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

[tebasuna51]

No progress. I make many test, but there are unpredictable peaks in sudden volume changes and are clipped with normaliced curves.

After a DRC film standard (with a theoretic max output -20 dB) is sure amplify +10 dB, but some sources go to 0 dB and others remains at -10 dB or less.

For ac3 decode, this problem (Normalize don't work after a DRC based in Sox filter), and don't now Dialog Normalization and DRC recommended in ac3 stream, make unusable this function.

If we need DRC decoding ac3 we can use ffdshow (with DirectShowSource). The output is equivalent to BeSweet-azid settings: -azid( -n1 -c normal)
After we can use Normalize without problems.

[dimzon]

@tebasuna51
brrr... seems like i realy misunderstood... maybe i need to read theoretical materials?
or i need step-by step explanation in easy english

[tebasuna51]

brrr... seems like i realy misunderstood... maybe i need to read theoretical materials?
or i need step-by step explanation in easy english
About peak problems or about correct method to apply DRC?

[dimzon]

About peak problems or about correct method to apply DRC?
About DRC itself. Sorry, I'm really far from DSP, and I even have very superfictial knowledge what is dB itself... I'm very confused, sorry..
Can you comment this post: http://forum.doom9.org/showthread.php?p=792680#post792680

[tebasuna51]

Can you comment this post: http://forum.doom9.org/showthread.php?p=792680#post792680
Yes is something like this. But:
"This transformation will increase volume for low-volume samples and keep volume untoched for hi-value samples."
is not exact, can be:
Low-volume increased, certain mid-range untouched and hi-volume decreased.

The Dolby defined curves are these. (http://pages.sbcglobal.net/wilsondr/ddcompprof.gif)

These curves are implemented in my initial post and can be used. But there are problems:

- The mid-range untouched is defined by Dialog Normalization in ac3 header:
the input audio (x-axis) must be attenuated before by 31 - (DialNorm) dB. For instance if DialNorm = -27 dB we need apply a Amplify(-4 dB) before use the appropriate curve.

- We don't know the appropriate curve. In ac3 stream each audio block have the needed amplify (+ or -) or don´t have nothing because this audio stream don't need DRC (option None in encoder).

- The theoretic max output is -20 dB (10% maxint, music & film standard), too low for standard audio equipments. And Normalize don't work after compand Sox function.

My first attempt to do "normalized" curves is similar to your curve: 100% (0 dB) input -> 100% output, "untouched for hi-value samples". Then I get another problem: "unpredictable peaks in sudden volume changes are clipped with normaliced curves".

To explain this problem a need a image. Maybe in other post.

[dimzon]

Yes is something like this. But:
"This transformation will increase volume for low-volume samples and keep volume untoched for hi-value samples."
is not exact, can be:
Low-volume increased, certain mid-range untouched and hi-volume decreased.

The Dolby defined curves are these. (http://pages.sbcglobal.net/wilsondr/ddcompprof.gif)

Hmm. Shifting this curves up we will transfer from "Low-volume increased, certain mid-range untouched and hi-volume decreased." to "Low-volume increased more, certain mid-range increased less and hi-volume untouched"...


These curves are implemented in my initial post with and can be used. But there are problems:

- The mid-range untouched is defined by Dialog Normalization in ac3 header:
the input audio (x-axis) must be attenuated before by 31 - (DialNorm) dB. For instance if DialNorm = -27 dB we need apply a Amplify(-4 dB) before use the appropriate curve.

- We don't know the appropriate curve. In ac3 stream each audio block have the needed amplify (+ or -) or don´t have nothing because this audio stream don't need DRC (option None in encoder).

- The theoretic max output is -20 dB (10% maxint, music & film standard), too low for standard audio equipments. And Normalize don't work after compand Sox function.

Brrr... Does I understand you right - you need actual mid-range magic value from AC3 header bcz this is not actual middle sound value == Int16.MaxValue/2 for Normalized audio but some another value?


To explain this problem a need a image. Maybe in other post.
Yes, picture, please!!!

