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

[siella]

You have used another approach for bitrate setting, but I will only integrate one ffdcaenc extension in Behappy.
I can not decide which is better, because I'm not interested in DTS encoding...
I've used ffdcaenc_213-20141209 (http://forum.doom9.org/showthread.php?p=1702067#post1702067) because it has extra options. like support multiple adio input and Reduced Bit Depth for DTS CD. well in the future maybe you will add multiple auido input for behappy with GetChannel of avisynth

Somehow I dislike the idea of writing help text in the config window...
And if the encoder not support the audio input then you see it on the exception thrown
I am a bit forgetful and lazy :)
An additional parameter for each option to specify if it should be shown in summary? Probably not...
{0} in <TitleFormatString> mean it will show all options ?

Another question, forexample nicauido and bassaudio are support ac3 decode. So will you choise only one? also the same question for ffmpeg and libav decode?
well i am thinking of making ffmpeg aac and ac3 encoder ext with more options.

[jones1913]

I am a bit forgetful and lazy
We all are, thats the whole point of GUIs. That we not have to remember all those annoying commandline switches.
I will think of a way supplying informations in config dialog, maybe a button which opens an info window or a tooltip?

{0} in <TitleFormatString> mean it will show all options ?
Yesss

Another question, forexample nicauido and bassaudio are support ac3 decode. So will you choise only one? also the same question for ffmpeg and libav decode?
Of course not, different encoders or decoders which support the same formats will stay in there. But I will not add several extensions for the same encoder in the BeHappy standard download package.
So you provided a ffdcaenc extension and TomKeller provided one, now we have to choose one for BeHappy download archive.

well i am thinking of making ffmpeg aac and ac3 encoder ext with more options.
Looking forward to this. I always wondered how ffmpeg AC3 encoder compares to Aften, or is ffmpeg using the Aften code?

[Richard1485]

I don't use these downmix now to avoid Normalize and low the volume of front channels.

Now I use:
flr = Getchannel(a, 1, 2, 3, 4)
blr = Getchannel(a, 5, 6)
slr = Getchannel(a, 7, 8)
sur = MixAudio(blr, slr, 1.0, 1.0).SoftClipperFromAudX(0.0)
return mergechannels(flr,sur)

Is this the same as eac3to -down6?

[tebasuna51]

So you provided a ffdcaenc extension and TomKeller provided one, now we have to choose one for BeHappy download archive.
With Siella .ext I can't obtain 1509.75 bitrate. For me the TomKeller one is ok.

Looking forward to this. I always wondered how ffmpeg AC3 encoder compares to Aften, or is ffmpeg using the Aften code?
ffmpeg AC3 encoder and Aften have the same developer jruggle, maybe there are some improvements in ffmpeg: http://forum.doom9.org/showthread.php?p=1522077#post1522077

BTW ffmpeg have some other problems:
- can't accept additional parameters like Aften
- poor STDIN support (without -readtoeof 1, wrong remap ignoring wav channelmask, ...)

I only can recommend ffmpeg encoder for stereo, multichannel output with BeHappy don't work.

EDIT: new ffmpeg versions need bitrate in b/s the .ext need add a 'k' in:
<Value>-ab {0}k</Value>

[tebasuna51]

Is this the same as eac3to -down6?

Nope, this is more advanced downmix using SoftClipperFromAudX(0.0) to avoid clip in surround channels without touch the front and LFE channels.

eac3to -down6 is the same than http://forum.doom9.org/showthread.ph...61#post1465361
Some peaks adding Side to Back channels can force (normalize) to down front volume.

[siella]

With Siella .ext I can't obtain 1509.75 bitrate. For me the TomKeller one is ok.
ffdcaenc doent support continues bitrate e.g. it cant encode 443.25 kpbs except 755.50 and 1509.75
How is this?
http://i.imgbox.com/Z5MsZD2M.png
@jones1913
tooltip will be fine.
If you think aften better i just will write aac enc ext. for ffmpeg.

