BeHappy - AviSynth based audio transcoding tool (UPD 19-07-2006) [Archive] - Page 3

tebasuna51

21st March 2007, 03:03

[EDIT] Here's the ac3 file I'm experimenting with. test file (http://www.mytempdir.com/1262380). btw, I also tried using DirectShow with the sonic audio decoder instead of ac3filter and got the same bloated output. Ugh...

Your test file is correct. And I can't reproduce your problem.

Really is not recommended use the DirectShowSource method because config differences between systems, player/convert options and difficult debug.

In my system I have ffdshow DSF to decode ac3 and work with BeHappy without this problem. Most sure is use the dedicated decoder NicAc3Source.

Sorry but I can't help you with your DirectShow configuration (is a mystery for me).

Chumbo

21st March 2007, 04:30

Your test file is correct. And I can't reproduce your problem.

Really is not recommended use the DirectShowSource method because config differences between systems, player/convert options and difficult debug.

In my system I have ffdshow DSF to decode ac3 and work with BeHappy without this problem. Most sure is use the dedicated decoder NicAc3Source.

Sorry but I can help you with your DirectShow configuration (is a mystery for me).
No worries, I really appreciate you trying to duplicate. I kinda' figured it was a filter issue and, I think, you validated that and am grateful to you for this. :) Many thanks.

shon3i

21st March 2007, 22:02

One more thing to do:

When is [..] for destination pressed, and common dialog show, should filename filed in common dlg, be filed with filename from destination textbox without full path, only filename.

Chumbo

22nd March 2007, 02:09

Let me take a look at that for you shon3i.

[EDIT] here you go shon3i:
2007-03-21 (Chumbo)
+ Destination/output dialog now prefills the file name

Latest BeHappyCAT (http://www.mytempdir.com/1265030) build.

shon3i

22nd March 2007, 08:24

Not working in my type.

http://img208.imageshack.us/img208/8388/untitledgi5.jpg

Maybe you didn't understand me.

Chumbo

22nd March 2007, 16:25

Not working in my type.
...
Maybe you didn't understand me.
Oh no, I understood perfectly. I may have copied the wrong BeHappy.exe in my haste. Sorry, let me double check it.

[EDIT] I'm an idiot. I uploaded the wrong file. It should be the 3/21 build. Here's the right link. BeHappyCAT (http://www.mytempdir.com/1265030).

shon3i

22nd March 2007, 20:00

Chumbo

22nd March 2007, 22:30

Thanks Chumbo, work like charm :)

@Chumbo, tebasuna51, Alwa, can we stop development now, and sum what we have here

BeHappy.exe 03/21/2007 version 0.1.9.35190
AvisynthWrapper.dll ? 16 or 32, you tebasuna51 said that all encoders support 32bit output except aud-x and CT. Did i were have problems using 32bit and CT encoder on 16bit audio?
And can somebody upload lastest versions of both thanks.

Aslo to upload lastest extensions.

What to use for ogg encoding and link?
What is last aften version and link?
What is good for wav2mono & wav2stereo extension?
What is last NicAC3?
What encoder for musepack?
Glad it worked. :)

btw, my project contains all the changes made by CAT crew. ;) I also have the latest utilities. I think I have twolame.exe too since that wasn't there initially. I can pack up all the parts including the two versions of avisynthwrapper some time tonight.

[EDIT] I packed my BeHappy program folder into this package (http://www.mytempdir.com/1265592). btw, I used your most recent install, but I updated all the files that needed updating, i.e., aften, extensions, etc. I included the original/old avisynthwrapper dll but the new one is the active one.

tebasuna51

23rd March 2007, 05:11

AvisynthWrapper.dll ? 16 or 32, you tebasuna51 said that all encoders support 32bit output except aud-x and CT. Did i were have problems using 32bit and CT encoder on 16bit audio?
AFAIK all encoders work with new AvisynthWrapper.dll and the added functions than work only when is necessary and not always like with old AvisynthWrapper.dll.
And can somebody upload lastest versions of both thanks.
Aslo to upload lastest extensions.

What to use for ogg encoding and link?
What is last aften version and link?
What is last NicAC3?
What encoder for musepack?
I agree with Chumbo package but :

BassAudio.extension (added .cda extension)
NicAudio.extension (added Normalize in mp123)

There are some files I don't know if are needed:
lame_enc.dll
wvgain.exe
wvselfx.exe
wvunpack.exe

And other I have new versions:
enc_aacPlus.exe 45.056 28/09/2006 12:03
enc_aacplus.dll 529.408 13/02/2007 19:28 Winamp v5.33
flac.exe 237.568 15/02/2007 17:38 v1.14
neroAacEnc.exe 843.776 12/02/2007 09:49 and SS2
oggenc2.exe 403.968 25/10/2006 14:30 v2.83
twolame.exe 294.912 22/03/2007 13:20 v03.10b
and
mppenc.exe 109.568 13/11/2006 02:00 v1.16

I use http://www.rarewares.org/ for:
LAME 3.97
twoLame 0.3.10b
Oggenc2.83 using aoTuVb5
Flac v1.14

Also:
http://kurtnoise.free.fr/index.php?dir=Aften/
http://www.wavpack.com/
http://www.musepack.net/

And this NicAudio.dll (http://nic.dnsalias.com/NicAudio_alpha3.zip)

What is good for wav2mono & wav2stereo extension?
Sometimes we need split multichannel input in mono/stereo wav's to be edited in stereo audio editors. Split in mono wav's is a BeSweet tool than can be accomplished with these utils (compressed are only 5 KB and are optional)

Here are (http://www.mytempdir.com/1265679) BassAudio.extension, NicAudio.extension and bass_cd.dll. This bass library (in AviSynth 2.5\plugins) enable BeHappy to open CD Audio directly.

序列人

23rd March 2007, 06:16

setarip_old

23rd March 2007, 07:41

This is often an indication that other memory is corrupt.Sounds like the program is suggesting that you may have some bad RAM...

alwa

23rd March 2007, 14:11

There are some files I don't know if are needed:
lame_enc.dll
wvgain.exe
wvselfx.exe
wvunpack.exe
BeHappy works fine without those files.

LAME 3.97 produces files with the wrong bitrate in ABR mode. Someone else had problems with it? At the moment i use ver. 3.96.1 which produces files with correct bitrate.

Aften (http://win32builds.sourceforge.net/aften/index.html)(Other link don't work right now)

@序列人: Can you reproduce the error or does it happen incidentally?

Chumbo

23rd March 2007, 15:41

Starting job bits0001.mpa->bits0001.m4a
Error: System.AccessViolationException: Attempted to read or write protected memory. This is often an indication that other memory is corrupt.
at BeHappy.AviSynthClip.dimzon_avs_init(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()

Some one can help me?
You can try alternative ways of feeding your audio:
- load your mpa into graphedt to see what your filter chain is. Your system may not be able to create a filter chain or even if it does, you may have to server it to behappy differently.

- server your mpa file via an AVS script with directshowsource. Make sure to use the fps parameter. If you still get the same error, add the framecount parameter

- use graphedt to save a grf file of your filter chain without the last output, i.e., default directsound device. So your audio output pin should be open. Create an avs script and use your file.grf as the input to directshowsource. Same here, use the fps parameter and you may have to use the framecount parameter.

The other thing to consider is what is your mpa file? Is it actually an ac3 file? If it is, rename it to ac3 and feed it accordingly. Is it a dts file? You can try renaming it to dts. Check out the thread on evodemux as there's plenty of info there on the demuxed audio tracks from both hd dvds and blu-ray dics.

[EDIT] Here's a sample of the avs line:
directshowsource("myfile.grf",fps=23.976,video=false,audio=true,framecount=xxxxx)
Of course set your fps according to your specific feature's setting. Some say fps and framecount are not necessary with audio files, but I guess you can check for yourself. I find it doesn't hurt to use them.

Chumbo

24th March 2007, 00:40

@shon3i,
Thanks to tebasuna and alwa, I updated my folder with the latest files (except enc_aacPlus.exe because I couldn't find the version tebasuna has). I also fed the help text to the encoder's xxxxHelp.txt files for convenience. I also put the avisynth plugin DLLs in their own folder for you.

Here's my updated folder (http://www.mytempdir.com/1266896) link if you want to use it. I hope it helps lessen the work you have to do.

tebasuna51

24th March 2007, 01:58

I updated my folder with the latest files (except enc_aacPlus.exe because I couldn't find the version tebasuna has).

