Capture 5.1 through S/PDIF

[ENunn]

I have this old Sound Blaster X-Fi Extreme with Toslink in and for funsies I want to try to record 5.1 audio. I know you have to record 24 bit audio but what else? Just tried Audacity and I don't know if I did it right. Is this what it's supposed to look like?


I tried decoding an exported 24-bit wav with BeSweet but it's really hard to use the GUI. Stuff disappears and when I did get it working it errors out.

I checked the MediaInfo on the exported wav and even though I had Audacity export a 24-bit file, it keeps exporting a 16-bit file. Don't know if that's normal or not.

General
Complete name                            : D:\recordings\5.1 test 96k.wav
Format                                   : Wave
File size                                : 2.48 MiB
Duration                                 : 4 s 514 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 4 608 kb/s

Audio
Format                                   : AC-3
Format/Info                              : Audio Coding 3
Format settings                          : Little
Muxing mode                              : SMPTE ST 337
Codec ID                                 : 1
Duration                                 : 4 s 514 ms
Bit rate mode                            : Constant
Bit rate                                 : 4 608 kb/s
Sampling rate                            : 96.0 kHz
Bit depth                                : 16 bits
Compression mode                         : Lossy
Stream size                              : 2.48 MiB (100%)

I would record 5.1 through HDMI through my Avermedia card but it doesn't seem detect 5.1 through my cable box's HDMI out, so I don't know if it's really outputting 5.1. I recorded this channel through firewire last year and it did have a 5.1 track though, so I'm not sure what's going on.

[usually_quiet]

I have this old Sound Blaster X-Fi Extreme with Toslink in and for funsies I want to try to record 5.1 audio. I know you have to record 24 bit audio but what else? Just tried Audacity and I don't know if I did it right. Is this what it's supposed to look like?


I tried decoding an exported 24-bit wav with BeSweet but it's really hard to use the GUI. Stuff disappears and when I did get it working it errors out.

I checked the MediaInfo on the exported wav and even though I had Audacity export a 24-bit file, it keeps exporting a 16-bit file. Don't know if that's normal or not.

General
Complete name                            : D:\recordings\5.1 test 96k.wav
Format                                   : Wave
File size                                : 2.48 MiB
Duration                                 : 4 s 514 ms
Overall bit rate mode                    : Constant
Overall bit rate                         : 4 608 kb/s

Audio
Format                                   : AC-3
Format/Info                              : Audio Coding 3
Format settings                          : Little
Muxing mode                              : SMPTE ST 337
Codec ID                                 : 1
Duration                                 : 4 s 514 ms
Bit rate mode                            : Constant
Bit rate                                 : 4 608 kb/s
Sampling rate                            : 96.0 kHz
Bit depth                                : 16 bits
Compression mode                         : Lossy
Stream size                              : 2.48 MiB (100%)

I would record 5.1 through HDMI through my Avermedia card but it doesn't seem detect 5.1 through my cable box's HDMI out, so I don't know if it's really outputting 5.1. I recorded this channel through firewire last year and it did have a 5.1 track though, so I'm not sure what's going on.

I can't answer your questions about your sound card and Audacity or BeSweet.

Most video capture devices can only capture stereo audio. Only a few video capture devices can capture 5.1 audio. A handful of older Hauppauge video capture devices could capture 5.1 channel AC3 using Toslink or HDMI with the right software.

There was also a discontinued Korean HDMI video capture device that could capture 5,1 channel AC3 and there is a Magewell HDMI video capture device that can capture 5.1 channel AC3 audio but the process requires a fair amount of technical expertise. https://forum.videohelp.com/threads/392907-Is-there-a-lossless-video-capture-card-that...ures-5-1-audio

Note that a cable box's HDMI or Toslink audio output is often controlled by settings in its internal menus. The older Xfinity box that I had provided a choice between 2 channel LPCM or the AC3 audio in the broadcast.

