Help to Capture AC-3 Does anybody know how or is it a huge seceret

[beeker1959]

Help
Does anybody have any info on how to capture an AC-3 signal from a Digital Set top box into the PC either via SPDIF or Coax

[SomeJoe]

Disclaimer: I have not tested any of this. However, I did some in-depth research on this subject a few weeks back and this is what I came up with. Use this info at your own risk.

Basically, you need 2 (possible 3) things to do this:

1. If the AC3 source is from something with an RF AC3 output (like a LaserDisc player), you will need an RF demodulator to convert the RF-modulated AC3 to SPDIF coax or optical (TosLink). The least expensive one available is the Yamaha APD-1 (http://www.angelfire.com/retro/clockz/yamaha/apd.html). I don't think these are made anymore, so you probably will have to get one off of e-Bay. Now, this piece is not necessary if your AC3 source is already from a box that puts out AC3 via SPDIF. Note: LaserDisc players typically also have an SPDIF coax/optical digital output, but this is NOT AC3 - it is only the digital 2-channel stereo from the LaserDisc.

A side note: SPDIF is a digital transport medium, originally defined by IEC60958. This original specification says that SPDIF is a digital PCM-only transport, capable of sending 2-channel stereo audio at 16, 20, or 24 bits, and at 32, 44.1, or 48 kHz.

Another specification was subsequently adopted, IEC61937, which says how non-PCM datastreams can be transported over SPDIF. This is how Dolby Digital (AC3) and DTS are sent via SPDIF. It is similar to a network protocol -- the 5.1 audio is compressed and stored inside DD or DTS frames. Then, these data frames are encapsulated inside the SPDIF digital headers.

2. Now, you need to capture the SPDIF stream into the computer. There are a few relatively inexpensive sound cards that can do this:

- M-Audio/Midiman Audiophile 2496 (http://www.midiman.net/products/m-audio/audiophile.php)
- AudioTrak Prodigy 7.1 (http://www.audiotrak.co.uk/products/prodigy71/index.shtml)
- Several other professional-level sound cards.
- The SoundBlaster Audigy/Audigy2 will NOT work. It post-processes SPDIF input streams, resulting in non-exact captures of the digital audio. While this still works for IEC60958 SPDIF PCM audio streams, it renders IEC61937 non-PCM datastreams useless.

After you install one of these soundcards, you'll run a professional sound editor (CoolEdit Pro or Sound Forge) and set the input the the SPDIF connector on the sound card, and begin recording.

The sound editor application knows NOTHING about IEC61937 non-PCM datastreams, and will think that what you've captured is PCM audio. When you play it back, it will sound like white noise.

3. Save that white noise as a .wav file, and run BeSplit on it with a type of ddwav. BeSplit will un-encapsulate the SPDIF framing from the file and will leave you with an .ac3 file.

That will get your AC3 audio captured digitally into the computer.


Some other threads on this subject: 1 (http://forum.doom9.org/showthread.php?s=&threadid=42568), 2 (http://forum.doom9.org/showthread.php?s=&threadid=25650).

[calinb]

[QUOTE]Originally posted by SomeJoe
- The SoundBlaster Audigy/Audigy2 will NOT work. It post-processes SPDIF input streams, resulting in non-exact captures of the digital audio. While this still works for IEC60958 SPDIF PCM audio streams, it renders IEC61937 non-PCM datastreams useless.

If you nuke the Creative drivers and get the KX drivers for the Audigy2, it seems to work very well in "Direct SPDIF Record" mode:

http://kxproject.spb.ru/

[calinb]

Now that I have my Audigy2 working with the KX Drivers, how can I get BeSplit -fix functionality working as an audio codec. I'd like to perform the function real-time as I capture AC3 audio and video from my mini dish receiver.

It would be convenient not to have to demux, run the command line, <BeSplit -core( -input wave1.wav -prefix c:\ -type ddwav -fix>, and remux.

Maybe this would be a good project for me to learn to program DirectShow. (Or maybe it can be done in graphedit.)

