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Old 3rd January 2009, 21:29   #821  |  Link
Seraphic-
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Nero AAC Codec 1.3.3.0 was released a few weeks ago.
Is neroAacEnc built into BeHappy or do you just have to put the neroAacEnc in the BeHappy "encoder" folder before BeHappy can encode NeroAAC. (i've been doing the latter)

Also, does anyone have any experiance with "save extra non audio information"?
Generally, if you are going directly from an audio editor like adobe premiere/audition to an audio encoder like BeHappy for NeroAAC, would it be recommended to disable or enable "save extra non audio information"?

Last edited by Seraphic-; 3rd January 2009 at 21:51.
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Old 3rd January 2009, 23:01   #822  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
You are writing always INT wav files.

In AviSynthAudioEncoder.cs (line 482) you need only write the correct Format_tag:
thanks for the trick...it seems to work fine now.


btw, did you noticed that with any sources, the bits per samples returned as info is always 32 ? even with 16 bits sources ?

to reproduce w/ BeHappy : take an ac3 (2.0) and transcode it to aac without applying any dsp. At the end, check the log, you'll see that the bps is always 32. Is it due to float conversion or is it a bug ?
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Old 4th January 2009, 04:21   #823  |  Link
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Quote:
Originally Posted by Kurtnoise13 View Post
btw, did you noticed that with any sources, the bits per samples returned as info is always 32 ? even with 16 bits sources ?

to reproduce w/ BeHappy : take an ac3 (2.0) and transcode it to aac without applying any dsp. At the end, check the log, you'll see that the bps is always 32. Is it due to float conversion or is it a bug ?
NicAudio/Bass decoders always output 32 bits float (also many others functions work in 32 float). Behappy select the high resolution supported by the encoder, most the times supply 32 bit float.

Only lossless formats can be preserved.

How do you know than one ac3 have 16 bitdepth sources?
Also the dts field header about source bitdepth can be wrong (Surcode write 24 when sources are 16)
And what is the problem when supply the best precission know?.
Encoders convert to float any input most the times. Then we can skip two conversions.
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Old 4th January 2009, 04:35   #824  |  Link
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Quote:
Originally Posted by Seraphic- View Post
Nero AAC Codec 1.3.3.0 was released a few weeks ago.
Is neroAacEnc built into BeHappy or do you just have to put the neroAacEnc in the BeHappy "encoder" folder before BeHappy can encode NeroAAC. (i've been doing the latter)
Yes NeroAacEnc can't be build with BeHappy the you must dowload and include in same folder than BeHappy or in the sibfolder "encoder".

Now only exist 1 version (for Windows) and you don't need check the 'Use SSE CPU instructions' (must disapear for next version)

Quote:
Also, does anyone have any experiance with "save extra non audio information"?
Generally, if you are going directly from an audio editor like adobe premiere/audition to an audio encoder like BeHappy for NeroAAC, would it be recommended to disable or enable "save extra non audio information"?
The "extra non audio information" is always ignored. Only if wav files are >4GB and the "extra info" is writed at end of file, after the 'data' chunk can be treated as audio data and produce a final click.
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Old 27th January 2009, 19:39   #825  |  Link
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i am encoding Madagascar 2 for my psp but the final audio (.m4a) has a really low volume...
i rip the 6ch.ac3 track > remove dialnorm (eac3to [other free options for this?]) > transcoded with behappy with nicac3source + downmix to stereo + normalize to 100% >.mp4 output is very low volume!
please advise...
_
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Old 28th January 2009, 01:20   #826  |  Link
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Quote:
Originally Posted by b66pak View Post
i am encoding Madagascar 2 for my psp but the final audio (.m4a) has a really low volume...
i rip the 6ch.ac3 track > remove dialnorm (eac3to [other free options for this?]) > transcoded with behappy with nicac3source + downmix to stereo + normalize to 100% >.mp4 output is very low volume!
please advise...
_
Your process is more or less correct but if you need the sound for low end audio equipment you need compress the dynamic range (less quality but more volume after normalize).
Then:
- You don' need remove DialNorm, NicAudio don't apply DialNorm.
- Open with BeHappy-NicAc3Source(DRC), Configure at (...)
Then the high volume are attenuated and low volume amplified (Dynamic Range Compresion)
- Downmix
- Normalize at end (must be the last DSP function).
Now are max amplified without overflow
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Old 28th January 2009, 18:36   #827  |  Link
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ok...thanks a lot for the help...

