Guide: Blu-ray MKV remux

[zgx]

Guide: Blu-ray MKV remux v1.0
Last updated 2007-03-29

Feedback and improvements are most welcome.


Download and install the needed software
Muxing/demuxing software
Haali Media Splitter (make sure you get the latest version)
http://haali.cs.msu.ru/mkv/

MKVtoolnix pre-builds
http://www.bunkus.org/videotools/mkvtoolnix/win32/pre/

Xport Transport Stream Demuxer
http://www.w6rz.net/


To compress LPCM tracks to FLAC
flac 1.1.4
http://flac.sourceforge.net/download.html

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


Instructions
When you install Haali Media Splitter click "Enable MPEG-PS support".



Blu-ray with MPEG2 and AC3/DTS

Step 1a: Making the MKV - if lucky
If you are lucky gdsmux will find all the streams that you want and you can complete the remux with just a few clicks.

Start gdsmux.exe (found under your Haali Media Splitter installation).
- Add the m2ts file containing your movie.
- If you find all your streams you want select them, put in an output filename and press "Start".

http://www.imageload.eu/out.php/i2270_1.png


Step 1b: Making the MKV - if less lucky
Sometimes gdsmux will not find the audio tracks that you want and you have to use an alternative way. In the example below we did not find the DTS track that we wanted:

http://www.imageload.eu/out.php/i2271_2.png

1. Demux the audio track:

Try the following commands:
xport -h 00000.m2ts 1 3 1
xport -h 00000.m2ts 1 3 2
xport -h 00000.m2ts 1 3 3
...
...

For each command you will see information about the audio tracks. For example:
Audio Bitrate = 448000, Audio Sampling Rate = 48000
Audio Mode = 3/2, bsid = 6, bsmod = 0

This information helps you find the track you want. Break with ctrl-break if you find a audio track you don't want.

What I have seen is that no information is given for certain DTS tracks. Instead you will just see:
ts rate = unspecified

To make sure you get the track you want break after a few seconds and then play the track you got. It's named "bits0001.mpa". You can also drag and drop the track into mkvmerge and it will display what kind of track you got. See example below:

http://www.imageload.eu/out.php/i2272_3.png

After you find the audio track you want let the xport command run so you get the complete track.


2. Create the MKV
Chances are good that gdsmux will not recognize the audio track you just demuxed so you will have to use mkvmerge instead.

- Drag and drop the m2ts file and the audio track to mkvmerge.
- Select a name for your new MKV file.
- You may also want to name the movie under "Global" and then "File/Segment title" and make some other smaller adjustments to your MKV.
- Press "Start Muxing"

http://www.imageload.eu/out.php/i2273_4.png



Blu-ray with H.264/AVC
I have not done that much testing with H.264/AVC but I know that I had some problems when I tried it out a month ago. Lately I have been successful using gdsmux that comes with Haali Media Splitter (the latest version from 2007-03-28). I don't have the CPU to play the H.264/AVC video to verify that it works but it seams ok. I don't know how well mkvmerge handle these files yet.



Saving your DTS-HD track for future use
If your Blu-ray features a DTS-HD track that you muxed into an MKV above you did only get a 1.5 Mbps "DTS core" of the DTS-HD. At the moment we cannot mux nor decode the entire DTS-HD track but we can save it for future use. If you have not already done so use xport to demux the DTS-HD track and save it for future use.



Blu-ray with FLAC (LPCM)
Step 1: Create you FLAC track

1. Extract the LPCM stream from Blu-ray
xport -h 00000.m2ts 1 3 2


2. Prepare file for WaveWizard
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 the program won't recognize it (perhaps because it's raw with big endian or something). Let’s convert it to a better format. There is probably an easy way to do this but I have not really found it so I use FLAC for the conversion.

flac -0 --force-raw-format --endian=big --channels=6 --bps=16 --sample-rate=48000 --sign=signed -o "temp.flac" - < bits0001.mpa
flac -d temp.flac -o temp.wav


3. Map channels
Start Wavewizard

- Enter Channel mapping (F2) and create a new mapping "0 1 2 5 3 4".
- Select your new channel mapping.
- Enter Preferences (F4) and make sure you enable channel mapping and make sure you set ouput to "Raw samples".
- Click "Convert" and create a new wave file.