[dimzon]

- The theoretic max output is -20 dB (10% maxint, music & film standard), too low for standard audio equipments. And Normalize don't work after compand Sox function.
Just one stupid question. Dosn't theoretic max output is -20 dB become actual max output is -20 dB when we using normalized audio as input? In this case (predictable max output) we doesn't need normalize AFTER, we must just Amplify to transfer actual max output is -20 dB to good value (i doesn't know wich value is equal to maxint for 16bit audio)

[tebasuna51]

Brrr... Does I understand you right - you need actual mid-range magic value from AC3 header bcz this is not actual middle sound value == Int16.MaxValue/2 for Normalized audio but some another value?
You can see in the picture. The mid-range is the linear part (1:1) in the curve and we need know DialNorm to apply the linear part to dialog values.

Imagine you have the dialog at -10 dB and apply the curve directly (20:1 near horizontal line), then the dialog are distorted. If you apply before -21 dB the dialog go to linear part and is not distorted (output at -31 dB).

Well, I don't like this obsolete Dolby system based in normalize the audio dialog at -31 dB, but ... work so.
The goal is maintain constant the volume across any source, but don't work because cheap audio equipment don't support this wide dynamic range and other sources (mp3, CD Audio, commercials in TV, ...) works with narrow dynamic range near the 0 dB.

[tebasuna51]

Just one stupid question. Dosn't theoretic max output is -20 dB become actual max output is -20 dB when we using normalized audio as input? In this case (predictable max output) we doesn't need normalize AFTER, we must just Amplify to transfer actual max output is -20 dB to good value (i doesn't know wich value is equal to maxint for 16bit audio)
I say:

"After a DRC film standard (with a theoretic max output -20 dB) is sure amplify +10 dB, but some sources go to 0 dB and others remains at -10 dB or less."

This is the peaks problem, be patient.:)

[dimzon]

After a DRC film standard (with a theoretic max output -20 dB) is sure amplify +10 dB, but some sources go to 0 dB and others remains at -10 dB or less.
Does it mean

Normalize()
DrcFilmStandard()


can go to 0 dB

[tebasuna51]

"After amplify +10 dB"

For instance, my test_channels.ac3 (only voice) after
Normalize()
DrcFilmStandard()
Amplify(10 dB)

go to 0 dB (can't accept further amplify without clipping)

[dimzon]

@tebasuna51
ok, i realize the problem, seems like i need to perform modification of NicAc3Source in order to implement build-in DRC control...
Shit, I hate C/C++ and I have not enought knowledge about AC3...
Will You help me with theory/math ?

Edit: Will try to asc Nic first: http://forum.doom9.org/showthread.php?t=108066

[tebasuna51]

ok, i realize the problem, seems like i need to perform modification of NicAc3Source in order to implement build-in DRC control...
I think this is the correct way.

Maybe is easy modify NicAudio.dll:

From a52dec-0.7.4/doc/liba52.txt
Dynamic range compression
-------------------------

void a52_dynrng (a52_state_t * state,
sample_t (* call) (sample_t, void *), void * data);

This function is purely optional. If you dont call it, liba52 will
provide the default behaviour, which is to apply the full dynamic
range compression as specified in the A/52 stream. This basically
makes the loud sounds softer, and the soft sounds louder, so you can
more easily listen to the stream in a noisy environment without
disturbing anyone.

If you do call this function and set a NULL callback, this will
totally disable the dynamic range compression and provide a playback
more adapted to a movie theater or a listening room.

If you call this function and specify a callback function, this
callback might be called up to once for each block, with two
arguments: the compression factor 'c' recommended by the bitstream,
and the private data pointer you specified in a52_dynrng(). The
callback will then return the amount of compression to actually use -
typically pow(c,x) where x is somewhere between 0 and 1. More
elaborate compression functions might want to use a different value
for 'x' depending wether c>1 or c<1 - or even something more complex
if this is what you want.

From NicAudio/m2_audio_ac3.cpp
bool m2AudioAC3Source::DecodeFrame() {
...
// Dynamic range compression
a52_dynrng(State, 0, 0);
AFAIK NicAudio cancel the default Dynamic range compression from liba52.
For test pourpose, only if you comment the a52_dynrng line and compile NicAudio.dll, I can test if default DRC from liba52 works ok.

After we can add a new parameter to NicAc3Source call, like DRC true/false and use:
if (!DRC) { a52_dynrng(State, 0, 0) };

[dimzon]

try http://www.mytempdir.com/487451

[dimzon]

btw
now I have perfect sample how to write AviSynth input audio plugins... Maybe I can join yet another MPASource into nicaudio and even add somethig like faadSource ;)

[tebasuna51]

try http://www.mytempdir.com/487451
Work!

Wait for complete test ...

