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

[Dayvon]

@ Dimzon

Sorry for dropping off the face of the earth man. I had every intention of helping out in some way, and then LIFE happened :o . Needless to say, BeHappy is rocking! You've been doing awesome work. Me and everyone else using it gives you thanks!

[fight2win]

for ac3 to aac conversion, how to increase volume of output file in behappy?

[tebasuna51]

for ac3 to aac conversion, how to increase volume of output file in behappy?
Use NicAc3Source with DRC and after Normalize at desired level.

[fight2win]

my behappy only says source as nicac3, not as nicac3drc, where to get it?

[dimzon]

my behappy only says source as nicac3, not as nicac3drc, where to get it?
choose nicac3 source AND click on [...] button against source selection ;)

[fight2win]

i have 3 little queries about ac3/dts to aac encoding:

1.if my source is 2-ch 192kbps ac3, which out of drc source,normalize to 100 percent and dpl2 downmix options should i use?

2.in what order should normalize and dpl2 downmix options should come, i mean, which should come first and which after that?

3. for muxing in mp4 container, should i use .aac conatainer output encoding or .m4a container output encoding?

thanks in advance, by the way, behappy rocks!

[shon3i]

1. for stereo encoding's don't use dominix
2. first dominix and then normalize (but i think is the same when is inverted)
3. mp4 is better choice, because sometimes must to singnal HE-AAC when muxing, and muxers can't detect propertly HE-AAC files.

[fight2win]

1. for stereo encoding's don't use dominix
2. first dominix and then normalize (but i think is the same when is inverted)
3. mp4 is better choice, because sometimes must to singnal HE-AAC when muxing, and muxers can't detect propertly HE-AAC files.

thanks!:)

[fight2win]

even when encoding ac3 6 channel or dts to 2-channel aac, should i use downmix to dpl2 ?

[tebasuna51]

2. first dominix and then normalize (but i think is the same when is inverted)
Is not the same. First downmix and after normalize.

When normalize over 6 channels the max peak of all channels is used to calculate the common gain applied to all channels.

After downmix, (to avoid clipping problems, the coeficients in downmix matrix must sum only 1.0) the resultant 2 channels never are normalized.

Only if all 6 input channels have a equal max peak at same time is the same.
even when encoding ac3 6 channel or dts to 2-channel aac, should i use downmix to dpl2 ?
Yes, why not?

[vlada]

Hello,

I have one very basic question: Why the NicAudio.dll isn't loaded? The script created by BeHappy doesen't work. It reports there is no command called NicAC3Source. I think there sould be something like LoadPlugin("NicAudio.dll"). Of course I have this file in my AviSynth plugin's directory. Why isn't BeHappy working for me?

Sorry for this very basic question, but I couldn't find the answer anywhere.

[dimzon]

Of course I have this file in my AviSynth plugin's directory.
Sometimes some software can modify HKEY_LOCAL_MACHINE\SOFTWARE\AviSynth registry key, check it to know wich ACTUAL dir is AviSynth plugin dir

[MacAddict]

Is not the same. First downmix and after normalize.
Maybe this should be default in the GUI then?

[tebasuna51]

Yes, for habitual jobs Normalize() is the last DSP to be used.
I don't remember if is the default but, if you use Move Down, the next time you use BeHappy your order is preserved.

[shon3i]

Again i haved some free time to make new fresh package (http://forum.doom9.org/showpost.php?p=829754&postcount=259) with last versions

[vlada]

dimzon> So if I have a plugin in AviSynth plugin's folder, there should be no need to load the plugin in script? I didn't know that.

Anyway I installed the package from shon3i and BeHappy started working. I have no idea what was wrong.

[shon3i]

Maybe you don't haved nicaudio.dll in avisynth plugins folder, or avisynth use other path for plugins(stored in registry, look dimzon's post above) and my setup use this path and find right plugins folder on system

[shon3i]

Fresh package (http://forum.doom9.org/showpost.php?p=829754&postcount=259) is out.

Now support BassAudio which means more input file types supported directly (wav/ogg/aiff/mp3/mp2/wma/flac/wv/ape/mpc/spx/aac/m4a/mp4/ac3)

but have some minor bugs and restrictions like

- aac decode don't work for 5.1 multichannel. Better to use DirectShowSource
- stereo ac3 (2.0) don't work. NicAC3 is better

And i found that ape (Monkey's Audio) don't work propertly

[Yanaran]

Just wanted to say thanks for this great tool, finally I get full volume ac3 for my DVDs without jumping through hoops. :)

[dimzon]

http://gotdotnet.com/
Now BeHappy is Featured Workspace

Hello,

Your project rocks! We're going to feature it on the GotDotNet homepage. Congrats and keep up the great work.