i use this .avs


#
NicAc3Source("F:\audio.ac3", DRC=1)
#
#
caaa4eafb6a2f44a1ae9bbae242a91a24=ConvertAudioToFloat(last)
#
function faaa4eafb6a2f44a1ae9bbae242a91a24(clip a)
{
#
flr = GetChannel(a, 1, 2)
#
fcc = GetChannel(a, 3)
#
lfe = GetChannel(a, 4)
#
lfc = MixAudio(fcc, lfe, 0.2071, 0.2071)
#
mix = MergeChannels(lfc, lfc)
#
lrc = MixAudio(flr, mix, 0.2929, 1.0)
#
blr = GetChannel(a, 5, 6)
#
return MixAudio(lrc, blr, 1.0, 0.2929)
#
}
#
faaa4eafb6a2f44a1ae9bbae242a91a24(caaa4eafb6a2f44a1ae9bbae242a91a24)
#
#
Normalize(100.0/100.0)
#

and the volume level is higher...i am a little confused....why is behappy using normalize function before the downmix and not after?

i also find that Normalize(200.0/100.0) is busting the volume even higher...is this wrong?

by the way the psp is not low end audio...you must use headphones for proper audio output...
_

L.E. it would be very nice if someone experimented will make a tutorial (or guidance) for proper audio transcoding...
_

L.E.2 i have a .mp2 (2channel 192k cbr) from a TV recording that i need to transcode to .m4a...it is proper to normalize it to 100%?

#
NicMPG123Source("F:\audio.mp2")
#
#
Normalize(100.0/100.0)
#

what is the difference between the above and below (beside the level of normalization)?

#
NicMPG123Source("F:\audio.mp2", true)
#
#
_

best regards...

Last edited by b66pak; 28th January 2009 at 20:45.
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Old 29th January 2009, 01:26   #828  |  Link
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Quote:
Originally Posted by b66pak View Post
...
and the volume level is higher...i am a little confused....why is behappy using normalize function before the downmix and not after?
In your sample Normalize is after.
The function definition can be at any place, only the execution line is important.
You have Up and Down buttons to put the DSP functions at desired order.

Quote:
i also find that Normalize(200.0/100.0) is busting the volume even higher...is this wrong?
Yes, the sound is cliped and distorted

Quote:
what is the difference between the above and below (beside the level of normalization)?
Nothing, if you don't need any other DSP function after you can use the decoder included normalize.
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Old 31st January 2009, 20:25   #829  |  Link
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how can i add delayaudio() and downmixing 3 or 4 or 5 channels to stereo to BeHappy's extensions?
_

L.E. a trim() extension too...
_

L.E.2 considering that mediainfo.dll is free for use in other apps it is posible to add an info button to display the audio track info?
_

Last edited by b66pak; 31st January 2009 at 21:06.
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Old 1st February 2009, 00:06   #830  |  Link
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Quote:
Originally Posted by b66pak View Post
how can i add delayaudio() and downmixing 3 or 4 or 5 channels to stereo to BeHappy's extensions?

L.E. a trim() extension too...
You have the Delay and Split (Trim) boxes in (2) Tweak section.

You always can write your own avs scripts. We can't cover all the situations.