Relevant thread:
http://forum.doom9.org/showthread.php?s=&threadid=28111&highlight=ddwav

[SomeJoe]

Originally posted by calinb
If you nuke the Creative drivers and get the KX drivers for the Audigy2, it seems to work very well in "Direct SPDIF Record" mode:

http://kxproject.spb.ru/

That's awesome. I read over those KX driver details today, and these drivers literally turn the SB Audigy into a professional level sound card. Custom routing, ASIO drivers, programmable filters, the works.

I had been holding off on buying a new sound card because some of the pro models are fairly pricey, but I might just pick up a used Audigy and give these drivers a whirl. ;)

(A side note: The Direct SPDIF Record function is necessary for bit-accurate captures of the SPDIF datastream, but only the Emu10K-2 DSP chip is capable of the bit-accurate capture, not the Emu10K-1. Though the KX drivers support both chips, you need an Emu10K-2 chip for this function, which is available only on the Audigy and Audigy2. The older SB Live!, PCI512, and various OEM models based on the Emu10K-1 won't work.)

Even a base Audigy MP3 or Audigy Gamer can be made to work for the capture application, despite the fact that they have no SPDIF coax or Toslink input. They do have a CD Digital input on the card, which is SPDIF, but not at the coax voltages. This (http://members.tripod.com/~Psych/coax-ttl-md.html) page has a circuit that's pretty easy and cheap to build to convert coax SPDIF to TTL-level SPDIF which can be plugged into the CD Digital input on the card. Not exactly elegant, but for less than $100, you can get AC3 capture on the computer. :D

[calinb]

Originally posted by SomeJoe
Even a base Audigy MP3 or Audigy Gamer can be made to work for the capture application, despite the fact that they have no SPDIF coax or Toslink input. They do have a CD Digital input on the card, which is SPDIF, but not at the coax voltages.

SomeJoe-

Good points about the Emu10K-2 vs. -1.

I have the basic Audigy2 card (no Audigy drive or EX box). I didn't even look at the logic voltage levels. The only digital input on my Audigy2 card is the CD-digital in. I have a Hoontech Digital XG card with the DB-1 bracket that has Toslink, coax, and AES/EBU I/O so I simply ran the coax out from the Hoontech to the Audigy2 Digital CD-input. It works perfectly (using digital "direct" mode in the Hoontech card, of course). If there's a logic level difference, I'm probably giving up noise margin, but it works!

Although the Hoontech driver goes bust on SCMS, it's a great digital I/O converter, especially considering the price (~$70 US). I purchased the Audigy2 specifically for the KX drivers and, given that I already had the Hoontech card, I didn't see any reason to spend the extra $100-$150 for an Audigy drive or the Platinum EX and all the extra digital I/O connections.

[e.lectronick]

Originally posted by calinb
[QUOTE]Originally posted by SomeJoe
- The SoundBlaster Audigy/Audigy2 will NOT work. It post-processes SPDIF input streams, resulting in non-exact captures of the digital audio. While this still works for IEC60958 SPDIF PCM audio streams, it renders IEC61937 non-PCM datastreams useless.

If you nuke the Creative drivers and get the KX drivers for the Audigy2, it seems to work very well in "Direct SPDIF Record" mode:

http://kxproject.spb.ru/

Hi, guys.
Just an update: while the older Audigy cards did not have a "Bit-Accurate" record mode via S/PDIF, the ZS cards do. Check this out:

http://www.soundblaster.com/products/Audigy2ZS_platinum_pro/compare.asp

There's actually a tab in the control panel using Creative's native drivers that allows you to select Bit Accurate recording in software. I'm getting one!

Here's a shot of the control panel itself:

http://www.xbitlabs.com/articles/multimedia/display/creative-audigy2-zs_7.html
-Erik.

[SeeMoreDigital]

Originally posted by SomeJoe
...This (http://members.tripod.com/~Psych/coax-ttl-md.html) page has a circuit that's pretty easy and cheap to build to convert coax SPDIF to TTL-level SPDIF which can be plugged into the CD Digital input on the card. Not exactly elegant, but for less than $100, you can get AC3 capture on the computer. :D Hi SomeJoe,

Unfortunately the link you provided does not work. I would like to build one of these little devices!