I have just tried convert 16 bit ac3 to dts but when i used bass and nic source read audio as 32 bit float but when i used directshowsouce read correct value?? (BeHappy_testing_2015-04-04)

[tebasuna51]

...I have just tried convert 16 bit ac3 ...
Don't exist 16 bit AC3, AC3 don't have bitdepth, have bitrate.

[jones1913]

I have just tried convert 16 bit ac3 to dts but when i used bass and nic source read audio as 32 bit float but when i used directshowsouce read correct value?? (BeHappy_testing_2015-04-04)
Isn't that related to this question you have already asked in the past:
I looked ac3 file with mediainfo and it gave 16 bit but when i try to decode with bassadio, ffaudio or nicac3 behappy show 32 bit float.

And LigH has given an accurate answer:
Again this confusing topic...

Most lossy audio compression technologies with psycho-acoustic spectrum masking (like MP2, MP3, AC3, AAC...) work in a floating-point mode, they will transform the samples into audio spectrums of brief blocks and store parameters of a Fourier transformation. Decoding them back into samples again will return floating point samples at first. They may be converted to integer samples later, or not, depending on the decoder.

I don't know which resolution of 16 bit MediaInfo reports. That may be the resolution of the input samples which the encoder used. But that doesn't matter. As soon as there is an AC3 bitstream, there are no more 16 bit integer samples. Only a sequence of encoded frequency spectrums in AC3 blocks.

Probably the DirectShow decoder on your system is doing a conversion to 16 bits, but we can't know which decoder is used on your system.

[siella]

Sorry I sad before I am a bit forgetful :)
There is lav filter on my system.
But really strange, why do ac3 header has bit depth information?
Also http://stnsoft.com/DVD/ac3hdr.html ?
that means when ac3 to dts converting all time need firts converting bit depth 16,24 or 32? Because dts has bit depth

I converted dts to ac3 and eac3 for testing with ffmpeg
Audio
Format : AC-3
Format/Info : Audio Coding 3
Mode extension : CM (complete main)
Format settings, Endianness : Big
Duration : 1h 52mn
Bit rate mode : Constant
Bit rate : 640 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Compression mode : Lossy
Stream size : 513 MiB (100%)

Audio
Format : E-AC-3
Format/Info : Audio Coding 3
Format settings, Endianness : Big
Duration : 1h 52mn
Bit rate mode : Constant
Bit rate : 768 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Compression mode : Lossy
Stream size : 615 MiB (100%)
eac3 hasnt bit depht but ac3 has

[tebasuna51]

But really strange, why do ac3 header has bit depth information?
Also http://stnsoft.com/DVD/ac3hdr.html ?

Like you can see don't exist bit depth info in AC3 header.

that means when ac3 to dts converting all time need firts converting bit depth 16,24 or 32? Because dts has bit depth
Only DTS-MA have bitdepth, and convert AC3 to DTS have no sense at all.

To standard DTS you can apply the LigH comment also.

The DTS header have a field to remember the original bitdepth of the source, but you can't trust this info.
Some encoders (Surcode, ffdcaenc,...) put always 24, and is ok because is always recommended decode to 24 bit int at least, no mather the original bitdepth of the source.

EDIT: Please don't trust in all than MediaInfo say. The ac3 'Bit depth' info is wrong.
https://sourceforge.net/p/mediainfo/bugs/908/

[siella]

@tebasuna51 thanks for information.

http://i.imgbox.com/nHDzUsEv.jpg

http://i.imgbox.com/Aq2LPwyH.jpg

http://i.imgbox.com/oUXDnkmH.jpg

Could anybody check this

[jones1913]

Could anybody check this
I had just a quick look at your extensions:

FFmpeg E-AC3: Looks OK.

FFmpeg MP2: We have also the standalone twolame encoder but thats OK.

