BeHappy - AviSynth based audio transcoding tool (UPD 19-07-2006)

[=Wolf=]

@tebasuna51
Yes....
- Open with NicAc3Source("17.ac3", DRC=0)
- Douwmix -> 2.0 with Stereo
- Normalize(100%)
- Encoded with Lame 128 Kb/s CBR
ONLI Lame broken... ogg, aac (32float) working is fine.... :(

[madshi]

The new options for rate/tempo/pitch have good names now (and helpful hints). Thank you guys!

[tebasuna51]

ONLI Lame broken... ogg, aac (32float) working is fine...
I use LAME 3.97 Release (http://www.rarewares.org/mp3.html)
Bundle: includes lame.exe, lame_enc.dll. (ICL9.1) 2006-10-03.

[tebasuna51]

AFAIK the actual ac3 free decoders (NicAudio, Azid, ffdshow, Ac3Filter, ...) can't decode DDP.
Only if you have installed in your system a DirectShow decoder filter capable to decode DDP, you can convert it to any other format.

[Rectal Prolapse]

Sonic Audio Decoder 4.2 can decode DD+ soundtracks.

After you install Sonic Cinemaster Decoder Pack 4.2, you can then use GraphEdit to make a .grf file, that can be loaded into BeHappy as a DirectShowSource.

Graph:

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

Hopefully this will work for you.

[idbirch2]

Hi, I'm trying to correct a gradual sync problem after a HD-DVD re-encode and am trying to stretch an AC3 file. Only ever so slightly as over a 2h21m film, it is only about 3.5s out by the end.

Anyway, when I try and do this with BeHappy 0.1.10.17947 I croaks straight away, in the log window, all I get is:

Found Audio Stream
Channels=6, BitsPerSample=16 int, SampleRate=48000Hz
Aften.exe -v 0 -b 384 -m 1 -readtoeof 1 -cmix 0 -smix 0 -dsur 0 -dnorm 31 -dynrng 5 - "H:\\poto-eac3to_happy.ac3"
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Error: System.ApplicationException: Abnormal encoder termination 1
at BeHappy.Encoder.encode()
#### Encoder StdErr ####

Aften: A/52 audio encoder
Version 0.06
(c) 2006-2007 Justin Ruggles, et al.

usage: aften [options] <input.wav> <output.ac3>
type 'aften -h' for more details.

Its as if the syntax used is wrong but how do I fix this? Thanks.

[tebasuna51]

"H:\\poto-eac3to_happy.ac3"

Seems the output file sintax is the problem

[idbirch2]

Sorry, I should have mentioned, I did spot that already and tried deleting the extra \ but the result is exactly the same. I can preview the output so I know the source filter is working. I don't suppose it matters but why does BH insert double \\'s because that wasn't an error on my part?

[Chumbo]

Sorry, I should have mentioned, I did spot that already and tried deleting the extra \ but the result is exactly the same. I can preview the output so I know the source filter is working. I don't suppose it matters but why does BH insert double \\'s because that wasn't an error on my part?
Below is an output from one of my many transcodes with BH and it never adds a double backslash. Internally, the code may use a "\\" to translate to "\" but that's standard.
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=48000Hz
Aften.exe -v 0 -b 640 -m 1 -readtoeof 1 -cmix 0 -smix 0 -dsur 0 -dnorm 27 -dynrng 5 - "E:\Media\audio\hf.ac3"
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Finalizing encoder

Check your input file path and name and make sure the input is okay. Try another drive and maybe even a simple file name, i.e., remove the dash and underscore, just to rule those things out.

[idbirch2]

Sorry, no joy. I moved the entire BeHappy folder to the root of C:\ in case it didn't like its location and also moved the target ac3 file to the same place, the result is the same. Here's how I have behappy set up:

http://img03.picoodle.com/img/img03/7/4/9/f_screenm_f2cf10b.gif (http://www.picoodle.com/view.php?srv=img03&img=/7/4/9/f_screenm_f2cf10b.gif)

What version of Aften are you using?

edit: I forgot, here's the output from this test:

Starting job test.ac3->test2.ac3
Found Audio Stream
Channels=6, BitsPerSample=16 int, SampleRate=48000Hz
Aften.exe -v 0 -b 384 -m 1 -readtoeof 1 -cmix 0 -smix 0 -dsur 0 -dnorm 31 -dynrng 5 - "C:\test2.ac3"
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Error: System.ApplicationException: Abnormal encoder termination 1
at BeHappy.Encoder.encode()
#### Encoder StdErr ####

Aften: A/52 audio encoder
Version 0.06
(c) 2006-2007 Justin Ruggles, et al.

usage: aften [options] <input.wav> <output.ac3>
type 'aften -h' for more details.