This is a very useful thread. I have an Audigy card and currently use a little Creative breakout box to capture 2Ch PCM audio from optical or electrical digital sources. However, being able to capture full 6Ch audio is going to be great fun!

Cheers and thanks you guys

[SomeJoe]

Hi SeeMoreDigital,

Yes, the link I provided seems to be broken.

I found another page (http://www.andrewkilpatrick.org/mind/spdif/) with a similar schematic. Try that one.


e.lectronick,

Nice to know that the new Audigy ZS cards can now capture bit-accurately. :) I see one of those in my future. :D

[rennervision]

Hello everyone. I've started looking into converting all of my laserdiscs to DVD and am curious if anyone has had any experience with this trick on a notebook PC card or an external card of some sort? I was considering the Soundblaster Audigy 2 ZS Notebook, but according to this link it does not appear it can record in bit accurate mode:

http://forums.creative.com/creativelabs/board/message?board.id=pcmciasb&message.id=600&query.id=65650#M600

Furthermore, it does not look like the KX drivers currently support 24/96 with this Audigy 2 card:

http://kxproject.lugosoft.com/faq.php?language=en#Q15

So, I continued researching and came across the M-Audio Transit which
appears that it might be able to do the job:

http://www.m-audio.com/products/en_us/Transit-main.html

However, M-Audio has told me I cannot capture AC-3 with this device. To be honest, I am not so certain what to believe. Creative has e-mailed me to say none of their Soundblaster products can do this either (which I know is not true). So I suspect this could be a feature no one outside of this forum knows about! :) Has anyone had any experience getting the M-Audio Transit or some other external card to capture AC-3?

I appreciate anyone who can share their expertise. I readily admit I am a
complete noob to all of this, so please let me know if I should just forget about external cards altogether. If there is a possibility that going the external route will create bandwidth issues, require additional compression when using USB 1.0, result in dropped framerates or anything else, please let me know. I am planning on using a RAID 0 system on a Toshiba Qosmio G25 (XP Media Center), so I think I'll be fine, but I honestly don't know for sure.

Thanks so much!

[calinb]

Hi, guys.
Just an update: while the older Audigy cards did not have a "Bit-Accurate" record mode via S/PDIF,
-Erik.
I was considering the Soundblaster Audigy 2 ZS Notebook, but according to this link it does not appear it can record in bit accurate mode:

Furthermore, it does not look like the KX drivers currently support 24/96 with this Audigy 2 card:

I've captured 48/16 AC3 off spdif outputs on my DVD player and my satellite STB with a plain (no Audigy drive) Audigy2 and the KX drivers. I'm not surprised that Creative says it can't be done because they don't provide the software to do it! In most cases, you'll need drivers that ignore SCMS, like KX, and you'll need a utility to strip the null packets from the resulting file, like besplit.exe. The KX driver capture IS bit accurate. If it were resampled, you could not recover and decode the AC3. During KX beta testing of the feature, I did loopback tests with KX and diff'ed the files to prove they were identical. The capture is bit-accurate (lossless), if it's set up correctly.

96/24 support in KX is spotty and I've never tried it.

[mic]

If I remember correctly, there are a couple of threads on bit accurate recording from Lasedisc -- not sure it's possible, but again if I remember correctly. :)

RE: Soundblaster and bit accurate...

Best info source I've found is the driverheaven.net forums. My personal opinion is that bit-acurate spdif recording is possible on *Some* cards... This is with creative or KX drivers -- I've seen good and bad reports with both. There's usually a pretty good variety for any of their models with various OEM & value versions plus updated designs -- hopefully someone has posted yes or no for your model number (on card itself), or with notebook adapter.

As far as Audigy2 goes, simple enough to jumper connector (if it has one) to turn on extra features including controls for bit-accurate transfer to give it a shot. See the above forum for pix & details & better drivers or KX. There's also a simple control panel mod that *might* work.