FFmpeg AAC: I dislike the way how it is configured. I suggest to use only native and faac with ffmpeg, and write an extension for standalone fdkaac encoder. (https://github.com/nu774/fdkaac)

[LigH]

standalone twolame encoder

RareWares: MP3 and MP2 - Others (http://www.rarewares.org/mp3-others.php)

[jones1913]

RareWares: MP3 and MP2 - Others
Yeah I mean we have already an extension for twolame encoder there. And I would prefer using the standalone encoders instead of ffmpeg if possible.

[tebasuna51]

RareWares: MP3 and MP2 - Others (http://www.rarewares.org/mp3-others.php)

Like I say in encoders\Readme.txt the TwoLame 0.3.12b in RareWares don't work with BeHappy (a regression, don't accept wav header by STDIN, only RAW).

TwoLAME 0.3.10b supplied with BeHappy work without problems.

Could anybody check this
I added ffmpeg AC3 (a copy of E-AC3) encoder with only the valid bitrates for AC3.
Seems work fine.

Like ffmpeg don't have a parameter like -readtoeof or -ignorelength, when the audio info sended is greater than 4 GB, typical for a mutichannel movie track, don't forget activate in [2] Tweak the header W64 (1).

[siella]

I'd added AC3 but before you sad aften is fine so i didint add. Also eac3 and ac3 has some diffirent commands. Like center and sorround mix level But in my opinion Aften is good and enough for ac3.
FFmpeg has a lot of audio encoder in inside. But i think we must add only native encoders. But actually some encoders improved by ffmpeg. So i am not sure.
Also I think jones1913 must moderate which will be add or not.
I made new extantion for helix mp3 encoder. It is good encoder for mp3.
http://i.imgbox.com/GChcSkFs.png

Also i changed mp2 encoder And removed aac library for ffmpeg
http://i.imgbox.com/w9Pf4ZuX.png

[jones1913]

FFmpeg has a lot of audio encoder in inside. But i think we must add only native encoders. But actually some encoders improved by ffmpeg. So i am not sure.
Also I think jones1913 must moderate which will be add or not.

If an encoder is available as standalone exe then we should prefer using this, because it is easier to check. When we place a ffmpeg.exe in encoder directory we never now for sure if it has the required libraries included.
Thats especially the case for non-redistributable encoders like fdk-aac.


I made new extantion for helix mp3 encoder. It is good encoder for mp3.

Cool.

Like ffmpeg don't have a parameter like -readtoeof or -ignorelength, when the audio info sended is greater than 4 GB, typical for a mutichannel movie track, don't forget activate in [2] Tweak the header W64 (1).

Fixed :-) see changelog:

- fixed plugin configuration summary if no radiobuttons are present
- added <Info> element for description and other informations about plugin, these infos are shown as messagebox
when the new little "?" button on logo area in config window is clicked (see QAAC plugin for sample)
- added <HeaderType> element to force specific audio data header written to encoder
(with same values as "Header" in [2]Tweak section on MainWindow)
- encoders which need several files can now arranged in subfolders

So extension system is in good shape now. Next time I spend time on this I make a code cleanup and upload to svn.

EDIT: removed invalid link


Also worth mentioning in this package:


- added an audio-only ffmpeg build with the encoders used by BeHappy, did some testing and noticed no problems so far. Only redistributable encoders are included.
Had problems with linking libtwolame so this encoder wont work for now...



- added lsmashsource.dll build with dcadec support included.
Had several linker warnings but seems to work fine, however I have not tested all dts variants, only a .dtshd file.
If there are problems with this build, the old one is still present as lsmashsource.bak in plugin folder.

[siella]

Finally I complated behappy jobs. And now realy i be happy :)
I checked and tested all encoder and extensions.
I made list for changes that i remember

