Wavewizard v0.54b

[bcn_246]

Do you think it would be possible to implement a function where you could set exactly the output time, by adding silence at the end/cropping the end audio?

I allways find this very useful when encoding to DVD to make sure all the streams are exactly the same lengh. I guess it would be best accurate to the millisecond.

Great tool by the way, keep it up.

Ben :)

[johnman]

I believe this should be possible to see upto the milisecond the legth of a file, but im not at home so i take a look at it this afternoon ...

[johnman]

It is possible to set the exact length of a file...
if you set the DELAY option on an change the filename you can change the length of the file.

[raquete]

If you have a question about the program, ask it in this thread so others might benefit from it.

is possible to encode one group of files? something like batch file?

thanks for this great encoder!

[raquete]

...one month later...

maybe the question was answered before but i can't find ...please,send me in the right direction or answer my doubt because i'm using Wavewizard following the "Steve's guide" and i'm encoding lots of AC-3 (one by one).

thanks.

[tebasuna51]

is possible to encode one group of files? something like batch file?
Is not possible with WaveWizard
...please,send me in the right direction or answer my doubt because i'm using Wavewizard following the "Steve's guide" and i'm encoding lots of AC-3 (one by one).
I don't know the "Steve's guide", but if you explain how use WaveWizard to encode ac3 (what kind of wav input, ac3enc.dll or SoftEncode) maybe there are another solution.

[raquete]

is not possible? (oh no)

here is the "Steve's guide" (Ursamtl):
http://forum.doom9.org/showthread.php?t=105684

read "Part 2: Merging or splitting files" that use Wavewizard.

:thanks: tebasuna51.

[tebasuna51]

here is the "Steve's guide" (Ursamtl):
http://forum.doom9.org/showthread.php?t=105684

read "Part 2: Merging or splitting files" that use Wavewizard.
I know this thread, just I post:
http://forum.doom9.org/showthread.php?p=792462#post792462
to modify the "Part 2: Merging or splitting files" because with the guide (first post) you obtain an ac3 with wrong channelmapping.
EDIT: Now the guide is correct. Sorry.

Really is difficult automatize this process (3 stereo input wav 32 bit float to encode to ac3 5.1), maybe with AviSynth and Bepipe ...

[ursamtl]

Actually that's no longer accurate. As I mentioned in my March 5 reply to you in the thread, I edited that section of the guide to correct the problem.

[tebasuna51]

Actually that's no longer accurate. As I mentioned in my March 5 reply to you in the thread, I edited that section of the guide to correct the problem.
Sorry ursamtl, is my fault:(
Your first post is ok now.
Bad memory and bad re-read :stupid:

[slothsquid]

Hi Johnman

Thanks for the wavewizard. It helps me a lot in transcoding wavs to DTS. I have a couple of questions

1. Do you any plan to integrate AF's tranzcode with ww?
2 Is there any way I can use WW in CLI?
3. It is more helpful, if you include the function to delete intermediate files in batch mode. As 6Ch wavs create huge files, harddisk filled so quickly.

Thanks
SS

[raquete]

slothsquid, (first post after register in Apr 2003...)
good "wish list".

@ tebasuna51
Really is difficult automatize this process (3 stereo input wav 32 bit float to encode to ac3 5.1), maybe with AviSynth and Bepipe ...
can you (maybe) do that? (please. :p )
now we have Aften to encode AC3-5.1 using the 6 channels(merged) from WW.

[tebasuna51]

can you (maybe) do that?
Now with Aften only need one step in WaveWizard (remmaping not needed).

There are also a double conversion not recommendable:
float -> Int 16 (AviSynth) -> float (Aften).

But if you have many conversions to do and/or don't want create the intermediate wav6 files ...

To automatize using AviSynth you need create .avs files like:
a = BassAudioSource("X:\Xpath\x_fLfR.wav")
fl = GetChannel(a, 1)
fr = GetChannel(a, 2)
b = BassAudioSource("X:\Xpath\x_CLFE.wav")
c = GetChannel(b, 1)
lfe = GetChannel(b, 2)
d = BassAudioSource("X:\Xpath\x_slsr.wav")
sl = GetChannel(d, 1)
sr = GetChannel(d, 2)
MergeChannels(fl, fr, c, lfe, sl, sr)

And after execute batch files like:
bepipe --script "Import(^X:\Xpath\x1.avs^)" | aften -b 448000 - x1.ac3
bepipe --script "Import(^X:\Xpath\x2.avs^)" | aften -b 448000 - x2.ac3
...