An alternative if bit-accurate not possible, with the card jumpered should be able to use spdif inputs & decode ac3 with the card and record the results.

[rennervision]

Thanks, mic. I found this thread in driverheaven.net...

http://www.driverheaven.net/showthread.php?t=61929&highlight=notebook

...and it pretty much confirms what I already knew that the KX drivers do not yet support the Audigy 2 ZS Notebook.

So has anyone had any luck capturing AC-3 with a USB or PC sound card for a laptop? (M-Audio's Transit was pretty much my last hope.) Or do I need to start considering internal sound cards for a desktop instead?

[bobcat56458]

I’ve just made a similar post on the “driverheaven.net > kx project audio driver support forum” in hopes of getting some answers to the below questions. I’ve resurrected this old thread because it had some useful information related to what I’m trying to achieve. I would like some information on recording the SPDIF coaxial digital audio from a Sony portable DAT tape recorder, model PCM-M1 to my Windows 98SE computer with an Audigy 1 Platinum EX soundcard, and the kx drivers version 3537. I have heard that it is possible to do bit for bit accurate transferring of the digital audio using the kX drivers. I’ve just ordered a digital I\O cable for the DAT recorder so I’ve not been able to test this feature out yet. I have some project studio DAT tapes that I would like to transfer to the computer for archiving, and editing. My questions are concerning how to best go about that process. What software will I need to record that bit for bit digital audio stream? Will it capture it as a digital stream, or as a wave file? Will it go through a digital to analogue conversion before being recorded? Will I be able to output by SPDIF the edited digital audio back to the DAT unit without an analogue to digital conversion. Any answers to these questions would be greatly appreciated. Also any tips on the kX drivers setup to accomplish this would be very helpful.

[calinb]

What software will I need to record that bit for bit digital audio stream?
Will it capture it as a digital stream, or as a wave file?
Will it go through a digital to analogue conversion before being recorded?
Will I be able to output by SPDIF the edited digital audio back to the DAT unit without an analogue to digital conversion.

Some time ago, I posted a few posts here and at Driverheaven on AC3 howto. Granted, they weren't too specific, but you may find them helpful. Your task will be similar, but you wish to do a full loop (return the edited file back to DAT).
I've never worked with DAT, but you should be able to do this.

I captured AC3 with KX to a wav file.

No conversion should be necessary, if setup correctly.

My tests for bit-accurate indicate you should be able to close the loop w/o losing bit-accuracy.

I created random files with a text editor, put wav headers on them (using besplit, maybe--can't recall) sent them out the Audigy2 SPDIF port and looped them back in the input SPIDIF port. When I diffed the files, after stipping the header, the code was the same. You could test files by doing a "loop and a half" and diff them. DAT >> PC (save file A) >> DAT >> PC (save file B). Then compare files A and B with a diff utility. May neeed to strip the headers and adjust offset with a hex editor first. If B contains the same data as A, you'll be set for bit-accurate conversion and editing.

My tests indicate you should be able to achieve your goals, unless the DAT deck presents problems. Sorry I don't have time to go find the details of my previous experiemtns with KX. Keep us up to date on your progress.

[bobcat56458]

Calinb: Thank you for the reply, and the information. I have not yet received the DAT coaxial digital cable, but I’ve done some tests streaming 2 channel Dolby Digital, and CD audio from a portable DVD player’s coaxial output and it worked fine. I was using the programs: Audacity, Cubase VST CE, and Wavelab lite. Has anyone been able to capture Dolby Digital 5.1 surround sound with any programs? That would be useful as a way of saving the DD5.1 surround sound from Directv Tivo programs that have it, like some HBO movies. I’m looking forward to archiving, and editing the DAT material that I have on tape, I’ll keep you posted.