[tebasuna51]

What version of Aften are you using?

edit: I forgot, here's the output from this test:
Channels=6, BitsPerSample=16 int, SampleRate=48000Hz
Aften.exe -v 0 -b 384 -m 1 -readtoeof 1 -cmix 0 -smix 0 -dsur 0 -dnorm 31 -dynrng 5 - "C:\test2.ac3"
...
Aften: A/52 audio encoder
Version 0.06

Please use the last shon3i package (http://forum.doom9.org/showthread.php?p=974646#post974646)

For Aften you need rev449 at least, because with 2h21m 6 channnls you need "-readtoeof 1" not supported by v0.06.

[idbirch2]

Thanks very much for your help. I did originally start out using build 449 but I must have been doing something else wrong then as having gone back to 449, BH is now working :)

[tauka]

hi guys, i need a little help. i have the newest behappy, and i try to convert an ac3 file from pal to ntsc 6ch wav, its 2h24min long.. the problem is this: i open the wav with any prog, it shows 2:04:17.. in delaycut in the info actually the correct length is shown (2h31min), but in the target file the 2:04:17 again.. what can be the problem? the wav is actually around 4,8gig.. any suggestion? is it possible at all to convert such a long track with behappy?
thanks

update: well, i actually managed to get the correct length, but only if i load in the six separate mono wavs and stretch those.. strange.. but the ntsc track isnt good, it goes out of sync at the end, but its correct in the beginning.. with adobe audition i get with time stretch perfect timing.. :s

[tebasuna51]

hi guys, i need a little help. i have the newest behappy, and i try to convert an ac3 file from pal to ntsc 6ch wav, its 2h24min long.. the problem is this: i open the wav with any prog, it shows 2:04:17..
Then don't trust in these prog's.
in delaycut in the info actually the correct length is shown (2h31min), but in the target file the 2:04:17 again.. what can be the problem? the wav is actually around 4,8gig.. any suggestion?
Wav files have a header field with a 4GB limit (or 2:04:17 for wav 16 bit int, 6 channel, 48 KHz) but you have more data until 4.8 GB or 2h31m.
Edit: see also this link (http://forum.doom9.org/showthread.php?p=974973#post974973)

Don't worry Aften can encode this long track with the parameter: -readtoeof 1.
If you want encode to aac, NeroAacEnc also support long files with -ignorelength parameter.
is it possible at all to convert such a long track with behappy?
Of course, and don't need the huge intermediate wav file. Just I make a test with an ac3 7846.336 seconds. ( 2 h. 10 m. 46.336 s.) and timestretch 25 -> 23.976 directly to ac3 and the result is 8181.472 seconds. ( 2 h. 16 m. 21.472 s.) with less than a frame error (32 ms) than 7846.336*25/23.976 = 8181,448

[thuongshoo]

Hi ! Thank all for continueing to developt BeHappy. I love all new feature.
I used to use OggdropXP. It can estimate bitrate in Q mode. I hope that a new version will has this feature.
Bye!

[thuongshoo]

tebasuna1! You seem to be the one which do last correct. Can you send a copy of newest version of BeHappy to me ?
Thank you!

[tebasuna51]

tebasuna1! You seem to be the one which do last correct. Can you send a copy of newest version of BeHappy to me ?
Last links resumed:
2007-03-24 Last Shon3i BeHappy package (http://www.box.net/shared/nkihizx1dh)

2007-03-31 Chumbo mod (http://www.mytempdir.com/1277580) for Timestretch and others v0.1.10

2007-04-01 BeHappy mod (http://www.mytempdir.com/1278153) to support new enc_aacPlus.exe and MP4mux


Don't forget last Aften 0.07 (http://forum.doom9.org/showthread.php?p=996683#post996683)

[thuongshoo]

Oh! Thanks tebasuna51!
I didn't say clearly. I like source code :)
These pic decribe my words

[tebasuna51]

Last BeHappy sources (http://www.mytempdir.com/1318681).

Use http://www.imageshack.us/ to share pic's and avoid the "Attachments Pending Approval".