The problem is only write paths and names of the files. To help this job I write 3wav1ac3.vbs
' This file (3wav1ac3.vbs) must be
' in the same folder than bepipe.exe and aften.exe
'
' You can Drag & Drop the *fLfR.wav file/s over this file or
' over a ShortCut created on Desktop.

' Autor: tebasuna
'
Dim wsh, args, fso, fe, fe2, fe3, fg, fa
title="3wav1ac3 Script"
' get input files
Set wsh = WScript.CreateObject("WScript.Shell")
Set args = WScript.Arguments
If args.Count = 0 Then
MsgBox "Special Aften ac3 encoder script." & vbCR & _
"Use drag & drop to specify the *fLfR.wav sources.", vbInformation, title
WScript.Quit
End If
' check path
Set fso = CreateObject("Scripting.FileSystemObject")
' Search aften.exe at same folder than 3wav1ac3.vbs
aften = WScript.ScriptFullName
camino = left(aften, len(aften) - len(WScript.ScriptName))
aften = camino & "aften.exe"
If Not fso.FileExists(aften) Then
MsgBox "Don't found aften!" & vbCR & "(must be at '" & aften & "')", vbCritical, title
WScript.Quit
End If
' Search bepipe.exe at same folder than 3wav1ac3.vbs
bepipe = camino & "bepipe.exe"
If Not fso.FileExists(bepipe) Then
MsgBox "Don't found bepipe!" & vbCR & "(must be at '" & bepipe & "')", vbCritical, title
WScript.Quit
End If
' Options to encode. Here we can put defaults or capture other parameters.
msg = "Bitrate to encode."
msg = msg & chr(13) & "Accepted values:"
msg = msg & chr(13) & "320, 384, 448, 512, 576 or 640 Kb/s"
p = InputBox(msg, "Bitrate", "448")
opts = " -b " & p & "000 - "
'
aften = Chr(34) & aften & CHR(34) & Chr(32)
bepipe = Chr(34) & bepipe & CHR(34)
bat = 0 ' bat not open
For i = 0 To args.Count-1
fe = args(i)
' check input file
If fso.FileExists(fe) Then
pre = Left(fe, Len(fe) - 8)
suf = LCase(Right(fe, 8))
If suf = "flfr.wav" then
fe2 = pre & "CLFE.wav"
fe3 = pre & "slsr.wav"
If fso.FileExists(fe2) Then
If fso.FileExists(fe3) Then
' write .avs file
Set fa = fso.CreateTextFile(pre & ".avs", True)
fa.WriteLine "a = BassAudioSource(" & Chr(34) & fe & Chr(34) & ")"
fa.WriteLine "fl = GetChannel(a, 1)"
fa.WriteLine "fr = GetChannel(a, 2)"
fa.WriteLine "b = BassAudioSource(" & Chr(34) & fe2 & Chr(34) & ")"
fa.WriteLine "c = GetChannel(b, 1)"
fa.WriteLine "lfe = GetChannel(b, 2)"
fa.WriteLine "d = BassAudioSource(" & Chr(34) & fe3 & Chr(34) & ")"
fa.WriteLine "sl = GetChannel(d, 1)"
fa.WriteLine "sr = GetChannel(d, 2)"
fa.WriteLine "MergeChannels(fl, fr, c, lfe, sl, sr)"
fa.Close
' write lines in batch file
If bat = 0 then
bepipe = bepipe & " --script " & Chr(34) & "Import(^"
Set fg = fso.CreateTextFile(pre & ".bat", True)
End if
bat = bat + 1
bepip2 = bepipe & pre & ".avs^)" & Chr(34) & " | "
bepip2 = bepip2 & aften & opts & Chr(34) & pre & ".ac3" & Chr(34)
fg.WriteLine bepip2
bepip2 = "del " & Chr(34) & pre & ".avs" & Chr(34)
fg.WriteLine bepip2
End If
End If
End If
End If
Next
if bat > 0 then
fg.Close
MsgBox "Finish. Run the batch file.", vbInformation, title
else
MsgBox "Don't found sources.", vbInformation, title
End If
WScript.Quit
Edit: Added line:
bepipe = Chr(34) & bepipe & CHR(34)

[raquete]

tebasuna51 (cool helper)
:helpful: :goodpost:

thank you.
;)

[raquete]

tebasuna51,
running the batch file (i add "pause" in the end) give this error:


C:\Arquivos de programas\AftenGUI-1.1>bepipe --script "Import(^e:\Xpath\script.avs^)" | aften -b 448000 - x1.ac3