Julie
Microsoft GotDotNet Team

[shon3i]

Congratulation dimzon, realy great work.

[thuongshoo]

Req1 : If I choose a source file, "destination file" textbox will be filled automatically. This thing only is done if "destination file" textbox is blank for currentl version

Req2: I want to choose multiple job when I need to delete jobs . After I delete a jobs, current item will be set to the fisrt jbo .

Thank you very much !

[Dark-Cracker]

hum just a little question about the difference beetween amplify and normalyze, i know the amplify factor (to amplify it at 100%) for my wav file, if i amplify it with this factor does it will be the same than normalize it at 100% ? i don't clearly see the difference beetween those 2 audio filter.

and i also want to know if there is some generical range or presets for the function normalize and amplify/amplifydb ?

perhaps some audio guru could enlight me on those silly questions :)

Bye.

Ps : very nice tool dimzon, and thanks for the people who give him life.

++

[Wilbert]

hum just a little question about the difference beetween amplify and normalyze, i know the amplify factor (to amplify it at 100%) for my wav file, if i amplify it with this factor does it will be the same than normalize it at 100% ?
Yes, that's correct.

I think the difference is as follows:

Normalize(volume=1.0) amplifies maximal without clipping (by calculating the maximum peak). Normalize(volume=0.5) amplifies to 50% of this maximal value. So, even in this case the audiolevels can be higher as before applying the filter.

Amplify(amount1=1.0) doesn't do anything. Amplify(amount1=2.0) doubles the audio levels, regardless whether there will be any clipping.

I'm sure someone will correct me if i'm wrong :)

[Dark-Cracker]

thank u for the explanation wilbert :) it's more clear now.

another question, does behappy will implement the filter AudioLimier (http://forum.doom9.org/showthread.php?t=108470) or it doesn't produce enought good results to be used ?

Ps : dimzon if this could help you to improve your nice tool here is a little trick (AVSAmp) to found the amplify full scale factor : http://forum.doom9.org/showthread.php?t=92120
not really sure if this could be usefull :) .

Bye.

[shon3i]

I think that this avsamp works like replaygain from foobar but i am not sure, but will be very usefull

[fight2win]

after nicac3source (drc), and then downmixing to dpl2, is it advisable to use normalize now, if yes, to what limit, i mean, 100% or more or less than that?also, when is audiolimiter gonna be implemented in behappy?

[tebasuna51]

after nicac3source (drc), and then downmixing to dpl2, is it advisable to use normalize now, if yes, to what limit, i mean, 100% or more or less than that?
After DRC (output may be at -11 dB) and DPL2 (BeHappy use a normalized matrix) is recommended to use Normalize with a 100% limit.

[Krizzz989]

My AC3 encodes are filled with noise, thanks in advance for any help.

BeHappy_20060532_v4
NicAudio_25_dll_20060314
ffmpeg CVS 2006-05-19
AviSynth 2.5.6a

BeHappy logStarting job 10.avi->10.ac3
Found Audio Stream
Channels=2, BitsPerSample=16, SampleRate=48000Hz
ffmpeg.exe -i - -y -acodec ac3 -ab 192 "F:\video\chobits\03\10.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 24 2006 08:10:52, gcc: 4.1.0 [Sherpya]
Input #0, wav, from 'pipe:':
Duration: N/A, bitrate: 1536 kb/s
Stream #0.0: Audio: pcm_s16le, 48000 Hz, stereo, 1536 kb/s
Output #0, ac3, to 'F:\video\chobits\03\10.ac3':
Stream #0.0: Audio: ac3, 48000 Hz, stereo, 192 kb/s
Stream mapping:
Stream #0.0 -> #0.0

video:0kB audio:34496kB global headers:0kB muxing overhead 0.000000%

Let me know if any further information is needed.

edit: Here's a sample http://www.zshare.net/download/sample-ac3.html

[tebasuna51]

My AC3 encodes are filled with noise, thanks in advance for any help.
The log seems ok.
For further analysis we need a sample of source audio and the .avs used ("Export AviSynth Script").