Select your desired downmix function:
Code:
function Dmix3Stereo(clip a) { # 3 Channels L,R,C or L,R,S
  flr = GetChannel(a, 1, 2)
  fcc = GetChannel(a, 3, 3)
  return MixAudio(flr, fcc, 0.5858, 0.4142)
}
function Dmix3Dpl(clip a) {    # 3 Channels only L,R,S
  flr = GetChannel(a, 1, 2)
  sl  = GetChannel(a, 3)
  sr  = Amplify(sl, -1.0)
  blr = MergeChannels(sl, sr)
  return MixAudio(flr, blr, 0.5858, 0.4142)
}
function Dmix4lStereo(clip a) { # 4 Channels L,R,C + LFE
  flr = GetChannel(a, 1, 2)
  fcc = GetChannel(a, 3, 3)
  lfe = GetChannel(a, 4, 4)
  clf = MixAudio(fcc, lfe, 0.2929, 0.2929)
  return MixAudio(flr, clf, 0.4142, 1.0)
}
function Dmix4qStereo(clip a) { #4 Channels Quadro L,R,SL,SR
  flr = GetChannel(a, 1, 2)
  blr = GetChannel(a, 3, 4)
  return MixAudio(flr, blr, 0.5, 0.5)
}
function Dmix4qDpl(clip a) {   # 4 Channels Quadro L,R,SL,SR
  flr = GetChannel(a, 1, 2)
  bl  = GetChannel(a, 3)
  br  = GetChannel(a, 4)
  sl  = MixAudio(bl, br, 0.2929, 0.2929)
  sr  = MixAudio(bl, br, -0.2929, -0.2929)
  blr = MergeChannels(sl, sr)
  return MixAudio(flr, blr, 0.4142, 1.0)
}
function Dmix4qDpl2(clip a) {  # 4 Channels Quadro L,R,SL,SR
  flr = GetChannel(a, 1, 2)
  bl  = GetChannel(a, 3)
  br  = GetChannel(a, 4)
  sl  = MixAudio(bl, br, 0.3714, 0.2144)
  sr  = MixAudio(bl, br, -0.2144, -0.3714)
  blr = MergeChannels(sl, sr)
  return MixAudio(flr, blr, 0.4142, 1.0)
}
function Dmix4sStereo(clip a) {# 4 Channels L,R,C,S
  flr = GetChannel(a, 1, 2)
  fcc = GetChannel(a, 3, 3)
  lrc = MixAudio(flr, fcc, 0.4142, 0.2929)
  blr = GetChannel(a, 4, 4)
  return MixAudio(lrc, blr, 1.0, 0.2929)
}
function Dmix4sDpl(clip a) {   # 4 Channels L,R,C,S
  flr = GetChannel(a, 1, 2)
  fcc = GetChannel(a, 3, 3)
  lrc = MixAudio(flr, fcc, 0.4142, 0.2929)
  sl  = GetChannel(a, 4)
  sr  = Amplify(sl, -1.0)
  blr = MergeChannels(sl, sr)
  return MixAudio(lrc, blr, 1.0, 0.2929)
}
function Dmix5Stereo(clip a) { # 5 Channels L,R,C,SL,SR -> Stereo
  flr = GetChannel(a, 1, 2)
  fcc = GetChannel(a, 3, 3)
  lrc = MixAudio(flr, fcc, 0.3694, 0.2612)
  blr = GetChannel(a, 4, 5)
  return MixAudio(lrc, blr, 1.0, 0.3694)
}
function Dmix5Dpl(clip a) {    # 5 Channels L,R,C,SL,SR -> dpl
  flr = GetChannel(a, 1, 2)
  fcc = GetChannel(a, 3, 3)
  lrc = MixAudio(flr, fcc, 0.3205, 0.2265)
  bl  = GetChannel(a, 4)
  br  = GetChannel(a, 5)
  sl  = MixAudio(bl, br, 0.2265, 0.2265)
  sr  = MixAudio(bl, br, -0.2265, -0.2265)
  blr = MergeChannels(sl, sr)
  return MixAudio(lrc, blr, 1.0, 1.0)
}
function Dmix5Dpl2(clip a) {  # 5 Channels L,R,C,SL,SR -> dpl II