-Added bass dll of ac3 opus,aix alac, adx, ofr and midi and also added bass ext.
-Changed readme.txt in plugin32 folder
-Added Strange search size,Bandwith and dc filter,lfe options to aften ext.
-Created help folder and i put all encoder help files as txt(maybe needed for extra command)
-Created aacPlus folder and move aacplus in to there also i writed needing dll.
-Added Cuetools Flaccl encoder and ext.
-Addded bitrate info for samplerate to lame ext
-Changed tta script for 6 ch 24 bit audio that is getting error because of more than 4gb
-Added dca ext. and encoder
- Added Float Bitrate option to dca and ffdca
-Added LWLibavAudio support thd, dtshd
-Added FFAudio source ext.
-Changed Ct-AAC ext.(Added tooltips and separated choosing ps)
-Revized FFmpeg ext.
-Added info to fhgmp3 ext.
-Replaced Radiobutton with Dropdown in flac ext.
-Added info in to Nero ext.
-Corrected nero ext.'s bitrate config
-Added FDK AAC Encoder and ext.
-Added extra config to twolame ext.
-Added lossy options for flac and wavpack
-Added alac encoder to qaac ext
-Added monkey ext. and encoder
-Added Tak ext. and encoder
-Added wma ext. and encoder
-Edit wavpack ext.
-Updated readme files
-Created multichs folder in other folder and putted input avs files for mono wavs. Now You can
put yours multiple wavs and into there, edit wavs name as channels name and than easly convert with using behappy avs input.

I release this with all encoders except nero, fhg mp3 encoders and fhgaac, aacplus,qaac dlls.

All extantions screen shoots
http://www.imagebam.com/gallery/l36oc79rxsrfravea5rx4d3xzken0yfl/
@jones1913
Can you add title name for exts also maybe make tab for group info table like
this photo
http://thumbnails108.imagebam.com/40846/90cd48408454716.jpg (http://www.imagebam.com/image/90cd48408454716)
Maybe next days behappy support multi language.
BTW i dont know why but in my home pc when i was reopening everytime behappy windows size was growing . When i delete BeHappy.State file turning normal. ( I am using windows 7 ) I tried to another 2 pc that installed win 8 and win 2008 and there is no problem.

Bug: wav write and pcm raw write doesnt work

http://www.mediafire.com/download/1k3s0zuw3k3hi0b/BeHappy_testing_2015-04-26+-+%28new+extansions+09.05.2015%29.zip

[jones1913]

I see you have done a lot work ,thanks! Some extensions have small cosmetic problems or typos but thats easy to fix.

I release this with all encoders except nero, fhg mp3 encoders and fhgaac, aacplus,qaac dlls.
Not sure but I think fdkaac is also not redistributable.


Can you add title name for exts also maybe make tab for group info table like this photo
No idea why you have no title in config window, here it looks as expected (win7, classic and aero).
14770


Not sure what you mean with "tab for group info table" but I wont spend more time in changing the gui.


BTW i dont know why but in my home pc when i was reopening everytime behappy windows size was growing . When i delete BeHappy.State file turning normal. ( I am using windows 7 ) I tried to another 2 pc that installed win 8 and win 2008 and there is no problem.
I have never seen this behavior on my system (also win7). Delete .state file can be recommened if you encounter problems after making big changes to extensions.


Bug: wav write and pcm raw write doesnt work
Ah I see, will check this soon.

Edit:
Another thing is that I personally dont like the preceding "[Lossy]" or "[Lossless]" at your extensions. It makes the encoder list a bit hard to read.
I have done already some work in the background to better control the growing list of encoders with a simple filter system, to show only encoders wich support eg. ac3 or aac or ...
But it is not ready to release yet. When this is done I'll take a break from working on BeHappy because the weather is becoming better here and I wont spend so much time on pc for the time being.

[jones1913]

Just tried to upload my sources to svn but it wont work anymore (subversion on linux terminal).
Even a simple checkout "svn co https://behappy.svn.codeplex.com/svn" fails with "Ungültiger Maschinenbefehl (Speicherabzug geschrieben)". Really great! :mad::mad::mad:

@siella
2 small problems with your package:
- If we include wrappers like qaac.exe in BeHappy package then the encoder is shown as "working" in gui encoder list because BeHappy checks only for the .exe , but it actually don't work because the needed .dlls are missing.
Best would maybe to not include this files (or rename to _qaac.exe ?).
- Before a job started a message box warned me about missing "OptimFrog.dll", altough it is present in plugins folder of your package. For what is this file needed?