[Krizzz989]

Here's a sample, demuxed from source avi using AVI-Mux GUI 1.17.6. http://www.zshare.net/audio/sample-mp3-hqw.html

avs script:
########################################
#Created by BeHappy v0.1.8.35345
#Creation timestamp: 7/11/2006 4:25:07 AM
########################################
#Source FileName:F:\video\chobits\03\10.avi
#Target FileName:F:\video\chobits\03\10.ac3
########################################

########################################
# [Source: DirectShowSource]
########################################
DirectShowSource("F:\video\chobits\03\10.avi")

EnsureVBRMP3Sync() # Some black magic to avoid desync

########################################
# [Encoder: ffmpeg AC3 @ 192 kbps]
########################################
6==Audiochannels(last)?GetChannel(last,1,3,2,5,6,4):last

[tebasuna51]

There are a new and interesting ac3 encoder Aften (http://forum.doom9.org/showthread.php?p=850635#post850635) based in ffmpeg libraries
Aften 0.01 : A/52 audio encoder
(c) 2006 Justin Ruggles

usage: aften [options] <input.wav> <output.ac3>
options:
[-h] Print out list of commandline options
[-v #] Verbosity (controls output to console)
0 = quiet mode
1 = show average stats (default)
2 = show each frame's stats
[-b #] CBR bitrate (default: none)
[-q #] VBR quality [1 - 1023] (default: 200)
[-w #] Bandwidth [0 - 60] (default: adaptive)
[-m #] Stereo rematrixing
0 = independent L+R channels
1 = mid/side rematrixing (default)
[-s #] Block switching
0 = use only 512-point MDCT (default)
1 = selectively use 256-point MDCT
[-c #] Center Channel Mix level
0 = -3 dB (default)
1 = -4.5 dB
2 = -6 dB
[-u #] Surround Channel Mix level
0 = -3 dB (default)
1 = -6 dB
2 = 0 dB
[-d #] Dolby Pro Logic Mode
0 = Ignored (default)
1 = Disabled
2 = Enabled
[-n #] Dialog Normalization level [0 - 31] (default: 31)
You can try in BeHappy with this aften.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">
<AudioEncoder UniqueID="a63940a0-0da0-11db-9cd8-0800200c9a66">
<Plugin>
<MultiOptionEncoder Type="BeHappy.Extensions.MultiOptionEncoder, BeHappy">
<DialogWidth>250</DialogWidth>
<ExecutableFileName>aften.exe</ExecutableFileName>
<TitleFormatString>aften AC3 @ {0}</TitleFormatString>
<SupportedFileExtension>ac3</SupportedFileExtension>
<Option>
<Name>448 kbps</Name>
<Value>-b 448000 - "{0}"</Value>
</Option>
<Option default="true">
<Name>384 kbps</Name>
<Value>-b 384000 - "{0}"</Value>
</Option>
<Option>
<Name>320 kbps</Name>
<Value>-b 320000 - "{0}"</Value>
</Option>
<Option>
<Name>256 kbps</Name>
<Value>-b 256000 - "{0}"</Value>
</Option>
<Option>
<Name>192 kbps</Name>
<Value>-b 192000 - "{0}"</Value>
</Option>
<Option>
<Name>160 kbps</Name>
<Value>-b 160000 - "{0}"</Value>
</Option>
</MultiOptionEncoder>
</Plugin>
</AudioEncoder>
</BeHappy.Extension>
To configure others parameters than bitrate (Dialog Normalization, Dolby Surround mode, ...) we need the help from Dimzon to make the appropriate Configuration Dialog (or you can edit aften.extension before run BeHappy).

[tebasuna51]

avs script:

DirectShowSource("F:\video\chobits\03\10.avi")

You can't use an avi like input for BeHappy.
BeHappy wait for a audio only file.
Use the mp3 extracted with AviMux.

[danpos]

You can't use an avi like input for BeHappy.
BeHappy wait for a audio only file.
Use the mp3 extracted with AviMux.

Nope, it can to do this using DirectShow since that the container have an audiostream. Take a look at the BeHappy's log from a quick conversion test that I did right now:


Starting job sample.avi->sample.ac3
Found Audio Stream
Channels=2, BitsPerSample=16, SampleRate=48000Hz
ffmpeg.exe -i - -y -acodec ac3 -ab 160 "D:\Compilation\SCRIPTS\Didees scripts\sample.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: 1536 kb/s
Stream #0.0: Audio: pcm_s16le, 48000 Hz, stereo, 1536 kb/s
Output #0, ac3, to 'D:\Compilation\SCRIPTS\Didees scripts\sample.ac3':
Stream #0.0: Audio: ac3, 48000 Hz, stereo, 160 kb/s
Stream mapping:
Stream #0.0 -> #0.0

video:0kB audio:6611kB global headers:0kB muxing overhead 0.000000%

JFYI. :)

See ya,

[tebasuna51]

Nope, it can to do this using DirectShow since that the container have an audiostream.
You are right, but the use of DirectShowSource is always unpredictable in BeHappy because different configurations (SPDIF out, Aud-X, ...).

