eac3to - audio conversion tool [Archive] - Page 66

umaximus

3rd February 2008, 01:26

No, it writes the message 'applying delay' when muxing, even tho its not listed under trueHD track..
Also, the extracted trueHD track got the '5ms' in title..

Eastern Promises
eac3to v2.20
eac3to feature_1.evo+feature_2.evo 2: video.mkv 3: truhd.truehd 3: audio.flac -nero
EVO, 1 video track, 3 audio tracks, 1:40:34
1: Joined EVO file
2: VC-1, 1080p24 /1.001
3: TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
4: E-AC3, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB, -16ms
5: E-AC3, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB, -16ms
Extracting primary video track...
Muxing video to Matroska...
Extracting audio track number 3...
Extracting audio track number 3...
Removing dialog normalization...
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Applying RAW/PCM delay...
This audio track contains more than 16 bit of information.
Encoding FLAC...
Creating/writing file "truhd.truehd"...
Creating/writing file "audio.24bit.flac"...

the raw trueHD track, after exctracting, is named ''truehd DELAY 5ms.truehd''

nautilus7

3rd February 2008, 01:34

Oh, then maybe it's a bug!

I'll have my Bourne Ultimatum disc next week to test my self and if necessary i'll send a sample to madshi (you can do it also if you don't mind), but first he should confirm this is a bug.

You can use .thd as extension for truehd track. It's sorter! :p

Thunderbolt8

3rd February 2008, 03:42

would it maybe also apply the delay for the flac track with the libav decoder?

nautilus7

3rd February 2008, 11:22

If there isn't a delay listed, eac3to shouldn't write a message that applying delay and not even write a delay value in the filename.

What i mention before about the encoders is done internally and doesn't apply to all kinds of files (flac and wav don't have that "problem"). I have only find this behavior to .ac3 and .dts where constant length frames are used.

rmtaibo

3rd February 2008, 14:27

this programs rocks!
But I have a problem. I'm transcoding DTS audio to AC3 from a DVD. All was fine until found a DTS file with 2 sec lenght. (247 Kb)

This error is related to eac3to or Sonic Audio Decoder?

Microsoft Windows [Versión 6.0.6000]
Copyright (c) 2006 Microsoft Corporation. Reservados todos los derechos.

D:\VIDEOS\******\VTS01>"C:\Program Files\EAC3TO\eac3to.exe" "D:\VIDEOS\******\VTS01\VTS_01_PGC_01_VOBID_005_1 - 0x8A - Audio - DTS - 6ch - 48kHz - DRC - English - DELAY -216ms.DTS" "D:\VIDEOS\******\VTS01\VTS_01_PGC_01_VOBID_005_1 - 0x8A - Audio - DTS - 6ch - 48kHz - DRC - English - DELAY -216ms.ac3"
-448 -216ms
The format of the source file could not be detected.

The VLC player, Nero ShowTime and PowerDVD plays this file without problem.
:thanks:

uk_dan2k

3rd February 2008, 17:34

It has to do with the spaces in the path/file names. Someone had the same error some weeks ago, but i don't really remember what the solution was.

Is the gui necessary? I mean it's not that difficult to use the cli, right? Btw, isn't the gui outdated?

ok typed it in command instead and it says getting sonic audio convertor failed

looking at the first page it says its from Sonic, but which program uses it made by sonic?

nautilus7

3rd February 2008, 18:24

nautilus7

3rd February 2008, 18:29

ok typed it in command instead and it says getting sonic audio convertor failedYou don't have the sonic decoder or it isn't working correctly. Run eac3to -test to see yourself.

looking at the first page it says its from Sonic, but which program uses it made by sonic?I can't understand what you 're writing here. Sonic is the default (best quality) decoder for DTS(-HD) tracks. If you don't have it, you can use the nero one (-nero in the command) or the free libav (-libav in the command).

tjf

3rd February 2008, 19:20

I have just finished remuxing bourne trilogy, and i noticed that the audio delay on Bourne Supremacy for english EAC3 is a bit off.
Eac3to reports it as +1001ms (also evodemux), but i found out that additional 300ms is needed for more accurate lip sync. Can anyone confirm this?

btw, remuxing with latest 2.20 version

I confirm this. Just finished it and the sound is off. Right now am remuxing with 300ms delay.

Edit: Sound is perfect now.

uk_dan2k

3rd February 2008, 20:11

You don't have the sonic decoder or it isn't working correctly. Run eac3to -test to see yourself.