[tebasuna51]

I didn't say clearly. I like source code :)
These pic decribe my words
Sorry I can't reproduce your problem (I haven't Visual C++), I only obtain this:
http://img263.imageshack.us/img263/8694/ctaacuf6.png (http://imageshack.us)

[NoX1911]

I'm trying to convert AC3 to AAC 6ch. The source is a DVB stream (live concert). I don't know if it has DRC. How can i figure that out (metadata)?

If i use 'NicAC3Source (DRC)' what compression rate will be used? Most standalone dvd players have 3 profiles (low, med, high).

[tebasuna51]

I'm trying to convert AC3 to AAC 6ch. The source is a DVB stream (live concert). I don't know if it has DRC. How can i figure that out (metadata)?
You need read the header like in http://forum.doom9.org/showthread.php?p=993860#post993860
Read the next posts also.
If i use 'NicAC3Source (DRC)' what compression rate will be used? Most standalone dvd players have 3 profiles (low, med, high).
The attenuation values supplied in the ac3 stream are used without correction. I don't know what is: low, med, high. Maybe half, full, double?

[NoX1911]

Ok.. just hoped there would be something like this where you could additionally modify the DRC (or change metadata/Line Mode Profile of AC3).

http://www.imagebanana.com/img/ykt0wrx8/ac3.png

http://www.imagebanana.com/img/a6x8zg0/drc.png

[tebasuna51]

Ac3Filter have a DRC function independent of the DRC data in ac3 stream. NicAudio is a decoder to apply only the DRC data in the stream, if you want addicional/different compression you can use SoxFilter.

[NoX1911]

I think its not a different compressor. It is intended by Dolby to have impact on DRC like this to optimize volume for different environments (small speakers (high compression/flat sound), big speakers (no compression/full dynamic)). That's the way 'dialogue normalization' works. There are official dolby whitepapers linked in the faq sticky (http://forum.doom9.org/showthread.php?t=56020). I think i read it there...

My settop dvd player has some static values (low, med, high) for DRC that do the same what Ac3Filter does (i think) so this must be intended by dolby.

The second screenshot i posted above holds some metadata for 'line mode profile'. These could be preset values for that additional compression and may be used by Nic's DRC method.

But i'm not sure on that...

Edit:
solved... neroaacenc.exe has to be moved to 'encoder' folder.

Beside of that BeHappy doesn't work with my scenario. I use BeHappy with aac update.
NicAc3Source(DRC) to load my Ac3 file. If i output as avs script it works fine with mpc. If i enqueue and start the batch process following error msg appears in the log:

Translation:
Das System kann die angegebene Datei nicht finden = File not found

Starting job ggg.ac3->ggg.mp4
Found Audio Stream
Channels=2, BitsPerSample=32 float, SampleRate=48000Hz
encoder\neroAacEnc.exe -ignorelength -q 0.3 -if - -of "F:\ggg.mp4"
Error: System.ApplicationException: Can't start encoder: Das System kann die angegebene Datei nicht finden ---> System.ComponentModel.Win32Exception: Das System kann die angegebene Datei nicht finden
bei System.Diagnostics.Process.StartWithCreateProcess(ProcessStartInfo startInfo)
bei System.Diagnostics.Process.Start()
bei BeHappy.Encoder.createEncoderProcess(AviSynthClip x)
--- Ende der internen Ausnahmestapelüberwachung ---
bei BeHappy.Encoder.createEncoderProcess(AviSynthClip x)
bei BeHappy.Encoder.encode()

solved... neroaacenc.exe has to be moved to 'encoder' folder.

[tebasuna51]

My settop dvd player has some static values (low, med, high) for DRC that do the same what Ac3Filter does (i think) so this must be intended by dolby.
Not exactly. Your standalone, and soft decoders like PowerDVD, Azid, ffdshow, NicAudio, ... read the gain/attenuation value put in the ac3 stream for each block (5.33 ms) and apply the full value or a part only, don't calculate the DRC needed analyzing the sound decoded.
AC3Filter analyze the sound and calculate the DRC required.
The second screenshot i posted above holds some metadata for 'line mode profile'. These could be preset values for that additional compression and may be used by Nic's DRC method.

These presets are used at encoding time. The encoder calculate the appropriate gain/attenuation in function of these presets and the actual volume and put this value in the ac3 stream for each block (5.33 ms). The decoders don't need calculate, only read the values proposed by the encoder.