http://www.imageload.eu/out.php/i2274_5.png

http://www.imageload.eu/out.php/i2275_6.png

http://www.imageload.eu/out.php/i2277_8.png

http://www.imageload.eu/out.php/i2276_7.png


4. Encode your FLAC
flac -8 --force-raw-format --endian=little --channels=6 --bps=16 --sample-rate=48000 --sign=signed -o "final.flac" - < temp.raw


Step2: Create your MKV
Start MKVmerge.
- Drag and drop the m2ts file and the audio track to mkvmerge.
- Select a name for your new MKV file.
- You may also want to name the movie under "Global" and then "File/Segment title" and make some other smaller adjustments to your MKV.
- Press "Start Muxing"

http://www.imageload.eu/out.php/i2278_9.png

[Rectal Prolapse]

Thanks zgx.

Any word on how to demux the subtitle tracks from a bluray disc, and then OCR them into SRT files?

[tomos]

just tried it now. dies on step 2 with flac.

i get the error - (No runtime statistics possible; please wait for encoding to finish...)
-: ERROR: got partial sample

and the resulting file is just over a gig - against the originals 3 gigs.

[zgx]

just tried it now. dies on step 2 with flac.

i get the error -

and the resulting file is just over a gig - against the originals 3 gigs.Hmm ok thats wierd. Are you sure you are using the exact same syntax (for example --endian=big)?

[zgx]

I also remember that on some of the step the process aborted when I connected to another computer over remote desktop from the computer I was "ripping" on.. really strange.. it happened every time I did it. Could perhaps be the error you had Tomos?

[tomos]

deffo. copied and pasted the line in as is. filename was the same

re: remote desktop. dont use it.

[zgx]

deffo. copied and pasted the line in as is. filename was the same

re: remote desktop. dont use it.When you used xport to demux the audio track did you get information that it was a LPCM track? It's not a DTS-HD track or something that you have?

[tomos]

it's casino royale and that has PCM audio, not DTS-HD

the 'sox' commands that i tried 1st (not your guide, the one mixed into the evob thread)

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

that gives me a wav file i can play fine but cant do anything with after (wavewizard wont accept it). the commands used in your guide also dont work for me :(

---

just redid everything and still the same. the flac method craps out. the sox method works upto the point of making a wav that can be played with sonics audio decoder only.

[toxik]

I can't seem to be able to demux Casino Royale's avc video stream properly. Can someone help me with how they did it. I tried both gdsmux and xport, both with no success. What command line should I use with xport?

[tomos]

xport -h "x:\whatever.m2ts" 1 1 1

the last one should be a 2 ( 1 1 2 ) if you want the LPCM track. the first digit is the (afaik) program stream. the 2nd is the video stream and the 3rd is the audio stream.

[toxik]

xport -h "x:\whatever.m2ts" 1 1 1

the last one should be a 2 ( 1 1 2 ) if you want the LPCM track. the first digit is the (afaik) program stream. the 2nd is the video stream and the 3rd is the audio stream.

that's what i used, the file I got was only 13GB though, where is the original m2ts is 35gbs. Surely, the audio can't make up the rest of the 20gbs right?

[zgx]

just redid everything and still the same. the flac method craps out. the sox method works upto the point of making a wav that can be played with sonics audio decoder only.Ok if you use the sox method so you have a WAV you can play. You can compress that WAV in MediaCoder to FLAC and then decompress it to WAV again and it will be readable in Wavewizard.

[Darth Pinous]

that's what i used, the file I got was only 13GB though, where is the original m2ts is 35gbs. Surely, the audio can't make up the rest of the 20gbs right?

I had the same problem, even if I only demuxed the LPCM soundtrack, so maybe it's because of the source...

Anyway, it works if you use graphedit with filesource(async) -> sonic HD demuxer -> Sonic Video Decoder, so if you want to reencode, it's still possible. (same goes for the LPCM soundtrack).

[tomos]

that's what i used, the file I got was only 13GB though, where is the original m2ts is 35gbs. Surely, the audio can't make up the rest of the 20gbs right?