[tebasuna51]

Just tried to upload my sources to svn but it wont work anymore (subversion on linux terminal).
Even a simple checkout "svn co https://behappy.svn.codeplex.com/svn" fails with "Ungültiger Maschinenbefehl (Speicherabzug geschrieben)". Really great! :mad::mad::mad:

Like you are the unique active BeHappy developer feel free to abandon Codeplex like source repository and select other at your convenience.

- Before a job started a message box warned me about missing "OptimFrog.dll", altough it is present in plugins folder of your package. For what is this file needed?

For the bass_ofr.dll decoder http://forum.doom9.org/showthread.php?p=1557337#post1557337

[siella]

- If we include wrappers like qaac.exe in BeHappy package then the encoder is shown as "working" in gui encoder list because BeHappy checks only for the .exe , but it actually don't work because the needed .dlls are missing.
Best would maybe to not include this files (or rename to _qaac.exe ?).
If itunes is installed it will work, but you can rename or remove.I think removing is best.

Not sure what you mean with "tab for group info table" but I wont spend more time in changing the gui.
Okey it is not necessary.

I have done already some work in the background to better control the growing list of encoders with a simple filter system, to show only encoders wich support eg. ac3 or aac or ...
it sounds good. I look forward to your last relase.

@tebasuna51
Before you made Dynamic Range copression (http://forum.doom9.org/showthread.php?p=779165#post779165)ext. for behhapy I've tried to create drc profile (https://github.com/nu774/qaac/wiki/Dynamic-range-compression) for qaac based on your extantion, but it is really mixed so i dont understand well,
Can you explain which value is what for
threshold, Compresion, knee, attack time, release time

[tebasuna51]

...I've tried to create drc profile (https://github.com/nu774/qaac/wiki/Dynamic-range-compression) for qaac based on your extantion, but it is really mixed so i dont understand well,
Can you explain which value is what for threshold, Compresion, knee, attack time, release time
In DynRanComp.extension I try to emulate the DD compression curves
http://web.archive.org/web/20051111165720/http://pages.sbcglobal.net/wilsondr/ddcompprof.gif
using the compand Sox effect. Syntax:

compand attack1,decay1{,attack2,decay2} [soft-knee-dB:]in-dB1[,out-dB1]{,in-dB2,out-dB2} [gain [initial-volume-dB [delay]]]

Then the Music Standard curve (yelow in the image):
# Music Standard Segments: (Noise) +12dB 2:1 = 20:1
# Points: ------- ------- ------- ------- ------- -----
SoxFilter("compand 0.1,0.3 -90,-90,-70,-58,-55,-43,-31,-31,-21,-21,0,-20 0 0 0.1")
have:
Attack: 100 ms
Decay: 300 ms
Hard-knee (don't exist soft-knee dB: before pairs in-dB,out-dB)
in-dB1,out-db1: -90,-90 (without change)
in-dB2,out-db2: -70,-58 (begin segment amplify 12 dB)
in-dB3,out-db3: -55,-43 (end segment amplify 12 dB)
in-dB4,out-db4: -31,-31 (begin segment without amplifly)
in-dB5,out-db5: -21,-21 (end segment without amplifly)
in-dB6,out-db6: -0,-20 (rest with limited output at -20 dB)
Gain: 0 dB
Initial volume: 0 dB
Delay: 100 ms

Like you can see the Sox compand accept many pairs of Threshold, Compresion (Ratio) values.

BTW you dont need emulate DD curves.

[siella]

Many thanks tebasuna51

If i understand correctly
-70,-58 mean 1.20:1 compression ?
so can i use sox Music value for qaac(only for downward compression) like
threshold:-31dBFS
20:1 compresion
knee width 0dB for hard knee
attack time 100ms
release time 300ms

i looked soxfilter v1.1 and i noticed that port from 12.17.9 version. I hope someone update it to 14.4.2 v.
Sox support stdin and maybe i can make sox extantion like encoder.