[calinb]

I was using the programs: Audacity, Cubase VST CE, and Wavelab lite. Has anyone been able to capture Dolby Digital 5.1 surround sound with any programs? That would be useful as a way of saving the DD5.1 surround sound from Directv Tivo programs that have it, like some HBO movies.
As I recall, I captured AC3 and analog s-video from my Dish (JVC) PVR7200 using Virtualdub, Dr.DivX, and ATV2000 (http://mikecrash.wz.cz/)

They all worked fine but it's a bit of a pain 'cause you must demux the captured program, "fix" the AC3 to remove null packets with besplit, and remux with the video (with any audio skew corrections) back to a file. I always wanted to write a DSF that would do realtime AC3fixup, but never had the time. DSPguru suggested that I leverage the AC3fix project code as a start: http://dspguru.doom9.net/ac3fix.rar

Thanks for the update!

[bobcat56458]

I received the digital coaxial SPDIF I\O cable today. I did some quick tests and both the stream from the DAT to the computer Audigy I Platinum EX soundcard, and the stream from the soundcard to the DAT work. I am able to have the kX mixer sliders all in the off position when recording on the computer and the audio recording program still is able to record the stream. This according to the kX manual shows it is recording a bit for bit accurate audio, and not doing any processing. The only doubt I have is that I’m wondering if it is going through a digital to analogue conversion before it gets to the audio recording program? I’m able to monitor the audio using the kX mixer and it sounds like the same audio on the DAT’s headphone output. I’m wondering if it should sound like a white noise kind of sound if it is not going through a digital to analogue conversion. So are my saved wave file computer copies true bit for bit accurate clones of the material on the DAT tapes, any thoughts?

[calinb]

I’m wondering if it should sound like a white noise kind of sound if it is not going through a digital to analogue conversion. So are my saved wave file computer copies true bit for bit accurate clones of the material on the DAT tapes, any thoughts?The DAT tapes contain PCM, so you should expect to hear sound in the monitor. If you monitor a different kind of source, (AC3 from a STB, for example) the montitor will render noise because the AC3 is not being decoded but, rather, played as PCM and stored as a wav file. Undecoded AC3 sounds much less pleasant than white noise, btw!

The only way to know for sure if your recording is bit accurate and error-free is to compare source and target files. You could use a diff utility or a MD5 sum utility like digestit. There are many others. The same check for bit accuracy should be done when ripping audio CDs using digital audio extraction too, btw.

[bobcat56458]

Calinb: Thank you for the reply, I will search the Internet and find digestit and run some tests. I’m also probing the minds of the members at the “driverheaven.net > kx project audio driver support forum” and a member TravelRec. Replied: “The recording path do not pass any processing, since it uses the WMM recording device. You can hear the sound that is coming through the playback device and of course, this is done by D/A conversion. You can proof this, if you feed a different sample rate compared with the selected one in your recording application - the recording will be pitched, ´cause it is not resampled by anything. I used the direct SPDIF recording feature many times for getting the clearest copy available from my old DAT-Tapes.”. So I ran his test and what he said held true. I also asked him, and the other forum members there these questions that you or other Doom9’s members might have some thoughts on: “Would you suggest that all forms of Dithering should be turned off in the computer audio editing program when capturing the digital stream, and that I capture the audio at 16 bits as to not alter the original audio that is 16 bits? On sending the computer audio back to the DAT for re-recording it back to tape. It appears that an analogue to digital conversion is unavoidable as I have to have the kx mixer sliders for "Master Volume", "Wave\PCM", & "Digital Front/SPIFO Output", all activated, and up, to receive an audio signal on the DAT recorder. Am I correct on this assumption? Any other forum members feel free to add your thoughts on these questions also. Any input is greatly appreciated, as I would like to get my DAT tapes transferred correctly the first time, to avoid any additional head wear on the portable DAT recorder, and myself!”. I’m sorry for highjacking this thread about AC-3 capturing, but I would never have gotten to this point in my project without the information, and links, provided here!