Aften: A/52 audio encoder
(c) 2006 Justin Ruggles

***************************************
BePipe by dimzon
***************************************
Script used:
# BEGIN
Import("e:\Xpath\script.avs")
# END


Scanning for Audio Stream...
ERROR: Can't find audio stream!
invalid wav file: -b

C:\Arquivos de programas\AftenGUI-1.1>pause
Pressione qualquer tecla para continuar. . ."

as the sources,paths,scripts are ok,the reason of the error is because my .waves are 48k-32bit?
:confused:

[tebasuna51]

To open in AviSynth 2.56 32 bit waves we need BassAudioSource, if you install the last BeHappy
shon3i package (http://forum.doom9.org/showthread.php?p=829754#post829754) you have BassAudioSource ready.

Sorry, I forget 1 line in 3wav1ac3.vbs (to work with paths with spaces):

'
aften = Chr(34) & aften & CHR(34) & Chr(32)
bepipe = Chr(34) & bepipe & CHR(34)
...
Edited in precedent post.

Now my log:
D:\Internet\AudioTest\usrmtl>"C:\Arquivos de programas\AftenGUI-1.1\bepipe.exe" --script "Import(^D:\Internet\AudioTest\usrmtl\Bueno_.avs^)" | "C:\Arquivos de programas\AftenGUI-1.1\aften.exe" -b 448000 - "D:\Internet\AudioTest\usrmtl\Bueno_.ac3"

Aften: A/52 audio encoder
(c) 2006 Justin Ruggles

***************************************
BePipe by dimzon
***************************************
Script used:
# BEGIN
Import("D:\Internet\AudioTest\usrmtl\Bueno_.avs")
# END


Scanning for Audio Stream...
Found Audio Stream
Channels=6, BitsPerSample=16, SampleRate=48000Hz (1)
Writing Header...
Writing Data...
Signed 16-bit 48000 Hz 6-channel
Done! 0 | bw: 60.0 | bitrate: 448.0 kbps


D:\Internet\AudioTest\usrmtl>pause
Presione una tecla para continuar . . .
(1) Original wav are 32 bit float, after BassAudio-AviSynth 16 bit Int.

In your log:
"invalid wav file: -b"

Maybe you need the last aften 0.3 or _dev from Kurtnoise (the parameters order are inverted).

[raquete]

oh boy....oh boy!!! :cool:

i had the aften 0.3 _dev in the AftenGUI folder,was only needed the "BassAudio.extension".
my "default bitrate" to encode AC3(music) is 512.
i edit the "batch" and the "3wav1ac3.vbs" to this bitrate.

look:
"C:\Arquivos de programas\AftenGUI-1.1>bepipe.exe --script "Import(^e:\Xpath\scri
pt.avs^)" | aften.exe -b 512000 - "e:\Xpath\x1.ac3"

Aften: A/52 audio encoder
(c) 2006 Justin Ruggles

***************************************
BePipe by dimzon
***************************************
Script used:
# BEGIN
Import("e:\Xpath\script.avs")
# END


Scanning for Audio Stream...
Found Audio Stream
Channels=6, BitsPerSample=16, SampleRate=48000Hz
Writing Header...
Signed 16-bit 48000 Hz 6-channel
Writing Data...
Done! 7 | bw: 27.3 | bitrate: 512.0 kbps


C:\Arquivos de programas\AftenGUI-1.1>pause
Pressione qualquer tecla para continuar. . ."


was not needed "drag and drop" over "3wav1ac3.vbs",
running only the "batch" give the result working perfectly.

thank you so much (again) tebasuna51
:cool: :cool:

[UVLaser]

Hi Johnman

Thanks for the wavewizard. It helps me a lot in transcoding wavs to DTS. I have a couple of questions

1. Do you any plan to integrate AF's tranzcode with ww?
2 Is there any way I can use WW in CLI?
3. It is more helpful, if you include the function to delete intermediate files in batch mode. As 6Ch wavs create huge files, harddisk filled so quickly.

Thanks
SS

:helpful: Good list. I am interested in #2 & 3. very much. I am having same problem. The batcher is not really useful me as I have small harddisk. I hope Johnman is still reading the posts.

:thanks:
UVLaser

[ReinerSchweinlin]

Hm, the download-link in post 1 is broken. Sorry for asking - I am lazy, but where can I download this tool?

