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  1. Member
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    Perhaps this question has been answered already, and if so, I apologize. I've read a variety of guides and postings, but not one that actually answered my question.

    I have a sound card (M-Audio Audiophile24/96) that has S/PDIF inputs and outputs (RCA Jack).

    What I would like to do is record a DTS and/or AC3 signal sent to this input. However, I'm a little uncertain as to what this signal is going to look like, how to capture it, save it, etc.

    What I normally use this input for is capturing digital audio (S/PDIF) produced by a synthesizer. This is easy -- Cool Edit and other sound editors treat the sound card's S/PDIF input as a stereo source, and can produce a WAV file. However, my understanding is that an AC3 or DTS source (such as a Laserdisc player or DVD Player) isn't quite the same as a simple 44.1 or 48 KHz 'digital stereo' signal produced by a synthesizer. For example, the .ac3 file extracted from a DVD just sounds like a lot of buzzing when looked at by a wave editor.

    So -- to the question. How can I capture a DTS or AC3 signal from the S/PDIF input on this card, save it, and then convert it to AC3 format to place on a DVD?

    Thanks!
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  2. SFAIK, you capture as a WAV file and then run it thru BeSplice as a DDWAV source, this converts it to a true AC-3 file.

    You have one of the two cards I have seen reported as being able to do this effectively. You may need a program called Total Recorder if your capture software does not offer the choice of SPDIF input.

    Make sure the source is set to output Digital Surround and not PCM. The original file should sound like pulsing static, after BeSplice it should be continous static. The pulsing is caused by blank intervals inserted to maintain time synch, apparently.

    I have a similar capture card to yours and am contemplating purchasing the same audio card. I have researched this off and on for a year or two. Am very interested in any results you obtain, and will be happy to help your efforts in any way I can.

    If I buy a $150 sound card and it doesn't work, I am in deep doodoo with the Domestic Financial Advisor.
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    can a WAV have that many channels, and will they continue through to the AC-3 format?
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  4. What you are capturing is a digital file, not a true audio file. Think of it like a ZIP file. The card has to be "bit-accurate" to do this. The BeSplice operation takes out the silent spaces, leaving you with a more or less standard AC-3 file.

    Read most of this on Doom9 and others. A couple of different techniques mentioned, will try and dig them up and post. The M-audio 24/96 and the Audigy 2 with special third-party drivers, and one other I can't remember, reported to be able to do this capture. Not much detailed information.

    No specific mention of synch to video issues, capture of DTS not discussed.

    I did get as far as recording PCM audio thru SPDIF-in on my Philips 706, using Total Recorder to specify input, synch was maintained using MMC 7.6. My card was only 16-bit, and I could not get a usable AC-3 file from it when I switched to Digital Surround (AC-3).
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    Hi Nelson,

    I'm about ready to give it a try (if I can get some free time!!). I suspect that I will need to record a monophonic .wav at whatever bitrate my sound card reports from the SPDIF link. I'm not sure whether or not I need to provide or accept clocking, but I imagine that the player will provide clock.

    If necessary, and I can get my hands on the specs, I'll write something.
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  6. Originally Posted by Perro Grande
    So -- to the question. How can I capture a DTS or AC3 signal from the S/PDIF input on this card, save it, and then convert it to AC3 format to place on a DVD?

    Thanks!
    Assuming the source for the Ac3 or DTS is a DVD player in the 1st place, why not just rip the DVD? It looks a lot simpler than what you are trying to do.
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    Hi Bugster,

    It sure would be easy if the source was DVD... RIP and GO, so to speak.

    However, the source is not a DVD. The source is other than a DVD -- such as an AC3 (or even DTS) encoded LaserDisc. I want to get the digital data stream right off the TOSLink/Coaxial cable and put it (or convert it) in whatever format is used by DVDs for their audio streams (AC3 5.1 or DTS 5.1).

