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Old 16th September 2005, 04:11   #21  |  Link
Rockaria
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Quote:
Originally Posted by bobcat56458
The only doubt I have is that Iím 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.
Quote:
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.

Last edited by Rockaria; 16th September 2005 at 04:28.
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Old 17th September 2005, 05:50   #22  |  Link
calinb
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Quote:
Originally Posted by bobcat56458
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.
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Old 29th September 2005, 01:50   #23  |  Link
bobcat56458
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Just an update, Iím 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. Iíve now figured out how to do this with some help from the link below, but Iíve not yet been able to capture both the DD5.1 audio, and the video at the same time.
http://www.videohelp.com/forum/viewt...14793eaeb262db
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Old 29th September 2005, 09:12   #24  |  Link
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Quote:
Originally Posted by bobcat56458
Iíve now figured out how to do this with some help from the link below, but Iíve 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!
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Old 30th September 2005, 22:40   #25  |  Link
bobcat56458
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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.
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Old 14th November 2005, 17:58   #26  |  Link
tyee
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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
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Old 16th November 2005, 20:57   #27  |  Link
calinb
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Quote:
Originally Posted by bobcat56458
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.
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Old 16th November 2005, 23:43   #28  |  Link
tyee
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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?

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Old 17th November 2005, 00:13   #29  |  Link
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Quote:
Originally Posted by 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
I 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.
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Old 18th November 2005, 12:30   #30  |  Link
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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.
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Old 18th November 2005, 18:06   #31  |  Link
calinb
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Quote:
Originally Posted by 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.
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*.

Last edited by calinb; 18th November 2005 at 18:14.
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Old 4th December 2005, 03:03   #32  |  Link
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Quote:
Originally Posted by 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
That is the $64,000 question. I've been asking this all over the internet for months.
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Old 7th June 2007, 02:37   #33  |  Link
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Does anybody know if is it possible to capture an ac3 stream trough SPDIF-in of my terratec aureon 5.1 fun?
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Old 7th June 2007, 17:00   #34  |  Link
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Give it a try

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
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Old 8th June 2007, 16:01   #35  |  Link
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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.

Last edited by dodone; 8th June 2007 at 16:12.
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Old 11th June 2007, 15:42   #36  |  Link
e.lectronick
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I don't think it'll work

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).

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Old 24th October 2007, 15:35   #37  |  Link
dodone
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Here again for another question
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
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Old 25th October 2007, 08:15   #38  |  Link
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Quote:
Originally Posted by dodone View Post
Here again for another question
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.
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Old 11th September 2009, 21:35   #39  |  Link
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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.
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Old 14th October 2009, 17:57   #40  |  Link
wonkey_monkey
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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
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