[raquete]

don't be lazy. :p
here in this same page have a link to the guide where you'll find WW:
http://forum.doom9.org/showpost.php?p=841747&postcount=107

[ReinerSchweinlin]

Thanx :)

[raquete]

anyone can help me to extract 6 mono channels-16bit from 5.1 dtswav with WW? ( tebasuna51,are you around? )
with tranzcodegui is easy but only for 32 bits waves.
i read the johnman "general info" http://forum.doom9.org/showpost.php?p=662842&postcount=5
but i can't put it working.(maps,etc..i'm lost)

thanks so much.

[Austin Forgotten]

anyone can help me to extract 6 mono channels-16bit from 5.1 dtswav with WW? ( tebasuna51,are you around? )
with tranzcodegui is easy but only for 32 bits waves.


Hi there, since I might not complete Tranzcode anytime soon since I've been working on a gui version which hopefully will be able to display DD/DTS decoded channel waveforms, where you would (again hopefully) be able to select section bounded by frames and cut/paste/trim/split visually and hope to add the playback as well. What was my point, oh ya since I ceased work on the console ver (called beta 4) in March (which still lacks the 'Two-Pass', and stereo down mixing), as the next step would be to create a dll from it. Anyway I just posted the beta 4 ver (March), on my little website. You can access the file here, use this link:

http://tranzcode.byethost22.com/files/Tranzcode_v0.40(beta).rar

I still need to document some added commands. In short, it will (last I tested way back then), decode .ac3 and ddwav as well as .dts, .cpt and dtswav, and you can specify the output bit-depth (8,16,24,32 bit integer, and 32 & 64 bit floating pt), as well as resample to either 44100, 48000 and 96000 Hz. It will also do batch file processing (use the '*' character as the file delimiter (also use double quotes for filenames containing spaces-of course), and also a means to list channel files if you want to mux a multichannel wav. It can also strip channels from multichannel wavs. and you can specify the speaker settings since it will create in wave extensible format. I haven't done this for awhile muxing, err.. actually used/tested it since March, so bare with me. Oh ya, it will produce an accurate info on the DD/DTS files using hmm.. I forgot, think it's the /i (file info calculated)i, and /im (file info measured) commands if I recall correctly and also produce an info file .tzf (I think :) ) which I would later use for editing using the gui. So I suppose I'll have to write a short guide on some of the new commands.

This may still work with TranzGUI but may only work as before which will be quite limited as to its capabilities.

Copy 'Tranzcode.exe into the folder where the file you want to decode is in (or copy it into your i.e. 'C:\WINDOWS\system32' dir)

So to decode to 16 bit, open a 'DOS' window (Start/Accessories/Command Prompt), Change dir 'cd' to the dir and issue the following command (example)


tranzcode "my file.dts" /16

or if you want to resample the output as well..

tranzcode "my file.dts" /16 /44 [edited: thats WRONG! :) it sould be /44100 or /48000 or /96000

corrected: tranzcode "my file.dts" /16 /44100

Good luck, hope not to many bugz :P


Ok I finally ran some testing (err guess I should have done that before posting it). No matter there were some log messages which you can ignore for now. I'll repack a new compile to get rid of any troubleshooting msgs and clean up any garbage I left in it, and add a list of some new commands with parameters.

Also I'll add those commands here and totally hijack this thread completely :P (sorry John).

BTW I would also like to mention that I thank Johnmann for giving me the inspiration, along with some pointers on how to incorporate SSRC (Shibatch Tools) into Tranzcode for resampling/bit depth conversion, dithering etc. Thanks guy!

[EDIT]

Ok it didn't take long to find bugs right away (I found 2 already) - which weren't there on preveous testing, so I messed something up.

bug 1) When Upsampling & Bit Depth Converting together, it fails (at least on DTSWav and DDWav)

bug 2) When decoding DDWav the progress display stays at 1% but actually decoded fine.

Argh.. it may take awhile since I haven't look at the code so long ago. Sorry folks never expected this... Really! Already tarnished my bad reputation :(

Ok it should be ok for now, uploaded a slightly older working version.

Seems the link is not working, so just go to the website and download it from there:

http://tranzcode.byethost22.com

Cheers

-AF

[tebasuna51]

anyone can help me to extract 6 mono channels-16bit from 5.1 dtswav with WW?
WaveWizard can't decode dts, then a dtswav can't be used like input for WaveWizard.

You can use WaveWizard to convert the TranzGUI output to 16 bit int, or DTSParser to convert dtswav to dts an after decode to 6 mono wav 16 int with BeHappy.