  flr = GetChannel(a, 1, 2)
  fcc = GetChannel(a, 3, 3)
  lrc = MixAudio(flr, fcc, 0.3254, 0.2301)
  bl  = GetChannel(a, 4)
  br  = GetChannel(a, 5)
  sl  = MixAudio(bl, br, 0.2818, 0.1627)
  sr  = MixAudio(bl, br, -0.1627, -0.2818)
  blr = MergeChannels(sl, sr)
  return MixAudio(lrc, blr, 1.0, 1.0)
}
function Dmix6Stereo(clip a) {
  flr = GetChannel(a, 1, 2)
  fcc = GetChannel(a, 3, 3)
  lrc = MixAudio(flr, fcc, 0.3694, 0.2612)
  blr = GetChannel(a, 5, 6)
  return MixAudio(lrc, blr, 1.0, 0.3694)
}
function Dmix6Dpl(clip a) {
  flr = GetChannel(a, 1, 2)
  fcc = GetChannel(a, 3, 3)
  lrc = MixAudio(flr, fcc, 0.3205, 0.2265)
  bl  = GetChannel(a, 5)
  br  = GetChannel(a, 6)
  sl  = MixAudio(bl, br, 0.2265, 0.2265)
  sr  = MixAudio(bl, br, -0.2265, -0.2265)
  blr = MergeChannels(sl, sr)
  return MixAudio(lrc, blr, 1.0, 1.0)
}
function Dmix6Dpl2(clip a) {
  flr = GetChannel(a, 1, 2)
  fcc = GetChannel(a, 3, 3)
  lrc = MixAudio(flr, fcc, 0.3254, 0.2301)
  bl  = GetChannel(a, 5)
  br  = GetChannel(a, 6)
  sl  = MixAudio(bl, br, 0.2818, 0.1627)
  sr  = MixAudio(bl, br, -0.1627, -0.2818)
  blr = MergeChannels(sl, sr)
  return MixAudio(lrc, blr, 1.0, 1.0)
}
function Dmix6StereoLfe(clip a) {
  flr = GetChannel(a, 1, 2)
  fcc = GetChannel(a, 3)
  lfe = GetChannel(a, 4)
  lfc = MixAudio(fcc, lfe, 0.2071, 0.2071)
  mix = MergeChannels(lfc, lfc)
  lrc = MixAudio(flr, mix, 0.2929, 1.0)
  blr = GetChannel(a, 5, 6)
  return MixAudio(lrc, blr, 1.0, 0.2929)
}
function Dmix6StereoLfe2(clip a) {
  flr = GetChannel(a, 1, 2)
  fcc = GetChannel(a, 3, 3)
  lrc = MixAudio(flr, fcc, 0.2929, 0.2071)
  lfe = GetChannel(a, 4, 4)
  lrc = MixAudio(lrc, lfe, 1.0, 0.2071)
  blr = GetChannel(a, 5, 6)
  return MixAudio(lrc, blr, 1.0, 0.2929)
}
function Dmix6DplLfe(clip a) {
  flr = GetChannel(a, 1, 2)
  fcc = GetChannel(a, 3, 3)
  lrc = MixAudio(flr, fcc, 0.2613, 0.1847)
  lfe = GetChannel(a, 4, 4)
  lrc = MixAudio(lrc, lfe, 1.0, 0.1847)
  bl  = GetChannel(a, 5)
  br  = GetChannel(a, 6)
  sl  = MixAudio(bl, br, 0.1847, 0.1847)
  sr  = MixAudio(bl, br, -0.1847, -0.1847)
  blr = MergeChannels(sl, sr)
  return MixAudio(lrc, blr, 1.0, 1.0)
}
function Dmix6Dpl2Lfe(clip a) {
  flr = GetChannel(a, 1, 2)
  fcc = GetChannel(a, 3, 3)
  lrc = MixAudio(flr, fcc, 0.2646, 0.1870)
  lfe = GetChannel(a, 4, 4)
  lrc = MixAudio(lrc, lfe, 1.0, 0.1870)
  bl  = GetChannel(a, 5)
  br  = GetChannel(a, 6)
  sl  = MixAudio(bl, br, 0.2291, 0.1323)
  sr  = MixAudio(bl, br, -0.1323, -0.2291)
  blr = MergeChannels(sl, sr)
  return MixAudio(lrc, blr, 1.0, 1.0)
}