I can't understand what you 're writing here. Sonic is the default (best quality) decoder for DTS(-HD) tracks. If you don't have it, you can use the nero one (-nero in the command) or the free libav (-libav in the command).

nero 7 or older isn't working correctly
sonic audio convertor not installed

the libav, seems to be working so far,

about the sonic what i meant was, what program uses the sonic decoder files, i.e. for the nero decoder files, nero 7 burning rom uses it.

nautilus7

3rd February 2008, 20:38

nero 7 or older isn't working correctly
sonic audio convertor not installed

the libav, seems to be working so far,

about the sonic what i meant was, what program uses the sonic decoder files, i.e. for the nero decoder files, nero 7 burning rom uses it.In order nero decoders to work you need to buy the nero blu-ray/hd dvd plug in. That's why you get this message.

Libav decoder comes with eac3to and works quite well.

Sonic decoders come with Sonic Cineplayer HD DVD Decoder (v4.3.0.169 is the latest). Read here (http://forum.doom9.org/showthread.php?t=133895) for info where to get.

rickardk

3rd February 2008, 21:23

nautilus7

3rd February 2008, 21:37

Do you get an error message? (No)
Is the processing completed instantly/after a few seconds or it takes time?
Convert the lpcm to another format to determine whether the problem is at the decoding or the encoding part.

rickardk

3rd February 2008, 22:38

No error message.
Takes about 10 minutes (FLAC).
Also tried converting to ac3. The resulting file is about 7MB.

nautilus7

3rd February 2008, 22:59

10 minutes? On what pc? Do you think the whole .pcm file is read? Are the output files (5MB & 7MB) playable?

Since the problem still exists with .ac3, i would say it has to do with decoding. Does eac3to detects the .pcm parameters correctly? Try setting them manually. Try also demuxing the .pcm track again, maybe using another program.

Final thing to do, if nothing works, upload a sample (check what output file size it results to).

Thunderbolt8

4th February 2008, 00:13

Trying to convert Stranger Then Fiction LPCM to FLAC.
The LPCM track is extracted with xport. 3.63GB in size.
The FLAC track created by eac3to is 5MB. Tried several times with the same result. Whats wrong here?
I already have remuxed that one a while ago and can therefore tell its at least no problem with the source :P

Thunderbolt8

4th February 2008, 00:21

rmtaibo

4th February 2008, 02:05

There's probably some garbage in the beginning of your dts file and eac3to can't determine its format. Run the file through delaycut and you will be fine.

eac3to still doesn't recognize the fixed DTS file with delaycut.

the audio file is 251 frames lenght (2 seconds). Can eac3to handle small files?

Attached can find the file. It is in rar (not zip) format. Change extension to rar again...
If someone can test and transcode successfully this file to AC3 5.1, please send me a PM.
Tell me if I'm doing something wrong...
It belongs to a Coldplay concert... And I'm making a PAL--> NTSC video transcoding and DTS 5.1-->AC3 5.1 audio transcoding.

I made this process several times before without problems, but this is the first time I get troubles with this small DTS file...

Thank you!
:thanks:

umaximus

4th February 2008, 03:01

good news for decoding with ffdshow:

Revision 1837 - Directory Listing
Modified Sat Feb 2 12:56:58 2008 UTC (34 hours, 18 minutes ago) by clsid2

Updated libavcodec.
H.264 decoding now supports interlaced + spacial direct mode.

am not home atm, but will test movies like babel and band of brothers 1st EP to check if the rainbow frames are gone now

I have just tested new version (clsid rev.1839) and rainbows in babel are still there. It must be something in the video header that changes at that time (56min22sec). Could you cut out particular sample with hex editor on that time?

Thunderbolt8

4th February 2008, 05:03

sure, I just dont know when Ill have time for that. maybe not before 1 week, though, if things go bad. but ill post it in the ffdshow thread.

nautilus7

4th February 2008, 14:50

eac3to still doesn't recognize the fixed DTS file with delaycut.

the audio file is 251 frames lenght (2 seconds). Can eac3to handle small files?

Attached can find the file. It is in rar (not zip) format. Change extension to rar again...
If someone can test and transcode successfully this file to AC3 5.1, please send me a PM.
Tell me if I'm doing something wrong...
It belongs to a Coldplay concert... And I'm making a PAL--> NTSC video transcoding and DTS 5.1-->AC3 5.1 audio transcoding.

