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

[tebasuna51]

it means BeHappy is started as 64-bit process and attept to load 32-bit dll...
to solve this problem we need rebuild BeHappy using target CPU x86 (not Any CPU)
it's also possible via special compiler key too
Hello, Dimzon. Thanks for your post, this question is out of my knowledge.

Last version was compiled with:
%windir%\Microsoft.NET\Framework\v2.0.50727\MSBuild BeHappy.csproj /t:Rebuild /p:Configuration=Release /p:Platform=x86 /p:OutputPath=.\Dist

and inside BeHappy.csproj also:
...
<PlatformTarget>x86</PlatformTarget>
...

The files (BeHappy.csproj and compile.bat) are in BeHappy source code.

[dimzon]

Hello, Dimzon. Thanks for your post, this question is out of my knowledge.

Last version was compiled with:
%windir%\Microsoft.NET\Framework\v2.0.50727\MSBuild BeHappy.csproj /t:Rebuild /p:Configuration=Release /p:Platform=x86 /p:OutputPath=.\Dist

and inside BeHappy.csproj also:
...
<PlatformTarget>x86</PlatformTarget>
...

The files (BeHappy.csproj and compile.bat) are in BeHappy source code.
Actually BeHappy works fine onto my Windows Server 2003 x64.
Unfortunally I have not Vista to debug this bug...

[dimzon]

finally System.BadImageFormatException solved
code (not binary) commited to codeplex

[cweb]

I had an error with the latest version, sorry I didn't keep track of it, converting mp3 to ac3 using the bundled aften, but I solved it by updating to the latest aften.

[Menedas]

I'm currently trying to time stretch a 6 channel FLAC, which I made from a eac3, and save it to a 6 channel WAV.

With version 0.1.0.35190 I get a WAV which seems to have asynchronous channels. It sounds not as it should.

With an upgrade to version 0.1.9.50202 I get this error message:

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

[tebasuna51]

I'm currently trying to time stretch a 6 channel FLAC, which I made from a eac3, and save it to a 6 channel WAV.

With version 0.1.0.35190 I get a WAV which seems to have asynchronous channels. It sounds not as it should.
Use you DirectShowSource in this version to open the LiW.flac?

With an upgrade to version 0.1.9.50202 I get this error message:

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

If you want use BassAudioSource instead DirectShowSource like flac decoder, you need in your 'c:\Program files\AviSynth 2.5\plugins' folder these 3 files (see the readme.txt in BeHappy\plugins folder):
BassAudio.dll
Bass.dll
Bass_flac.dll

[Menedas]

Use you DirectShowSource in this version to open the LiW.flac?

No I also use BassAudio here. With DirectShow Source I get this error:
Error: BeHappy.AviSynthException: DirectShowSource: couldn't open file C:\LiW.flac:
Unsupported format.
at BeHappy.AviSynthClip..ctor(String func, String arg, AviSynthColorspace forceColorspace, AviSynthScriptEnvironment env)
at BeHappy.Encoder.encode()



If you want use BassAudioSource instead DirectShowSource like flac decoder, you need in your 'c:\Program files\AviSynth 2.5\plugins' folder these 3 files (see the readme.txt in BeHappy\plugins folder):
BassAudio.dll
Bass.dll
Bass_flac.dll

Its still in the AviSynth plugins directory, else it would not work in version version 0.1.0.35190, too.

[tebasuna51]

Its still in the AviSynth plugins directory, else it would not work in version version 0.1.0.35190, too.

Yep, for this my first question.

And your SO, Vista, 32/64 bits?

[Menedas]

And your SO, Vista, 32/64 bits?

XP Pro 32

[tebasuna51]

XP Pro 32
I don't know what is the problem.

Try inserting the line (or equivalent):
LoadPlugin("C:\Program files\Avisynth 2.5\plugins\BassAudio.dll")
in the avs generated with 'Export Avisynth script'

[Menedas]

I have tried to copy the bassAudio.dll to different paths, thats why it is D root here, but I always get this message:

Error: BeHappy.AviSynthException: LoadPlugin: unable to load "D:\BassAudio.dll"

AviSynth is correct installed. For example MeGUI works.