I don't remember from where I download the "AACPlus v2 Encoder (using Winamp 5.3 enc_aacplus.dll)" but here (http://www.mytempdir.com/1266951) is.

shon3i

24th March 2007, 10:21

Thanks, i alredy have lastest enc_aac files, plus i use 5.33/5.34 which is CT 8.03

shon3i

24th March 2007, 11:45

New package is out

Link (http://www.box.net/shared/nkihizx1dh)

i've add whole changelog later

buzzqw

24th March 2007, 12:04

thanks shon3i!

BHH

Deckard2019

24th March 2007, 14:35

New package is out
Thank you but could you just provide a zip or rar file instead ?

shon3i

24th March 2007, 14:39

Thank you but could you just provide a zip or rar file instead ?
Why?

I don't see the point?

buzzqw

24th March 2007, 14:59

not an installer...

BHH

Deckard2019

24th March 2007, 16:14

not an installer...
Thank you buzzqw ;)

shon3i

24th March 2007, 16:44

Then use all files from BeHappy folder and just zip that's all, including avs audio plugins, nothing simpler ;)

BeHappy

aften.exe
aud-x.extension
audxlib.dll
AviSynth Plugin DLLs
AvisynthWrapper.dll
BassAudio.extension
BeHappy.exe
BeHappy.exe.config
ConvertSample.extension
DownMix.extension
DuplicateChannels.extension
enc_aacplus.dll
enc_aacPlus.exe
enc_AudX_CLI.exe
ffmpeg.exe
ffmpeg.extension
flac.exe
flac.extension
lame.exe
libmmd.dll
MP4Box.exe
mppenc.exe
musepack.extension
neroAacEnc.exe
neroAacEnc_SSE2.exe
NicAudio.extension
nscrt.dll
oggenc2.exe
SSRC.extension
twolame.exe
twolame.extension
UpMix.extension
wav2mono.exe
wav2mono.extension
Wav2stereo.exe
Wav2stereo.extension
wavpack.exe
wavpack.extension

Avs Plugins

bass.dll
bassAudio.dll
bass_aac.dll
bass_ac3.dll
bass_ape.dll
bass_cd.dll
bass_flac.dll
bass_mpc.dll
bass_spx.dll
bass_wma.dll
bass_wv.dll
NicAudio.dll
soxfilter.dll

you can use this my package exe and run in administrative mode with "BeHappy_20070324.exe" /a, then setup will extract all files from package to C:\ without real install.

This package is designed for all ppl's who want install and encode easly, for that purpose, i planing to make package who will include lastest AviSynth and Net Framework 2.0.

buzzqw

24th March 2007, 16:58

no...no need to include .net2 just offer to download

BHH

Chumbo

24th March 2007, 23:08

no...no need to include .net2 just offer to download

BHH
Yeah, I agree on this one. :) You should be able to build your package to require the .net frmwrk 2 and then download it if it needs it. It's too much overhead to download for folks who don't have a fast connection and already have it.

madshi

30th March 2007, 19:52

tebasuna51

30th March 2007, 20:37

Open your track in BeHappy and apply the timestretch needed and other changes, instead Enqueue use Export AviSynth Script.

Edit with Notepad (or similar) the avs changing:
"tempo" by "rate" in line:

TimeStretch(last, tempo=95.904)

The new line:

TimeStretch(last, rate=95.904)

Now open the avs with BeHappy and encode (uncheck the DSP's functions now).

Chumbo

30th March 2007, 20:38

Hi guys,

I'm trying to mux audio tracks from my German PAL DVDs to HD-DVD. For that to work I need of course to change the German audio track from 25fps to 23.976fps. I've tried "TimeStretch", but it's more intelligent than I like. It seems to do pitch correction. But that's *BAD* in my case, cause by far most German audio tracks have been sped up without pitch correction. So basically I need a stupid and simple stretch, much less sophisticated than BeHappy's TimeStretch.

Can anybody help me, please?

Thanks!
If you export the avs script, you'll note that all BeHappy does is do the math and put a line in the resulting avs script similar to this:
########################################
# [DSP: TimeStretch - 25 -> 23.976]
########################################
TimeStretch(last, tempo=95.904)
I know BeSweet is outdated, but it does frame conversion if you want to try it and compare.

[EDIT] I knew tebasuna would have the answer. :)

tebasuna, what do you think about updating the UI to include a checkbox for pitch in the stretch dialog? If checked, then tempo is used, otherwise rate is used.

madshi

30th March 2007, 21:07

Thank you guys! :) That should do the trick for me. Of course having this option in the GUI would be great, but as long as I can make it work, I'm happy.

tebasuna51

30th March 2007, 23:47

tebasuna, what do you think about updating the UI to include a checkbox for pitch in the stretch dialog? If checked, then tempo is used, otherwise rate is used.
There are three options with TimeStretch:
- rate: tempo and pitch changed
- tempo: only tempo changed
- pitch: only pitch changed

this third option can be used to repair the tracks in this case if we don't need change the fps in video.

Maybe for the next version.

Chumbo

31st March 2007, 01:02

There are three options with TimeStretch:
- rate: tempo and pitch changed
- tempo: only tempo changed
- pitch: only pitch changed

this third option can be used to repair the tracks in this case if we don't need change the fps in video.

Maybe for the next version.
From the way you broke it down, sounds like the default behavior is what madshi needed, i.e., tempo, because he didn't want pitch change. The suggestion to use rate would actually do both. I just want to make sure I understand. Thanks.

Here's the usage from the help file (just like you broke it down):
TimeStretch allows changing the sound tempo, pitch and playback rate parameters independently from each other, i.e.:
Sound tempo can be increased or decreased while maintaining the original pitch.
Sound pitch can be increased or decreased while maintaining the original tempo.
Change playback rate that affects both tempo and pitch at the same time.
Choose any combination of tempo/pitch/rate.

tebasuna51

31st March 2007, 01:52

From the way you broke it down, sounds like the default behavior is what madshi needed, i.e., tempo, because he didn't want pitch change. The suggestion to use rate would actually do both. I just want to make sure I understand.

The normal usage is 'tempo': modify the audio duration (preserving the pitch) to maintain the sync with a video with a new duration because is played at other fps. If the video is modified adding or deleting frames to change the fps, but maintain the duration, the audio don't need change.

But now madshi say: "cause by far most German audio tracks have been sped up without pitch correction", then the pitch is incorrect with the old fps, and tempo is incorrect for the new fps.

Chumbo

31st March 2007, 02:08

The normal usage is 'tempo': modify the audio duration (preserving the pitch) to maintain the sync with a video with a new duration because is played at other fps. If the video is modified adding or deleting frames to change the fps, but maintain the duration, the audio don't need change.

But now madshi say: "cause by far most German audio tracks have been sped up without pitch correction", then the pitch is incorrect with the old fps, and tempo is incorrect for the new fps.
Yeah, that makes sense which is what I thought. Thanks.

btw, I went ahead and started the next round of changes. ;) I had time so I added the UI for those options.

2007-03-30 (Chumbo)
+ added rate control to the TimeStretch DSP module - you can select rate, pitch or both
+ updated the version to 0.1.10.* from 0.1.9.*. With all the changes we made we should've update the version sooner

Get latest BeHappy update (http://www.mytempdir.com/1276813). Please test and report any problems.

=Wolf=

31st March 2007, 07:58

I has tried from AC3 5.1 448kbps-> Mp3 Sterio 2.0 and here that at me has left....
http://home.farlep.net/~wolf2/17_jpn.mp3

old version is work fine....

p.s sorry for my bad English :(

madshi

31st March 2007, 11:06

Hey, that's what I call a quick implementation of a feature suggestion! Thanks, Chumbo!! :)

May I suggest a renaming of the options, though? I think the names are not clear enough. Basically "change frame rate and pitch" can mean two things:

(1) It could mean: Do no pitch correction, so that the pitch is practically changed during conversion.

(2) It could mean: Do pitch correction, so that the pitch after the conversion is the same as before.

Basically I'm not sure when you say "change pitch" do you mean "do pitch correction"? Or do you mean "don't do pitch correction" (the latter of which would result in a pitch change during conversion).

I'd suggest these labels for the 3 options:

(1) Change frame rate without pitch correction.
(2) Change frame rate with pitch correction.
(3) Perform pitch correction without frame rate change.

Alternatively you could also replace the 3 radio buttons with 2 check buttons:

(1) Pitch correction.
(2) Run length correction (or "run time correction" or "frame rate correction").