[thuongshoo]

This is a new version of BeHappy
http://www.box.net/shared/5tyvsczuxo
:)

[alwa]

This is a new version of BeHappy
http://www.box.net/shared/5tyvsczuxo
:)

@thuongshoo:You should list your changes.
I've noticed chages in OggVorbisEncoder.cs and CodingTechnologiesAAC.cs, please document them. :rolleyes:

Maybe the mod should have it's own repository, something like a project page on sf.net.

[tebasuna51]

In CodingTechnologiesAAC.cs only size parameters to avoid the problem in post #568.

In OggVorbisEncoder.cs size parameters and:
if (Quality <= 4) ApproximateBitrate = (double)(Quality + 2)*16 + 32;
if ((Quality > 4) && (Quality <= 8)) ApproximateBitrate = (double)(Quality - 4) * 32 + 128;
if (Quality > 8 && Quality <= 9) ApproximateBitrate = (double)(Quality - 8) * 64 + 256;
if (Quality > 9) ApproximateBitrate = (double)((Quality - 9)) * 179.8 + 320;
rbtnVBR.Text = string.Format("Variable Bitrate Q={0} approximate {1} kbs", Quality,ApproximateBitrate);

I don't use ogg files and don't know if is correct or not, but maybe the number of channels need to be considered.

[thuongshoo]

I've noticed chages in OggVorbisEncoder.cs and CodingTechnologiesAAC.cs, please document them.
@alwa:I'm sorry!
CodingTechnologiesAAC.cs: I resized and changed position of component to avoid the problem in post #568
OggVorbisEncoder.cs: added "ApproximateBitrate" value.
I don't use ogg files and don't know if is correct or not, but maybe the number of channels need to be considered.
@Tebasuna51: I tested 2 file and feel Ok. Everyone please re-test :)

It is quite right if I use oggenc2 while oggenc show a lower bitrate.
I'm using MediaInfo to observer information of media file
This feature come from OggDropXPd
http://img149.imageshack.us/img149/4330/approximatebitrateix7.th.png (http://img149.imageshack.us/my.php?image=approximatebitrateix7.png)

Link to download oggenc2 ( I prefer oggdropXPd V.1.8.9 using libVorbis v1.1.2 with IMPULSE_TRIGGER_PROFILE Option)
and OggDropXP
http://www.rarewares.org/ogg.html
Good luck! :)

[tebasuna51]

@Tebasuna51: I tested 2 file and feel Ok. Everyone please re-test

I say the approximate bitrate is only for stereo audio:

StereoTrack -> Ogg Q=3 approx. 112 kbs -> Real 113 Kb/s OK.
StereoTrack -> Downmix to mono -> Ogg Q=3 approx. 112 kbs -> Real 62 Kb/s
StereoTrack -> Upmix to 5.1 -> Ogg Q=3 approx. 112 kbs -> Real 441 Kb/s

I propose only a literal change:
Variable Bitrate Q=3 (approx. 112 kb/s for stereo)

[MuLTiTaSK]

this tool is amazing i mostly use it to convert .ac3 files i extract from my dvd's to .mp3 format and it works great

but now i want to use it to convert a 224Kbps CBR 2CH Stereo 48KHz .mpa i extracted with DGIndex from one of my tuner caps to .ac3 format so i can import it into a dvd authoring program

i never did this before so i'am not sure what encoder to use and what settings to set in BeHappy i tried with ffmpeg but not sure if i did it right

i'am using lastest Shon3i BeHappy package with 2007-04-01 BeHappy mod http://forum.doom9.org/showthread.php?p=998715#post998715

below is the log i got after the encode was done
hope someone can help me do this conversion right thanks in advance

