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  1. Hi guys,

    I read a lot in the videohelp forum and found some threads, for example from Nelson,
    that try to do exactly what I try to do, but with no success.
    My setup: I have a Creative PCIe SB X-Fi with driver Version 2.40.8 .
    I now want to record the 5.1 DD Sound from a PS3 or an Apple TV, connected with an optical cable to the spdif input of the sound card.
    The only thing, that I get, is an extremly loud noise.
    I tried different recording programs like audacity, wavestudio, cdwave, vlc, etc. Always the same problem.
    I also tried to use Besplit and Besweet to fix the recorder track, but the result is empty.
    Please help me.

    Stefan
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    Capture as a 2.0 WAV file, then decode with AC3Filter, OR use the CLI application bsconvert.exe.
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  3. Thank you for your quick answers.
    But I'm desperate, it's not working.
    The original recorded wav is like a hissing sound,
    and when I use bsconvert, it sounds a little bit like R2D2.
    original wav file
    wav file after bsconvert
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    Then you are doing something very-wrong.

    The file "cdwave.wav" is detected (by MediaInfo) as a *6-channel WAV file*@4608kbps@48kHz.

    It should be detected as 2 channels, either 1411.2kbps@44.1kHz or 1536kbps@48kHz.

    Goldwave says the 'fact' chunk is missing --- which means the WAV header is borked.

    Also, bsconvert says "MPEG Audio Stereo 44100"
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  5. I managed to get a 2channel 1536kbps@48kHz file, at least Mediainfo tells me that.
    But still the same problem.
    Any ideas what I can do?

    2ch48khz1536kbps
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    The AC3 syncword is back to front and for some reason it's beginning at the 9th byte of each frame rather than the first. Your PCM file has the wrong Endianness, it should be read as Big rather than Little, and I'm wondering if maybe somehow the input is being detected as 5.1 PCM and the channels are being mixed up.


    -Edit- 16 bit 5.1, with both the channels and Endianness reversed?
    Last edited by ndjamena; 26th Oct 2014 at 20:42.
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  7. Formerly 'vaporeon800' Brad's Avatar
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    You might try Dolby's own AC3Rec program.

    Click image for larger version

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    But from my own adventures in SPDIF recording (1 & 2 for example) I am curious how you're going to overcome copy protection.
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    There are definitely AC3 frames in there in the midst of all the 0s, 1s and -1. I managed to reverse the endianness thusly:

    ffmpeg -acodec pcm_s16be -i D:\test2.wav -acodec pcm_s16le -f wav d:\test3.wav

    after using a hex editor to alter the wav header to read 5.1/16 bit, which, assuming it actually was 16 bit, should have taken me at least part of the way there. Now I just need to figure out the correct way to remap the channels. The ninth byte would be the 5th channel, so do I just reverse 1 to 5 and leave the 6th channel alone or is it more messed up than that?
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  9. Is the noise you get a continuous hiss, or is it rapidly pulsating?

    If not rapidly pulsating, then you are doing something wrong. MUST record in 24-bit. 2-channel. 16-bit recording will NOT work.

    Been a long time since I did this, but still have the original system. Only the Creative Console would record properly, had to set 24-bit separately. Bit of a PITA. Did not find any other app which would record properly.

    Some apps will recognize the original recording as a DD-Wav, but the besweet processing was necessary to get usable file. AC3Filter was also able to process this file in some ways, IIRC it allowed live playback. Didn't end up using it for some reason, I forget why. The Besweet, or BeSplit, whichever, removed the gaps which are necessary for timing purposes, but also prevent most apps from recognizing the audio file.

    There was a second method involving VLC which somebody else proposed, I recall verifying that it did actually work but that's about it.

    You probably already know this, but turn your volume WAY DOWN before testing any playback of captured files.
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    This:
    Code:
    78 0B
    pretty much makes it impossible to get anything playable out of your wav file. It's the Syncword from one of the AC3 frames and it should read 77 0B. It looks as though all those random 1s and -1s scattered throughout the wav file may be there on purpose and are actually mixed with the AC3 data to make it unplayable (unless of course somewhere in your pipeline there's an analogue stage causing the problem...).

    So what we're facing is the SPDIF has converted the stream blindly to 5.1 16 bit pcm, remapped the channels, switched the endian mode then incremented/subtracted 1 from random samples throughout the entire file. Finally it converted that blindly to 24 bit stereo before feeding it to your recording program.

    Everything but the +/-1 I could have dealt with.

    Depending on where this processing is happening Nelsons methods may still work, but your current method will definitely not.
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  11. ndjamena - You are not understanding what is being done here. This IS an analog recording of an AC-3 file, as a compressed, stereo, PCM file. That is how it is transmitted over the optical cable, intended for decoding and decompression by the receiving device.

    It is not 5.1 until it is decompressed and decoded. The blank areas are inserted by design for timing purposes. It is playable by a very few programs, which recognize it as a "DD-Wav". Most can not.

    There is no bitrate conversion. The recording MUST be 24-bit, bit-accurate to allow for conversion. Only one or two cards can do this. BeSweet removes the padding, and renders a recognizable AC-3 file at whatever bitrate and channel content the original file contained.