Actually I think I'd prefer the 2 check buttons over 3 radio buttons. That's the most intuitive option set IMHO. Of course having both check buttons unchecked would simply do no modification to the audio stream at all.

------------

Btw, after doing some tests yesterday, the "rate" conversion is really exactly what I needed. The converted audio tracks are practically perfect. Perfect sync, perfect pitch.

Don't know if this is a known problem, but with the "tempo" method, the sync was not perfect. The final audio file was a few seconds too short. I believe it played slightly too "fast". No such problems with the "rate" option. Also I noticed that the whole conversion process needs significantly less time when using the "rate" option. That was to be expected, of course, cause it does much simpler manipulations to the audio file.

Anyway, for all people trying to convert PAL audio tracks to 23.976, I suggest this approach:

(1) Check whether the PAL track has too high pitch (this is the case for by far most PAL tracks, but not for all).
(2) If the pitch of the PAL track is too high, use the "rate" option.
(3) If the pitch is not too high, use the "tempo" option. However, you might have sync problems this way.

tebasuna51

31st March 2007, 12:13

The parameters 'tempo', 'pitch' and 'rate' are well know not only by AviSynth users but also by BeLight-BeSweet users then the name must appear in the options.

I'm opposed to use the words 'frame rate' when speak about audio, the real audio frame rate is never changed with this options.

Then I propose:

(1) Tempo changed preserving pitch.
(2) Pitch changed preserving tempo.
(3) Rate, tempo and pitch changed.

@madshi: "with the "tempo" method, the sync was not perfect. The final audio file was a few seconds too short."
Sometimes the exact change needed is not the exact relation between video framerates, then we need use the 'custom' option.

tebasuna51

31st March 2007, 12:25

I has tried from AC3 5.1 448kbps-> Mp3 Sterio 2.0 and here that at me has left....
http://home.farlep.net/~wolf2/17_jpn.mp3

old version is work fine....
Please send the original ac3 5.1 to try reproduce your problem.

Here with an ac3 5.1 448 kb/s, a downmix DSP and encode with Lame work fine.

=Wolf=

31st March 2007, 12:50

@tebasuna51
Here. Take it

http://file4.webfile.ru/1362905/17.ac3
48000Hz 448 kb/s tot , 6 chnls (3/2 .1)

madshi

31st March 2007, 13:55

The parameters 'tempo', 'pitch' and 'rate' are well know not only by AviSynth users but also by BeLight-BeSweet users then the name must appear in the options.
Agreed, that makes sense.

I'm opposed to use the words 'frame rate' when speak about audio
Yeah, makes sense to me, too.

Then I propose:

(1) Tempo changed preserving pitch.
(2) Pitch changed preserving tempo.
(3) Rate, tempo and pitch changed.
Again for me "pitch changed" is not clear. It could mean "pitch corrected" in the sense "changed=corrected". Or it could mean "pitch changed because it was not corrected". I suggest making use of the terms "pitch correction" and "no pitch correction", as that is a fixed term everybody understands.

Sometimes the exact change needed is not the exact relation between video framerates, then we need use the 'custom' option.
"Rate" gave me perfect sync where "Tempo" give me incorrect sync with the same audio track.

Chumbo

31st March 2007, 16:19

tebasuna51

31st March 2007, 16:44

Again for me "pitch changed" is not clear. It could mean "pitch corrected" in the sense "changed=corrected". Or it could mean "pitch changed because it was not corrected". I suggest making use of the terms "pitch correction" and "no pitch correction", as that is a fixed term everybody understands.
Pitch changed means more acute or more deep we don't know if is correct or incorrect.

This functions can be used for other purpose than audio movie tracks. For instance we can need change the pitch, preserving tempo, to accommodate music for different karaoke singers.

"Rate" gave me perfect sync where "Tempo" give me incorrect sync with the same audio track.
Yes, yes, the 'tempo' algorithm is more complex, and more inexact, like you say, for that we need any custom correction and try with different values (really I don't found seconds of difference, only ms, in movie tracks).

tebasuna51

31st March 2007, 16:46

@tebasuna51
Here. Take it

http://file4.webfile.ru/1362905/17.ac3
48000Hz 448 kb/s tot , 6 chnls (3/2 .1)

Sorry I get a "404 - Not Found" error with this link.

alwa

31st March 2007, 16:48

@tebasuna51
Here. Take it

http://file4.webfile.ru/1362905/17.ac3
48000Hz 448 kb/s tot , 6 chnls (3/2 .1)
link fix (http://webfile.ru/1362905)
I can't reproduce your issue.
I'm using NicAC3Source/Bass -> Downmix DSP -> LAME and the output is fine.

Again for me "pitch changed" is not clear. It could mean "pitch corrected" in the sense "changed=corrected". Or it could mean "pitch changed because it was not corrected". I suggest making use of the terms "pitch correction" and "no pitch correction", as that is a fixed term everybody understands.
then:
(1) Tempo changed, pitch correction.(hint: changes the length of the audio track while preserving the original pitch)
(2) Pitch changed preserving tempo.(hint: manipulates pitch, but Track length will kept unchanged)
(3) Rate, tempo and no pitch correction.(hint: changes the length of the audio track without preserving the original pitch )
?

Are the details right(i may be completely wrong)? :rolleyes:

Chumbo

31st March 2007, 17:10

...then:
(1) Tempo changed, pitch correction.(hint: changes the length of the audio track while preserving the original pitch)
(2) Pitch changed preserving tempo.(hint: manipulates pitch, but Track length will kept unchanged)
(3) Rate, tempo and no pitch correction.(hint: changes the length of the audio track without preserving the original pitch )
?

Are the details right(i may be completely wrong)? :rolleyes:
Thanks alwa. I'll add the hints too.

[EDIT]2007-03-31 (Chumbo)
+ updated TimeStretch rate control radio button labels per suggestions by tebasuna, alwa and madshi.
+ added hints/popup text to rate control radio buttons. I wound up using the AVISynth text for the hint text as it's straightforward.

Latest BeHappy update (http://www.mytempdir.com/1277580)

=Wolf=

1st April 2007, 01:45

I have tried to make 5.1 -> 2.0 so here that has left....
http://webfile.ru/1362905 <- ac3

AC3 5.1 -> MP3
Starting job 17_jpn.ac3->17_jpn.mp3
Found Audio Stream
Channels=2, BitsPerSample=32 int, SampleRate=48000Hz
lame.exe -b 128 -h -S --silent - "C:\FTP\Incoming\17_jpn.mp3"
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Finalizing encoder
Complete

AC3 5.1 -> Ogg
Starting job 17_jpn.ac3->17_jpn.ogg
Found Audio Stream
Channels=2, BitsPerSample=32 float, SampleRate=48000Hz
oggenc2.exe -Q --quality 3 -o "C:\FTP\Incoming\17_jpn.ogg" -
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Finalizing encoder
Complete

tebasuna51

1st April 2007, 03:10

I have tried to make 5.1 -> 2.0 so here that has left....
Seems all perfect.

1) Ac3 5.1 decoded by NicAc3Source produce 32 float. The downmix function works in 32 float, perfect. Now we want encode with Lame. Warning Lame don't support 32 float, the best quality supported is 32 int, then we need a conversion to 32 int. Done. All the process is make with the best quality possible.

2) Ac3 5.1 to Ogg. All the same but oggenc2 accept 32 float. Congratulations! we don't need convert the 32 float and is delivered to the encoder directly.

I test your sample and all is ok here.

=Wolf=

1st April 2007, 03:17

@tebasuna51
Then why at me such result turns out?
At work lame...
http://home.farlep.net/~wolf2/17_new.mp3

tebasuna51

1st April 2007, 03:45

Here (http://www.mytempdir.com/1278134) is your sample with:
- Open with NicAc3Source("17.ac3", DRC=0)
- Douwmix -> 2.0 with DLP II
- Normalize(45%) to be comparable with your sample
- Automatic conversion to 32 int
- Encoded with Lame 128 Kb/s CBR (like your sample)

Only your mp3 is broken? The ogg play fine?

tebasuna51

1st April 2007, 04:41

BeHappy update (http://www.mytempdir.com/1278153) :
- Changes to support new enc_aacPlus.exe from Shon3i. (NeroDigitalEncoder.cs, CodingTechnologiesAAC.cs)

- New enc_aacPlus.exe version to support enc_aacPlus.dll from WinAmp 5.33/5.34 and new MP4mux instead MP4Box. (enc_aacPlus.exe, MP4mux.exe)

=Wolf=

1st April 2007, 06:55

@tebasuna51
Yes....
- Open with NicAc3Source("17.ac3", DRC=0)
- Douwmix -> 2.0 with Stereo
- Normalize(100%)
- Encoded with Lame 128 Kb/s CBR
ONLI Lame broken... ogg, aac (32float) working is fine.... :(

madshi

1st April 2007, 11:08

The new options for rate/tempo/pitch have good names now (and helpful hints). Thank you guys!

tebasuna51

1st April 2007, 11:23

ONLI Lame broken... ogg, aac (32float) working is fine...
I use LAME 3.97 Release (http://www.rarewares.org/mp3.html)
Bundle: includes lame.exe, lame_enc.dll. (ICL9.1) 2006-10-03.

tebasuna51

4th April 2007, 11:56

AFAIK the actual ac3 free decoders (NicAudio, Azid, ffdshow, Ac3Filter, ...) can't decode DDP.
Only if you have installed in your system a DirectShow decoder filter capable to decode DDP, you can convert it to any other format.