[dimzon]

btw
there are
// Dynamic range compression
dts_dynrng(State, 0, 0);

in m2audio_dts.cpp

what does you know about DRC for DTS ?

[tebasuna51]

what does you know about DRC for DTS ?
Sorry, I don't know DTS format.

First test with new NicAc3Source: the DRC info in ac3 stream is applied, now work exactly like BeSweet -azid( -c normal )

Seems OK.

[dimzon]

Sorry, I don't know DTS format.
Ok

First test with new NicAc3Source: the DRC info in ac3 stream is applied, now work exactly like BeSweet -azid( -c normal )

Seems OK.
Very well. Any idea how to implement other DRC modes?

[tebasuna51]

Very well. Any idea how to implement other DRC modes?
AFAIK liba52 only apply the attenuation found in ac3 stream, don't have any algorithm to do DRC, remember (previous post):
"If you call this function and specify a callback function, this callback might be called up to once for each block, with two arguments ..."

But I think is not necesary, now work like ffdshow, Cyberlink PowerDVD 6, ...

[dimzon]

But I think is not necesary, now work like ffdshow, Cyberlink PowerDVD 6, ...
I really want to cover all besweet functionality. Seems like I must write some custom callback to realize additional DRC modes. Can You help me to write it (i need math and valid coefficients)

Talking about DRC in DTS, maybe it's possible to use same function in it? Does You have DTS samples for testing (i have not)

[dimzon]

http://forum.doom9.org/showthread.php?p=793573#post793573

[tebasuna51]

I really want to cover all besweet functionality.
Nice. The most important for me is accept any kind of wav input and split in mono waves (to send after to good ac3 encoders).
Maybe also cover WaveWizard functionality accepting wav > 4GB and WAV_FORMAT_EXTENSIBLE headers.

Seems like I must write some custom callback to realize additional DRC modes. Can You help me to write it (i need math and valid coefficients)
I think you need not easy algorithms. Just I learn a few with Compand Sox function and there are involved parameters like attack, dekay, delay, ... and I don't understand very much how work. See this image (http://www.mytempdir.com/489584)

I think is not a priority, with your new NicAudio.dll is enough for MeGUI.

About NicAc3Source I detect a bug decoding 5.0 (without LFE) ac3 streams. Make a wav 6 channels (with LFE empty) but wrong mapping C_SL_FR_FL_SR_LFE, need Getchannels(4,3,1,6,2,5).

Talking about DRC in DTS, maybe it's possible to use same function in it? Does You have DTS samples for testing (i have not)
I have a swedish test channel but decoded with NicAudio, foobar, Tranzcode or ffdshow (with errorrs) is always the same. Maybe don't have DRC (optional in DTS, like in Ac3).
I can't test DTS because I don't own a DTS encoder. Maybe other user can test your new NicAudio.dll.

[tebasuna51]

I still have problems with BeHappy new version.

Using like input this Test1.avs (3 stereo wav's to wav 6 channels)
flfr = WavSource("g:\Test_FL_FR1.wav")
fl = GetChannel(flfr, 1)
fr = GetChannel(flfr, 2)
clfe = WavSource("g:\Test_C_LFE1.wav")
c = GetChannel(clfe, 1)
lfe = GetChannel(clfe, 2)
slsr = WavSource("g:\Test_SL_SR1.wav")
sl = GetChannel(slsr, 1)
sr = GetChannel(slsr, 2)
MergeChannels(fl, fr, c, lfe, sl, sr)

With BeHappy v0.1.0.30756 (20051229) work fine, but with BeHappy v0.1.0.28107 (20060206) don't work, I get:
Starting job Test1.avs->Test.wav
Found Audio Stream
Channels=6, BitsPerSample=16, SampleRate=48000Hz
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Error: BeHappy.AviSynthException: AudioStreamSource
at BeHappy.Encoder.encode()

Please, if you need make a new BeHappy version for include NicAudio options, and is easy to modify:
- Don't include EnsureVBRMP3Sync(), or make it optional with a checkbox in Tweak section.
- Don't include the two AudioDubEx() if the checkbox Split is not marked.
- Accept negative values in Amplify.

Thanks.

[dimzon]

Maybe also cover WaveWizard functionality accepting wav > 4GB and WAV_FORMAT_EXTENSIBLE headers.
Isn't possible to accept such files just using DirectShowSource instead WavSource?

split in mono waves (to send after to good ac3 encoders).
I'm planning to perform some reverse-engineering of Dolby AC3 encoder Dll's in order to write my own CLI frontend for it (acceptable to encode 6ch WAV directly) but I plan to do it much more later...