settings i used in BeHappy
http://i12.tinypic.com/4ka87t5.png


Starting job test T01 DELAY 0ms.mpa->test T01 DELAY 0ms.ac3
Found Audio Stream
Channels=2, BitsPerSample=32 int, SampleRate=48000Hz
ffmpeg.exe -i - -y -acodec ac3 -ab 192 "H:\Caps\test T01 DELAY 0ms.ac3"
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Finalizing encoder
Complete
#### Encoder StdErr ####
FFmpeg version CVS, Copyright (c) 2000-2004 Fabrice Bellard
configuration: --enable-mingw32 --enable-memalign-hack --enable-gpl --enable-a52 --enable-dts --enable-mp3lame --enable-faac --enable-amr_nb --enable-faad --enable-amr_wb --enable-pp --enable-x264 --enable-xvid --enable-theora --enable-libogg --enable-vorbis
libavutil version: 49.0.0
libavcodec version: 51.9.0
libavformat version: 50.4.0
built on May 13 2006 18:31:30, gcc: 4.1.0 [Sherpya]
Input #0, wav, from 'pipe:':
Duration: N/A, bitrate: 3072 kb/s
Stream #0.0: Audio: pcm_s32le, 48000 Hz, stereo, 3072 kb/s
Output #0, ac3, to 'H:\Caps\test T01 DELAY 0ms.ac3':
Stream #0.0: Audio: ac3, 48000 Hz, stereo, 192 kb/s
Stream mapping:
Stream #0.0 -> #0.0

video:0kB audio:145896kB global headers:0kB muxing overhead 0.000000%

[tebasuna51]

@MuLTiTaSK
What is the problem?
Seems the conversion is ok. You have a 'H:\Caps\test T01 DELAY 0ms.ac3' about 145896 kB.
Don't worry about info from ffmpeg, all is ok.

Maybe you can use the last Aften (http://kurtnoise.free.fr/index.php?dir=Aften/&file=aften_rev511.zip) like encoder with more options than ffmpeg.

[shon3i]

yep, everythig is normal like log says, if file is playable, there is nothing to do.

btw new package will be redy, at the end of this week

[MuLTiTaSK]

@MuLTiTaSK
What is the problem?
Seems the conversion is ok. You have a 'H:\Caps\test T01 DELAY 0ms.ac3' about 145896 kB.
Don't worry about info from ffmpeg, all is ok.

Maybe you can use the last Aften (http://kurtnoise.free.fr/index.php?dir=Aften/&file=aften_rev511.zip) like encoder with more options than ffmpeg.

i wasnt sure if BeHappy was reporting a error by looking at the log i was a little confused thank you guys for clearing that up for me

would i get better quality using aften to encode the .mpa to .ac3?

i notice it had more settings but i did'nt know which ones to set so i used ffmpeg ;/

would these settings be correct for my conversion thanks again
http://i13.tinypic.com/4lpfh1x.png

[tebasuna51]

These default settings are ok, equivalents to ffmpeg.

Aften is based in same code than ffmpeg but have some improvements, maybe not noticeable but there are.

If you want ensure the max volume without clipping you can use the DSP function Normalize 100%

[MuLTiTaSK]

worked like a charm thanks alot tebasuna51 & everyone else keeping this great program alive

shon3i your packages help a great a deal thank you very much for building them saves new users to the program alot of grief

i'am still surprised why more people dont use this gem

is there a reason BeHappy is not listed on sites like?

http://www.digital-digest.com/software/topcategory-23.html
http://www.videohelp.com/tools/sections/audio-encoders

[thuongshoo]

StereoTrack -> Ogg Q=3 approx. 112 kbs -> Real 113 Kb/s OK.


StereoTrack -> Downmix to mono -> Ogg Q=3 approx. 112 kbs -> Real 62 Kb/s
62 = 112/2
StereoTrack -> Upmix to 5.1 -> Ogg Q=3 approx. 112 kbs -> Real 441 Kb/s
I don't know how 5.1 file is encoded.
I used word "Approximate Bitrate" , not "Bitrate". We are using "Q mode"

[tebasuna51]

I don't know how 5.1 file is encoded.
I used word "Approximate Bitrate" , not "Bitrate". We are using "Q mode"

5.1 (441 Kb/s) is encoded with Q mode and literal:

"Variable Bitrate Q = 3 approximate bitrate 112 kbs"

And I propose a literal like:

"Variable Bitrate Q = 3 (approx. 112 Kb/s for stereo)"

[paranoid87]

are you sure? teba? that the 5.1 is encoded witht the Q mode?

