EVOB De/Multiplexers

[Deckard2019]

No, this will never work!!!

Ok. Wrong copy/paste (I don't use this method as I use BeHappy to bypass RAW/WAV file creation).
But what about ffdshow ? I still don't understand why/when it's used in eac3toac3 ...

[madshi]

Originally Posted by Deckard2019

But what about ffdshow ? I still don't understand why/when it's used in eac3toac3 ...

The reason is that the Sonic Audio Decoder is very unreliable without adding ffdshow to the processing stages. I found that the Sonic Audio Decoder sometimes outputs 16 bit, sometimes 24 bit. Furthermore more often then not only 2 channels contained valid information while the remaining 4 channels were garbage. This all doesn't happen with ffdshow in the chain. Don't ask me why. I don't really know.

[Deckard2019]

Originally Posted by madshi

The reason is that the Sonic Audio Decoder is very unreliable without adding ffdshow to the processing stages. I found that the Sonic Audio Decoder sometimes outputs 16 bit, sometimes 24 bit. Furthermore more often then not only 2 channels contained valid information while the remaining 4 channels were garbage. This all doesn't happen with ffdshow in the chain. Don't ask me why. I don't really know.

Ok. Thank you. The 4 crappy channels thing happened to a friend of mine.
With same source, same tools, it doesn't happen to me ...

Outputting directly AC3 from ffdshow is still considered as a wrong method ?
(For now, I use BeHappy/Aften with .grf embedded in avs script. grf only contains orbitlee filter and sonic one).

[MichalHabart]

Originally Posted by Deckard2019

Ok. Thank you. The 4 crappy channels thing happened to a friend of mine.
With same source, same tools, it doesn't happen to me ...

Outputting directly AC3 from ffdshow is still considered as a wrong method ?
(For now, I use BeHappy/Aften with .grf embedded in avs script. grf only contains orbitlee filter and sonic one).

Yes, ffdshow can be used only for changing bit depth of output. AC3 created directly by ffdshow has wrong channel order

[Jack-Bauer]

Originally Posted by MichalHabart

AC3 created directly by ffdshow has wrong channel order

Matrix filter can be inserted in beetween sonic and ffdshow, allowing to set its matrix as desired.

Jack

[zgx]

Originally Posted by MichalHabart

Well, LPCM does not seem to work at the moment. With xport or graphedit i get raw file with same size but when i use sox to create wav, i receive 77 minutes instead of 100 and the sound inside is horrible (sounds like when you call to fax machine )

Does anyone manage to get it work?

I did get it to work with Graphedit. I selected the m2ts file with File Source (Async) and selected the LPCM track in Sonic HD Demuxer.

File Source (Async) -> Sonic HD Demuxer -> Sonic Cinemaster Audio Decoder 4.2 -> Dump

Then I converted the raw file to wav and got a working file:
sox -r48000 -t .raw -c 6 -3 -s file.raw file.wav


I'm now trying out the new version of xport (since I don't trust Sonic Audio Decoder). It finds a LPCM track and dumps it:

LPCM Audio Mode = 3/2+lfe
LPCM Audio Bits/sample = 16
LPCM Audio Sample Rate = 48000

But don't know what to do with the dump. Have tried all sorts of options in Sox and BeSweet but I can't convert it to a working wav.

[MichalHabart]

Originally Posted by zgx

I did get it to work with Graphedit. I selected the m2ts file with File Source (Async) and selected the LPCM track in Sonic HD Demuxer.

File Source (Async) -> Sonic HD Demuxer -> Sonic Cinemaster Audio Decoder 4.2 -> Dump

Then I converted the raw file to wav and got a working file:
sox -r48000 -t .raw -c 6 -3 -s file.raw file.wav


I'm now trying out the new version of xport (since I don't trust Sonic Audio Decoder). It finds a LPCM track and dumps it:

LPCM Audio Mode = 3/2+lfe
LPCM Audio Bits/sample = 16
LPCM Audio Sample Rate = 48000

But don't know what to do with the dump. Have tried all sorts of options in Sox and BeSweet but I can't convert it to a working wav.

Well, that is exactly the problem i have. but it seems that raw dump is not working, i should get 5GB file but i get only 3,5GB. So thatswhy my soxed wav is not going to work. have you tried to load it to wavewizard? it should be able to tell you length and channels in that wav file.

[drmpeg]

The LPCM output of xport is big endian (just like it's stored in the .m2ts file).

This sequence works:

xport -h2 movie.m2ts 1 1 2

sox -B -r48000 -t .raw -c 2 -2 -s bits0001.mpa bits.wav

besweet -core( -input bits.wav -output bits.ac3 ) -ac3encode

This sequence fails:

xport -h movie.m2ts 1 1 2

sox -B -r48000 -t .raw -c 6 -2 -s bits0001.mpa bits.wav

besweet -core( -input bits.wav -output bits.ac3 ) -ac3encode ( -6ch )

Error 58: Error : Unknown Input-File Format : "bits.wav".

Seems to me that sox can't create a 6 channel .wav file that besweet understands.

Ron

[madshi]

Originally Posted by Deckard2019

Outputting directly AC3 from ffdshow is still considered as a wrong method ?

I don't know. I've read in this thread that it gives inferior results. And I read that Aften is considered to be a good AC3 encoder, so that's why I did it this way.

[madshi]

Originally Posted by drmpeg

Seems to me that sox can't create a 6 channel .wav file that besweet understands.