    It is tricky to setup properly, VERY picky on hardware requirements, but definitely works, I recorded several dozen and a couple others have done it, as well. It is an analog recording of a digital file being transmitted. The recording is 24-bit, stereo, 48k, but this has nothing whatsoever to do with the specs of the final content.

    i would have to look up the BeSweet correction commands, it should be referenced in my original post. However, if playback of the original recording is continuous, and not rapidly pulsating, something is not being done correctly. IIRC this indicated either a 16-bit recording, rather than 24-bit, or that the card is NOT bit-accurate. Such files never converted at all.

    Again, for anyone trying or testing this, turn your volume WAY DOWN, at the hardware level, before attempting playback of the recorded file and DO NOT use headphones. The sound is EXTREMELY loud and can damage eardrums or equipment. The file plays back at MAX volume and is immune to most software volume controls.

    EDIT: - The SB X-fi is one of only two cards reported to do this correctly, and the only one I can verify. Various other SB, Phillips, and M-audio cards failed, though the M-audio had a defective port and could not really be tested. 24-bit, bit-accurate, with a SPDIF is mandatory, although no guarantee of success. The other reported working card was a Terratec, IIRC. Also, the guy with the VLC method reported success with an early model MAC, I forget which model.
    Last edited by Nelson37; 29th Oct 2014 at 09:17.
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    Grrr, I just checked the original file I downloaded and it looks like he recorded it as 16 bit, I thought I'd altered it to say that in the header.
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  13. From his description of the sound, that's what I thought. It's just garbage and will never convert. He PM'd me days ago and I responded, haven't heard back and he hasn't returned here.

    i remember that you had to double-check the 24-bit setting in the Creative software, it would keep changing to 16-bit at some point. There was more than one area you had to specify SPDIF settings, and only one of the included Creative progs would record properly, the other one did not.

    Pretty sure I did get it to work with another prog, but I do recall I stayed with the Creative software, not sure why. I did verify the VLC method, but I think that was only for the conversion and not the original recording process. Not certain there, it's been a long time ago.

    As mentioned, I still have the same PC, though last fired it up over a year ago. It still has my last ATI cap card in it, AGP slot.

    If you like a puzzle, try this one - Sat tv, HD prog with AC-3 captured with above method. Same prog captured in SD MPG with ATI card thru S-video and standard audio outputs at same time. The SD audio and HD audio should be identical, but they are not. Gradually decreasing synch, with the AC-3 file roughly a second shorter per 40 minutes. Fixable, but PITA.

    Either, A - the HD broadcast is at a slightly different frame rate, or the SD output changes it, same thing I guess, OR, B - something in the conversion process alters the file slightly, but consistently over a wide range of bitrates and play-time lengths.

    Towards the end I was able to capture anamorphic widescreen, DVD ready, from HD source, 23.97 fps, with surround sound and subtitles. Had to write a little program to convert the ATI text to subtitles, and much hassle getting all to work, but damn it was nice.

    Much easier to just DL the damn thing, but where's the fun in that?
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  14. Formerly 'vaporeon800' Brad's Avatar
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    Originally Posted by Nelson37 View Post
    The recording MUST be 24-bit
    I've recorded in 48/16 with Audacity and successfully read it back as ddwav. But this was on a VIA chipset, not Creative, and only on XP.
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  15. A 5.1 source from an SPDIF input?? Everything I read on the subject said this was not possible, my own tests agree. From a VIA soundcard? Did you need to use a BeSweet module to convert to usable audio?

    A straight stereo PCM, output as such and recorded the same way, sure. This file is output as 5.1 ac-3, recorded as stereo PCM, then a brief conversion back to the original 5.1.

    TBH, I would need to see a LOT more detail before I could call your report credible. NO ONE, and I repeat, NO ONE, has ever reported anything remotely similar being successful, and I spent quite a bit of time researching this project. Found less than 5-6 people reporting success of any kind, all together.
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    If there's so few devices that can record ac3 from SPDIF, it might be worth noting that the OP is using an X-Fi PCIE and not an original X-Fi.

    (There are definitely ac3 frames in the file he posted as the presence of the Syncword proves. I can only guess what it's been through to make it unreadable, though.)
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  17. Not sure what is contained in a file that doesn't work, makes sense that there would be some recognizable fragments.

    Don't remember if anyone used a pci-e x-fi card. There was some mention of an audigy model with hacked drivers, also.
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  18. Formerly 'vaporeon800' Brad's Avatar
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    Originally Posted by Nelson37 View Post
    A 5.1 source from an SPDIF input?? Everything I read on the subject said this was not possible, my own tests agree. From a VIA soundcard? Did you need to use a BeSweet module to convert to usable audio?
    I was mainly capturing 2.0 AC3 bitstreams from my satellite box's SD channels via optical, using the app I attached above. Sometimes I would also capture 5.1 AC3 640kbps. I successfully tested the ddwav method, but since BeSplit was far less convenient than AC3Rec I didn't use it day-to-day.

    TBH, I would need to see a LOT more detail before I could call your report credible. NO ONE, and I repeat, NO ONE, has ever reported anything remotely similar being successful, and I spent quite a bit of time researching this project. Found less than 5-6 people reporting success of any kind, all together.
    I already linked to two of my posts from the time, earlier in the thread. Unfortunately Imageshack nuked everything on free accounts, so the screenshots are gone. The user "jimwhite" is who originally tested some cards and suggested the one I bought. Similar: OT.com thread talking about 44.1/16 capture of 5.1 DTS CD
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