[Cornucopia]

Toslink (aka optical fiber-based SPDIF, which is a consumer variant of AES3) only supports a limited bitrate (~2.5Mbps). So it is not capable of carrying 5.1+ multichannel audio in uncompressed form.
It is capable of carrying COMPRESSED (ac3 or dts) MC audio, but only when that compressed stream has been re-packetized to match the data rate of a standard stereo uncompressed stream. And then, it masquerades itself AS such a stereo stream.

So to capture, you would need to:
1. Record the 2 ch incomming "lpcm" stream
2. Save the stream as a "wav"
3. Convert the "wav" (aka repacketized compressed file) to a normal compressed file. So, dts-wav -> dts, or ac3-wav -> ac3.

Then you will have your normal 5.1. It will be compressed, no getting around that when using toslink.
Note that the removal of the null packets does not change the compressed data itself in any way, so no further degradation or loss.



Scott

[usually_quiet]

Toslink (aka optical fiber-based SPDIF, which is a consumer variant of AES3) only supports a limited bitrate (~2.5Mbps). So it is not capable of carrying 5.1+ multichannel audio in uncompressed form.
It is capable of carrying COMPRESSED (ac3 or dts) MC audio, but only when that compressed stream has been re-packetized to match the data rate of a standard stereo uncompressed stream. And then, it masquerades itself AS such a stereo stream.

So to capture, you would need to:
1. Record the 2 ch incomming "lpcm" stream
2. Save the stream as a "wav"
3. Convert the "wav" (aka repacketized compressed file) to a normal compressed file. So, dts-wav -> dts, or ac3-wav -> ac3.

Then you will have your normal 5.1. It will be compressed, no getting around that when using toslink.
Note that the removal of the null packets does not change the compressed data itself in any way, so no further degradation or loss.



Scott

The 5.1 channel AC3 audio masquerading as 2 channel PCM WAV file doesn't sound like normal audio when played does it? If so, it should be possible to tell if 2 channel PCM audio is being captured by playing it.

There is one thing about the Hauppauge HDMI capture devices that capture AC3 that I forgot to mention. If they receive 2-channel LPCM audio, it is captured in AAC format rather than WAV.

[ENunn]

Most video capture devices can only capture stereo audio. Only a few video capture devices can capture 5.1 audio. A handful of older Hauppauge video capture devices could capture 5.1 channel AC3 using Toslink or HDMI with the right software.

My Live Gamer 4k supports recording at 5.1 in RECentral, don't know if there's a way to unlock that through other software. The cable box doesn't seem to want to output 5.1 even though I'm on a supported channel/show/etc. It's not worth it to record in RECentral anyways because you can't record interlaced on the LG4K so :P

Toslink (aka optical fiber-based SPDIF, which is a consumer variant of AES3) only supports a limited bitrate (~2.5Mbps). So it is not capable of carrying 5.1+ multichannel audio in uncompressed form.
It is capable of carrying COMPRESSED (ac3 or dts) MC audio, but only when that compressed stream has been re-packetized to match the data rate of a standard stereo uncompressed stream. And then, it masquerades itself AS such a stereo stream.

So to capture, you would need to:
1. Record the 2 ch incomming "lpcm" stream
2. Save the stream as a "wav"
3. Convert the "wav" (aka repacketized compressed file) to a normal compressed file. So, dts-wav -> dts, or ac3-wav -> ac3.

Then you will have your normal 5.1. It will be compressed, no getting around that when using toslink.
Note that the removal of the null packets does not change the compressed data itself in any way, so no further degradation or loss.

I'm not worried about it being compressed or uncompressed.
What should I used to convert the wav though? I've been exporting as a 24-bit WAV and on some recordings just sends a 24-bit PCM file according to MediaInfo.