[tebasuna51]

If i understand correctly
-70,-58 mean 1.20:1 compression ?
Nope, in this segment there are a gain of 12 dB then the ratio is 1:12

so can i use sox Music value for qaac(only for downward compression) like
threshold:-31dBFS
20:1 compresion
knee width 0dB for hard knee
attack time 100ms
release time 300ms
With only 1 segment you can emulate the last part of Music Ligth with:
threshold:-20dBFS
10:1 compresion

Range input [-20 to 0 dB] go to range output [-20 to -10 dB]

Plain last segments (Music Std, Film Light, Film Std) can't be emulated with this method.

[siella]

Thanks again tebasuna51
I think it is not necessary. I will make sox extension.

[tebasuna51]

@jones1913
Please read http://forum.doom9.org/showthread.php?p=1722472#post1722472

To upgrade to SoundTouch library to 1.8.0 (multichannel timestretch better support) we need load the new TimeStretch.dll plugin and change TimeStretch() to TimeStretchPlugin().
Thanks.

[jones1913]

Like you are the unique active BeHappy developer feel free to abandon Codeplex like source repository and select other at your convenience.
OK, but I wanted to avoid that. Anyway I found a working svn client and just uploaded my sources with 2 small changes:

- fixed: wav and raw writer are now working again
- fixed: glitch if no radiobuttons are present in config window

A build is not needed yet for these small changes.

To upgrade to SoundTouch library to 1.8.0 (multichannel timestretch better support) we need load the new TimeStretch.dll plugin and change TimeStretch() to TimeStretchPlugin().
Thanks.
Ok.

[jones1913]

@siella

I have uploaded your extensions to svn rep (https://behappy.codeplex.com/SourceControl/changeset/76678) with slight changes:

- modified some encoder logos (reduced size, added transparency)
- fixed some typos and added empty lines in tooltips for better readability
- ffaudiosource: removed info text about haali splitter because newer versions doesn't rely on haali anymore

Now we have 6 AAC encoders, 4 MP3 encoders, 2 AC3 encoders and a varied mix of exotic encoders in the list. :cool::cool::cool:

[siella]

Thanks jones1913
I look forward your last build release

[jones1913]

Pre-release: EDIT: dead link removed

- added file type filters to show only certain encoder/source plugins (use right click on source/encoder combobox)
- added <IsLossless> field to extensions
- cleaned up about-tab a bit

14807

@siella
There is a problem with Helix MP3 encoder: When the process reaches 99%, then the encoder throw an error and delete the encoded file.
I have tried with various encoder settings. Is it working for you?
Log:
Starting job test.flac -> test.mp3
Found Audio Stream
Channels=2, BitsPerSample=16 int, SampleRate=48000Hz
encoder\hmp3.exe - "C:\Dokumente und Einstellungen\jones\Eigene Dateien\media\test.mp3" -V80 -hf2 -C0 -O1 -U2 -X2 -SBT450 -TX0
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Finalizing encoder
Error: System.ApplicationException: Abnormal encoder termination 13237
bei BeHappy.Encoder.encode()
#### Encoder StdErr ####

file-file MPEG Layer III audio encode v5.1 2005.08.09
Copyright 1995-2005 RealNetworks, Inc.

Usage: mp3enc <input> <output> [options]
<input> and/or <output> can be "-", which means stdin/stdout.

Example:
mp3enc input.wav output.mp3

Options:
-Nnsbstereo -Sfilter_select -Aalgor_select
-C -X -O
-D -Qquick -Ffreq_limit -Ucpu_select -TXtest1
-SBTshort_block_threshold -EC
-h (detailed help)

<press any key to stop encoder>
PCM input file: -
MPEG ouput file: C:\Dokumente und Einstellungen\jones\Eigene Dateien\media\test.mp3
pcm file: channels = 2 bits = 16, rate = 48000 type = 0
Layer III mode 1 STEREO 48000Hz VBR-80 hf2

-------------------------------------------------------------------------------
Frames | Bytes In / Bytes Out | Progress | Current/Average Bitrate
13237 | 61032960 / 6080904 | 371517% | 75.08 / 153.14 Kbps
-------------------------------------------------------------------------------
Compress Ratio 37015.485756%





@tebasuna
I noticed the wavsplit.exe build is older (1.0.0.3) than the source code in BeHappy/others folder (1.0.1.1) ??