I made this process several times before without problems, but this is the first time I get troubles with this small DTS file...

Thank you!
:thanks:Can you upload the sample on a fileshare site? It takes ages to approve an upload here!

rmtaibo

4th February 2008, 17:35

Can you upload the sample on a fileshare site? It takes ages to approve an upload here!

it was uploaded in previous post... It was 222 Kb lenght (very small file)
eac3to didn't recognize it as DTS (maybe too small)
But the file is valid. Maybe eac3to need improved por small files (not so usefull, but sometimes one find this small files in some videos)
Found a solution. Foobar2000 did the job.
DTS 5.1-->WAV 6 ch with foobar2000 and foo_input_dts.dll plugin
WAV 6 ch-->AC3 5.1 with eac3to

nautilus7

4th February 2008, 17:44

No, it's not uploaded yet. It says pending approval.

I am curious why it didn't work with eac3to, so can you upload it somewhere? Thanks!

laserfan

4th February 2008, 19:08

I can't seem to get eac3to v2.20 to demux to a 2nd drive e.g. "eac3to d:\input.evo e:\ -demux" doesn't work--eac3to writes the videotrack & audiotracks to the input file location.

How to demux to another drive? :confused:

nautilus7

4th February 2008, 20:03

Yes, you can't this way apparently. eac3to syntax with -demux switch is this:

eac3to input.file -demux where no output file is accepted/needed. I am surprised it didn't return an error with e:\ in the command.

Maybe in a future update the -demux switch can be used with an optional destination path. Something like:

eac3to input.file -demux e:\folder1\folder2

Currently, you can demux to a different path like this (unless madshi accept my suggestion :p ):

eac3to d:\input.evo 1: e:\output1.vc1 2: e:\output2.thd 3: e:\output3.eac3 4: ... etc

laserfan

4th February 2008, 22:02

Yes, you can't this way apparently....Currently, you can demux to a different path like this...Hey, thanks, I didn't realize there was an alternate way. I'll give that a try.

bertox

5th February 2008, 06:27

simple question: How can i encode an wav source to flac? command lines to different flac compression levels (0 to 8)?
Thanks in advance.

nautilus7

5th February 2008, 09:23

You can't choose a compression level in eac3to. The highest level is used.

rickardk

5th February 2008, 09:43

Remuxing Miami Vice and eac3to reports alot of gaps and overlaps.

Video has a gap of 4 frames at playtime 0:06:10.
Video overlaps for 4 frames at playtime 0:06:19.
Video has a gap of 4 frames at playtime 0:43:34.
Video overlaps for 4 frames at playtime 0:44:01.
Video has a gap of 4 frames at playtime 1:01:48.
Video overlaps for 4 frames at playtime 1:01:58.
Video has a gap of 4 frames at playtime 1:02:45.
Video overlaps for 4 frames at playtime 1:03:01.
Video has a gap of 4 frames at playtime 1:29:17.
Video overlaps for 4 frames at playtime 1:29:51.
Video has a gap of 4 frames at playtime 1:30:06.
Video overlaps for 4 frames at playtime 1:30:16.
Video has a gap of 4 frames at playtime 1:31:25.
Video overlaps for 4 frames at playtime 1:31:55.

Is this due to errors in the stream or is eac3to's VC-1 muxing not able to handle this title?

(The EVO plays back perfect. No errors in the eac3 stream)

madshi

5th February 2008, 09:57

I think you have to wait a few days, because something tells me madshi is writing his H264/AVC splitter!!! :)
Not yet. Been busy with other stuff...

Strange issue here with both EVOdemux and eac3to. They both tell me that you me and dupree has 6 audio tracks. I know that the movie actually has 5 tracks. Here is where it starts to get real weird. When I read the .xpl in EVOdemux the sixth audio track disappears. Eac3to still demuxed 6 audio tracks. When I try to convert the mysterious sixth track to .ac3 the resulting file is 500 MB but has no sound. :confused: I know that the read .xpl function in EVOdemux is a little wonky (on the same movie it also told me there was a separate VC-1 track which disappeared upon .xpl reading) but I have never encountered eac3to doing this kind of thing too. So what is this mysterious sixth track? Is the .ac3 that eac3to made from it just 500 MB of silence?
It seems that the EVO file contains a valid 6th track which has nothing but silence in it. Don't ask me why it's this way, but I'm quite sure that this is not a bug in eac3to. Just ignore/delete the empty track.

