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[MAC] 7.1 separate channel editor?

Kal-El

Kal-El

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Does anyone have any software that allows you to edit channels separately?
When I open my m2ts file (with 7.1) in Audacity, I only get 6 channels.
I looked into it, and apparently if you have ffmpeg Audacity can open 5.1 channels but it can't open 7.1?
It looks as if it's joining (or ditching) the leftover channels and only shows 6 of them?

Does anyone know of any software (MAC preferably) that can handle 7.1 channel files?
 
Kal-El said:
Does anyone have any software that allows you to edit channels separately?
When I open my m2ts file (with 7.1) in Audacity, I only get 6 channels.
I looked into it, and apparently if you have ffmpeg Audacity can open 5.1 channels but it can't open 7.1?
It looks as if it's joining (or ditching) the leftover channels and only shows 6 of them?

Does anyone know of any software (MAC preferably) that can handle 7.1 channel files?
Click to expand...

What is the filetype? DTS? You can always convert to mono WAV files (8?).
 
ThrowgnCpr said:
What is the filetype? DTS? You can always convert to mono WAV files (8?).
Click to expand...
It's DTS-HD. It's not so much that I can't convert but that I can't convert enough.
I only see 6 channels in Audacity, but there ARE 8. When I put my BDMV file in MakeMKV it clearly shows 8:

SCbrvAo.png


It's 1536 kb/s according to VLC (stream 1). The others are the 5.1 audio channels, subtitles, menu audio bits etc.
So I do Open Disc > Open BDMV Folder and then go to my folder.
It finds it, and it can open it just fine. Everything's accounted for.
When I drag the stream 1 m2ts file into Audacity that works too.
I then get a little pop-up asking me which stream I want.
The top one says 1536 (the bits per second) so I guess that's the one I need because all the other ones are 640 bits per second.

Sg1nqbS.png


I'm not really in the mood to waste HDD space on a 1:1 MKV copy of my BR rip with the FLAC file I need, which I then have to extract AGAIN into WAV files so I can start work on them.

Is there a way to skip all of that and get a tool that reads the 8.1 channels the way it's supposed to?

Btw no rush since I'm only experimenting with a next edit. Nothing major. Just getting the feel of things in case my edit is approved.

This is what I get in Audacity:

cahzsfo.png
 
I think I may be getting somewhere with this...
Set up my Windows VM to work with eac3to and downloaded all of the plugins (mkvmerge, tsmuxer, xport, etc).
Set the source folder of my BDMV stream to be shared on my VM and then ran eac3to.
Now I've got 8 different wav channels being extracted straight onto my Mac HDD rather than my VM HDD (because of the shared folder) and for now they look to be +1GB per channel. Don't know where they'll end up in size and if it will work.

Will post back here with results. Might be a bit of a cumbersome workaround but couldn't care less if this works :)

UPDATE: Here's the log:

eac3to v3.29
command line: "C:\Users\Caviar Admin\Desktop\eac3to329\eac3to.exe" "G:\00001.m2ts" 2: "G:\mosoutput.wavs" -16 -44100 -8
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 12 subtitle tracks, 2:23:03, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: DTS Master Audio, English, 7.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
3: AC3, English, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
4: AC3, French, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
5: AC3, Italian, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
6: AC3, Spanish, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
7: Subtitle (PGS), English
8: Subtitle (PGS), French
9: Subtitle (PGS), Italian
10: Subtitle (PGS), Vietnamese
11: Subtitle (PGS), Dutch
12: Subtitle (PGS), Danish
13: Subtitle (PGS), Finnish
14: Subtitle (PGS), Icelandic
15: Subtitle (PGS), Norwegian
16: Subtitle (PGS), Swedish
17: Subtitle (PGS), French
18: Subtitle (PGS), Italian
[a02] dts, 48000, 7.1
[a02] Extracting audio track number 2...
[a02] Decoding with libDcaDec DTS Decoder...
[a02] Writing WAVs...
[a02] Creating file "G:\mosoutput.L.wav"...
[a02] Creating file "G:\mosoutput.R.wav"...
[a02] Creating file "G:\mosoutput.C.wav"...
[a02] Creating file "G:\mosoutput.LFE.wav"...
[a02] Creating file "G:\mosoutput.BL.wav"...
[a02] Creating file "G:\mosoutput.BR.wav"...
[a02] Creating file "G:\mosoutput.SL.wav"...
[a02] Creating file "G:\mosoutput.SR.wav"...
[a02] The original audio track has a constant bit depth of 24 bits.
Video track 1 contains 205777 frames.
eac3to processing took 16 minutes, 40 seconds.
Done.
Captain Khajiit you're quite knowledgeable on all things aec3to. Does that look okay to you? Plan was to extract to lossless wavs. Crap... just noticed I extracted to 16bit rather than the original 24... I suppose that doesn't matter that much?
 
I would always point eac3to to a playlist rather than an M2TS (unless there is a good reason not to do so). Go here.

The log looks all right. Why are you resampling your audio? Leave it at 48kHz.

... just noticed I extracted to 16bit rather than the original 24...
Click to expand...

How can you say that you just noticed when you tried to specify 16 bits by adding -16? With up-to-date versions of eac3to, the switch should really be -down16. Check your WAVs in MediaInfo and see if they are 16- or 24-bit files.

You might notice little difference, but it never hurts to retain high quality throughout the editing process, and you will probably need to run eac3to again to keep the original sampling-rate anyway. Make sure that your NLE can support 24-bit audio and 7.1. If it cannot support the latter, add -down6.
 
Captain Khajiit said:
I would always point eac3to to a playlist rather than an M2TS (unless there is a good reason not to do so). Go here.

The log looks all right. Why are you resampling your audio? Leave it at 48kHz.



How can you say that you just noticed when you tried to specify 16 bits by adding -16? With up-to-date versions of eac3to, the switch should really be -down16. Check your WAVs in MediaInfo and see if they are 16- or 24-bit files.

You might notice little difference, but it never hurts to retain high quality throughout the editing process, and you will probably need to run eac3to again to keep the original sampling-rate anyway. Make sure that your NLE can support 24-bit audio and 7.1. If it cannot support the latter, add -down6.
Click to expand...

More specifically I meant that I had overlooked the fact it was 24bit originally. I set a few settings manually to be certain I'd get the right output. That also explains the resampling. I'll start another pass. I knew I should talk to you, thanks for the info! :)
 
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