[tebasuna51]

I noticed the wavsplit.exe build is older (1.0.0.3) than the source code in BeHappy/others folder (1.0.1.1) ??
Thanks for your pre-release.

Yep, seems I forget modify the .exe, I experience some problems with stdin in command line usage (never with BeHappy) and I make a little change.
Changelog:
2008-11-06 v1.0.1.1
Cosmetic changes and continue until input buffer is full.

Here is the 1.0.1.1 wavsplit.exe

[EDIT] wavsplit.exe included in wav2Util.7z here (https://forum.doom9.org/showthread.php?p=1624209#post162420)

[jones1913]

I've just uploaded the previous pre-release as new stable release to codeplex. (https://behappy.codeplex.com/releases/view/615867)

changes since last stable release:

2015-06-21 (jones1913) 0.2.8.19896
* newer wavsplit.exe

2015-05-31 (jones1913) r76729
+ added file type filters to show only certain encoder/source plugins (use right click on source/encoder combobox)
+ added <IsLossless> field to extensions
* cleaned up about-tab a bit

2015-05-24 (jones1913) r76678
+ added siella's extensions

2015-05-17 (jones1913) r76645
* time stretch dsp now use the TimeStretchPlugin() function

2015-05-17 (jones1913) r76644
* enhanced extension system and replaced all extensions (commented samples in "BeHappy/extensions/extension_specs/*")
+ added calculator to internal timestretch dsp plugin

2014-12-27 (jones1913) r76062
* start next job only if previous running job has properly initialized

2014-12-03 (jones1913) r75896
+ encoder plugin item in combobox is grayed out if the needed encoder executable is missing
+ added a button to automatically set the recommened maximum of simultanously running jobs

[tebasuna51]

Thanks jones1913.

[Yamcha]

Finally got the program working after a few tries but can't encode. I followed all instructions from the readme files. Using a Dell Dimension 2350 with 1 meg of memory with Windows XP Pro 32 bit.

Log: Starting job 01b.wav -> 01b.mp3
Error: System.Runtime.InteropServices.SEHException (0x80004005): External component has thrown an exception.
at BeHappy.AviSynthClip.dimzon_avs_init_2(IntPtr& avs, String func, String arg, AVSDLLVideoInfo& vi, AviSynthColorspace& originalColorspace, AudioSampleType& originalSampleType, String cs)
at BeHappy.AviSynthClip..ctor(String func, String arg, AviSynthColorspace forceColorspace, AviSynthScriptEnvironment env)
at BeHappy.Encoder.encode()

[tebasuna51]

Please post the log.

[Yamcha]

Please post the log.

Look up.

[jones1913]

@Yamcha

We need some more informations:

- AviSynth installed? If yes which version?
- selected source filter?
- selected encoder?

[Yamcha]

@Yamcha

We need some more informations:

- AviSynth installed? If yes which version?
- selected source filter?
- selected encoder?

Not sure what you mean by Source Filter. I didn't run a script, straight from file.
I don't have FFMpeg installed to system btw.

AviSynth 2.5.8.5
Wav to Mp3 (FhG)

EDIT: Instead image:
[1] Source
C:\Wav\01b.wav
WavSource

[3] DSP (none)

[4] Destination
C:\Wav\01b.mp3
MP3 FHG - CBR mode 128 Kb/s. High Quality

[tebasuna51]

Maybe MP3 FHG is not correctly installed in your system ("External component has thrown an exception")

Try with MP3 Lame.

[GMJCZP]

Try run for first time the archive mp3sencoder.exe and follow the onscreen instructions.