Quote:
L.E.2 considering that mediainfo.dll is free for use in other apps it is posible to add an info button to display the audio track info?
I suppose, yes.

You can try, the BeHappy code is public and free. But you can always ask to MediaInfo before open BeHappy
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Last edited by tebasuna51; 14th July 2012 at 01:27. Reason: two typos flr -> lrc
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Old 1st February 2009, 22:48   #831  |  Link
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thank you very much...this is very usefull...the reason for asking for it is that if you don't know about this avs scripts and try to downmix anything but 5.1 you get an error (in megui is worse because you get same number of channels as you input!!!)...also i suggest to rename "downmix to stereo" to "downmix 5.1 to stereo" or "5.1 to stereo" to avoid this confusion...
i also noticed that megui don't use anymore EnsureVBRMP3Sync() when transcoding from .mp3...is this obsolete?
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Old 2nd February 2009, 01:16   #832  |  Link
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Quote:
Originally Posted by b66pak View Post
i also noticed that megui don't use anymore EnsureVBRMP3Sync() when transcoding from .mp3...is this obsolete?
I don't know if is obsolete because there are changes in buffer sizes in last AviSynth releases.

The decoder used with BeHappy/MeGUI (NicAudio) don't need this tool, maybe with DirectShowSource, but using DirectShow we can't know the decoder used.
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Old 2nd February 2009, 16:35   #833  |  Link
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thanks for the downmix preset tebasuna51!

update in automkv!

BHH
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Old 17th February 2009, 16:05   #834  |  Link
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I made some changes to fix a crash that was occurring on my system and may affect others. The executable is available in this download which also includes the up to date release notes and the changed class file.

http://www.mediafire.com/?dyndymymnmo
Code:
2009-02-16
+ Added stability by handling exceptions in the main form's saveConfiguration method.
+ Added checks in the same method to make sure that items added to any collections do not already exist.
+ Project solution updated to Visual Studio 2008
@tebasuna51,
I sent you an email regarding the changes and the little mess I created on codeplex, i.e., extra Change Sets that can be removed if possible. Not sure if you got it or not.
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Old 20th February 2009, 12:28   #835  |  Link
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@Chumbo
Restored the sources in Codeplex, but I don't know how delete the empty/bad extra Change Sets.
Could you explain your crash (OS, AviSynth version, ...)?
I always compile using .NET (compile.bat), I don't know if the change to Visual Studio 2008 can affect others.

Some minor changes added:
- Kurtnoise fix for AvisynthWrapper.cs
- Low limit for NicAacEnc to 8 Kb/s (NeroDigitalEncoder.cs)
- SSRC SpeedUp and SlowDown methods (SSRC.extension)
- NicAc3Source internal downmix (simple DolbyProLogic) to stereo, this work with any source channels. (NicAudio.extension)
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Old 20th February 2009, 14:55   #836  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
@Chumbo
Restored the sources in Codeplex, but I don't know how delete the empty/bad extra Change Sets.
Could you explain your crash (OS, AviSynth version, ...)?
I always compile using .NET (compile.bat), I don't know if the change to Visual Studio 2008 can affect others.

Some minor changes added:
- Kurtnoise fix for AvisynthWrapper.cs
- Low limit for NicAacEnc to 8 Kb/s (NeroDigitalEncoder.cs)
- SSRC SpeedUp and SlowDown methods (SSRC.extension)
- NicAc3Source internal downmix (simple DolbyProLogic) to stereo, this work with any source channels. (NicAudio.extension)
I don't know that we can. I've tried everything over the last few days to get rid of the change sets that are not needed. Anyway, I hope you didn't get rid of the change in set 18494 because that one is fine.

The crash I was getting was this:

Which is why I put checks in the fix I added to make sure the collection items check for already existing items prior to adding because that's what's causing this exception.

This started happening after one of the builds last year but I was too lazy to check into it until now. It happened every time I closed the app which meant I lost all my state changes, e.g., the queue.
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Old 20th February 2009, 16:33   #837  |  Link
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<AudioSource UniqueID="58ab9132-50c8-11dc-8314-0800200c9a66"> is defined for RaWavSource in NicAudio.Extension.
Maybe you have a old RaWav.extension not needed now.
Also RaWav.dll must be deleted in AviSynth plugins, now is fully integrated in NicAudio.dll

The set 18532 is the 18494 with the 4 changes in my post.
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Old 20th February 2009, 19:21   #838  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
<AudioSource UniqueID="58ab9132-50c8-11dc-8314-0800200c9a66"> is defined for RaWavSource in NicAudio.Extension.
Maybe you have a old RaWav.extension not needed now.
Also RaWav.dll must be deleted in AviSynth plugins, now is fully integrated in NicAudio.dll

The set 18532 is the 18494 with the 4 changes in my post.
Good to know and you're probably right. Thanks.
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Old 21st February 2009, 19:20   #839  |  Link
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@tebasuna51 thank for update but where can i find it?
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Old 22nd February 2009, 03:05   #840  |  Link
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Quote:
Originally Posted by b66pak View Post
@tebasuna51 thank for update but where can i find it?
_
There are very little changes to do a new official release.

You have the BeHappy.exe with Chumbo changes in the Chumbo post (set 18494).

The Kurnoise patch don't affect to BeHappy, the NicAudio.extension and SSRC.extension can be downloaded from CodePlex (Source Code) and put in \extensions folder.

And, if you need the low limit for NeroAAcEnc to 8 Kb/s instead 16 Kb/s, you always can download the full set 18532 and double click to 'compile.bat' to obtain the last BeHappy.exe (in \release subfolder).
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BeHappy, AviSynth audio transcoder.
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