[Rockaria]

The only doubt I have is that I’m wondering if it is going through a digital to analogue conversion before it gets to the audio recording program?The stream from spdif-in is exactly the same DIGITAL signal as the original SPDIF transmitter was sending. No analog concept is intervening and the SPDIF link layer has another stream & protocol inside.
The capture device(component) for the spdif-in in general enforces the input stream recognized as 16bit 44.1khz stereo PCM regardless of the actual format, wrapping the stream into a 16bit 44.1khz 2ch wav file when recoding.
It has no problem when the original format is PCM (16bit 44.1k stereo), but for other raw formats such as AC3 or DTS, you will have to use the besplit tool to strip out the wave wrapper.
set id=%1%
BeSplit -core( -input "%id%.wav" -output "%id%.ac3" -type ddwav -fix )
BeSplit -core( -input "%id%.wav" -output "%id%.dts" -type dtswav -fix ) or use the besliced-drop-dts-fix

In graphedit to capture spdif-in:
sound card capture device -> wav dest-> file writer
So if any driver or utility for the card cannot bypass the capture device, to decode the original raw formats, the route explained above(or similar with different recoding tool)) is the only option to take.

[calinb]

Calinb: Thank you for the reply, I will search the Internet and find digestit and run some tests. Travelrec is a KX expert! That's a good suggestion--look for the pitch change. A diff (with a Windows verson of this time-honored unix/linux utility) or a digestit sum, will tell you if there are any errors in the stream too. However, you may need to "line up" the data and remove any offset or headers with a hex editor first. In this respect, my suggestion may prove somewhat difficult. The first several hundred bytes and last several hundred bytes in the file aren't very important. Everything else should match, though. I'll try to find some time to give some thought to your other questions. It appears that your DAT deck is not capable of bit-accurate capture back from a PC source.

[bobcat56458]

Just an update, I’m doing well with my DAT tape to computer by SPDIF project. I captured digitally 10 tapes so far, and no problems with tape dropouts. In an earlier post on this thread I inquired about being able to capture a digital DD5.1 audio stream. I’ve now figured out how to do this with some help from the link below, but I’ve not yet been able to capture both the DD5.1 audio, and the video at the same time.
http://www.videohelp.com/forum/viewtopic.php?t=190946&sid=cdacc09e69042e5e9114793eaeb262db

[calinb]

I’ve now figured out how to do this with some help from the link below, but I’ve not yet been able to capture both the DD5.1 audio, and the video at the same time.[/URL]That's progress, bobcat56458! If you're capturing analog video, say from an s-video output, you should be able to use the aps I mentioned above. Mike Crash's ATV2000 software was the one I frequently used to capture 5.1 AC3 and video from my pizza dish receiver. Howver, though I captured them together to an .avi or .mkv file, the audio still needed to be "fixed" to remove the null packets. That required demuxing, fixing, and remuxing (usually with an audio +/- delay to regain sync.) That was the hassle. Now I capture all my broadcasts in the digital domain via firewire, DVB-S, or ATSC capture cards. Much easier!

[bobcat56458]

I've now figured out how to capture DD5.1 audio and video at the same time. If you are interested how I did this go to the link on my last post here, and on page 2 of that thread you will get the information on how to do this with the kX drivers, and other freeware tools. I did a test capture of 7 minutes from my portable DVD player and had no problems with audio\video sync. Now I have to get a 10 foot optical SPDIF cable so I can try it with my Directv Tivo. I'm hopeing that the tivo outputs SPDIF/RAW digital streams, and not SPDIF/PCM which I can't use to do this.

[tyee]

Anyone know if the M-audio transit will work properly to do this dolby digital and dts capture. I called their tech support and they said it does not resample on it's input. Anyone know for sure?

tyee

[calinb]

I've now figured out how to capture DD5.1 audio and video at the same time. <snip>That's good news, bobcat56458. I've always been able to capture the audio at the same time, but the problem is whether or not I can play it back without additional processing! If the AC3 contains padding/null packets, then it won't play back with any of my players without demuxing it, stripping the nulls, and remuxing it--not to mention the excessive file size with the nulls!

I found that my Echostar JVC 7200 PVR contains nulls in the SPDIF ouput but my Motorola DSR-922 receiver, connected to my 10' C-band BUD, does not contain nulls.

tyee,
I really don't understand all this concern about "resampling" of compressed digital audio. If the card resampled it, I believe the audio would be corrupted and unplayable. You'd know it straight away! Removing null packets is okay, sure, but not resampling the actual "data." In order to resample compressed audio to a compressed audio output, you'd have to decode it, resample it, and then re-encode it. Maybe you realize that resampling results in a unusable recording that's what you're concerned about.