    The rub is that I don't know the digital format of the bitstream coming out of the TOSLink/Coaxial output... And herein lies the confusion. If I can get the specs for the data stream for AC3/DTS and the AC3/DTS file format, I could write something to do this.
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  8. Originally Posted by Perro Grande
    Hi Bugster,

    It sure would be easy if the source was DVD... RIP and GO, so to speak.

    However, the source is not a DVD. The source is other than a DVD -- such as an AC3 (or even DTS) encoded LaserDisc.
    Fair enough, I was just curious. I forgot that some Laser Discs carried Ac3 audio, never having owned an LD player. Sorry I can't help, but good luck with your endeavours.
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  9. Apparently the AC3 stream will be either 44 or 48 Khz, the software should autodetect. You should NOT be able to change the volume.

    Examples - 48k wav using SoundForge 4.5, saved as RAW, CoolEdit 48k stereo saved as RAW, 16-bit, Motorola PCM.

    The command line is "BeSplit -core( -input wave1.wav -prefix c:\ -type ddwav -fix" to remove the padding. Rename to ".AC3".

    The other card was the Audiotrak Prodigy 7.1, w/optical input. Also one report of onboard C-media sound working.

    Audigy2 works with KX drivers, apparently.

    Don't know how this will work with concurrent video capture, I used Total Recorder but not together with MMC to produce a WAV file. My Philips died so I can't test SPDIF input anymore.
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  10. Awesome! Some one has finally answered this long awaited question! The only problem is now I have to buy the $150 sound card, $180 RF (De?)Modulator, $80 LD player w/ DTS or AC-3 RF output, and the AC-3 SW Episodes 4-6! :P

    But I guess it is worth knowing for all those people who had the money to spend in the mid 90's, and still have the equipment.

    I think that DVD writers (standalone) should start coming out with optical/coaxial in. After all HDTV is broadcast is DD, but then the quality of HDTV is lost when converting to DVD. I guess those who spend the big $$$ on HDTV will have to wait till BluRay or something else to come out.
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  11. Now what about the problems with audio sync????
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  12. I just posetd another reply like this, but I've found that in most current DVD authoring progs that you have to be working in 48k audio. If you capture at a different rate, you need to change the sample rate (not resample) the audio. It has been my experience that if I start a video project with captured at time audio that is other than 48k, I have to change the sample rate. Which is a real drag if you're doing Foley work with 44.1 samples because you have to resample them all and then paste them for Roxio, MYDVD and DVD Architect to burn and render in sync as they all render to NTSC 48k and upsample instead of convert sample rate. At least for me, and I could be missing an easy workaround but I haven't found it yet.
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  13. Originally Posted by c_hernandez32
    The only problem is now I have to buy the $150 sound card, $180 RF (De?)Modulator, $80 LD player w/ DTS or AC-3 RF output, and the AC-3 SW Episodes 4-6! :P
    Or you could just wait another few years for the DVDs to come out. Lucas filmed the 2nd trilogy first because the technology wasn't developed for him to film the 1st trilogy to his vision and standard. Certainly he isn't going to leave the "best" copies of the 2nd trilogy in VHS or LD (which isn't any better than 8-track right? -- I'm talking about availability, not quality). I don't care what DVD sites have been rumoring. SW 4,5,6 will be on DVD or the next medium of the future.
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    Yep...

    I could wait... But I'm not going to.

    If I wait, I learn nothing. I'm having fun learning about various aspects of digital audio and video. Frankly, it is less about the destination than the journey at this point.
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  15. Ok I think I found an easy way to record DD 5.1 and maybe DTS audio off of LD's. The program CD Wave, a shareware with no limitations can record DD 5.1 wavs. While I don't have a soundcard with TOSlink or S/PDIF inputs so I can't be sure if it will actually work. Then you have the problem of converting to AC3 from WAV. But if it works you are one step closer to a true and complete LD to DVD conversion. Also maybe since WAV is uncompressed, you could record DTS and convert to DTS audio and still keep the high quality audio. Well good luck and let us know.

    Also just a thought, you could get a AC-3 RF demodulator (standalone) with six RCA outputs and record each channel alone.
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    Well, my first attempt was a dismal failure...