I think you need not easy algorithms. Just I learn a few with Compand Sox function and there are involved parameters like attack, dekay, delay, ... and I don't understand very much how work. See this image (http://www.mytempdir.com/489584)
I'm just talking about standart DRC in addition to normal: Neavy/Normal/Light/Inverse


About NicAc3Source I detect a bug decoding 5.0 (without LFE) ac3 streams. Make a wav 6 channels (with LFE empty) but wrong mapping C_SL_FR_FL_SR_LFE, need Getchannels(4,3,1,6,2,5).
Seems like I found bug:
case A52_3F2R:
LFE = Flags && A52_LFE;
DecFlags = A52_3F2R;
if (LFE)
DecFlags |= A52_LFE;
ChannelCount = 6;
break;

In this case LFE variable is always = true.
Replaced to
LFE = 0!=(Flags &A52_LFE);
Same for DTS, try this binary: http://www.mytempdir.com/489796

I still have problems with BeHappy new version.
I have found same problem just 3 days ago. Unfortunally I can't say nothing about it except it's internal AviSynth problem (exeption is thrown by AviSynth) and it will occured sometimes (not every time). Try to open yor avs script using Import, not WavSource...


Please, if you need make a new BeHappy version for include NicAudio options, and is easy to modify:
- Don't include EnsureVBRMP3Sync(), or make it optional with a checkbox in Tweak section.
- Don't include the two AudioDubEx() if the checkbox Split is not marked.
- Accept negative values in Amplify.

All Your suggestions will be implemented in next build BUT I'm not planning to release new BeHappy version during at least next 2 weeks, sorry...

[dimzon]

Sorry, I don't know DTS format.

The DTS Coherent Acoustics standard (ETSI 102 114 v1.2.1), as published by the ETSI, is available here (http://pda.etsi.org/pda/queryform.asp) (search for DTS Coherent Acoustics)

[tebasuna51]

The DTS Coherent Acoustics standard (ETSI 102 114 v1.2.1), as published by the ETSI, is available here (http://pda.etsi.org/pda/queryform.asp)[/COLOR][/B])
After read this I check my test.dts and is encoded without DRC.

Then if anybody want do the test before a new BeHappy version support this issue, can create a new file in BeHappy folder:

NicAudioDRC.extension
<?xml version="1.0"?>
<BeHappy.Extension xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xmlns="http://workspaces.gotdotnet.com/behappy">
<AudioSource Name="NicAc3SourceDRC" UniqueID="79c27720-ab72-11da-a746-0800200c9a66">
<Script>NicAc3Source("{0}", DRC=1)</Script>
<SupportedFileExtension>ac3</SupportedFileExtension>
</AudioSource>
<AudioSource Name="NicDtsSourceDRC" UniqueID="880150e0-ab72-11da-a746-0800200c9a66">
<Script>NicDtsSource("{0}", DRC=1)</Script>
<SupportedFileExtension>dts</SupportedFileExtension>
</AudioSource>
</BeHappy.Extension>
With this there are two new input source methods:
NicAc3SourceDRC
NicDtsSourceDRC

The ac3 method work fine, and the bug with ac3 5.0 is also corrected.

Isn't possible to accept such files just using DirectShowSource instead WavSource?
Yes, configuring ffdshow Audio Decoder -> Codecs -> WAV -> All supported, I can open wav 32bit float, WAVE_FORMAT_EXTENSIBLE, ... (still not tested > 4 GB).
Is not a easy way, because I need uncheck other functions used for play and after do the conversion recheck them, but work.

I have found same problem just 3 days ago. Unfortunally I can't say nothing about it except it's internal AviSynth problem (exeption is thrown by AviSynth) and it will occured sometimes (not every time). Try to open yor avs script using Import, not WavSource...
Yes sometimes work. If you think the problem is solved using Import maybe we need a new source method:

AvsSource.extension
<?xml version="1.0"?>
<BeHappy.Extension xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xmlns="http://workspaces.gotdotnet.com/behappy">
<AudioSource Name="AvsSource" UniqueID="0aa78710-aafd-11da-a746-0800200c9a66">
<Script>Import("{0}")</Script>
<SupportedFileExtension>avs</SupportedFileExtension>
</AudioSource>
</BeHappy.Extension>
and disable .avs like extension for WavSource.