[DenisKanone]

Gratulation for this great Programm. I use BeHappy since the first Release. With WinXP 32bit (NET.Framework 2) works the Programm perfect. But with Win VISTA 32bit (NET.Framework 3) i have a little Problem. The Programm encode the File correct, but after the Encode-Processe BeHappy close from self without Error-Reporting. When I will use Batch-Encoding (more Jobs), then BeHappy encode the first File (first Job) and then close from self.
Then I want me a new Feature: Fade In/Out with seperatly Preferences.

Sorry for my "School-English". I hope all understand me.

[dimzon]

I found some little bugs in the last released BeHappy ver.0.1.9.50202.
1.enc_aacplus limit bitrate is 16kbps not 8kbps,this is just converse to neroAacEnc.
2.neroAacEnc limit bitrate is 8k not 16,but the different neroAacEnc_SSE limit also is 16k no longer support 8k.
3.when I new get the behappy and download the neroaacenc form internet, put in behappy encoder folder, but its not work if I choose p4 support.bocs the original file name is "neroAacEnc_SSE.exe" not "neroAacEnc_SSE2.exe".So I must rename the codec.
All the really little bugs,but hope correct in next version,make BeHappy perfect..
if received,give me reply pls.
Chris
2008-06-28 14:12
tebasuna51, just rebuild last sources and make new relesase...

Reference: http://forum.doom9.org/showthread.ph...75#post1146075

I'm using windows vista 64, and i've got this bug!

Where I can find the binary of your bugfix?

Thank you

[tebasuna51]

tebasuna51, just rebuild last sources and make new relesase...

Done. New BeHappy v0.2.2.30338 released.

Please report if all bugs are solved

[Menedas]

Still have the same bug. :(

[tebasuna51]

Still have the same bug. :(

I don't know what is your problem, try this:

Put this Test1.exe (http://www.sendspace.com/file/yhulvl) in D:\, DoubleClick (execute) and accept decompress in D:\.

DoubleClick D:\Test1\Test1.bat and say me what happen.

[Menedas]

It opens cmd without output and one hidden window with the title "Incorrect BASS.DLL version" I could only see in the task bar.

[tebasuna51]

It opens cmd without output and one hidden window with the title "Incorrect BASS.DLL version" I could only see in the task bar.
Check if you have in your system other bass.dll with a version lower than 2.4

The three dll files in Test1 are 2.4 and work fine for me. Sorry.

[shon3i]

New package is out:

Based on BeHappy 0.2.2.30338 (2008-06-28)
All encoders and avsplugins are upto date.
New installer system: Inno Setup, to downsize package size, Installshield uses bigger overhead.

Link (http://tom.niko.users.sbb.co.yu/BeHappy_20080704.exe), Mirror (http://www.box.net/shared/epiguyhkco), Alternative site (tom.niko.users.sbb.co.yu)

[Menedas]

Check if you have in your system other bass.dll with a version lower than 2.4


Ok, I found the Problem. It was the bass_flac.dll in my AviSynth Plugin Directory which came for the previous BeHappy Package (installer). When I removed the bass_flac.dll the Test works. When I install the new Package from shon3i I get the error, that OptimFROG.dll could not been found, but process finished successful.

I will test later to convert my flac and tell the results.

[shon3i]

When I install the new Package from shon3i I get the error, that OptimFROG.dllConfirmed. Just remove bass_ofr.dll from avs plugins dir.

[chriszxl]

use enc_aacplus output .aac format...I can't give the --mpeg4aac order by Behappy..So only version2 aac I can get...not version 4.
but if output .mp4 or .m4a format mux by mp4box/mp4mux the defualt is aac version4

[tebasuna51]

use enc_aacplus output .aac format...I can't give the --mpeg4aac order by Behappy..So only version2 aac I can get...not version 4.
but if output .mp4 or .m4a format mux by mp4box/mp4mux the defualt is aac version4

Yep, the default output of last Winamp CT dll's is mpeg2, before was mpeg4.

Only when go to be put in mp4 container we change the header flag to mark mpeg4 (the data stream is always the same).

If there are any other reason to select mpeg2/4 flag we can put a checkbox in Configure CT-AAC GUI.