Sorry I don't know dtswav -> 6 mono wav 16 bit int in only 1 step.

Edit: I don't see the Austin Forgotten post. We can try with Tranzcode v0.40

[johnman]

hello ppl, guess who is back :)
as you might have seen development of ww sort of halted last couple of months but i just wanted to drop by to see what has happened. Who knows what the future brings :). iirc the last thing i was working on was vst chaining and multichannel recording but to be honest i am not really sure :D.

[raquete]

@ johnman,welcome back.

@ tebasuna51 & Austin Forgotten

:thanks: so much!!

[johnman]

I have taken a look at the code and i was busy trying to chain vst's together. I wonder, is anyone still using the program ? even needfull things seems unreachable so i wonder if it would be worth the effort to atleast finish the vst chaining + multichannel recording?

It has been a while and it takes me a lot of effort to understand my 1000's lines of code :sly:

And to be honest, the program is to complex to really grasp. I should have focussed more on functionality because now its a sort mess with the mapping and the vst chaning and merging of files.

[raquete]

I wonder, is anyone still using the program ?
i can be sure!

[johnman]

thats nice but if thats all, i dont think further development would be usefull :rolleyes:

[ursamtl]

thats nice but if thats all, i dont think further development would be usefull :rolleyes:

Oh I think a lot of people are using ww. I've seen several comments to that effect. As for needfulthings, you should ask daphy about that. Last I heard, he was quite busy with work but I'm sure he hasn't completely given up. As for me, I intend to get back into surround once the autumn sets it. Lately I've been exploring, of all things, analog! I heard some transfers of vinyl LPs done from quality turntables and I must say they sounded better than anything I've ever heard on commercial CDs. So right now I'm exploring the possibility of purchasing a new turntable! Given the improved soundstage, etc., I'm actually excited about what kind of source these could be for 5.1!

[tebasuna51]

I wonder, is anyone still using the program ?.
I make a search
Key Word(s): WaveWizard
Posts Made By: tebasuna51
and I found 30 threads where I recommend the use.

[Skelsgard]

I make a search
Key Word(s): WaveWizard
Posts Made By: tebasuna51
and I found 30 threads where I recommend the use.

Same here, there´s pretty much no audio-related posts of mine where I don´t recommend (or actually order) to use WW.
Ww is my butt-saver mandatory wonder audio tool for anything surround related.
Keep the great work.. (please!!! :D )

[johnman]

It is somewhat of a strange and unique program and it is capable of doing somethings most other programs can't. But its also not very user friendly, but when you get it to do what you want its kind of cool :).

I've got a couple of free days now so i'll have a look if i can jumpstart my sourcecode and get it to work again.

now let me get a mirror up, and daphy.. if you are back again send me a msg ;).


!tebunasa, i know you have noticed a couple of bugs long long time ago, ill come to you soon about it and i hope you remember some of the bugs you found. Im afraid i dont :(. I even have a bit of a problem to use ww :P.

edit 2. my computer broke so i only have a onboard soundcard on a very old computer that produces a lot of hum out of my speakers but it will have to do for now i guess :P.


New Download link (http://www.rarewares.org/wavewiz/wavewizardv0.54b.zip)

[Skelsgard]

But its also not very user friendly
Really? The first time I used it I actually thought is was incredibly simple and right-to-the-point (no unnecesary blowed-up BS), and I´m like every other user.

[johnman]

Really? The first time I used it I actually thought is was incredibly simple and right-to-the-point (no unnecesary blowed-up BS), and I´m like every other user.

To REALLY understand ALL functionality it takes a while. i.e. if you wanna merge 3 stereo file to a single 6 ch file and encode the output in 1 step to dts... its possible but i bet most ppl need a few tries to get it to work :P.

But i agree that it is to the point :). But i would like it to be less functional and more usable. More functionallity in't always better imo.

[tebasuna51]

tebasuna, i know you have noticed a couple of bugs long long time ago, ill come to you soon about it and i hope you remember some of the bugs you found.
I only remember a problem in ursamtl upmix method (http://forum.doom9.org/showthread.php?p=792462#post792462):
"Seems that Stream manipulation and Channelmapping don't work together, the merged file in pass 5 is always in correct wav order with Channel mapping enabled or disabled."

But is solved in two steps for ac3enc.dll, and in one step without mapping thanks to the new encoder Aften with remapping included.