Aften revision 449 is happy with sox created 6 channel wav files, as long as you use the Aften "-readtoeof 1" parameter.

[MichalHabart]

Originally Posted by drmpeg

The LPCM output of xport is big endian (just like it's stored in the .m2ts file).

This sequence works:

xport -h2 movie.m2ts 1 1 2

sox -B -r48000 -t .raw -c 2 -2 -s bits0001.mpa bits.wav

besweet -core( -input bits.wav -output bits.ac3 ) -ac3encode

This sequence fails:

xport -h movie.m2ts 1 1 2

sox -B -r48000 -t .raw -c 6 -2 -s bits0001.mpa bits.wav

besweet -core( -input bits.wav -output bits.ac3 ) -ac3encode ( -6ch )

Error 58: Error : Unknown Input-File Format : "bits.wav".

Seems to me that sox can't create a 6 channel .wav file that besweet understands.

Ron

And have you tried to open that multiwav in something else, for example in wavewizard?
I was succesfull in this. probably just besweet has these problems. I was able to demux 6 channel lpcm to raw from Terminator 1 Bluray and converted it to wav and now i made DTS track from it. It sounds wonderful

[The1n]

Do i have to change the channelmapping if im just going to convert eac3 to aac in megui, im using graphedit and loads the graph with avisynth script.
When i play the track in graphedit all channels sound ok, but when i encode it to aac multichannel suddenly the center is gone.

The1n

[zgx]

Perfect lossless LPCM 5.1 to FLAC conversion
It's really easy but has taken me quite a few hours to get it all right.

Programs needed:
xport by drmpeg @doom9
http://www.w6rz.net/xport.zip

sox v13.0.0
http://sourceforge.net/projects/sox

Wavewizard 0.45b
http://www.rarewares.org/wavewiz/wavewizardv0.54b.zip

MediaCoder 0.5.1
http://mediacoder.sourceforge.net/download.htm

flac 1.1.4
http://flac.sourceforge.net/download.html


Step 1 - Extract the LPCM stream from Blu-ray
xport -h movie.m2ts 1 3 2

Explaination:
"1 3 2" selects program 1, video stream 3 and audio stream 2.
Since video stream 3 does not exist you will only demux audio (change to 1 if you want video as well).
The LPCM stream should most often be number 2.


Step 2 - Map channels
sox -B -r48000 -t .raw -c 6 -2 -s bits0001.mpa file.wav

Start Wavewizard and enter channel mapping (F2).
- Create a new mapping "0 1 2 5 3 4".
- Select your new mapping.
- Click "Enable channelmapping"
- Click "Convert" and create a new file.

Explaination:
The channels in the LPCM file are mapped "L,R,C,BL,BR,LFE" as opposed to "L,R,C,LFE,BL,BR". This is corrected with Wavewizard but in order for it to accept the file you need to use sox first.


Step 3 - Encode your FLAC
- Install MediaCoder 0.5.1. Make sure you replace "flac.exe" under the "codecs" directory so you get version 1.1.4.
- Drag and drop your file(1).wav to the MediaCoder window.
- Under Audio select "FLAC" as Encoder and press F5 to start the encode.

Explanation:
FLAC doesn't like 4+ GB wave files but somehow it works with MediaCoder.


Step 4 - Enjoy your FLAC
Enjoy your new FLAC!

May I sugest using mkvmerge (http://www.bunkus.org/videotools/mkvtoolnix/) to mux your FLAC with your video stream to a Matroska container.

[moshmothma]

Hmm, I get to the wavewizard part and get the following error when I try to convert:

Error on: "file.wav". This file is not recognized by wavewizard.

[zgx]

Originally Posted by moshmothma

Hmm, I get to the wavewizard part and get the following error when I try to convert:

Error on: "file.wav". This file is not recognized by wavewizard.

Sorry made a typo. Should of course be "sox -B -r48000 -t .raw -c 6 -2 -s bits0001.mpa file.wav" and not "-3".

[LRN]

I can't beleive they're using LPCM...They must be completely dazzed by BluRay disc sizes! There's NO reason to but more than 500 kilobits on one channel, and if there IS, they could have used FLAC, WAVPack, damn MLP or something! But not the LPCM...it's so DULL.

P.S. How about fullscale "X.1 E-AC3 -> 5.1 AAC" guide?

P.P.S. Where i could get Sonic 4.2 decoder? 4.1 doesn't works in shown graphs

[Rectal Prolapse]

Wow zgx - just what I was looking for! Hopefully there won't be any issues with FLAC audio decoding past the 4GB mark - when I use WAV I get silence once that barrier is surpassed.

[madshi]

Originally Posted by zgx

Perfect lossless LPCM 5.1 to FLAC conversion

Just for my interest: How big is the FLAC compressed audio track file compared to the original LPCM one? Thanks!

[LRN]

20-30%, iirc

[zgx]

Originally Posted by madshi

Just for my interest: How big is the FLAC compressed audio track file compared to the original LPCM one? Thanks!

About 25% of the original. For "16 bit/48 kHz/6 channel" movie soundtracks I have gotten files from 1100 to 1900 Kbps so it doesn't take more space then a 1.5 Mbps DTS track does. The only downside is that you need 6 analog RCA cables or HDMI in order to get lossless sound to your transciever. You can of course let AC3filter transcode to AC3 for S/PDIF ouput. Then it won't be lossless but will still sound really good.