Rectal Prolapse

4th April 2007, 18:23

Sonic Audio Decoder 4.2 can decode DD+ soundtracks.

After you install Sonic Cinemaster Decoder Pack 4.2, you can then use GraphEdit to make a .grf file, that can be loaded into BeHappy as a DirectShowSource.

Graph:

File Source (Async.) -> Sonic HD Demuxer -> Sonic Audio Decoder 4.2

Hopefully this will work for you.

idbirch2

8th April 2007, 18:17

Hi, I'm trying to correct a gradual sync problem after a HD-DVD re-encode and am trying to stretch an AC3 file. Only ever so slightly as over a 2h21m film, it is only about 3.5s out by the end.

Anyway, when I try and do this with BeHappy 0.1.10.17947 I croaks straight away, in the log window, all I get is:

Found Audio Stream
Channels=6, BitsPerSample=16 int, SampleRate=48000Hz
Aften.exe -v 0 -b 384 -m 1 -readtoeof 1 -cmix 0 -smix 0 -dsur 0 -dnorm 31 -dynrng 5 - "H:\\poto-eac3to_happy.ac3"
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Error: System.ApplicationException: Abnormal encoder termination 1
at BeHappy.Encoder.encode()
#### Encoder StdErr ####

Aften: A/52 audio encoder
Version 0.06
(c) 2006-2007 Justin Ruggles, et al.

usage: aften [options] <input.wav> <output.ac3>
type 'aften -h' for more details.

Its as if the syntax used is wrong but how do I fix this? Thanks.

tebasuna51

8th April 2007, 21:08

"H:\\poto-eac3to_happy.ac3"

Seems the output file sintax is the problem

idbirch2

8th April 2007, 22:15

Chumbo

9th April 2007, 00:41

Sorry, I should have mentioned, I did spot that already and tried deleting the extra \ but the result is exactly the same. I can preview the output so I know the source filter is working. I don't suppose it matters but why does BH insert double \\'s because that wasn't an error on my part?
Below is an output from one of my many transcodes with BH and it never adds a double backslash. Internally, the code may use a "\\" to translate to "\" but that's standard.
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=48000Hz
Aften.exe -v 0 -b 640 -m 1 -readtoeof 1 -cmix 0 -smix 0 -dsur 0 -dnorm 27 -dynrng 5 - "E:\Media\audio\hf.ac3"
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Finalizing encoder

Check your input file path and name and make sure the input is okay. Try another drive and maybe even a simple file name, i.e., remove the dash and underscore, just to rule those things out.

idbirch2

9th April 2007, 11:16

Sorry, no joy. I moved the entire BeHappy folder to the root of C:\ in case it didn't like its location and also moved the target ac3 file to the same place, the result is the same. Here's how I have behappy set up:

http://img03.picoodle.com/img/img03/7/4/9/f_screenm_f2cf10b.gif (http://www.picoodle.com/view.php?srv=img03&img=/7/4/9/f_screenm_f2cf10b.gif)

What version of Aften are you using?

edit: I forgot, here's the output from this test:

Starting job test.ac3->test2.ac3
Found Audio Stream
Channels=6, BitsPerSample=16 int, SampleRate=48000Hz
Aften.exe -v 0 -b 384 -m 1 -readtoeof 1 -cmix 0 -smix 0 -dsur 0 -dnorm 31 -dynrng 5 - "C:\test2.ac3"
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Error: System.ApplicationException: Abnormal encoder termination 1
at BeHappy.Encoder.encode()
#### Encoder StdErr ####

Aften: A/52 audio encoder
Version 0.06
(c) 2006-2007 Justin Ruggles, et al.

usage: aften [options] <input.wav> <output.ac3>
type 'aften -h' for more details.

tebasuna51

9th April 2007, 11:51

What version of Aften are you using?

edit: I forgot, here's the output from this test:
Channels=6, BitsPerSample=16 int, SampleRate=48000Hz
Aften.exe -v 0 -b 384 -m 1 -readtoeof 1 -cmix 0 -smix 0 -dsur 0 -dnorm 31 -dynrng 5 - "C:\test2.ac3"
...
Aften: A/52 audio encoder
Version 0.06

Please use the last shon3i package (http://forum.doom9.org/showthread.php?p=974646#post974646)

For Aften you need rev449 at least, because with 2h21m 6 channnls you need "-readtoeof 1" not supported by v0.06.

idbirch2

9th April 2007, 12:52

Thanks very much for your help. I did originally start out using build 449 but I must have been doing something else wrong then as having gone back to 449, BH is now working :)

tauka

25th April 2007, 04:08

tebasuna51

25th April 2007, 19:58

hi guys, i need a little help. i have the newest behappy, and i try to convert an ac3 file from pal to ntsc 6ch wav, its 2h24min long.. the problem is this: i open the wav with any prog, it shows 2:04:17..
Then don't trust in these prog's.
in delaycut in the info actually the correct length is shown (2h31min), but in the target file the 2:04:17 again.. what can be the problem? the wav is actually around 4,8gig.. any suggestion?
Wav files have a header field with a 4GB limit (or 2:04:17 for wav 16 bit int, 6 channel, 48 KHz) but you have more data until 4.8 GB or 2h31m.
Edit: see also this link (http://forum.doom9.org/showthread.php?p=974973#post974973)

Don't worry Aften can encode this long track with the parameter: -readtoeof 1.
If you want encode to aac, NeroAacEnc also support long files with -ignorelength parameter.
is it possible at all to convert such a long track with behappy?
Of course, and don't need the huge intermediate wav file. Just I make a test with an ac3 7846.336 seconds. ( 2 h. 10 m. 46.336 s.) and timestretch 25 -> 23.976 directly to ac3 and the result is 8181.472 seconds. ( 2 h. 16 m. 21.472 s.) with less than a frame error (32 ms) than 7846.336*25/23.976 = 8181,448

thuongshoo

1st May 2007, 17:52

Hi ! Thank all for continueing to developt BeHappy. I love all new feature.
I used to use OggdropXP. It can estimate bitrate in Q mode. I hope that a new version will has this feature.
Bye!

thuongshoo

3rd May 2007, 08:52

tebasuna1! You seem to be the one which do last correct. Can you send a copy of newest version of BeHappy to me ?
Thank you!

tebasuna51

3rd May 2007, 14:27

tebasuna1! You seem to be the one which do last correct. Can you send a copy of newest version of BeHappy to me ?
Last links resumed:
2007-03-24 Last Shon3i BeHappy package (http://www.box.net/shared/nkihizx1dh)

2007-03-31 Chumbo mod (http://www.mytempdir.com/1277580) for Timestretch and others v0.1.10

2007-04-01 BeHappy mod (http://www.mytempdir.com/1278153) to support new enc_aacPlus.exe and MP4mux

Don't forget last Aften 0.07 (http://forum.doom9.org/showthread.php?p=996683#post996683)

thuongshoo

5th May 2007, 10:47

Oh! Thanks tebasuna51!
I didn't say clearly. I like source code :)
These pic decribe my words

tebasuna51

5th May 2007, 13:38

Last BeHappy sources (http://www.mytempdir.com/1318681).

Use http://www.imageshack.us/ to share pic's and avoid the "Attachments Pending Approval".

tebasuna51

7th May 2007, 16:21

I didn't say clearly. I like source code :)
These pic decribe my words
Sorry I can't reproduce your problem (I haven't Visual C++), I only obtain this:
http://img263.imageshack.us/img263/8694/ctaacuf6.png (http://imageshack.us)

NoX1911

13th May 2007, 18:50

I'm trying to convert AC3 to AAC 6ch. The source is a DVB stream (live concert). I don't know if it has DRC. How can i figure that out (metadata)?