I have just finished remuxing bourne trilogy, and i noticed that the audio delay on Bourne Supremacy for english EAC3 is a bit off.
Eac3to reports it as +1001ms (also evodemux), but i found out that additional 300ms is needed for more accurate lip sync. Can anyone confirm this?
I can confirm this problem. Will need to find out why this happens. Don't know yet...

btw, i also noticed on Bourne Ultimatum and Eastern Promises, that when using ''eac3to feature_1.evo+feature_2.evo'', the trueHD track is not showing any delay, but when eac3to starts muxing it, it applys +5ms of delay. Not that is noticable, but still, why is doing it?
Sometimes there are some "garbage" TrueHD frames in the beginning of a TrueHD track which the TrueHD decoders ignore. In order to make audio delay as perfect as possible, eac3to applies an additional small delay in such cases.

I'm transcoding DTS audio to AC3 from a DVD. All was fine until found a DTS file with 2 sec lenght. (247 Kb)
eac3to expects a minimum length of 300KB for DTS tracks. The reason is that one single DTS frame could eventually be near to 300KB. Maybe I can improve eac3to to handle smaller files, too, but I don't really find it very important, to be honest... ;)

simple question: How can i encode an wav source to flac? command lines to different flac compression levels (0 to 8)?
"eac3to source.wav dest.flac"

As nautilus7 wrote, there's no parameter to switch compression levels. eac3to always uses the highest level (8).

madshi

5th February 2008, 10:06

The Fugitive HD DVD did not work before. But plays perfect remuxed with eac3to v2.20!!
Nice... :)

Is there a way to solve video gaps?
Only title with video gap before was Born on the 4th of July. But with later eac3to version gap was not there.

Gap of 3 frames in Matrix Revolutions...
This gap in Revolutions was reported before as being at the beginning of the credits. It's quite possible that this is a real gap. But since usually credits are terribly boring there's not really a good way to check whether audio sync after the gap is still correct or not. My advice for Revolutions would be to play the MKV file and check the timecode which eac3to reported for the gap to see if the final remuxing result is ok or not. If you don't see any obvious artifacts/stuttering caused by the gap, everything should be just fine. Of course you can also redo the remux with the "-ignoreGaps" parameter. But I think you will not notice any difference, because (as I said) usually it's impossible to check audio sync in credits.

Remuxing Miami Vice and eac3to reports alot of gaps and overlaps.

Video has a gap of 4 frames at playtime 0:06:10.
Video overlaps for 4 frames at playtime 0:06:19.
[...]
Video has a gap of 4 frames at playtime 1:31:25.
Video overlaps for 4 frames at playtime 1:31:55.
<sigh> Will need to find a better way for handling gaps. Probably I'll end up doing two pass encoding for those rare movies with *real* gaps in them. Miami Vice doesn't seem to have any real gaps, just very unstable timestamps. Whenever there is a gap/overlap combination with the same frame number for both the gap and the overlap, the gap never was a real gap but just bad timecodes in the EVO file. In this situation you can remux with the "-ignoreGaps" parameter. A future eac3to version will automatically take care of such problems as with Miami Vice. However, when there are real gaps (such as seemingly the case with Matrix Revolutions) this future eac3to version will probably need to trash the first muxing and redo it with proper gaps.

Trying to convert Stranger Then Fiction LPCM to FLAC.
The LPCM track is extracted with xport. 3.63GB in size.
The FLAC track created by eac3to is 5MB. Tried several times with the same result. Whats wrong here?
Can you please post the full eac3to log? Does your harddisk have enough space?

madshi

5th February 2008, 10:12

You are correct. It is just matrixing the original PCM and setting a couple flags. As for the matrixing, I haven't looked into exactly how to do it. But here are the flags:
AftenContext.meta.xbsi2e = 1;
AftenContext.meta.dsurexmod = 2;
Definitely, the 7th. Center surround channel in a 6.1 DD EX stream is matriced using both rear surround channels, and I'm also sure it is adding one to another CS= (LS + RS) x A
That's because DD EX has to be compatible with AC3 5.1, which playback the CS on both LS and RS channels so it's an addition and not a susbstraction.
The "A" factor is unknown (by me) and also de flags.
Maybe looking at how Sound Forge does it can help (http://forum.doom9.org/showthread.php?t=124721)
Thanks, Justin and K-rnivoro. Now I'd just need to figure out the correct matrixing parameters. Also I'm wondering:

(1) Would I need to apply a 90° phase shift? Because I don't have any code for that.
(2) How to handle 7.1 input? Would it be correct to average the two back channels together and then handle the track just like a normal 6.1 track?

rickardk

5th February 2008, 10:18

Thanks for all answers!

