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LD AC-3 Capture Help
#1
I have a Nakamichi DE-1 RF Demodulator, and a Audiophile 2946. I was able to easily capture the digital 2.0 LPCM over coax digital-out with Audacity. But whenever I try to get 5.1 it just never works right for me.

I've seen a few people on this forum say they've used this particular soundcard so maybe they can help.

My M-Audio Control panel for my soundcard will display that SPDIF meters always at max volume (don't know if this is good or bad?) and has the sync source set to locked and correctly picks up the audio Hz at 48000

I've tried recording it in multiple different softwares and none seem to work, at most I can get it 2 record 2 channels with white noise (but not 5.1 channels).

If anyone could help with this with steps, programs, etc used it would help greatly. My worst fear is that the AC-3 Demodulator might not be working correctly  Dodgy
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#2
The AC-3 output of a laserdisc demodulator is a 384kb/s signal 'padded' within a 48kHz 16bit PCM stream. If you were to zoom in on the captured stream in audio editing software you will see little packets of information every 32ms with silence inbetween, these are the AC-3 frames.
Once the stream (ddwav) is captured you will have to convert it to a true AC-3 file using besplit. This is when you find out whether your method is bitperfect or not!
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#3
(2018-08-27, 07:02 PM)zoidberg Wrote: The AC-3 output of a laserdisc demodulator is a 384kb/s signal 'padded' within a 48kHz 16bit PCM stream. If you were to zoom in on the captured stream in audio editing software you will see little packets of information every 32ms with silence inbetween, these are the AC-3 frames.
Once the stream (ddwav) is captured you will have to convert it to a true AC-3 file using besplit. This is when you find out whether your method is bitperfect or not!

Will there captured stream only show 2 channels of audio while recording though?

https://forum.fanres.com/thread-82-post-...l#pid21433
The person here has said they've captured AC3 successfully with this soundcard so I don't think thats the issue.
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#4
Yes the recorded stream will be stereo. Remember you're not capturing the AC-3 directly just it's container, you have to extract it with besplit
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#5
This is what the waveform looks like while recording: https://i.imgur.com/m6wwVxC.png

and BeSplit output is this

[00:00:00:000] +------- BeSplit -----
[00:00:00:000] | Input : test.wav
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] | Channels Count: 3, Bitrate: 640kbps
[00:00:00:000] | Output : test2.ac3
[00:00:00:000] +---------------------
[00:00:00:000] | Writing test2.ac3
[00:00:00:000] +---------------------
[00:00:00:000] Operation Completed !
[00:00:00:000] <-- Process Duration

-------------
Which for some reason seems to get different bitrate/channels each time I do it.
What software do you use for your recordings? and what exact besplit command.
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#6
The besplit command line should look like this:

Code:
BeSplit -core( -input input.wav -output output.ac3 -type ddwav -fix )

You have to capture to 48kHz 16bit and be sure that no part of the chain is resampling/dithering the audio. I use a m-audio usb device and capture with reaper software.
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#7
(2018-08-27, 07:44 PM)Phobicpanda Wrote: This is what the waveform looks like while recording: https://i.imgur.com/m6wwVxC.png

and BeSplit output is this

[00:00:00:000] +------- BeSplit -----
[00:00:00:000] |  Input : test.wav
[00:00:00:000] |  Source Sample-Rate: 48.0KHz
[00:00:00:000] |  Channels Count: 3, Bitrate: 640kbps
[00:00:00:000] |  Output : test2.ac3
[00:00:00:000] +---------------------
[00:00:00:000] |  Writing test2.ac3
[00:00:00:000] +---------------------
[00:00:00:000] Operation Completed !
[00:00:00:000] <-- Process Duration

-------------
Which for some reason seems to get different bitrate/channels each time I do it.
What software do you use for your recordings? and what exact besplit command.

If you're getting anything other than "Channels Count: 5, Bitrate: 384kbps" it means that there is likely to be some resampling or sync issue happening. For some reason BeSplit only reports 5 channels, even thought it's really 6. My advice is to go in and change some of the settings in your M-Audio control panel before recording. I've found that changing settings and the returning to the correct format right before recording can help fix sync issues with my M-Audio card.

You can also use bsconvert.exe (here: http://www.ac3filter.net/wiki/AC3Filter_tools#bsconvert) to get your ac3 stream unpacked. I like it because it's less typing. ex:

bsconvert input.wav output.ac3
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#8
(2018-08-27, 10:52 PM)schorman Wrote:
(2018-08-27, 07:44 PM)Phobicpanda Wrote: This is what the waveform looks like while recording: https://i.imgur.com/m6wwVxC.png

and BeSplit output is this

[00:00:00:000] +------- BeSplit -----
[00:00:00:000] |  Input : test.wav
[00:00:00:000] |  Source Sample-Rate: 48.0KHz
[00:00:00:000] |  Channels Count: 3, Bitrate: 640kbps
[00:00:00:000] |  Output : test2.ac3
[00:00:00:000] +---------------------
[00:00:00:000] |  Writing test2.ac3
[00:00:00:000] +---------------------
[00:00:00:000] Operation Completed !
[00:00:00:000] <-- Process Duration

-------------
Which for some reason seems to get different bitrate/channels each time I do it.
What software do you use for your recordings? and what exact besplit command.

If you're getting anything other than "Channels Count: 5, Bitrate: 384kbps" it means that there is likely to be some resampling or sync issue happening. For some reason BeSplit only reports 5 channels, even thought it's really 6. My advice is to go in and change some of the settings in your M-Audio control panel before recording. I've found that changing settings and the returning to the correct format right before recording can help fix sync issues with my M-Audio card.

You can also use bsconvert.exe (here: http://www.ac3filter.net/wiki/AC3Filter_tools#bsconvert) to get your ac3 stream unpacked. I like it because it's less typing. ex:

bsconvert input.wav output.ac3

Thanks for the bsconvert link.
You have this M-Audio card (or a similar one?). Could you tell me what settings work for you and or which ones you are referring to changing before recording?

My control panel is really simplistic and doesn't have many options, even the one in the manual has more options lol.
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#9
Hi, not sure if it makes a difference, but I use Sound Forge rather than Audacity to capture. As mentioned already, you have to capture at 16/48, but the M-audio control panel also has to be set at 48kHz, and it should also have a "sync source" setting that needs to be set to "External - SPDIF" (this setting is also required for external PCM capture).

Oh, and have you confirmed that your demodulator works (like when it's connected to a receiver)?
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#10
(2018-08-28, 03:12 AM)BusterD Wrote: Hi, not sure if it makes a difference, but I use Sound Forge rather than Audacity to capture. As mentioned already, you have to capture at 16/48, but the M-audio control panel also has to be set at 48kHz, and it should also have a "sync source" setting that needs to be set to "External - SPDIF" (this setting is also required for external PCM capture).

Oh, and have you confirmed that your demodulator works (like when it's connected to a receiver)?

Yea, the sync source is set to External SPDIF which makes it go to "locked" status.
I don't have a receiver to test it on, but the fact that it is recognizing the 48kHz and actually feeding it a signal should mean that its working I think.

My lord... Sound Forge worked.
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