If i use 'NicAC3Source (DRC)' what compression rate will be used? Most standalone dvd players have 3 profiles (low, med, high).

tebasuna51

13th May 2007, 19:58

I'm trying to convert AC3 to AAC 6ch. The source is a DVB stream (live concert). I don't know if it has DRC. How can i figure that out (metadata)?
You need read the header like in http://forum.doom9.org/showthread.php?p=993860#post993860
Read the next posts also.
If i use 'NicAC3Source (DRC)' what compression rate will be used? Most standalone dvd players have 3 profiles (low, med, high).
The attenuation values supplied in the ac3 stream are used without correction. I don't know what is: low, med, high. Maybe half, full, double?

NoX1911

13th May 2007, 23:51

Ok.. just hoped there would be something like this where you could additionally modify the DRC (or change metadata/Line Mode Profile of AC3).

http://www.imagebanana.com/img/ykt0wrx8/ac3.png

http://www.imagebanana.com/img/a6x8zg0/drc.png

tebasuna51

14th May 2007, 00:38

Ac3Filter have a DRC function independent of the DRC data in ac3 stream. NicAudio is a decoder to apply only the DRC data in the stream, if you want addicional/different compression you can use SoxFilter.

NoX1911

14th May 2007, 00:56

I think its not a different compressor. It is intended by Dolby to have impact on DRC like this to optimize volume for different environments (small speakers (high compression/flat sound), big speakers (no compression/full dynamic)). That's the way 'dialogue normalization' works. There are official dolby whitepapers linked in the faq sticky (http://forum.doom9.org/showthread.php?t=56020). I think i read it there...

My settop dvd player has some static values (low, med, high) for DRC that do the same what Ac3Filter does (i think) so this must be intended by dolby.

The second screenshot i posted above holds some metadata for 'line mode profile'. These could be preset values for that additional compression and may be used by Nic's DRC method.

But i'm not sure on that...

Edit:
solved... neroaacenc.exe has to be moved to 'encoder' folder.

Beside of that BeHappy doesn't work with my scenario. I use BeHappy with aac update.
NicAc3Source(DRC) to load my Ac3 file. If i output as avs script it works fine with mpc. If i enqueue and start the batch process following error msg appears in the log:

Translation:
Das System kann die angegebene Datei nicht finden = File not found

Starting job ggg.ac3->ggg.mp4
Found Audio Stream
Channels=2, BitsPerSample=32 float, SampleRate=48000Hz
encoder\neroAacEnc.exe -ignorelength -q 0.3 -if - -of "F:\ggg.mp4"
Error: System.ApplicationException: Can't start encoder: Das System kann die angegebene Datei nicht finden ---> System.ComponentModel.Win32Exception: Das System kann die angegebene Datei nicht finden
bei System.Diagnostics.Process.StartWithCreateProcess(ProcessStartInfo startInfo)
bei System.Diagnostics.Process.Start()
bei BeHappy.Encoder.createEncoderProcess(AviSynthClip x)
--- Ende der internen Ausnahmestapelüberwachung ---
bei BeHappy.Encoder.createEncoderProcess(AviSynthClip x)
bei BeHappy.Encoder.encode()

solved... neroaacenc.exe has to be moved to 'encoder' folder.

tebasuna51

14th May 2007, 03:16

My settop dvd player has some static values (low, med, high) for DRC that do the same what Ac3Filter does (i think) so this must be intended by dolby.
Not exactly. Your standalone, and soft decoders like PowerDVD, Azid, ffdshow, NicAudio, ... read the gain/attenuation value put in the ac3 stream for each block (5.33 ms) and apply the full value or a part only, don't calculate the DRC needed analyzing the sound decoded.
AC3Filter analyze the sound and calculate the DRC required.
The second screenshot i posted above holds some metadata for 'line mode profile'. These could be preset values for that additional compression and may be used by Nic's DRC method.

These presets are used at encoding time. The encoder calculate the appropriate gain/attenuation in function of these presets and the actual volume and put this value in the ac3 stream for each block (5.33 ms). The decoders don't need calculate, only read the values proposed by the encoder.

thuongshoo

16th May 2007, 04:27

This is a new version of BeHappy
http://www.box.net/shared/5tyvsczuxo
:)

alwa

16th May 2007, 18:08

This is a new version of BeHappy
http://www.box.net/shared/5tyvsczuxo
:)

@thuongshoo:You should list your changes.
I've noticed chages in OggVorbisEncoder.cs and CodingTechnologiesAAC.cs, please document them. :rolleyes:

Maybe the mod should have it's own repository, something like a project page on sf.net.

tebasuna51

16th May 2007, 21:30

In CodingTechnologiesAAC.cs only size parameters to avoid the problem in post #568.

In OggVorbisEncoder.cs size parameters and:
if (Quality <= 4) ApproximateBitrate = (double)(Quality + 2)*16 + 32;
if ((Quality > 4) && (Quality <= 8)) ApproximateBitrate = (double)(Quality - 4) * 32 + 128;
if (Quality > 8 && Quality <= 9) ApproximateBitrate = (double)(Quality - 8) * 64 + 256;
if (Quality > 9) ApproximateBitrate = (double)((Quality - 9)) * 179.8 + 320;
rbtnVBR.Text = string.Format("Variable Bitrate Q={0} approximate {1} kbs", Quality,ApproximateBitrate);

I don't use ogg files and don't know if is correct or not, but maybe the number of channels need to be considered.

thuongshoo

17th May 2007, 11:56

I've noticed chages in OggVorbisEncoder.cs and CodingTechnologiesAAC.cs, please document them.
@alwa:I'm sorry!
CodingTechnologiesAAC.cs: I resized and changed position of component to avoid the problem in post #568
OggVorbisEncoder.cs: added "ApproximateBitrate" value.
I don't use ogg files and don't know if is correct or not, but maybe the number of channels need to be considered.
@Tebasuna51: I tested 2 file and feel Ok. Everyone please re-test :)

It is quite right if I use oggenc2 while oggenc show a lower bitrate.
I'm using MediaInfo to observer information of media file
This feature come from OggDropXPd
http://img149.imageshack.us/img149/4330/approximatebitrateix7.th.png (http://img149.imageshack.us/my.php?image=approximatebitrateix7.png)

Link to download oggenc2 ( I prefer oggdropXPd V.1.8.9 using libVorbis v1.1.2 with IMPULSE_TRIGGER_PROFILE Option)
and OggDropXP
http://www.rarewares.org/ogg.html
Good luck! :)

tebasuna51

17th May 2007, 13:08

@Tebasuna51: I tested 2 file and feel Ok. Everyone please re-test

I say the approximate bitrate is only for stereo audio:

StereoTrack -> Ogg Q=3 approx. 112 kbs -> Real 113 Kb/s OK.
StereoTrack -> Downmix to mono -> Ogg Q=3 approx. 112 kbs -> Real 62 Kb/s
StereoTrack -> Upmix to 5.1 -> Ogg Q=3 approx. 112 kbs -> Real 441 Kb/s

I propose only a literal change:
Variable Bitrate Q=3 (approx. 112 kb/s for stereo)

MuLTiTaSK

22nd May 2007, 18:55

this tool is amazing i mostly use it to convert .ac3 files i extract from my dvd's to .mp3 format and it works great

but now i want to use it to convert a 224Kbps CBR 2CH Stereo 48KHz .mpa i extracted with DGIndex from one of my tuner caps to .ac3 format so i can import it into a dvd authoring program

i never did this before so i'am not sure what encoder to use and what settings to set in BeHappy i tried with ffmpeg but not sure if i did it right

i'am using lastest Shon3i BeHappy package with 2007-04-01 BeHappy mod http://forum.doom9.org/showthread.php?p=998715#post998715

below is the log i got after the encode was done
hope someone can help me do this conversion right thanks in advance

settings i used in BeHappy
http://i12.tinypic.com/4ka87t5.png

Starting job test T01 DELAY 0ms.mpa->test T01 DELAY 0ms.ac3
Found Audio Stream
Channels=2, BitsPerSample=32 int, SampleRate=48000Hz
ffmpeg.exe -i - -y -acodec ac3 -ab 192 "H:\Caps\test T01 DELAY 0ms.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: 3072 kb/s
Stream #0.0: Audio: pcm_s32le, 48000 Hz, stereo, 3072 kb/s
Output #0, ac3, to 'H:\Caps\test T01 DELAY 0ms.ac3':
Stream #0.0: Audio: ac3, 48000 Hz, stereo, 192 kb/s
Stream mapping:
Stream #0.0 -> #0.0

video:0kB audio:145896kB global headers:0kB muxing overhead 0.000000%

tebasuna51

22nd May 2007, 20:30

shon3i

22nd May 2007, 20:38

yep, everythig is normal like log says, if file is playable, there is nothing to do.

btw new package will be redy, at the end of this week

MuLTiTaSK

22nd May 2007, 20:52

@MuLTiTaSK
What is the problem?
Seems the conversion is ok. You have a 'H:\Caps\test T01 DELAY 0ms.ac3' about 145896 kB.
Don't worry about info from ffmpeg, all is ok.