[Cornucopia]

Toslink only supports up to 20bit LPCM. A compressed stream, such as ac3 or dts, is gonna be distinct from and independent of that limit. But by and large, most compressed streams (and the encoders that create them) seem to have 16bit source limitations, so am not sure you'd really be getting much more by trying to push for end to end 24bit support.
Not only that, but you say you don't care it's compressed, but that compression could be considered to be doing more damage to signal quality than the limiting from 24 to 16bit.

@usually_quiet, yes capture and direct, non-decoded playback would yield what would most likely be called digital "NOISE".


Scott

[Cornucopia]

Bsconvert used to be able to do it.


Scott

[usually_quiet]

The cable box doesn't seem to want to output 5.1 even though I'm on a supported channel/show/etc.

With my older Comcast Xfinity cable box, the HDMI audio settings (2-channel PCM or pass-through (for AC3)) were accessed via a hidden setup menu. I think there was a set-up menu for Toslink as well. Have you checked online to see if there is any guidance available from your provider about settings that control the cable box's audio output for its digital connections?

[ENunn]

Sorry for the months late reply

With my older Comcast Xfinity cable box, the HDMI audio settings (2-channel PCM or pass-through (for AC3)) were accessed via a hidden setup menu. I think there was a set-up menu for Toslink as well. Have you checked online to see if there is any guidance available from your provider about settings that control the cable box's audio output for its digital connections?

Yep, nothing. There's some devices option in the settings but it doesn't really show anything.


For the TOSLINK output, if I change the audio coding format to Stereo, audio is fine but it's sped up. Everything else is just typical digital buzzing or whatever.

Bsconvert used to be able to do it.

I really don't know what I'm doing here. I dragged my WAV file onto bsconvert and nothing happened.

General
Complete name                            : D:\recordings\toslinktest.wav
Format                                   : Wave
File size                                : 3.42 MiB
Duration                                 : 6 s 219 ms
Overall bit rate                         : 4 609 kb/s

Audio
Format                                   : PCM
Codec ID                                 : 1
Duration                                 : 6 s 219 ms
Bit rate                                 : 4 608 kb/s
Channel(s)                               : 2 channels
Sampling rate                            : 96.0 kHz
Bit depth                                : 24 bits
Stream size                              : 3.42 MiB (100%)

[ENunn]

Over a year later (wow already) and I'm still trying to find a good solution. I asked on another site and someone suggested me to get an Arduino based analyzer with a plugin for some software. https://github.com/pfrench42/saleae_spdif

I'll try it out once I get the analyzer but this looks incredibly complicated to set up.

I did another test recording on a channel that's 5.1 but I still cannot decode anything I record. Do I just not have the right SB for this? I don't know what I'm doing wrong. I'm capturing 48k/24bit with Audacity and decoding with BeSweet. Still jumbled garbage. Unless it's just DD ruining it.


[Attachment 60461 - Click to enlarge]

General
Complete name                            : D:\recordings\test toslink 8 26 21-2.wav
Format                                   : Wave
File size                                : 14.6 MiB
Duration                                 : 53 s 60 ms
Overall bit rate                         : 2 304 kb/s

Audio
Format                                   : PCM
Codec ID                                 : 1
Duration                                 : 53 s 60 ms
Bit rate                                 : 2 304 kb/s
Channel(s)                               : 2 channels
Sampling rate                            : 48.0 kHz
Bit depth                                : 24 bits
Stream size                              : 14.6 MiB (100%)

Can't seem to upload the file as an attachment right now so Google Drive it is: https://drive.google.com/file/d/10s_yX-YgiK7Uxu5x16xilRaFr0e5YMyt/view?usp=sharing

Does anyone have any other solutions?

[ENunn]

I was able to find a solution. I had to route my Sound Blaster through ASIO4All and record a 48k 16-bit WAV through Reaper, then decode with bsconvert. I have found no other working solution. I tried recording directly with both Amarec and Virtualdub2 and they both produce invalid files. ffmpeg doesn't work either nor does GraphStudioNext. If anyone has any other solutions, please let me know.