I'm planning to perform some reverse-engineering of Dolby AC3 encoder Dll's in order to write my own CLI frontend for it (acceptable to encode 6ch WAV directly) but I plan to do it much more later...
Great notice. We need a good free ac3 encoder better than ffmpeg, ac3enc, ...

But split in mono wavs is also needed for other reasons and I think is really easy to do, don't forget this please.

And at last, but not the last, thanks for accept my suggestions for new BeHappy release.

[dimzon]

@tebasuna51
does you tested new binary (5.0 issue)?

[scharfis_brain]

Without having read the whole thread: is nicaudio capable of putting out more than 16bit audio sampling precision?

[tebasuna51]

does you tested new binary (5.0 issue)?
Yes:
"The ac3 method work fine, and the bug with ac3 5.0 is also corrected."

[tebasuna51]

Without having read the whole thread: is nicaudio capable of putting out more than 16bit audio sampling precision?
Ac3 is a old codec and, AFAIK, internal precision is not greater than 16 bit, you can convert the output to any precision for further conversions.

[scharfis_brain]

internal precision 'may' be only 16bit, but due to the internal dynamic range control of AC3 (in German it is "Hüllkurve") you won't loose precision in silent scenes when decoding to 24bit.

Also you may decode every psycoacoustic coding to arbitary bit depths because of its frequency nature. Of couse this precision is faked a bit, but it is much better than going from 16bit to 24 (or higher) bit after decoding.

[dimzon]

Yes:
"The ac3 method work fine, and the bug with ac3 5.0 is also corrected."
Oh sorry, thanx a lot for testing. Does it mean I can do new stable release of NicAudio?

Talking about AC3 compression - I have not plans to do it right now but can you tell me wich AC3 encoder does You prefer?

[tebasuna51]

Does it mean I can do new stable release of NicAudio?
Yes for me. Maybe any dts test, but seems dts_dynrng work like the equivalent ac3.

Talking about AC3 compression - I have not plans to do it right now but can you tell me wich AC3 encoder does You prefer?
About free encoders you can see: http://forum.doom9.org/showthread.php?p=759503#post759503

- DarkAvenger (HeadAC3he) seems solve the low volume issue (50%) in their ac3enc.dll.

- DRC info must be generated in ac3 stream.

- Others parameters: DialNorm, MixLevels, ... must be user defined.

With non-free encoders, I only test the old and slow Sonic Foundry SoftEncode.

[dimzon]

@tebasuna51
FYI
fresh NicAudio http://forum.doom9.org/showthread.php?t=108112
fresh BassAudio http://forum.doom9.org/showthread.php?t=108254

Can You play with / test BassAudio ?

[NorthPole]

@tebasuna51

I read your post in this thread http://forum.doom9.org/showthread.php?p=759503#post759503

Thanks for the work...very interesting. What program did you use to test the ac3 output levels?

Did you use wavewizard with the dark avenger ac3enc.dll or were you using the headAC3he frontend?

[tebasuna51]

What program did you use to test the ac3 output levels?
We need decode to wav the ac3 stream with BeHappy-NicAudio or BeSweet-Azid without DRC, ota-normalize, downmix,...
After any wav editor Audacity, Goldwave, Audition...

Did you use wavewizard with the dark avenger ac3enc.dll or were you using the headAC3he frontend?
The HeadAC3he frontend is needed because have the Gain parameter. The ac3enc.dll from HeadAC3he don't work with WaveWizard, but the used is supplied for Dark Avenger to Johnman.

AFAIK the problem is how send this Gain parameter to ac3enc.dll, because is applied internally.

[NorthPole]

@tebasuna51

Thanks for the reply. Sorry about the off thread topic. I have been doing the same as you. Decoding to wave and checking the levels with cooledit. I have been using ffmpeg from http://celticdruid.no-ip.com/xvid/

@Dimzon or maybe @Dayvon

The new logo looks good.

I'll try the bassAudio decoder and let you know if I have any problems.

[tebasuna51]

BASSAUDIO TEST

Seems work ok with mp2, mp3, ogg, wma and stereo aac (LC, HE and PS)
Also work with ac3 5.1 (without DRC), 6 channel ogg and wma.
And with wav stereo and multichannel int16, int32, float32 with canonical or WAVE_EXTENSIBLE_FORMAT headers.