Maybe you can use the last Aften (http://kurtnoise.free.fr/index.php?dir=Aften/&file=aften_rev511.zip) like encoder with more options than ffmpeg.

i wasnt sure if BeHappy was reporting a error by looking at the log i was a little confused thank you guys for clearing that up for me

would i get better quality using aften to encode the .mpa to .ac3?

i notice it had more settings but i did'nt know which ones to set so i used ffmpeg ;/

would these settings be correct for my conversion thanks again
http://i13.tinypic.com/4lpfh1x.png

tebasuna51

22nd May 2007, 21:05

These default settings are ok, equivalents to ffmpeg.

Aften is based in same code than ffmpeg but have some improvements, maybe not noticeable but there are.

If you want ensure the max volume without clipping you can use the DSP function Normalize 100%

MuLTiTaSK

22nd May 2007, 22:55

worked like a charm thanks alot tebasuna51 & everyone else keeping this great program alive

shon3i your packages help a great a deal thank you very much for building them saves new users to the program alot of grief

i'am still surprised why more people dont use this gem

is there a reason BeHappy is not listed on sites like?

http://www.digital-digest.com/software/topcategory-23.html
http://www.videohelp.com/tools/sections/audio-encoders

thuongshoo

27th May 2007, 13:34

StereoTrack -> Ogg Q=3 approx. 112 kbs -> Real 113 Kb/s OK.

StereoTrack -> Downmix to mono -> Ogg Q=3 approx. 112 kbs -> Real 62 Kb/s
62 = 112/2
StereoTrack -> Upmix to 5.1 -> Ogg Q=3 approx. 112 kbs -> Real 441 Kb/s
I don't know how 5.1 file is encoded.
I used word "Approximate Bitrate" , not "Bitrate". We are using "Q mode"

tebasuna51

27th May 2007, 14:24

I don't know how 5.1 file is encoded.
I used word "Approximate Bitrate" , not "Bitrate". We are using "Q mode"

5.1 (441 Kb/s) is encoded with Q mode and literal:

"Variable Bitrate Q = 3 approximate bitrate 112 kbs"

And I propose a literal like:

"Variable Bitrate Q = 3 (approx. 112 Kb/s for stereo)"

paranoid87

16th June 2007, 18:05

are you sure? teba? that the 5.1 is encoded witht the Q mode?

tebasuna51

16th June 2007, 22:43

are you sure? teba? that the 5.1 is encoded witht the Q mode?
I'm sure about the commandline:

Channels=6, BitsPerSample=16 int, SampleRate=48000Hz
oggenc2.exe -Q --quality 3 -o "G:\Test.ogg" -

and the result is a ogg 5.1 with bitrate 441 (Foobar Properties)

chros

21st June 2007, 19:08

Last links resumed:
2007-03-24 Last Shon3i BeHappy package (http://www.box.net/shared/nkihizx1dh)
I'm using this version, but encoding to flac doesn't manage:
Starting job audio.ac3->audio.flac
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=48000Hz
flac.exe --best --force -o "F:\audio.flac" --silent --force-raw-format --endian=little --channels=6 --bps=24 --sample-rate=48000 --sign=signed --input-size=7172361216 -
Writing PCM data to encoder's StdIn
Error: System.IO.IOException: Using the pipe has finished.
source: ac3 , with NicAC3Source , filters: none, flac: best

tebasuna51

22nd June 2007, 01:56

I'm using this version, but encoding to flac doesn't manage...

Works for me with small samples (--input-size=183057408):
Starting job Test.ac3->Test.flac
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=48000Hz
flac.exe --best --force -o "D:\Test.flac" --silent --force-raw-format --endian=little --channels=6 --bps=24 --sample-rate=48000 --sign=signed --input-size=183057408 -
Writing PCM data to encoder's StdIn
Finalizing encoder
Complete

But also crash with big sources (--input-size=7068791808):
Starting job 136_min.ac3->136_min.flac
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=48000Hz
flac.exe --best --force -o "G:\136_min.flac" --silent --force-raw-format --endian=little --channels=6 --bps=24 --sample-rate=48000 --sign=signed --input-size=7068791808 -
Error: System.ApplicationException: Can't start encoder: Cannot process request because the process (1096) has exited. ---> System.InvalidOperationException: Cannot process request because the process (1096) has exited.
...

The same source can be transcoded to wav or aac without problems, then seems a flac encoder problem.

buzzqw

22nd June 2007, 08:10

i can confirm the bug using ffmpeg for piping to faac.
sothing is wrong in faac pipe input

BHH

tebasuna51

22nd June 2007, 11:38

i can confirm the bug using ffmpeg for piping to faac.
sothing is wrong in faac pipe input

Are you talking about faac or flac?

buzzqw

24th June 2007, 19:12

ops.. misread... i mean faac not flac

BHH

jordisound

3rd July 2007, 16:07

Chumbo

3rd July 2007, 19:40

tebasuna51

3rd July 2007, 20:00

I've tried to convert an ac3 4 channels from an old DVD to 4 wavmono using behappy (installer 20070315). After process only appears FL, FR.
Using besweet i've got the 4 channels FL, FR, C, S (SL, SR are the same, and LFE is empty.)
So, behappy is not able to convert 4.0 to 4 wavmono? or have I to change the configuration?

The problem is in the ac3 decoder (NicAudio.dll) and is detected in this thread (http://forum.doom9.org/showthread.php?p=977363#post977363)

When the ac3 is 3/1 (FL, C, FR, S) NicAudio make an automatic downmix to:
3/1.0 -> stereo downmix
lt = 0,4xFL + 0,3xC + 0,3xS
rt = 0,4xFR + 0,3xC + 0,3xS

If the ac3 is 2/2 (FL, FR, SL, SR) NicAudio works ok.

- To decode properly an ac3 3/1.0, with free tools, you need use Azid.exe directly with:

azid.exe -d3/1 -ol,r,c,sl input.ac3 output.wav

The output.wav is a 4 channel (FL, FR, C, S) wav and you can split in mono wav's with WaveWizard or Wav2mono (1)

- Also you can use Foobar2000 to Convert to a WAVE_FORMAT_EXTENSIBLE wav or, if you use Wav2mono with Foobar, directly to 4 mono wav's (1)

(1) The wav's are named FL, FR, BL, BR and must be FL, FR, C, S

jordisound

3rd July 2007, 23:17

ok. i'll try and tell you if it works. thank you tebasuna.

edit:
it works.
The wav's are named FL, FR, BL, BR and must be FL, FR, C, S
exact! this is very important!

bjt

9th July 2007, 17:57

Where I can download the newest version of behappy ?

BeHappy Workspace （http://workspaces.gotdotnet.com/behappy ）has been phased-out.

tebasuna51

9th July 2007, 20:20

Where I can download the newest version of behappy ?

BeHappy Workspace （http://workspaces.gotdotnet.com/behappy ）has been phased-out.
Yes GotDotNet is out and the BeHappy author Dimzon is also out of this forum from long time.

The last 'official' release is from 2006-07-19, but there are mod's here (http://forum.doom9.org/showthread.php?p=998715#post998715) and here (http://forum.doom9.org/showthread.php?p=1003847#post1003847).

Search always in this thread.

BTW, I make a copy of GotDotNet info, if anybody need old sources ...

fight2win

15th July 2007, 07:47

Yes GotDotNet is out and the BeHappy author Dimzon is also out of this forum from long time.

The last 'official' release is from 2006-07-19, but there are mod's here (http://forum.doom9.org/showthread.php?p=998715#post998715) and here (http://forum.doom9.org/showthread.php?p=1003847#post1003847).

Search always in this thread.

BTW, I make a copy of GotDotNet info, if anybody need old sources ...

sir, mytempdir links are removed file,pls can u update the mytempdir links,please?

tebasuna51

16th July 2007, 02:27

sir, mytempdir links are removed file,pls can u update the mytempdir links,please?