My issue now is syncing. Most of the time I'm able to sync fine, but Spectum usually inserts other ads during the commercials, which causes the capture to stop momentarily. This makes syncing the whole thing impossible. Is there any way I can get around this? Can I add padding to the affected areas? Any help is appreciated.

[ENunn]

I was able to find a solution. I had to route my Sound Blaster through ASIO4All and record a 48k 16-bit WAV through Reaper, then decode with bsconvert. I have found no other working solution. I tried recording directly with both Amarec and Virtualdub2 and they both produce invalid files. ffmpeg doesn't work either nor does GraphStudioNext. If anyone has any other solutions, please let me know.

My issue now is syncing. Most of the time I'm able to sync fine, but Spectum usually inserts other ads during the commercials, which causes the capture to stop momentarily. This makes syncing the whole thing impossible. Is there any way I can get around this? Can I add padding to the affected areas? Any help is appreciated.

Still looking for a solution. Just had a dropout recording a channel and now syncing the entire video is pretty much impossible now. I tried converting with both bsconvert and BeSweet, any dropouts will just skip the blank space and go to the next audio frame, which for my purposes I don't want. I tried looking at stuff like DelayCut and I really don't know what I'm supposed to do, if that'll even help. If anyone has any ideas, please share.

[sebus]

The easiest I found (and tested quite few units/solutions) is to use cheap Hifime UR23 SPDIF Optical to USB converter and Audacity build with ASIO.
On Windows 10 it requires nothing beyond plug, play, record and the resulting .wav processed by ffmpeg as per this

sebus

[ENunn]

The easiest I found (and tested quite few units/solutions) is to use cheap Hifime UR23 SPDIF Optical to USB converter and Audacity build with ASIO.
On Windows 10 it requires nothing beyond plug, play, record and the resulting .wav processed by ffmpeg as per this

sebus

I'll consider picking up that box, thanks.

I did a test capture with my Sound Blaster through Amarec and exported the audio in Virtualdub, but it still refuses to decode properly. I don't know why Reaper with ASIO4ALL is somehow the only program that'll work properly. I'm capturing at 48k 16-bit, I really don't see why it shouldn't work. I'll probably pick up the Hifime, although do I still have to deal with syncing issues? If so, then I'll still be in the same boat as I am now. I can record the signal just fine but if the channel goes out or if a local ad comes on, then it's nearly impossible to sync audio with the video without doing more pointless work.

[ralfito]

The easiest I found (and tested quite few units/solutions) is to use cheap Hifime UR23 SPDIF Optical to USB converter and Audacity build with ASIO.
On Windows 10 it requires nothing beyond plug, play, record and the resulting .wav processed by ffmpeg as per this

sebus

sebus,
it seems I am too stupid for this. I have the UR23 running (working fine in Stereo), but AC3 is giving me grief. In Audacity, what host do you select? The UR23 does not support ASIO, does it? Any other options I select, gets me a "scrambled" noise signal (I guess that is normal at this stage?). When running ffmpeg, all I seem to achieve is a

Input #0, spdif, from 'input.wav':
  Duration: N/A, bitrate: N/A
Output #0, ac3, to 'out_audio_ac3.ac3':
Output file #0 does not contain any stream

Anything special I having to consider when saving that .wav file?

Thanks so much
ralfito

[ralfito]

So after 3 days in, here's what is finally working for me (me thinks that everyone is having different systems with different obstacles):

With the UR23, no need for ASIO; Windows WASAPI works just fine.
What really made it work, were the quality settings in Audacity: Set the Real-time Conversion settings to Best Quality and Dither None. Voila!
I am running the UR23 at 48 kHz, 24-bit. Equal settings in Audacity, saving WAV as 16-bit PCM.
MediaInfo was a great help.

ralfito

[sebus]

Best is to build Audacity with ASIO support as per this:

https://gist.github.com/SteveALee/da24c2be633340b8791066dd98eb5d0b?permalink_comment_i...omment-4430980