    I started by connecting my soundcard's S/PDIF input to the Coaxial output on the LD/DVD combo player I have. I allowed the sound card to use the LD player as the sync source and set it to 48K. As expected, I was able to record the PCM digital stereo flawlessly.

    My next step was to move to the AC3 S/PDIF connector. This particular LD player has both the AC-3 RF output and a S/PDIF AC3 output. Again, I set the card to sync from the player, 48KHz. I also set the data type to Non-Audio.

    As expected, the data coming out of this port sounds like nothing more than buzzing. As expected, the more audio content, the more busy this port became. It looked exactly like what happens when you try to open an AC3 file in CoolEdit...

    I recorded about 2 minutes of this stuff in cool edit. I saved it two ways -- "Raw (AC3)" and as a .wav file. I tried this with both a 1-channel capture and a 2 channel capture (stereo, eg).

    The file saved in "raw" format, although it had an AC3 extension, would not play. My software DVD player will play a legit AC3 file, but it would not accept this file.

    Then I tried the .wav file using BeSplit to convert from a "DDWAV" format. I used the syntax that appears in the example file. It ran, did its thing, and produced an AC3 file. My software DVD player accepted this as a valid AC3 file, but there was no sound...

    So I'm heading back to the drawing board... So, does anyone know where I can find the technical specs on the AC3 file format?
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    More information:

    1) I'm not sure exactly how standard the "ddwav" format really is. When I convert an AC3 file to a DDWAV using BeSweet, and try to go back again using BeSplit (-ddwav -fix), it only transcodes a small section of the file, always to complete silence. I end up with about a 10 second, completely silent AC3 file when trying to reverse the process. Thus far, I've stuck to 2 Channel AC3 streams and files. Is DDWAV made solely for 6 channel? I dunno...

    2) What is coming in the S/PDIF input "looks" like a "ddwav" file produced by BeSweet. Specifically, it contains bursts of data, evenly and regularily spaced. Between the bursts is a stream of 0's and FF's. I suspect this is timing and control in some fashion. So, the effect is burst - silence - burst - silence. When "Listening" to this file as if it were a wave, you get a buzzing sound.

    3) An AC3 file has none of the burst characteristics. It is a solid stream of data - continuous. If it is packetized, it is not done in the segmented fashion as the S/PDIF or DDWAV file. When "listening" to this file, it sounds much more like white noise than the buzzing of a DDWAV.

    4) I suspect (but lack proof) that if I could somehow programatically join only the data sections of one of these ddwav-style files (either captured or created) and added an AC3 file designator (two specific bytes at the beginning of the file), it would work as an AC3 file -- but I have no evidence to really support this theory.

    The theory of going from ddwav to AC3 sounds reasonable, and is probably correct. However, the tool at hand (BeSweet) is expecting something, or something is causing it to bail out of the transcoding early.
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  18. Perro, you seem to be getting the same results that I did. I also was recording a 2-channel AC-3, (from cable), that may be the issue. Though the card I used has not been listed as able to do "bit-accurate" capture.

    At the time I did this I did not have an AC-3 capable stereo and was only able to test using WinDVD, though I did verify it could play a valid AC-3 file. Now I have an AC-3 amp but my Phillips 706 card, with Digital input, died shortly thereafter. It is possible I may have something to test with tomorrow morning!!!!

    Can you try the same test with a true 6-channel AC-3 captured from a DVD, perhaps?

    Also, what did you use to view the file?
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    Hi Nelson,

    I have not been able to do too much more on this project over the past few days -- holidays and all.

    I've had the same (lack of) luck with two independent, but verified 6-ch AC3 sources via Coaxial SPDIF -- DVD and LD. So far, no success.

    With BeSweet 1.4 (although not 1.5x) I can go from a *ripped* AC3 to a "DDWAV" and back to AC3 using BeSplit. With a captured stream, though, no such luck... Simply capturing a WAV file of an AC3 source does not see to equal a "DDWAV' file as far as BeSplit is concerned.