The sample.mp3 provided by Krizzz is encoded without problems using NicMPG123Source like decoder and ffmpeg encoder

[danpos]

You are right, but the use of DirectShowSource is always unpredictable in BeHappy because different configurations (SPDIF out, Aud-X, ...).

I'm agreed with you in this point. :)

See ya,

[Robot]

Is it possible to convert a 5.1 AAC file into a AC3 file that I can use in DVDLab?

I followed the install directions, but when I hit start I get this error:

Starting job theusvsjohnlennon_h720p_track2.aac->theusvsjohnlennon_h720p_track2.ac3
Error: BeHappy.AviSynthException: Required Avisynth 2.5
at BeHappy.AviSynthClip..ctor(String func, String arg, AviSynthColorspace forceColorspace, AviSynthScriptEnvironment env)
at BeHappy.Encoder.encode()

Thanks

[tebasuna51]

@Robot, in your log:
"Error: BeHappy.AviSynthException: Required Avisynth 2.5"

At first post of this thread:
"System requirements
...
* Microsoft .NET Framework Version 2.0
* Avisynth v2.56"

[Robot]

@Robot, in your log:
"Error: BeHappy.AviSynthException: Required Avisynth 2.5"

At first post of this thread:
"System requirements
...
* Microsoft .NET Framework Version 2.0
* Avisynth v2.56"

Yeah, I've downloaded both. :confused:

Do I need t do something else to connect Avisynth to BeHappy?

[tebasuna51]

Do I need t do something else to connect Avisynth to BeHappy?
After install AviSynth v2.56 restart your computer and run BeHappy.

For further analysis we need the .avs used ("Export AviSynth Script").

To decode aac 5.1 you need DirectShow properly configured (maybe ffdshow Audio Decoder with 6 chan 16 bit int output and "Don't use WAVEFORMATEXTENSIBLE header...")

[Robot]

I re-installed Avisynth and restarted. I get the same error. I exported the script.

########################################
#Created by BeHappy v0.1.5.526
#Creation timestamp: 7/18/2006 6:25:24 AM
########################################
#Source FileName:C:\Documents and Settings\J\Desktop\NTS\johnlennon\theusvsjohnlennon_h720p_track2.aac
#Target FileName:C:\Documents and Settings\J\Desktop\NTS\johnlennon\theusvsjohnlennon_h720p_track2.ac3
########################################

########################################
# [Source: AviSynth]
########################################
Import("C:\Documents and Settings\J\Desktop\NTS\johnlennon\theusvsjohnlennon_h720p_track2.aac")

EnsureVBRMP3Sync() # Some black magic to avoid desync

########################################
# [Encoder: ffmpeg AC3 @ 448 kbps]
########################################
6==Audiochannels(last)?GetChannel(last,1,3,2,5,6,4):last

Thanks

[dimzon]

Robot
Wich OS did You use?
Are you able to play obtained script via mplayerc?

[alwa]

[Source: AviSynth]
How is that possible? :confused: It's an aac-file, not an avs-file...
Try to use DierectShowSource or BassAudio, then it should work properly.

[Robot]

Robot
Wich OS did You use?
Are you able to play obtained script via mplayerc?

I use Windows XP

and I have no idea what the next thing means.
I'm a novice. :D

[tebasuna51]

In BeHappy [1] Source you must select DirectShowSource instead AviSynth after the input file.

[dimzon]

In BeHappy [1] Source you must select DirectShowSource instead AviSynth after the input file.
And don't forget about AAC splitter :D

[dimzon]

http://img232.imageshack.us/img232/274/aftenam4.png
Aften support (not tested yet, please report)

[Robot]

I'm lost. No matter what I do, I get this error:

Starting job theusvsjohnlennon_h720p_track2.aac->theusvsjohnlennon_h720p_track2.ac3
Error: BeHappy.AviSynthException: Required Avisynth 2.5
at BeHappy.AviSynthClip..ctor(String func, String arg, AviSynthColorspace forceColorspace, AviSynthScriptEnvironment env)
at BeHappy.Encoder.encode()

:confused:

[nfm]

Can somebody compile this app into one package with all plugins etc. I don't even know where to begin, everything is all over the place. :(

[thuongshoo]

@nfm : I often install K-lite codec pack , and plugin which is need for Behappy is at gotdotnet