[Yamcha]

Maybe MP3 FHG is not correctly installed in your system ("External component has thrown an exception")

Try with MP3 Lame.

I don't have mp3sEncoder 'Installed', I assumed it was a stand alone command line encoder. I have the FhG ACM hack called Radium 1.2 installed. That never gave me a problem as long as I use EAC to encode/decode with it.
I use LFE Lame Frontend v1.8 for Lame mp3s.

This is the error reported through EAC: C:\MP3sEncoder\mp3sEncoder.exe -if 2.wav -of 2.mp3 -br 128

Error: Could not open 'wav' file '2.wav'.

[LigH]

So please give us a detailed analysis of this "2.wav", e.g. using MediaInfo. Is it at all a WAV file? Does it contain compressed audio?

[jones1913]

Not sure what you mean by Source Filter. I didn't run a script, straight from file.
For every file BeHappy generates an AviSynth script in the background an opens the file using the selected (AviSynth)source filter.


I don't have FFMpeg installed to system btw.
I don't have mp3sEncoder 'Installed', I assumed it was a stand alone command line encoder.
To clear this a bit up: BeHappy doesnt rely on any encoders 'installed' on system. It only uses encoders which stay in the "BeHappy/encoders/*" directory.
Some encoders are included in the download package (lame, ffmpeg). Others cant be included due to their licensing conditions (fhg mp3sencoder, most aac encoders), these must be copied manually by the user to encoders directory. And all of them are command line encoders.

The next step is: So please give us a detailed analysis of this "2.wav", e.g. using MediaInfo. Is it at all a WAV file?

[Yamcha]

General
Complete name: 02.wav
Format: Wave
File size: 43.6 MiB
Duration: 4mn 19s
Overall bit rate mode: Constant
Overall bit rate: 1 411 Kbps

Audio
Format: PCM
Format settings, Endianness: Little
Format settings, Sign: Signed
Codec ID: 1
Duration: 4mn 19s
Bit rate mode: Constant
Bit rate: 1 411.2 Kbps
Channel(s): 2 channels
Sampling rate: 44.1 KHz
Bit depth: 16 bits
Stream size: 43.6 MiB (100%)
================================================
########################################
#Created by BeHappy v0.2.8.19896
#Creation timestamp: 6/30/2015 8:44:13 PM
########################################
#Source FileName:C:\Wav\02.wav
#Target FileName:C:\Wav\02.mp3
########################################

########################################
# [Source: WavSource]
########################################
WavSource("C:\Wav\02.wav")

########################################
# [Encoder: MP3 Helix - CBR Mode per chnl 112 kb/s; HF Encode:HF2]
########################################
16==Audiobits(last)?last:ConvertAudioTo16bit(last)

[jones1913]

Your wave file looks regular.

Helix mp3 encoder plugin is probably broken, I have mentioned it (http://forum.doom9.org/showpost.php?p=1724791&postcount=1181), but siella (author of the plugin) has not yet responded.

Have you tried another encoder? Lame plugin should work in any case.

[Yamcha]

I get the same error message regardless of encoder. I tried every one in the encoder folder.

[yonta]

extensions\e_FFmpeg.ext seems to have a typo at line 339. I think -c:a aac -strict -2 should be removed from the line.
And the included ffmpeg.exe doesn't seem to support .m4a output.

[jones1913]

extensions\e_FFmpeg.ext seems to have a typo at line 339. I think -c:a aac -strict -2 should be removed from the line.

You're right, Thanks.

And the included ffmpeg.exe doesn't seem to support .m4a output.

Can this be configured somehow? However I have fixed this for now by appending "-f mp4" to the command line.
https://behappy.codeplex.com/SourceControl/changeset/77012
Seems its not possible to download single files from codeplex, the download button always downloads the complete source code package.

I get the same error message regardless of encoder. I tried every one in the encoder folder.

Is AviSynth in general working for you?

[Yamcha]

Is AviSynth in general working for you?

I wouldn't know, don't know anything about avisynth. I used EAC and CEP2 for audio encoding before.