    As far as looking at the file goes, I've used cool edit for viewing general patterns, and a Hex Editor to look at the actual data.

    I think I'm going to look in to getting the SDK for my sound card and see if I can look at the SPDIF stream more directly.
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  20. Has any one tried capturing via CDWIN? It can capture 6 channel wavs, but I'm not sure that BeSweet will convert them to AC3.

    What kind of AC-3 demodulator are you using? I know some have six seperate audio channel outputs. So you could create your own AC-3, maybe in Vegas, and simply record two channels at a time.
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    Yes, I've tried CDWAVE. It produced the same results, essentially, as all the others.

    There are several RF Demods out there that produce 6 discrete analog outputs. I have not gone this route yet -- I **know** I can make that work. Unfortunately, I don't have a demod of this variety. I'm still grabbing SPDIF stuff and running straight into a brick wall.
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  22. Interesting topic and good discussion...keep it going!

    I've been trying to do something similar, capture movies with DD5.1 sound from satellite. I'm having some successes, but I'm not there yet. Maybe with some more experimentation we can help each other out.

    Here's what I've discovered so far about capturing DD via TOSLink.

    1) You just about absolutely need a "bit-perfect" capture card. I've got the Terratec 6-fireLT, which should fit the bill. I'm pretty sure the M-Audio cards with digital I/O should work also.

    2) The AC-3 stream that comes from satellite (and presumably from LD also) has a certain bitrate, which is less than the video stream. To keep in sync with the video, "filler" data is streamed along with the audio data to keep the frame sizes of the AC-3 stream fixed. This extra padding needs to be stripped from the captured audio before muxing it with the video for DVD purposes (hence the need for BeSplit, or similar programs to "fix" the audio into a proper AC-3 file).

    3) You need to capture the audio stream as a STEREO WAV, 48Kbps, 16 bit. I've had the best luck so far with Virtual Dub. After capture, I export the audio as a WAV file and then run it through BeSplit:

    BeSplit.exe -core( -input xxx.wav -prefix yyy -fix -logfile BeSplit.txt -type ddwav )

    with xxx.wav as the inputfilename, yyy as the output folder name.

    The resultant file SHOULD be a valid AC-3 file suitable for muxing back with the video after transcoding it to MPEG2. Here's where my problem lies. If I play the resultant muxed movie with PowerDVD, my receiver recognizes and plays the audio as Dolby Digital 5.1, but I get periodic (like millisecond duration) dropouts or "skips" in the audio. Sometimes it will go several minutes with perfect audio, and then I'll get a little millisecond of silence. Either my sound card is not getting a "bit perfect" capture, or Besplit isn't stripping the padding properly, or something else is wrong with my capture. It's almost like when you record audio and get clipping, but AFAIK there should be any clipping when recording a digital stream.

    I'm not dropping any video frames during capture, and as far as I can tell, I'm capturing the audio fine according to Virtual Dub (CPU at 9% use during capture, on a defragged drive).

    I'm sooo close yet not there.

    =caduceus=
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    Hi Caduceus,

    That is good info! You've done better than I've done thus far. I can sometimes get BeSplit to give me a valid AC3 file, but I always end up getting a file that contains only a few seconds of silence...

    In looking at the files, the raw captures off the SPDIF input "look" like the ddwav files that besweet produces -- burst of data, silence, burst, silence, etc.

    It strikes me that the capture should probably be single channel and not stereo -- SPDIF is one baseband channel, after all. However, I agree that BeSplit might be looking for a stereo WAV file, as I've had no luck whatsoever with the mono captures.

    I'll be doing some more tests in the next few days now that the holidays are over. I'll report my results.

    Cheers!
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  24. Perro Grande wrote:

    It strikes me that the capture should probably be single channel and not stereo -- SPDIF is one baseband channel, after all. However, I agree that BeSplit might be looking for a stereo WAV file, as I've had no luck whatsoever with the mono captures.
    The raw Dolby Digital stream comes through in 2-channels. If you capture in mono, you're losing information because your capturing software is converting the two channel stream into one channel.