Working links:

2007-03-24 Last Shon3i BeHappy package (http://www.box.net/shared/nkihizx1dh)

Last BeHappy mod (http://www.box.net/shared/5tyvsczuxo).

About enc_aacplus you can see this thread (http://forum.doom9.org/showthread.php?p=977901#post977901).

I think you have all the necessary.

fight2win

16th July 2007, 08:34

Working links:

2007-03-24 Last Shon3i BeHappy package (http://www.box.net/shared/nkihizx1dh)

Last BeHappy mod (http://www.box.net/shared/5tyvsczuxo).

About enc_aacplus you can see this thread (http://forum.doom9.org/showthread.php?p=977901#post977901).

I think you have all the necessary.

thanks

buzzqw

16th July 2007, 10:30

@tebasuna51

why not open a new thread , with the new links .. and ask for developers support?

BHH

tebasuna51

16th July 2007, 12:43

why not open a new thread , with the new links .. and ask for developers support?

Developers for what?

What is the TODO list for BeHappy?

BeHappy is only a GUI to write avs's and encoder parameters for inexpert users. We never can replace a expert user writing avs's and never can put all parameters (see Aften for instance) of a encoder without confuse the newbies.

Actually my only doubt about BeHappy is use the last enc_aacplus or not (see the thread mentioned (http://forum.doom9.org/showthread.php?p=977901#post977901)).

Other question is AviSynth Audio Encoding with more general issues, common with others tools to convert avs's to audio like BePipe, Wavi (http://forum.doom9.org/showthread.php?p=1019016#post1019016) or SoundOut (http://forum.doom9.org/showthread.php?t=120025).

I'm testing Bass libraries 2.3 with a new BassAudio.dll (http://www.mybigdir.com/1893) and there are some improvements about aac multichannel decoding.
To be continued...

tebasuna51

17th July 2007, 18:43

Decoding the aac multichannel (in .mp4 container), with Bass_aac 2.3.0.1 the channel mapping is ok. But ...

- Decoding .aac, instead .mp4, SoundOut and BeHappy/BePipe crash (Fatal stack overflow error with BePipe). Wavi work but there are some artifacts.

- Decoding .mp4, Wavi and SundOut work with a few artifacts but BeHappy/BePipe have a lot of artifacts.

- Also I get artifacts decoding two channel aac, Bass 2.2 works fine.

I can't recommend the use of Bass 2.3 for aac or ac3.

I can't see changes with Bass 2.2 decoding wav (WAVE_FORMAT_EXTENSIBLE), ogg, mp2, mp3 or wma then using Bass 2.2 at least aac stereo can be used.

fight2win

17th July 2007, 19:37

does bassaudio enhance ac3 to aac encoding in any way?

tebasuna51

17th July 2007, 20:10

does bassaudio enhance ac3 to aac encoding in any way?
Nope, you can use NicAc3Source like decoder and Nero or CT for encoding ac3 -> aac.

We are searching a dedicated multichannel aac AviSynth decoder to use instead the problematic DirectShowSource method.

Bluedeep

18th July 2007, 10:47

I'm trying to convert ac3 file on raw pcm.

It show this error:

Starting job VTS_01_1 T01 2_0ch 192Kbps DELAY -192ms.ac3->VTS_01_1 T01 2_0ch 192Kbps DELAY -192ms_0f4e7507bf9748eaaa7d22b8485d4cb8.dat
Error: BeHappy.AviSynthException: unexpected character ""
in BeHappy.AviSynthClip..ctor(String func, String arg, AviSynthColorspace forceColorspace, AviSynthScriptEnvironment env)
in BeHappy.Encoder.encode()

Any idea?

Thanks

EDIT: I've tried also Last BeHappy mod, but there is an windows error crash (I think is AvisynthWrapper.dll )

EDIT2: Oh my bad! I used Directshow and now works ;)

DiGiT@LON€

9th August 2007, 17:28

Hello, I'm italian and I'm trying to convert some audio tracks with BeHappy (I'm sorry about some mistakes for the language).
Unfortunately, I'm having problems.
Everytimes I try to convert (from some format to another one), an error occurs and the tool is closed by windows XP.
Here's a screenshot before the quit of the tool.
http://img128.imageshack.us/img128/5185/errorbehappybi8.th.jpg (http://img128.imageshack.us/my.php?image=errorbehappybi8.jpg)

I have 0.1.9.35190 version, with Avisynth 2.8 and Net Framework 2.0.
How can I resolve?
Thanks.

vlada

9th August 2007, 17:36

Avisynth 2.8? I don't think there is such a version. You probably meant 2.58? If it's so, I would try the latest stable version which is 2.57.

DiGiT@LON€

9th August 2007, 18:42

Avisynth 2.8? I don't think there is such a version. You probably meant 2.58? If it's so, I would try the latest stable version which is 2.57.I'm a totally careless. I installed 2.08 version.
But I have installed 2.57 version and the problem has occurred again.
What can I do?:(

DiGiT@LON€

9th August 2007, 22:54

I have reinstalled BeHappy and now it is running perfectly.

tebasuna51

13th August 2007, 21:40

A new release (http://www.mytempdir.com/2002112) of Behappy, for shon3i request.

Maybe shon3i can offer a new full release, but here are sources, last .exe's from BeHappy team and links to external free tools.

The changelog is:

2007-08-13 (Tebasuna)
+ New changes to support new enc_aacPlus.exe from Shon3i: select between Mp4Box and Mp4Mux. (CodingTechnologiesAAC.cs)
+ New FraunhoferMp3.extension to support free mp3sEncoder.exe from Fraunhofer (http://www.all4mp3.com/tools/sw_fhg_cl.html).
+ New wavSplit.extension and wavSplit.exe to replace wav2mono and wav2stereo
+ Only literals changes. (AssemblyInfo.cs, OggVorbisEncoder.cs)

The test are welcome.

ACrowley

24th August 2007, 10:29

tebasuna51

24th August 2007, 14:15

Is there any Way/ Plugin/Mod to output mono waves instead of mutichannel interleaved wave ?

I use NicsAC3Source and Timestretch on PAL AC3s and it would be nice to have 6 mono wave Output .
For now i use wavewizard to split interleaved to mono

In precedent Shon3i package there are Wav2mono and Wav2Stereo .exe and .extension.

In the new release in my precedent post there are WavSplit .exe and .extension to replace Wav2mono and Wav2Stereo.
You need only put the files in BeHappy folder, run BeHappy, select WavSplit in BeHappy (4) Destination, instead Wav Writer, and configure (...) mono/stereo.

ACrowley

24th August 2007, 21:28

THX

what is the difference to old wave2mono Output ? Outputs 6 mono waves too...

I found Wavesplit.extension ins \Behappy\src Folder, works fine..nice. Thats all i need :)

Behappy Default Wave output is 32Bit right ?
For AC3 i set it to "Convert Sample to 16bit"

Other Question :
Now i prefer NicAC3Source 1.92 over Azid, because Behappy/NicAudio is more Updated
Is there any Quality Difference in AC3 to Wave decoding ? I dont think so...

And with NicDTSSource compared with Tranzcode 0.4 ?

tebasuna51

25th August 2007, 04:39

what is the difference to old wave2mono Output ? Outputs 6 mono waves too...

None, only reduce the files needed.

Behappy Default Wave output is 32Bit right ?

Nope, now BeHappy don't touch the bitdepth (before the output is always 16 int) if the output encoder accept the bitdepth.

The 32 bit float come from NicAc3Source.

AviSynth can change also the bitdepth if any function don't support the actual.

For AC3 i set it to "Convert Sample to 16bit"

Never, the first thing Aften do is convert to 32 float then you have a double conversion 32 float -> 16 int -> 32 float.
Let the 32 float.

Now i prefer NicAC3Source 1.92 over Azid, because Behappy/NicAudio is more Updated
Is there any Quality Difference in AC3 to Wave decoding ? I dont think so...

Is updated but not in AC3 decoder. I think the quality is the same, only for special num_channels 3, 4, 5 Azid is still necesary.

And with NicDTSSource compared with Tranzcode 0.4 ?

I don't know very much about DTS.