[masscamp24]

I download the installer package and tried to use ffmpeg ac3. The program crashed I got this message "Application has failed to start because optimFROG.dll was not found. Re-installing may fix the problem" I reinstall but get the same message. What does this mean. I'm useing windows xp system32.

PS I get the same message when I start VDUBMod and try to use WMP2 via AviSynth 2.58. I never had this problem before useing these programe. Can someone please help.

[Menedas]

Look three posts above your post.

[masscamp24]

That was quick I love this forum. Thanks Mate

[Menedas]

Now that I can handle flac files with Bass again, I tried to convert my 6 channel flac to aac. The resulting file is still does have async channels.

[shon3i]

Now that I can handle flac files with Bass again, I tried to convert my 6 channel flac to aac. The resulting file is still does have async channels.
What setting you use, CT or Nero AAC, can you split this flac and upload small segment here? You can aslo try to decode flac to wav, and report what you get?

[Menedas]

I also created a wav with the same problem. Output format seems not the problem. I think the TimeStretch is the problem, because if I only just convert the flac to aac without TimeStretch the channels are synchron.

Here is the header of the flac. Hope it works without the footer:
http://www.megaupload.com/?d=2UAALBSP

You can hear that it is unsync at 28 sec, when the narrator starts to talk.

I time stretched from 23,976 to 25.

[chriszxl]

When the source sampling rate is lower than 32000,use the enc_aacplus code will be error,as:
Error: System.IO.IOException:
at System.IO.__Error.WinIOError(Int32 errorCode, String maybeFullPath)
at System.IO.FileStream.WriteCore(Byte[] buffer, Int32 offset, Int32 count)
because of enc_aacplus only allows the source as 32000-48000,hope can present the correct prompt in Behappy,or Behappy can auto choose a resample rate to fit to encaacplus is better...
and another problem is I cant choose the bitrate to 8kbps when use neroaacenc,but actually neroaacenc can recoding to 8kbps(neroaacen_sse can not),bcos I really care about the file size when I make some stuff to my handphone...so fix it pls..

[tebasuna51]

When the source sampling rate is lower than 32000,use the enc_aacplus code will be error,as:
Error: System.IO.IOException:
at System.IO.__Error.WinIOError(Int32 errorCode, String maybeFullPath)
at System.IO.FileStream.WriteCore(Byte[] buffer, Int32 offset, Int32 count)
because of enc_aacplus only allows the source as 32000-48000,hope can present the correct prompt in Behappy,or Behappy can auto choose a resample rate to fit to encaacplus is better...
BeHappy is a GUI for AviSynth audio and don't check the input sources.
and another problem is I cant choose the bitrate to 8kbps when use neroaacenc,but actually neroaacenc can recoding to 8kbps(neroaacen_sse can not),bcos I really care about the file size when I make some stuff to my handphone...so fix it pls..
Maybe in next release.

[Deckard2019]

Just a small note : ffmpeg need more details for bitrates. So in ffmpeg.extension, "Value" tags become :
... <Value>-i - -y -acodec mp2 -ab 224kb "{0}"</Value> ...
... <Value>-i - -y -acodec mp2 -ab 192kb "{0}"</Value> ...

[Menedas]

Does someone checking my problem with the asynchronous channels after time stretch?

[tebasuna51]

Just a small note : ffmpeg need more details for bitrates. So in ffmpeg.extension, "Value" tags become :

Well, not in my version.

We can't support all ffmpeg versions/compiles.

Everybody is free to adapt ffmpeg.extension to their ffmpeg version.

I can't recommend ffmpeg to encode audio (float samples, channelmap bugs, ...)

[tebasuna51]

I also created a wav with the same problem. Output format seems not the problem. I think the TimeStretch is the problem, because if I only just convert the flac to aac without TimeStretch the channels are synchron.

Here is the header of the flac. Hope it works without the footer:
http://www.megaupload.com/?d=2UAALBSP.

Sorry but I can't recover the wav file.

BTW I need the original wav file (first minute) to try reproduce the problem.

[Menedas]

Whats the problem with the flac file? It is the original. Don't wanna have a very large wav lying around and have to upload it. Just use the flac and create a wav (or what ever format you like) with behappy and with time stratch as I said with the flac. It should work. As it does here. The conversion will take a while, I know. The same time as when you have the whole flac file.