Anyway, it works if you use graphedit with filesource(async) -> sonic HD demuxer -> Sonic Video Decoder, so if you want to reencode, it's still possible. (same goes for the LPCM soundtrack).

good thing i checked. mine was short too. something freaky with the m2ts source - has to be. dumping with graphedit/hd pump have me a 24.6 GB h264 file that's ok.

[Darth Pinous]

flac -0 --force-raw-format --endian=big --channels=6 --bps=16 --sample-rate=48000 --sign=signed -o "temp.flac" - < bits0001.mpa

Strange, when I use this line, FLAC stops the encode as soon as the FLAC output reaches 2GB... I tested with sign=signed and sign=unsigned, but no difference... 2 GB is the limit for me... Where could it come from ? I used FLAC 1.1.4 from the link you post.

[klona]

... 2 GB is the limit for me...

Sure there isn't any 2GB FAT32 limitation on your system ? Perhaps you already know but NTFS is needed for more than 2GB files.

[zgx]

Strange, when I use this line, FLAC stops the encode as soon as the FLAC output reaches 2GB... I tested with sign=signed and sign=unsigned, but no difference... 2 GB is the limit for me... Where could it come from ? I used FLAC 1.1.4 from the link you post.2 GB FLAC ouput .. sounds like a huge LPCM file hmm. Dumb question but are you sure it's a LPCM track you are trying to encode and not something else like a DTS track? You could also try to increase the compression with -8 instead of -0 but 1.1.4 should support larger files then 2 GB.

[Chumbo]

Quick question regarding Casino Royale. I dumped the lpcm track via Graphedt which gave me a 4.6Gig track. I'm able to convert it via sox, but then I get a wav that's only 19min 55sec in length. I can convert the wav just fine to a 5.1 ac3 but it's still 19:55 in length. Anyone else experience the same problem?

My graphedt chain is simple:
file source (async) -> sonic hd demuxer -> dvb hdtv dump

I made sure to select the right stream under the demuxer filter properties before proceeding. I also tried sticking the audio decoder between the demuxer and dump plus several variations of having ac3filter too. Always with the same result.

I was able to dump the main 5.1 track just fine.

[tomos]

2 GB FLAC ouput .. sounds like a huge LPCM file hmm. Dumb question but are you sure it's a LPCM track you are trying to encode and not something else like a DTS track? You could also try to increase the compression with -8 instead of -0 but 1.1.4 should support larger files then 2 GB.

redownloaded FLAC again, deffo v1.1.4

stalls on 2GB files. gives an error saying the disk is full (as a guess) but there's 70 gigs left on the drive so that's not the issue

flac 1.1.4, Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007 Josh Coalson
flac comes with ABSOLUTELY NO WARRANTY. This is free software, and you are
welcome to redistribute it under certain conditions. Type `flac' for details.

(No runtime statistics possible; please wait for encoding to finish...)

-: ERROR during encoding
state = FLAC__STREAM_ENCODER_CLIENT_ERROR

An error occurred while writing; the most common cause is that the disk is full.

[Rectal Prolapse]

FLAC command-line program is known to have a limit of 2GB output.

MediaCoder 0.5 might do what you want - it gets around the limitation somehow (using named pipes?).

[zgx]

redownloaded FLAC again, deffo v1.1.4

stalls on 2GB files. gives an error saying the disk is full (as a guess) but there's 70 gigs left on the drive so that's not the issueI'm getting the same problem myself now on a huge 24 bit LPCM track. MediaCoder usually works but not sure how to adjust the parameters there so I'll get 24 bit and big endian.

[Darth Pinous]

My graphedt chain is simple:
file source (async) -> sonic hd demuxer -> dvb hdtv dump

I was able to dump the main 5.1 track just fine.

Interesting graph. Do you get raw PCM with it (i.e. you need to remap channels after that) ?

What is your sox command line ?


Edit : Just tested this graph on a M2TS with 24/48 LPCM soundtrack. I used Sox with the "-B" option and it's the first time that the resulting WAV EXT file is readable (i.e. neither horrible fax-machine sound, nor complete silence). When I dump the audio with xport, I can't obtain a valid WAV out of Sox, no matter the options I choose.

One other point... Does anyone know if it's possible to build a MKV file with a 24 bits Wavpack soundtrack ? I tried it, and the MKV splitter (Haali) sees the soundtrack as a 32 bits soundtrack (which produces a VERY horrible sound).