[tebasuna51]

are you sure? teba? that the 5.1 is encoded witht the Q mode?
I'm sure about the commandline:

Channels=6, BitsPerSample=16 int, SampleRate=48000Hz
oggenc2.exe -Q --quality 3 -o "G:\Test.ogg" -

and the result is a ogg 5.1 with bitrate 441 (Foobar Properties)

[chros]

Last links resumed:
2007-03-24 Last Shon3i BeHappy package (http://www.box.net/shared/nkihizx1dh)
I'm using this version, but encoding to flac doesn't manage:
Starting job audio.ac3->audio.flac
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=48000Hz
flac.exe --best --force -o "F:\audio.flac" --silent --force-raw-format --endian=little --channels=6 --bps=24 --sample-rate=48000 --sign=signed --input-size=7172361216 -
Writing PCM data to encoder's StdIn
Error: System.IO.IOException: Using the pipe has finished.
source: ac3 , with NicAC3Source , filters: none, flac: best

[tebasuna51]

I'm using this version, but encoding to flac doesn't manage...


Works for me with small samples (--input-size=183057408):
Starting job Test.ac3->Test.flac
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=48000Hz
flac.exe --best --force -o "D:\Test.flac" --silent --force-raw-format --endian=little --channels=6 --bps=24 --sample-rate=48000 --sign=signed --input-size=183057408 -
Writing PCM data to encoder's StdIn
Finalizing encoder
Complete

But also crash with big sources (--input-size=7068791808):
Starting job 136_min.ac3->136_min.flac
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=48000Hz
flac.exe --best --force -o "G:\136_min.flac" --silent --force-raw-format --endian=little --channels=6 --bps=24 --sample-rate=48000 --sign=signed --input-size=7068791808 -
Error: System.ApplicationException: Can't start encoder: Cannot process request because the process (1096) has exited. ---> System.InvalidOperationException: Cannot process request because the process (1096) has exited.
...

The same source can be transcoded to wav or aac without problems, then seems a flac encoder problem.

[buzzqw]

i can confirm the bug using ffmpeg for piping to faac.
sothing is wrong in faac pipe input

BHH

[tebasuna51]

i can confirm the bug using ffmpeg for piping to faac.
sothing is wrong in faac pipe input

Are you talking about faac or flac?

[buzzqw]

ops.. misread... i mean faac not flac

BHH

[jordisound]

I've tried to convert an ac3 4 channels from an old DVD to 4 wavmono using behappy (installer 20070315). After process only appears FL, FR.
Using besweet i've got the 4 channels FL, FR, C, S (SL, SR are the same, and LFE is empty.)
So, behappy is not able to convert 4.0 to 4 wavmono? or have I to change the configuration?
Thanxs!

[Chumbo]

I've tried to convert an ac3 4 channels from an old DVD to 4 wavmono using behappy (installer 20070315). After process only appears FL, FR.
Using besweet i've got the 4 channels FL, FR, C, S (SL, SR are the same, and LFE is empty.)
So, behappy is not able to convert 4.0 to 4 wavmono? or have I to change the configuration?
Thanxs!
You have to feed your source using an AVS script and add the GetChannel command to grab each channel separately. Add each script to the queue and then run it to produce each wav file.

[tebasuna51]

I've tried to convert an ac3 4 channels from an old DVD to 4 wavmono using behappy (installer 20070315). After process only appears FL, FR.
Using besweet i've got the 4 channels FL, FR, C, S (SL, SR are the same, and LFE is empty.)
So, behappy is not able to convert 4.0 to 4 wavmono? or have I to change the configuration?

The problem is in the ac3 decoder (NicAudio.dll) and is detected in this thread (http://forum.doom9.org/showthread.php?p=977363#post977363)

When the ac3 is 3/1 (FL, C, FR, S) NicAudio make an automatic downmix to:
3/1.0 -> stereo downmix
lt = 0,4xFL + 0,3xC + 0,3xS
rt = 0,4xFR + 0,3xC + 0,3xS

If the ac3 is 2/2 (FL, FR, SL, SR) NicAudio works ok.

- To decode properly an ac3 3/1.0, with free tools, you need use Azid.exe directly with:

azid.exe -d3/1 -ol,r,c,sl input.ac3 output.wav

The output.wav is a 4 channel (FL, FR, C, S) wav and you can split in mono wav's with WaveWizard or Wav2mono (1)

- Also you can use Foobar2000 to Convert to a WAVE_FORMAT_EXTENSIBLE wav or, if you use Wav2mono with Foobar, directly to 4 mono wav's (1)

(1) The wav's are named FL, FR, BL, BR and must be FL, FR, C, S

[jordisound]

ok. i'll try and tell you if it works. thank you tebasuna.

edit:
it works.
The wav's are named FL, FR, BL, BR and must be FL, FR, C, S
exact! this is very important!