Or do you mean that you don't have that bug with your created time stretched audiofile?

[Deckard2019]

Well, not in my version.
I'm talking about the version found in shon3i's package.

Everybody is free to adapt ffmpeg.extension to their ffmpeg version
Sure. shon3i too ;)

[tebasuna51]

I'm talking about the version found in shon3i's package.
...
Sure. shon3i too ;)

Yep, I think the next shon3i package come without ffmpeg.

[tebasuna51]

Whats the problem with the flac file?
I can't decode it.

[Menedas]

I wonder. Normally I'm the one who has problems :)

Ok, here is the header as wav file:
http://www.megaupload.com/?d=KRR980YC

[tebasuna51]

I wonder. Normally I'm the one who has problems :)

Ok, here is the header as wav file:
http://www.megaupload.com/?d=KRR980YC

Thanks. Now I have 33.9816 sec of wav 5.1 file with begin talk at 28 sec like you say (in FL, FR and FC channels in sync.).

I try a Timestretch 23.976 -> 25 and obtain a 32.5897 sec. wav file with Rate and Tempo modes, and the same time 33.9816 with Pitch mode.

In Tempo and Rate modes now begin talk at 27 sec. and in all modes FL, FR and FC are sync.

What async. you detect?

[Menedas]

Hard to tell how it sounds. Maybe its again only a problem on my side. So here is the file I get when I make a Timestretch on the header file I've uploaded:
http://www.megaupload.com/?d=PB0SI21B

I'm using the Rate Control "Tempo changed, pitch correction". Hope you can hear the difference to the original, else tell me which software do you use to hear it. I have tested it with foobar2000, ffdshow and vlc. All with the same problem.

[tebasuna51]

Hard to tell how it sounds.
...
Hope you can hear the difference to the original, else tell me which software do you use to hear it. I have tested it with foobar2000, ffdshow and vlc. All with the same problem.

Sorry, I can't hear the difference (Foobar2000). There are, of course, a pitch change, but async?

But, also, I can't see the diference. I use audio editors (Audacity, Wavosaur, ...) to see the exact ms.

[Menedas]

I have tested the Timeshifted audio file on a different computer with Linux. I can also hear there that the sound is wrong. And its not a problem of pitch. I have tried all three time shift options with the same result. For me its easy to hear. I can hear that the channels are not sync which results in an echo and some other effect. Maybe you should use a headphone. Also the ABX comparison of foobar2000 may help you to hear the difference.

[tebasuna51]

I have tested the Timeshifted audio file on a different computer with Linux. I can also hear there that the sound is wrong. And its not a problem of pitch. I have tried all three time shift options with the same result. For me its easy to hear. I can hear that the channels are not sync which results in an echo and some other effect. Maybe you should use a headphone. Also the ABX comparison of foobar2000 may help you to hear the difference.

Is well know than a timestretch transformation is not perfect (the Rate option with only samplerate change can be the more accurate).

I always say, also, than the audio never is the culprit of different video framerate problems. The video must be changed to mantain the same duration than original and the audio don't need change. This is the correct way and I recommend you use this method.

[Menedas]

To just change the frame rate the video is not a solution for me, because I mostly have 25fps and 23,976fps audio which I both need. So one of them have to be time shifted.

And as I said, I tested all three time shift modes. Also the one option without pitch correction you mentioned makes it wrong, so the explanation seems not accurate to me.

[tebasuna51]

To just change the frame rate the video is not a solution for me, because I mostly have 25fps and 23,976fps audio which I both need. So one of them have to be time shifted.

Try with not free soft: TimeFactory (http://www.prosoniq.com/main/timefactory-2-windows/)

And as I said, I tested all three time shift modes. Also the one option without pitch correction you mentioned makes it wrong, so the explanation seems not accurate to me.

?

[Menedas]

The "?" means that you don't understand my writing or my conclusion? Can't you still don't hear it?

[tebasuna51]

The "?" means that you don't understand my writing or my conclusion? Can't you still don't hear it?

Like I say you the Rate transform can't introduce audible artifacts like the Tempo and Pitch can do.

[Menedas]

Of which DSP do you talk now? Not Timestretch?