Now is not necesary DirectShowSource for wav, wma, ogg or stereo aac files.

PROBLEMS

With ac3 2.0 make a wav 6 channels R_e_e_L_e_e (e=empty channel).

With aac 5.1 sometimes make a wav6 with all channels equal to a distorted Center channel, and sometimes:
Starting job Test6.aac->Test6.wav
Found Audio Stream
Channels=6, BitsPerSample=16, SampleRate=48000Hz
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Error: System.NullReferenceException: Object reference not set to an instance of an object.
at BeHappy.AviSynthClip.dimzon_avs_getaframe(IntPtr avs, IntPtr buf, Int64 sampleNo, Int64 sampleCount)
at BeHappy.Encoder.encode()

or

Error: System.NullReferenceException: Object reference not set to an instance of an object.
at BeHappy.AviSynthClip.dimzon_avs_destroy(IntPtr& avs)
at BeHappy.AviSynthClip.System.IDisposable.Dispose()
at BeHappy.Encoder.encode()

[dimzon]

PROBLEMS

With ac3 2.0 make a wav 6 channels R_e_e_L_e_e (e=empty channel).

With aac 5.1 sometimes make a wav6 with all channels equal to a distorted Center channel...

Hm. Thanx a lot, seems like this is BASS problem



and sometimes
How often does this bug reproducable. Does it only for 5.1 AAC decoding via bassAudio?

[dimzon]

With aac 5.1 sometimes make a wav6 with all channels equal to a distorted Center channel, and sometimes
Can you provide small samples to send it to BASS developers?

[dimzon]

I'm planning to write AudioLimiter plugin

output = tahh(input*factor) / tanh(factor)

http://img64.imageshack.us/img64/3035/tmp1141993187p629f25xl.gif
factor = 1
factor = 1.5
factor = 2
factor = 3
factor = 4


Does anybody know another sutable formulas?

[dimzon]

Yet another filter to play with / test - AudioLimiter
http://forum.doom9.org/showthread.php?t=108470

[tebasuna51]

Can you provide small samples to send it to BASS developers?
Here is: Test6HE.7z (http://www.mytempdir.com/506506)
The source Test6HE.aac is decoded ok by Foobar v0.8.3 for instance.

[dimzon]

Here is: Test6HE.7z (http://www.mytempdir.com/506506)
The source Test6HE.aac is decoded ok by Foobar v0.8.3 for instance.
please test this binary with your AAC
http://www.un4seen.com/filez/2/bass_aac.dll

[tebasuna51]

please test this binary with your AAC
http://www.un4seen.com/filez/2/bass_aac.dll
Sorry, same output.
Works for you?

[gino25]

i try to transcode a file wav (44100 hz, mono) into a aac+ 16kbps.

In [1] i select my wav file as "Wavsource", in [2] and [3] all are empty. In [4] i select aac+ 16kbps as adts aac

but i have this error

Starting job bri.wav->bri.aac
Error: BeHappy.AviSynthException: Script error: there is no function named "AudioDubEx"
at BeHappy.AviSynthClip..ctor(String func, String arg, AviSynthColorspace forceColorspace, AviSynthScriptEnvironment env)
at BeHappy.Encoder.encode()

[gino25]

ah this is the avc generated by behappy

########################################
#Created by BeHappy v0.1.0.28107
#Creation timestamp: 11/03/2006 14.39.50
########################################
#Source FileName:C:\Documents and Settings\Enea\Desktop\bri.wav
#Target FileName:C:\Documents and Settings\Enea\Desktop\bri.aac
########################################

########################################
# [Source: WavSource]
########################################
WavSource("C:\Documents and Settings\Enea\Desktop\bri.wav")

EnsureVBRMP3Sync() # Some black magic to avoid desync

########################################
# [BeHappy: Create fictive 1000fps video for triming]
########################################
AudioDubEx(BlankClip(length=Int(1000*AudioLengthF(last)/Audiorate(last)), width=32, height=32, pixel_type="RGB24", fps=1000), last)

########################################
# [Encoder: Coding Technologies AAC+ @ 16 kbps as ADTS AAC]
########################################


########################################
# [BeHappy: Kill video]
########################################
AudioDubEx(Tone(), last)

[dimzon]

@gino25
instal AviSynth 2.56

[gino25]

@gino25
instal AviSynth 2.56

thank you