[tyee]

I believe my concern about resampling was regarding a thread about bit perfect 44.1khz dts cd playback. This test is used to confirm non resampling of a soundcard. If it resamples to 48khz, this dts playback will get corrupted.

So, I was wondering if I wanted to capture an ac3 or dts bitstream, (like bobcat56458), into my soundcard via spdif, and it was a creative live card that resamples everything to 48khz, would this also corrupt my capture?? Now that I've written it out, I guess it would not prevent my capture because ac3 and dts are both 48khz on DVD and laserdisc aren't they?

tyee

[calinb]

I believe my concern about resampling was regarding a thread about bit perfect 44.1khz dts cd playback. This test is used to confirm non resampling of a soundcard. If it resamples to 48khz, this dts playback will get corrupted.

So, I was wondering if I wanted to capture an ac3 or dts bitstream, (like bobcat56458), into my soundcard via spdif, and it was a creative live card that resamples everything to 48khz, would this also corrupt my capture?? Now that I've written it out, I guess it would not prevent my capture because ac3 and dts are both 48khz on DVD and laserdisc aren't they?

tyeeI don't have a laserdisc, but yeah, guess it is too. My Audigy cards even do bit accurate 48KHz sampling with the Creative drivers, if there's no SCMS, of course. I think I remember seeing a 44.1 KHz setting too, but it said "PCM only" or something like that. Don't really know why it would care, if I sampled from 44.1KHz DTS CD source instead of PCM. How would it "know," for that matter. :)

[bobcat56458]

Just a note to save someone alse some time. I found out that a DirecTV Tivo outputs a SPDIF/PCM digital audio stream, so that put an end to my hopes of capturing DD5.1 sound from my DirecTV Satellite reciever\Tivo, I don't know if it's the same for EchoStar.

[calinb]

Just a note to save someone alse some time. I found out that a DirecTV Tivo outputs a SPDIF/PCM digital audio stream, so that put an end to my hopes of capturing DD5.1 sound from my DirecTV Satellite reciever\Tivo, I don't know if it's the same for EchoStar.Guess I was somewhat lucky with my old JVC/Echostar 7200 PVR. At least it had AC3 (with SCMS :() at the SPDIF output. KX Drivers handled it okay, but it also had null packets that had to be stripped, as I mentioned above.

I believe there may still be a way to capture your AC3, but it would require an extra PCI card. If the DirectTV audio is unencrypted (as are the music channels on Dish) you could get a DVB-S PCI card that can handle the unencrypted audio streams, like a Twinhan 1020A. They can be found for as little as $75, including shipping. You could get a splitter and send a signal to the PCI card. The place to learn about it is the DVB/mpeg2 forums at www.satforums.com. I've never tried to record a DBS program with a PCI card. My Twinhan 102G can record the AC3 from PBS feeds but the 102G, reportedly, doesn't work with D*.

[rennervision]

Anyone know if the M-audio transit will work properly to do this dolby digital and dts capture. I called their tech support and they said it does not resample on it's input. Anyone know for sure?

tyee

That is the $64,000 question. I've been asking this all over the internet for months. :(

[dodone]

Does anybody know if is it possible to capture an ac3 stream trough SPDIF-in of my terratec aureon 5.1 fun?

[e.lectronick]

The one thing you need to be certain of is whether your card will perform a "Bit-accurate" capture of the ac-3 stream. In other words, will it record the stream as is, without adding any information to the signal (no extra bits or volume modifications). If it will, then you need to capture it as a 48KHz 16 bit .raw file and use BeSplit to remove the padding (null packets sent along with the stream in real time) and restore it to an .AC3 file. Refer to Rockaria's post in this thread from 15th September 2005 23:11 to see how.