[tomos]

I'm getting the same problem myself now on a huge 24 bit LPCM track. MediaCoder usually works but not sure how to adjust the parameters there so I'll get 24 bit and big endian.

good to know it's not just me :)

[Chumbo]

Interesting graph. Do you get raw PCM with it (i.e. you need to remap channels after that) ?

What is your sox command line ?


Edit : Just tested this graph on a M2TS with 24/48 LPCM soundtrack. I used Sox with the "-B" option and it's the first time that the resulting WAV EXT file is readable (i.e. neither horrible fax-machine sound, nor complete silence). When I dump the audio with xport, I can't obtain a valid WAV out of Sox, no matter the options I choose.

One other point... Does anyone know if it's possible to build a MKV file with a 24 bits Wavpack soundtrack ? I tried it, and the MKV splitter (Haali) sees the soundtrack as a 32 bits soundtrack (which produces a VERY horrible sound).
This works with everything except the LPCM track when it's done. Here's my sox command:
sox -B -r 48000 -t .raw -c 6 -2 -s casino.lpcm casino.wav

btw, I stopped the sox command at around 2.5Gig, just to test. Sure enough the length of the wav file was around 1hr 31min I think. I just don't get why, when it's done, the 4.6G file only shows a 19min 55sec play length. Makes no sense.

btw, after sox, you have to patch the header like this:
WavNotEx -p casino.wav

Then I use an avs script to get the channels in the right order like this:
wavsource("casino.wav")
GetChannels(1,2,3,6,4,5)

[Rectal Prolapse]

MediaCoder usually works but not sure how to adjust the parameters there so I'll get 24 bit and big endian.

Ah right - I forgot that MediaCoder only handles 16 bits.

My best solution was to switch to WavPack - and use BeHappy for the encode. The latest modded BeHappy version supports 24 bit audio.

Unfortunately, I've had issues with WavPack playback - I had to add the FFDShow Audio Processor after the WavPack decoder to keep sync.

[madshi]

Drag and drop the m2ts file and the audio track to mkvmerge.
That's confusing me. I thought that:

(1) m2ts is a transport stream? and that
(2) mkvtoolnix cannot parse transport streams?

What am I missing? Thanks!

[DreckSoft]

Blu-ray with H.264/AVC
I have not done that much testing with H.264/AVC but I know that I had some problems when I tried it out a month ago. Lately I have been successful using gdsmux that comes with Haali Media Splitter (the latest version from 2007-03-28). I don't have the CPU to play the H.264/AVC video to verify that it works but it seams ok. I don't know how well mkvmerge handle these files yet.


I tried remuxing two Blu-Ray with H264 video to MKV.

Seeking is impossible in the resulting MKV no matter which way I use.

It won't work with MKVMerge, GDSMux, GraphEdit Haali Splitter => HaaliMuxer. Neither m2ts => MKV nor m2ts => 264|ac3|dts => mkv works

No idea whats wrong here.

[Darth Pinous]

btw, I stopped the sox command at around 2.5Gig, just to test. Sure enough the length of the wav file was around 1hr 31min I think. I just don't get why, when it's done, the 4.6G file only shows a 19min 55sec play length. Makes no sense.


Funny, I've got the same 19min55sec length with the sox result of a file made with the graph FileSource(Async) -> Sonic HD Demuxer -> Sonic Audio Decoder -> Dump.

Anyway, when I load the file in Wavewizard, length is OK and it's possible to make a Wave PCM, and then to make a Wavpack file with correct length.

[Chumbo]

Funny, I've got the same 19min55sec length with the sox result of a file made with the graph FileSource(Async) -> Sonic HD Demuxer -> Sonic Audio Decoder -> Dump.

Anyway, when I load the file in Wavewizard, length is OK and it's possible to make a Wave PCM, and then to make a Wavpack file with correct length.
Very cool. Thanks so much for sharing that info. Hopefully it will be helpful to others. :) I've never used wavewizard, but it looks like I'm going to have to try it now. ;)

[LOGiC]

good thing i checked. mine was short too. something freaky with the m2ts source - has to be. dumping with graphedit/hd pump have me a 24.6 GB h264 file that's ok.


