much smaller than .DSF files (by ~2.5x). Since foobar will play the .ISO files and I'm not trying to play the tracks on anything else I don't see any reason to do anything other than .ISO.
 

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That's pretty nice; I like being able to archive my physical media so if I had SACDs I'd probably look into this more. I never got into SACDs so I don't have any in my collection to warrant getting into it though.
 

Stereodude

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I ripped all 47 of my SACDs to .ISOs. 152GB for all of them. I hadn't bought one in probably close to a decade. I probably listen to less than one a year. Now that they're files I can play on a computer that might change.
 
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