    =caduceus=
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  25. Caduceus - I had a similar problem capturing PCM files thru SPDIF, the audio contained periodic "pops". These were detectable listening to the original signal. My source was Digital Cable, cable company told me it was interference of some sort on the line. After my complaint, which was right after Digital became available, the problem largely but not completely went away.

    Have you tried outputting the captured, unaltered wav file thru SPDIF to the amp to test for the sound drops?

    Can you increase buffers on the card, or in VDUB?

    You are the first person I have been able to converse with who has reported any kind of successful capture of AC-3. THANK YOU!

    Does the capped file synch OK with the video?
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    So you're using VDub to capture the video and the audio is coming along with the resulting AVI?

    Sorry if this is a really stupid question, but I don't (can't) use VDub to do my captures (I'm using firewire-based capture devices) so I'm a little unfamiliar with capturing in it.

    Yes, I've tried firing the raw captured file out to my amp -- no luck. In fairness, I tried this with my laptop not my main computer. However, the software interpreted the file as a .WAV, and proceeded to send the "buzzing" sound out the TOSLink port as PCM... Oh well... I'll give this another shot.
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  27. Nelson37,

    Have you tried outputting the captured, unaltered wav file thru SPDIF to the amp to test for the sound drops?
    Yes, I have a Dish PVR, so I capture everything to the PVR first, then capture to PC later on. The movies, when played from the PVR, are supposed to be an exact copy of the original satellite feed. When played via SPDIF passthrough to my Logitech Z680s (with built in DD/DTS decoders), the skips are not present in the original stream. I get noise/garbage if I try to play the captured WAV prior to fixing it with BeSplit. You'd think the decoder would see it the same. The extra padding data seems to not let the receiver decode it properly....why, I have no idea.

    Can you increase buffers on the card, or in VDUB?
    Yeah, I really messed around with that for a while...not in VDub, but in the Terratec mixer settings. This DID seem to make a difference. The optimal buffer size setting for me was 3msec. Five msec was second best. Anything smaller or larger seemed to increase the pops. Note that it's essential to lock the master clock on the sound card to the EXTERNAL device that you're recording from, or you can't record "bit-for-bit" from what I understand.

    You are the first person I have been able to converse with who has reported any kind of successful capture of AC-3. THANK YOU! Does the capped file synch OK with the video?
    Well, I'm certainly no expert at this...I searched the net for days trying to find bits of info. I still haven't been able yet to do a "perfect" AC-3 recording, and I haven't yet tried to capture a full movie (and probably won't until I can eliminated the blips). Timing of video and audio with short clips seemed to be OK, although as I said, I haven't done a long capture yet. Soft Encode is great for reviewing the AC-3 streams after running them through BeSplit.

    It could be my capture card or drivers. I've heard some complaints about clicks and pops when recording with the 6-fire in other forums. I'd love to hear a success story from someone with the same audio card, or even M-Audio or others and find out if a bit accurate AC-3 capture really can be done. In the meantime, I'll keep trying.
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  28. Perro Grande,

    So you're using VDub to capture the video and the audio is coming along with the resulting AVI?
    Yes. I have to first set the recording source in the system mixer settings to digital input (SPDIF) before recording, and lock the sound card's master clock to my PVR's digital output.

    I went with Virtual Dub because I didn't have any luck with ATI's MMC (ATI's TV Tuner/capture software). For the benefit of others, I'll mention that process I stumbled through. MMC, as designed, does not let you record via digital input. My only choices were analog sources (microphone, CD, Line-In, etc), and when you fire up MMC, it automatically changes the recording source in your mixer (to analog Line-In in my case). I "fooled" MMC by switching the recording source in the system mixer to SPDIF-In AFTER starting up MMC. After my capture is done, I switch the recording source back to whatever it was before (otherwise MMC can lock up).