Tomos, Sir, may I please ask you which graph you exactly used to get the H264-Stream out of the m2ts-source in case of the casino royale blu ray ? Was ist just File Source (async) --> HDTV Pump ?

Thank you very much in advance.

[mikeyakame]

i had no problems with casino royale, used graphedit to create a graph for the video using haali's latest media splitter (supports m2ts fully now) -> cyberlink h264/avc decoder. that gives me the video stream for encoding, and if i attach vmr9 to it, playback is perfect, and with the audio, i have it encoded to 640kbps DD5.1 from the 2mbps LPCM stream, im pretty sure i created a graph in graphedit, with, file source async (m2ts) -> sonic hd demuxer -> sonic audio decoder -> dump (.raw) then used sox to convert to 16-bit wav. wave wizard to split 6 mono channels, (0,1,2,5,3,4), through these in sonic audio transcoder (dolby digital 5.1 creator) set the dialog norm levels and all that, and encoded a 640kbps ac3...took me no longer than 45mins to do all that.

[tomos]

afaik, sonics audio decoder doesnt do LPCM (at least v4.2 doesnt IMO). casino royale has a standard AC3 track though so that's probably that you have

might have been easier to just dump that straight off the bat.

[Darth Pinous]

afaik, sonics audio decoder doesnt do LPCM (at least v4.2 doesnt IMO). casino royale has a standard AC3 track though so that's probably that you have

might have been easier to just dump that straight off the bat.

What do you mean by "sonics audio decoder doesnt do LPCM" ?
I made a Wavpack reencode of the LPCM soundtrack using this filter in graphedit.

Just to be sure, I made an extract of the LPCM soundtrack via xport (just 30 minutes), then made a wavpack reencode of it, and compare the two WV files : I couldn't hear any differences between them...

[tomos]

sonics demuxer picks the AC3 track not the LPCM afaik.

[Darth Pinous]

sonics demuxer picks the AC3 track not the LPCM afaik.
I'm surprised... I heard others reencodes from AC3 or E-AC3 soundtracks @ 640 Kbps, and the sound is never that good, that clear...

[Chumbo]

i had no problems with casino royale, used graphedit to create a graph for the video using haali's latest media splitter (supports m2ts fully now) -> cyberlink h264/avc decoder. that gives me the video stream for encoding, and if i attach vmr9 to it, playback is perfect, and with the audio, i have it encoded to 640kbps DD5.1 from the 2mbps LPCM stream, im pretty sure i created a graph in graphedit, with, file source async (m2ts) -> sonic hd demuxer -> sonic audio decoder -> dump (.raw) then used sox to convert to 16-bit wav. wave wizard to split 6 mono channels, (0,1,2,5,3,4), through these in sonic audio transcoder (dolby digital 5.1 creator) set the dialog norm levels and all that, and encoded a 640kbps ac3...took me no longer than 45mins to do all that.
I've done the exact same thing, on other titles with no issues. Casino Royale is the only one. I've used the exact chain that you use and the resulting wav, after sox, is a 19min 55sec wav file which makes no sense.

I've never used Wavewizard, but tried it and it shows the wav info correctly, however, when I select Convert, it displays the following error: "CB error on: 'myfile.wav'. Unsupported sampletype. The only sampletype supported is IEEE Float."

So I have to ask, are you sure you encoded the LPCM track? Did you go into the HD Demuxer and select that track? Because if you did not do that, then you did not encode the LPCM track. The default audio track for CR is AC3 5.1 448kbps.

There's two of us now that are having the exact same problem with the LPCM track, so we'd love to know how you were able to get it to work and not us. Many thanks.

I think I've tried every combination I can think of using different splitters, demuxers and audio decoders. I even tried not using the audio decoders and just dump the stream straight from the demuxer and that didn't work either. btw, my first attempt was using the exact filter chain you used.

Any how, it looks like I'm almost there since wavewizard recognizes the file with the right time stamp, but I think that's because, in Preferences, I have "Ignore size in header" checked when greater than 4 GB.

[Chumbo]

sonics demuxer picks the AC3 track not the LPCM afaik.
The demuxers always default to the first track. If the filter properties allow you to select a track, then you have to do that before you dump the file. The Sonic Demuxer 4.2 does allow you to select the audio stream.