There is a small stutter in Revolutions. I will redo with "-ignoreGaps".

Will also redo Miami Vice.

After demuxing the LPCM track from Stranger Then Fiction with TSremux, eac3to (v2.13) produced a perfect flac.

The_Keymaker

5th February 2008, 18:09

I Will release an updated version of EAC3toGUI shortly incorporating new options. Your patience is appreciated.

The_Keymaker

kurt

5th February 2008, 18:10

I Will release an updated version of EAC3toGUI shortly incorporating new options. Your patience is appreciated.

The_Keymaker
your work is much appreciated :thanks:
(as well as madshis' :))

bertox

5th February 2008, 18:33

@madshi:

there's no parameter to switch compression levels. eac3to always uses the highest level (8)

Can you implement this?? Take longest time to compress in this level....and the players "stressed" to much unnecessarily.

Maybe two (or three) options only is fine...4 and 8.

4 is sufficient to me.

@The_Keymaker:

I Will release an updated version of EAC3toGUI shortly incorporating new options. Your patience is appreciated.

Can you add MLP files as source???? Since eac3to YES support them..i'm really tired of renaming every MLP extension to *.thd or *.dtshd...to be accepted in this GUI. Please. I won't to see "unsupported file type" anymore......please. Remember allowing to re-map channels like .thd files too.... :D

Thanks.

nautilus7

5th February 2008, 19:23

Can you implement this?? Take longest time to compress in this level....and the players "stressed" to much unnecessarily.

Maybe two (or three) options only is fine...4 and 8.

4 is sufficient to me.I believe you can use the standalone flac.exe to encode as you wish.

bertox

5th February 2008, 19:29

I believe you can use the standalone flac.exe to encode as you wish.
...well...i see

I actually using flac.exe..., or Foobar..., or dbpoweramp converter...

But i want to use eac3to to do simple one task in my MLP files decoding...; to flac...and not to wav and to flac later. 1 movement against 2.:)

Let's madshi answer to me..please.

Thanks anyway.

nautilus7

5th February 2008, 19:59

Madshi is the one who decides of course (he writes the code :D), but he asked some months ago if someone needed a lower compression in flac and everybody was happy with the way things are. Maybe if there is enough interest he 'll add support for that.

bertox

5th February 2008, 20:23

aha....just only two options...4 and 8...nothing more.., please.:D

The_Keymaker

5th February 2008, 20:56

Please download the latest version (v1.52) of EAC3toGUI from the following link:

http://www.sendspace.com/file/v2cj36

Version 1.52 changes and upgrades:
1. Added support for *.mlp source files
2. -core option supported (via DTS section)
3. -down2 option added
4. -blu-ray option added (in channel remapping section)
5. -mixlfe option added
6. -override option added

I will work on adding more options and enhancements soon. In the meantime you can always use the "Other Options" tab to input options manually.

Please report any bugs or suggestions for improvements.

Regards,
The_Keymaker

bertox

5th February 2008, 21:50

Yeeaaaahhhh baby!!!!!!!! :D

But, why aren't channels mapper to re-order channels for MLP???

Please fix it.

Thanks.

The_Keymaker

5th February 2008, 22:12

EAC3toGUI Version 1.53 with enabled channel channel mapping for *.mlp files.

http://www.sendspace.com/file/smzxno

The_Keymaker

bertox

5th February 2008, 22:44

Yeeaaaahhhh baaaabbbyyyyyyy!!!!!!!!!!!!!!!!!!!:D.:D.:D.:D

Now, what is "conversion options 2" blank tab??

Why -big and -24 default options to MLP files? This is new thing for MLP files...You just source MLP files like THD files who has -big and -24 options??

Thanks for all.