    I found I could easily capture digital audio (PCM streams only) with this method, and it's my preferred way of capturing non-Dolby satellite programs for burning to DVD.

    However, capturing DD encoded programs with MMC was impossible, at least as far as I tried. I got AC-3 white noise only in my captures, and no combination of Besplit, AC3fix, etc. would work. I suspect it's maybe a codec issue but I didn't really dig into what the root of the cause was. I switched to Virtual Dub, and after running the audio through BeSplit, actually got something for my DD decoder to decode.

    Sorry if this is a really stupid question, but I don't (can't) use VDub to do my captures (I'm using firewire-based capture devices) so I'm a little unfamiliar with capturing in it.
    There are no stupid questions here...we're all learning. I don't have firewire, so I don't have the first clue about using that method for capturing video.

    Yes, I've tried firing the raw captured file out to my amp -- no luck. In fairness, I tried this with my laptop not my main computer. However, the software interpreted the file as a .WAV, and proceeded to send the "buzzing" sound out the TOSLink port as PCM... Oh well... I'll give this another shot.
    Make sure the WAV file is a 48K, 16-bit stereo DDWAV, and that you ran it through BeSplit first to convert into an AC-3.

    Believe me, I know that "buzzing" sound. To ANYBODY testing AC-3 files...warning....keep your speaker volume down so you don't blow your speakers, and be careful with headphones....I already learned that (painful) lesson!
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    and be careful with headphones....I already learned that (painful) lesson!
    Me too! My wife has witnessed, on several occasions, a quick shedding of the headphones (with a motion not unlike that used by a dog to scratch his ears) followed by a stream of colorful anglo-saxon monosyllabic words. Good thing I have durable headphones and durable ears...

    Back to the subject at hand...

    I just cannot get consistent behavior from this stuff! I'm not even bothering to capture video -- that I can do. Just audio... Yesterday's efforts brought more confusion and frustration.

    I tried, again, to capture an AC3 feed from the coaxial output of my DVD player. I ran it into the S/PDIF input of my M-Audio sound card. I allowed the sound card to lock to the DVD player as source, which it did at 48KHz. I then captured 16-bit stereo WAV (48K) files using three tools: Cool Edit, GoldWave, and CDWave. I also tried using CDWave's "5.1" setting.

    In all cases, I captured 20 seconds of the AC3 stream. I started with the DVD paused so that I could get any sync or message bits that might be in the stream.

    The results, regardless of the capture tool, were the same:

    1) BeSplit bailed out after only transcoding a couple hundred milliseconds.
    2) BeSplit produced no errors to the screen or to the logfile
    3) BeSplit produced a tiny, silent, 0-duration AC3 file.

    Late last night, I noticed a few things that I wanted to change in this setup. I'll try it again today...
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    SUCCESS SUCCESS SUCCESS!!!!

    I **finally** managed to get a working 5.1 AC3 file via S/PDIF capture! Resulting file appears to be flawless, has no pops, clicks, or anything else. All 6 channels are present, and the timing appears to be accurate. Of course, more testing is required, but this is promising.

    Here is what I did (not too different than previous attempts):

    1) Upgraded my M-Audio drivers to the latest (December 26th) edition (I am using an Audiophile 24/96 card)

    2) Set the card as follows:
    • * interpret the S/PDIF signal as non-audio information
      * 48K S/PDIF sample rate
      * Locked to DVD/LD player
      * SCMS set to "Original"
      * 512 sample buffer
      * Independent
      * Disable audio app use of monitor mixer... UNchecked

    3) Used CDWave (a nice little utility available at www.cdwave.com) and captured the input as a 16-bit Stereo PCM wave file. The 5.1 setting does NOT work for this method.

    4) Ran BeSplit with the -fix and -type ddwav set

    e.g.

    BeSplit -core( -input infile.wav -prefix out_prefix -type ddwav -fix )

    It does not matter if the capture is started before the audio stream is running, or if it is started in mid-stream.

    I'm going to try these settings with the other wave capture utilities I have and see how this works... I will report back as I find more info.
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