[Darth Pinous]

I've done the exact same thing, on other titles with no issues. Casino Royale is the only one. I've used the exact chain that you use and the resulting wav, after sox, is a 19min 55sec wav file which makes no sense.

I've never used Wavewizard, but tried it and it shows the wav info correctly, however, when I select Convert, it displays the following error: "CB error on: 'myfile.wav'. Unsupported sampletype. The only sampletype supported is IEEE Float."

So I have to ask, are you sure you encoded the LPCM track? Did you go into the HD Demuxer and select that track? Because if you did not do that, then you did not encode the LPCM track. The default audio track for CR is AC3 5.1 448kbps.

There's two of us now that are having the exact same problem with the LPCM track, so we'd love to know how you were able to get it to work and not us. Many thanks.

I think I've tried every combination I can think of using different splitters, demuxers and audio decoders. I even tried not using the audio decoders and just dump the stream straight from the demuxer and that didn't work either. btw, my first attempt was using the exact filter chain you used.

Any how, it looks like I'm almost there since wavewizard recognizes the file with the right time stamp, but I think that's because, in Preferences, I have "Ignore size in header" checked when greater than 4 GB.

It's best to check "Ignore size in header" and select below the "always" option, to be sure that wavewizard always ignore the size, and select WAV PCM for the output.

Strange error, though. When you check the properties of the file in Wavewizard, does it see the file as a 16 bits file or a 32 file ?
Is it WAV EXT ?

[Chumbo]

It's best to check "Ignore size in header" and select below the "always" option, to be sure that wavewizard always ignore the size, and select WAV PCM for the output.

Strange error, though. When you check the properties of the file in Wavewizard, does it see the file as a 16 bits file or a 32 file ?
Is it WAV EXT ?
It sees it as a 16bit file. The "ignore..." option is checked and I have to use WAVE PCM because ww tells me it's the only option to choose.

I took snapshots of my wavewizard. Here you go:

http://img204.imageshack.us/img204/2811/wwmaindt9.jpg (http://img204.imageshack.us/img204/2811/wwmaindt9.jpg)

http://img214.imageshack.us/img214/6918/wwpropertiesta9.jpg (http://img214.imageshack.us/img214/6918/wwpropertiesta9.jpg)

http://img214.imageshack.us/img214/8867/wwprefspk5.jpg (http://img214.imageshack.us/img214/8867/wwprefspk5.jpg)

http://img214.imageshack.us/img214/8831/wwerrorqk1.jpg (http://img214.imageshack.us/img214/8831/wwerrorqk1.jpg)

[Darth Pinous]

It sees it as a 16bit file. The "ignore..." option is checked and I have to use WAVE PCM because ww tells me it's the only option to choose.

Are you trying to use the conversion batcher in the process (i.e. in Edit -> Conversion Batcher, the "Send jobs to batcher" and/or "Start batcher when finished" options are checked) ? If so, try without using it by unchecking these options.

Personally, I only make a WAV PCM file, with sometimes a 16 bits conversion for 24/48 PCM and/or put an audio delay to get in sync with the video (for example, I had to put a -650ms delay for CR), then I use wavpack in command line to reencode it.

[Chumbo]

Are you trying to use the conversion batcher in the process (i.e. in Edit -> Conversion Batcher, the "Send jobs to batcher" and/or "Start batcher when finished" options are checked) ? If so, try without using it by unchecking these options.

Personally, I only make a WAV PCM file, with sometimes a 16 bits conversion for 24/48 PCM and/or put an audio delay to get in sync with the video (for example, I had to put a -650ms delay for CR), then I use wavpack in command line to reencode it.
My hero! That did the trick. Being that I really haven't used the program before, I really should have tried those options. Doh! Nice call.

I hope this helps someone else who is running into this issue. :)

[EDIT] Just to follow up. Even though Wavewizard processed the file, it was still read as a 19min 55sec wave. Man this is just nuts. Darth, were you able to use the wav file after going through ww? Was it shown as the correct length? Or did you wind up using wavepack like you mentioned? Thanks.

[Darth Pinous]

My hero! That did the trick. Being that I really haven't used the program before, I really should have tried those options. Doh! Nice call.