-Erik
P.S. it seems that more and more sound cards these days are able to perform "bit-accurate" capture. Some are quite cheap. The M-Audio cards all do this and have terrific sounding converters to boot. I love my 2496

[dodone]

Thank you for your reply but I'm sorry I'm a kind on newbie on audio matters...
I made the capture using graphedit and i got a pcm 44khz file of an ac3 sat movie.
What should I do now? Convert the samplerate and use besplit?
If i use besplit on that file i get a 360KB ac3 file.

[e.lectronick]

If you captured a 44.1khz pcm file, then I don't think what you're trying to do will work. The problem is, your soundcard and software are trying to force the incoming bitstream to fit into the PCM format. 44.1khz 16 bit "Pulse Code Modulation" is the standard for CD digital audio. This is what it normally expects to see coming in through its digital input jacks. Your soundcard is trying to fit a square peg into a round hole. Instead, it needs to be able to simply record exactly what is being input -nothing more, nothing less (this is the bit-for-bit, or "Bit-Accurate" capture, I mentioned before) The ac3 bitstream being sent through the cable into your S/PDIF input jack is a 48khz 16 bit format. Since it is different from PCM, it will not tolerate any interpretation or alteration made by your soundcard. When your soundcard/software receives a PCM signal, it knows just which bits to alter to adjust the signal strength, eq, and other appropriate parameters for optimum sound quality. However, if it tries to do this to a format in which the altered bits aren't the right ones, then the whole sound file gets screwed up and becomes unplayable.
Hope this clarifies things a bit.

Try this: Set your soundcard capture parameters to 48khz, and 16bit, see if there's any setting in the control panel which allows you to do bit-accurate or bit-for-bit capture (or something similar). See if your recording software allows you to save the file as a .raw file. .Raw just means it has no specific format. Using this file, you can then go in with BeSplit, and strip away everything that isn't part of the .ac3 file ("null packets" of non-information are inserted into the realtime stream for the purposes of transmitting through your digital cables. They just fill up space to keep the transimission in real time. They don't belong in the file stored on your HDD).

-Erik

[dodone]

Here again for another question :o
As i'm to buy a new pc, i'm wondering if is there any motherboard with integrated audio with spdf in and multichannel acquisition. I'm looking for an intel775 with 650 or 680 nvidia chipset.
Thanks in advance for any answer :)

[patul]

Here again for another question :o
As i'm to buy a new pc, i'm wondering if is there any motherboard with integrated audio with spdf in and multichannel acquisition. I'm looking for an intel775 with 650 or 680 nvidia chipset.
Thanks in advance for any answer :)

Abit IP35 Pro, but it uses Intel P35 chipset.

[PDub]

I know this is a huge 2 year bump in this thread, but I thought I should let people know this.

I was able to capture 5.1 audio with a audigy 2zs. I used the method of capturing a wav file detailed in this thread and converted it with besplit. I originally tried using audacity, but got stuttering audio after converting. I tried Creative's own WaveStudio to record, and it worked perfectly after converting it with besplit. I tested capturing from two 5.1 sources. One was my digital cable box, and one from my xbox360, which was the main reason for me wanting to do this. Both worked great.

I'm not sure if any of the my settings made a difference, but I will list them for you.

Format in wavestudio- PCM 16-bit 48000 kHz stereo
Through AudioHQ/Device Controls
-Digital Input - DolbyDigital/DTS SPDIF/In Decode
-Bit Accurate Mode - Enable Bit Accurate Recording box checked
-Sampling Rate - 48 kHz(Don't think it matters since it's for Digital Output)
-Decoder - SPDIF Passthrough (I had all the boxes in the "Installed Decoder" box unchecked, such as Dynamic Range Compression, etc, before choosing Passthrough, just in case the Decoder selection doesn't matter either.

Hope this helps people. If anyone wants to hear a sample, point me to an ac3 cutter so I can trim the file down.

[davidhorman]

I've written a simple program to record AC3 over SPDIF because the audio editors I tried insisted on mangling the bits when it came to saving the file. It removes all padding and checks the CRCs of each AC3 frame. If anyone wants the source code, or a modified .exe that isn't specific to my soundcard, PM me or post here. It will only work on 48kHz 384kbit/s streams though.

David