Maybe for a new release you need include the new ac3/aac encoders Aften, Nero, CT (winamp).
And don't forget a simple remapping option Dolby Prologic II downmix (http://forum.doom9.org/showthread.php?p=864319#post864319):
6->2#mix#inactive#Dolby Prologic II downmix
0.325 0.000 0.000 0.325 0.230 0.230 0.000 0.000 -0.282 0.163 -0.163 0.282

[johnman]

I have taken a look at the project and i really should have made more documentation. I have a little problem figuring it all out again and since i was busy with several things at once some functionallity doesn't work anymore so maybe i have to go back to the latest working release which is a real bummer since i came a long way with recording sound from the soundcard and vst's functionality and some other stuff :(.

[daphy]

Hiho,

Johnman, nice to see you back on this 'stage' :D
:)

[johnman]

Thank you daphy, nice to be back :)
I went deep into the code and i am almost back on track from where i left of some time ago.

[Robot]

I’m hoping someone can help me with this problem.

I have a 6-channel PCM wav that I’m trying to turn into an ac3 file. The problem I’m having is with the channel mapping. I cannot figure out how to map them correctly for ac3. No matter what I do, I seem to either get the center channel switched with a rear channel or the left switched with the center, etc. It’s driving me nuts!

I’m using Wavewizard to re-map the channels from the wav file. The wav file was originally encoded from a 6-channel AAC file if that helps.

Thanks

[Skelsgard]

Your output WAV PCM must be in this order for correct channel mapping: L-R-C-LFE-Ls-Rs.
What are you using for AAC --> WAV decoding?
If the tool doesn´t have a correct maping on decoding, it is gonna get you a while till you figure out what order is being used, as AAC is C-L-R-Ls-Rs-LFE (the LFE order might be wrong though, I don´t remember).
The AC3 encoder (i.e Aften) assumes that the order is right, so you get this:
C-L-R-Ls-Rs-LFE (a wrongly decoded WAV from AAC)
L-R-C-LFE-Ls-Rs (a correct WAV mapping)

So C goes to L, L to R, R to LFE, and so on and so forth.

Check your AAC decoding.
As last resource, strip into mono streams the 6-ch WAV (with Ww :D ), determine wich mono channel is wich, and remux with WAV order.

Cheers

[Robot]

This is the process I’ve been using.

1. I take the 6-channel AAC file (ripped from a QT file)

2. I put the AAC file into the Nero encoding tool and output it to a PCM WAV file.

3. I take the WAV and put it into Wavewizard (where I get stuck with the channel mapping).

4. I then take the newly created WAV and encode it to AC3 via Aften.

Thanks.

[Skelsgard]

Does the first WAV decoded thru Nero plays with correct mapping?

[Robot]

Does the first WAV decoded thru Nero plays with correct mapping?

Unfortunately, I don’t know. I don’t have a 5.1 system on my computer. I can only tell if the mapping is right if I burn to DVD. :(

[raquete]

if you have one wave(interleaved-6 channels),open in nero wave editor only to check the "map".

[Ebobtron]

does this help?
AAC center
front left channel
front right channel
rear left channel
rear right channel
LFE
Wav
front left
front right
front center
LFE
rear left
rear right

oops never mind

[Skelsgard]

A fast way I use to see the mapping is to play the WAV thru AC3filter, wich has an equalizer that shows you the levels of each audio channel. This way avoids having to strip the channels and check each channel individually.
Usually the channel with little to no level througought the track is the LFE, the one that is almost always high specially on dialogs is the Center. Left and Right have normally the same levels at the same time, and peaks in the surround channels are not as frequent as the Frontal Left and Right, but there are still more than in the LFE. Also, Left comes always before Right. If u have channel 0 and channel 1, it´s bound to be that #0 is L and #1 is Right (the same with surround channels).
This might sound complicated but you only need to see it once and you´ll understand inmediately.

Cheers.

[raquete]

This might sound complicated ..is complicated and the levels of channels don't "obey" rules...less LFE of course,all depend of the source and his separations between channels..opening one short interleaved 6-channels in nero wave editor is easy to "see" the waveforms and to hear the channels(all together of separated)...will be faster and secure but...taste is taste. :-)

[Skelsgard]

Opening with AC3filter implies no need to trim, export, wait till the file is loaded into the audio app and peak-searched.
You just open it. It doesn´t get any faster, easier or less energy consuming than that.

Cheers.

[raquete]

right Skelsgard,seems easy to recognize when you know the source but for unknow...well,i like and use AC3filter but it only show me levels and not waveforms then again,for unknow sources is hard only whatching levels...i lose something in your hints? trust,i really want to know how you use in this case.thanks! :)