I hope this helps someone else who is running into this issue. :)

[EDIT] Just to follow up. Even though Wavewizard processed the file, it was still read as a 19min 55sec wave. Man this is just nuts. Darth, were you able to use the wav file after going through ww? Was it shown as the correct length? Or did you wind up using wavepack like you mentioned? Thanks.

The WAVE PCM file out of WW is still wrong if you try to play it (I think that on my computers, I can't even play the file), but that doesn't bother Wavpack, which makes a nice and readable file with correct length.

By the way, could anyone get something readable with the PCM 24/48 soundtrack from The Prest1ge ? In graphedit, even if I try to render the file using the usual working graph FileSource(Async) -> Sonic HD Demuxer -> Sonic Audio Decoder -> Defaut DSound Device, I only have noise (while the other AC3 soundtracks are OK) ?

[zgx]

By the way, could anyone get something readable with the PCM 24/48 soundtrack from The Prest1ge ? In graphedit, even if I try to render the file using the usual working graph FileSource(Async) -> Sonic HD Demuxer -> Sonic Audio Decoder -> Defaut DSound Device, I only have noise (while the other AC3 soundtracks are OK) ?I'm working with The Prestige at the moment also. The dump from xport is not valid for me but "FileSource(Async) -> Sonic HD Demuxer -> Dump" gives me a valid file to work with. But since the file is so huge I cannot give it the valid wave header that is needed for wavewizard. But seams to be working if I split the file up in smaller parts first.

[tomos]

the only problem getting it working in a much easier way (for me anyway) is sonics demuxer being a PITA

the graph, source - sonic demux - sonic audio decoder, handles LPCM fine and plays back nice. the problem is that you have to choose LPCM manually every time so the graph cant be saved and used in an avisynth script.

demuxing doesnt work either. graphedit closes down straight away when trying to connect those pins

[Huntta]

Sorry for a nooby question, but if it's GetChannels(1,2,3,6,4,5) for 5.1, would it then be GetChannels(1,2,3,8,5,6,4,7) for 7.1?

[Darth Pinous]

Sorry for a nooby question, but if it's GetChannels(1,2,3,6,4,5) for 5.1, would it then be GetChannels(1,2,3,8,5,6,4,7) for 7.1?

If you're talking about converting LPCM 7.1 to WAV 7.1, yes, it should be that.

[Huntta]

OK thanks.

Darth Pinous, I've been having the same problem with Prestige:
whatever I do, I keep getting just noice. Any input zgx how did you manage to get it correctly?

[Chumbo]

OK thanks.

Darth Pinous, I've been having the same problem with Prestige:
whatever I do, I keep getting just noice. Any input zgx how did you manage to get it correctly?
I'm assuming you used SOX on the file right? If so, did you use the right endian setting? If you used -B (big endian) and you're getting noise, then try -L (little endian).

[Darth Pinous]

OK thanks.

Darth Pinous, I've been having the same problem with Prestige:
whatever I do, I keep getting just noice. Any input zgx how did you manage to get it correctly?
Well, I used the "FileSource -> Sonic HD Demuxer -> DVB Reference Dump" graph and it worked, but the first time I dumped the audio with the same graph I had only noise, don't know why, because I didn't change/install anything on my PC... Maybe I choose another soundtrack in Sonic HD Demuxer properties, then changed for the correct soundtrack before the dump, and Sonic did'nt make the change...

I had problems with video, too... When I tried to dump the video with xport, then mux it in a MP4 with YAMB, the picture looked messy... Using a graph with FileSource -> Sonic HD Demuxer -> Sonic Video Decoder did the trick, though...

[Darth Pinous]

I'm assuming you used SOX on the file right? If so, did you use the right endian setting? If you used -B (big endian) and you're getting noise, then try -L (little endian).

About LPCM and Sox, just a detail that I haven't read but could be interesting :

- if you use xport to dump the LPCM soundtrack, or graphedit with FileSource(Async) -> Sonic HD Demuxer -> Dump, the outputted raw audio seems to be big endian, so it's necessary to use Sox with -B option.

- if you use graphedit with FileSource(Async) -> Sonic HD Demuxer -> Sonic Audio Decoder -> Dump, the outputted raw audio seems to be little endian (Sonic audio dec seems to not only put the channels in correct order, but to change the endian type too), so it's necessary to use Sox without -B or -L option.