Snowknight26

6th February 2008, 04:21

4.1 MLP files are converted to 5 channel FLAC files.

The order of the MLP is Lf, Rf, LFE, Ls, Rs, but the FLAC is (according to ffdshow) Lf, C, Rf, Ls, LFE.
If I convert the FLAC to WAVs, it extracts Lf, Rf, C, Rs, Lf (its not what ffdshows, but its wrong nontheless).

eac3to.exe "track-01-[Lf Rf LFE Ls Rs].mlp"
MLP, 4.1 channels, 24 bits, 96khz

eac3to.exe "track-01-[Lf Rf LFE Ls Rs].mlp" track01.flac
MLP, 4.1 channels, 24 bits, 96khz
Encoding FLAC...
Creating/writing file "track01.flac"...
eac3to processing took 17 seconds.
Done.

eac3to.exe track01.flac
FLAC, 5 channels, 0:01:12, 24 bits, 96khz

eac3to.exe "track-01-[Lf Rf LFE Ls Rs].mlp" track01.wavs
MLP, 4.1 channels, 24 bits, 96khz
Writing WAVs...
Creating/writing file "track01.SL.wav"...
Creating/writing file "track01.R.wav"...
Creating/writing file "track01.L.wav"...
Creating/writing file "track01.SR.wav"...
Creating/writing file "track01.C.wav"...
eac3to processing took 9 seconds.
Done.

bertox

6th February 2008, 05:51

@Snowknight26:

In ffdshow you can change the order of channels as yout own taste.

Another thing...try to change de order of channels. I'm using eac3togui downmixing a 5.0 source without problems.

Place in this order in channel mapper: 0,1,4,2,3

You want to try using SoundForge too, for graphical viewing the orders of channels, cause' very often, in 4.1 or 5.0 sources they are truly bad accommodate!! Compare this with a 5.1 file in SoundForge. You can see that in the 4.1 source the LFE channel stays in number 5 channel (Lf Rf Ls Rs LFE), despite of its name (track-01-[Lf Rf LFE Ls Rs].mlp); when in 5.1 source it always stays in the meddle of fronts and rears channels (Lf Rf (C) LFE Ls Rs). The 5.1 mix show you the mix matrix as it should be. You follow it, then you get it...:D

Hope that helps.

Snowknight26

6th February 2008, 06:08

According to an ffmpeg mailing list, MLP 4.1 is "left, right, lfe, surround left, surround right"; this holds true for my file as well.

Plus, if I try to change the channel mapping, I have to include the 6th channel which happens to be nonexistant. Not only that, but, say, if I hadn't known the original mapping, how am I supposed to relate 0,1,2,3,4,5 to FL,FR,C,LFE,SR,SL respectively or whatever the correct # to channel relation is?

bertox

6th February 2008, 06:29

According to an ffmpeg mailing list, MLP 4.1 is "left, right, lfe, surround left, surround right"

ok...but...very often, in 4.1 or 5.0 sources they are truly bad accommodate!!

Now.... if I hadn't known the original mapping, how am I supposed to relate 0,1,2,3,4,5 to FL,FR,C,LFE,SR,SL respectively or whatever the correct # to channel relation is?
You better...try using SoundForge too, for graphical viewing the orders of channels

If you use SoundForge then you fastly recognize all channels, is very easy, believe me. SoundForge let's you visualize in a graphical mode the waves of an sound and listen every single channel. You'll recognize every separate channel doing some probes and using your brain.
Try it.:)

if I try to change the channel mapping, I have to include the 6th channel which happens to be nonexistant

When you re-mapp the channels with eac3togui, no 6th channel added at all. Just eac3to omits it.:) I say 0,1,4,2,3....without number 5...just leave it.

tebasuna51

6th February 2008, 12:26

4.1 MLP files are converted to 5 channel FLAC files.

The order of the MLP is Lf, Rf, LFE, Ls, Rs, but the FLAC is (according to ffdshow) Lf, C, Rf, Ls, LFE.
If I convert the FLAC to WAVs, it extracts Lf, Rf, C, Rs, Lf (its not what ffdshows, but its wrong nontheless).

Seems flac only support one kind of 5 channel: Lf, Rf, C, Ls, Rs
flac 1.2.1, Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007 Josh Coalson
...
5e_LRFlr.wav: ERROR: WAVEFORMATEXTENSIBLE chunk with unsupported channel mask=0x003B
...
5e_LRCFc.wav: ERROR: WAVEFORMATEXTENSIBLE chunk with unsupported channel mask=0x010F
...
5e_LRClr.wav: wrote 632393 bytes, ratio=0,066 =>OK (channel mask=0x0037)

Then only 3/2 files can be encoded. Don't work with 3/1.1 or 2/2.1 like your MLP 4.1 is.

Edit: Curious, the encoded like channel mask=0x0037 (FL, FR, FC, BackLeft, BackRight) are decoded by flac like channel mask=0x0607 (FL, FR, FC, SideLeft, SideRight)