ACrowley

25th August 2007, 09:21

Nope, now BeHappy don't touch the bitdepth (before the output is always 16 int) if the output encoder accept the bitdepth.
The 32 bit float come from NicAc3Source.
AviSynth can change also the bitdepth if any function don't support the actual.
Never, the first thing Aften do is convert to 32 float then you have a double conversion 32 float -> 16 int -> 32 float.
Let the 32 float.
AH, ok NicAC3 Decoder output 32bit..allright
I dont use Aften :) so its no double Conversion
I use Behappy/NicAC3Source only to decode AC3 to wave / Timestretch ( no Pitch Correction).
For AC3 encoding i use a only cert DD Encoder.
For example the cert DD Encoder from Sonic Scenarist accepts only 16bit Wave source.
I think 16bit Output from standard DVD AC3 is enough ? In my case its AC3 Audio from DVD which is 48khz 16bit.
However, NicsAC3Source outputs 32bit and maybe i should leave it to avoid the 16bit conversion

Interesting, what is Default Azids Bitdepth Output ?
And when i set it to 16bit in Besweet, is it "Conversion" too?

thx

tebasuna51

25th August 2007, 11:02

I think 16bit Output from standard DVD AC3 is enough ? In my case its AC3 Audio from DVD which is 48khz 16bit.

AC3 don't have bitdepth because the samples are in frequency domain instead time domain. Maybe the source is 16 bit but not the ac3.

The dynamic range in ac3 is equivalent to 24 bits depth.

Interesting, what is Default Azids Bitdepth Output ?
And when i set it to 16bit in Besweet, is it "Conversion" too?
The default is 16 bits but azid.exe have the parameter:
-F FILE_TYPE, --filetype=FILE_TYPE
----------------------------------

Default: wav

Selects the file type to generate. Possible values are:

o wav. Generates "normal" 16-bits wav.
o wav24. Generate 24-bit integer wav.
o wav32. Generate 32-bits integer wav
o wav_float. Generate 32-bits floating-point (IEEE) wavs.
o pcm. Generate 16-bit pcm (equal to wav, only without
the wav-header)
o pcm32. Generate 32-bit integer PCM
o pcm_float. Generate 32-bits floating-point (IEEE) PCM
output.

ACrowley

25th August 2007, 11:20

AC3 don't have bitdepth because the samples are in frequency domain instead time domain. Maybe the source is 16 bit but not the ac3.

Thats what i mean...more or less

So, What which Bitdepth do you recommend "generally" to decode AC3 to Wave ? I decode without DRC/DialoNorm

I think 32bit is good and you can make nothing wrong,right ? Especially for NicsAC3SOurce to avoid extra 32-16 bit Conversion

tebasuna51

25th August 2007, 13:25

So, What which Bitdepth do you recommend "generally" to decode AC3 to Wave ? I decode without DRC/DialoNorm

I think 32bit is good and you can make nothing wrong,right ? Especially for NicsAC3SOurce to avoid extra 32-16 bit Conversion

If your ac3 encoder support 32 bit float, or you need make some edit, of course is the best option.

But if your encoder only support 16 int ...

ACrowley

25th August 2007, 20:13

If your ac3 encoder support 32 bit float, or you need make some edit, of course is the best option.

But if your encoder only support 16 int ...

Yeah, i use Sony Vegas Dolby Digital Pro Encoder 7
It takes 32bit Input

cweb

4th September 2007, 12:30

The main behappy site is down ...

Nikos

4th September 2007, 13:41

Yeah, i use Sony Vegas Dolby Digital Pro Encoder 7
It takes 32bit Input

Yes the Sony Vegas accept 32 bits wavs input but i think it converts to 16 or 24 bits.

http://blue-whitegt.com/covers/vegas.jpg

tebasuna51

4th September 2007, 13:42

@cweb
Yes...
Use the last shon3i package and mods.
Read this post (http://forum.doom9.org/showthread.php?p=1023021#post1023021) and next ones.

cweb

4th September 2007, 13:48

@cweb
Yes...
Use the last shon3i package and mods.
Read this post (http://forum.doom9.org/showthread.php?p=1023021#post1023021) and next ones.
thanks

ACrowley

4th September 2007, 14:51

@Nikos

No, Vegas dont convert anything.
Because you simply drag and drop your 6 mono waves into it.
Theres no internal Wave Processing or somewaht in this Case
Only when you export/edit the waves ,the internal ProjectSetting is active
So Vegas dont touch the Waves. When you disable View-"Waveform and Frames" ,you can see theres absolutly no internal preprocessing/conversion.
And to display the Waveform Vegas simply read the Input. Theres no writing/Conversion too

32Bit Float Wave after drag&drop into the 5.1 Project :
http://img252.imageshack.us/img252/2396/sshot4nd0.th.png (http://img252.imageshack.us/my.php?image=sshot4nd0.png)
You can see its still 32Bit "before" you render it to AC3 .

@tebasuna51
You told me Behappy dont touch the Input wave Bitdepth ,right ?
But when i run a 16 bit (mono) wave i get 32bit Output from 16Bit Source by default ?
"Channels=1, BitsPerSample=32 float, SampleRate=48000Hz"

Mhh, not nice when i have to select a extra 16 bit Conversion to get my 16bit back from Behappy

tebasuna51

4th September 2007, 17:46

@tebasuna51
You told me Behappy dont touch the Input wave Bitdepth ,right ?
But when i run a 16 bit (mono) wave i get 32bit Output from 16Bit Source by default ?
"Channels=1, BitsPerSample=32 float, SampleRate=48000Hz"

Mhh, not nice when i have to select a extra 16 bit Conversion to get my 16bit back from Behappy

Actually Behappy only touch the bitdepth if the output encoder don't support it, never for uncompressed output.

From AviSynth docs:
"Starting from v2.5 the audio samples will be automatically converted if any filters requires a special type of sample. This means that most filters will accept several types of input, but if a filter doesn't support the type of sample it is given, it will automatically convert the samples to something it supports."

If you use Timestretch, also from AviSynth docs:
"SoundTouch is used in float sample mode."

And if you want 16 bit output a final ConvertAudioTo16bit() must be included explicitly.

ACrowley

4th September 2007, 20:18

AH ,Ok Thx

Nikos

4th September 2007, 21:30

Thanks ACrowley for the explanation about 32 bit wavs in Vegas.

bagge1

16th October 2007, 01:16

tebasuna51

16th October 2007, 10:14

The links are expired on mytempdir.com. Can somebody please upload Behappy again?

Here (http://www.mytempdir.com/2041184) is a new link to last BeHappy.

Only one change:
2007-10-16 (Tebasuna)
+ New RaWav.extension and AviSynth plugin RaWav.dll. Open files > 4 GB with uncompressed formats like:
wav (WAVE_FORMAT_EXTENSIBLE and others), bwf, raw, au, aif, w64, RF64, or caf (see RaWav_readme)

bagge1

16th October 2007, 10:57

Here (http://www.mytempdir.com/2041184) is a new link to last BeHappy.

Only one change:
2007-10-16 (Tebasuna)
+ New RaWav.extension and AviSynth plugin RaWav.dll. Open files > 4 GB with uncompressed formats like:
wav (WAVE_FORMAT_EXTENSIBLE and others), bwf, raw, au, aif, w64, RF64, or caf (see RaWav_readme)

Thank you, tebasuna51!

Chumbo

4th January 2008, 03:41

I think I found the problem with the stdout stuff. I've removed the link until I get it fixed. It may be tomorrow. Sorry.

Here is another BeHappy update.

Changes:
2008-01-03 (Chumbo)
+ Updated the DSP Move buttons to behave according to selection and position.
+ Added a multiple file feature for the new Aften to create ac3 files from multiple wav sources. I'm sure there's a better way to implement this, but I put it together quickly. Note that this feature is for Aften only. It will launch a command window when executed as I couldn't get the process object to work for some reason. So if anyone can check out the code and get the process stdout redirection working that would be great. That way the output status can be displayed within BeHappy like it normally does. Here are the rules:

When you select Multiple Mono File List,

only the Aften encoder is allowed
avisynth is not used
DSP is disabled along with all other avisynth features

The input file must be a plain text file with either .src, .txt or .lst extension
The input file format is to have each mono wav file on its own line and the file name must have the following indicators in the file name:
_fl = front left
_fr = front right
_c = front center
_lfe= LFE
_sl = surround left
_sr = surround right

See the included SampleMultiMonoFile.txt for an example input file (see below)

C:\Documents and Settings\userx\My Documents\For encoding tests\audio\Test_FL.wav
C:\Documents and Settings\userx\My Documents\For encoding tests\audio\Test_FR.wav
C:\Documents and Settings\userx\My Documents\For encoding tests\audio\Test_C.wav
C:\Documents and Settings\userx\My Documents\For encoding tests\audio\Test_SL.wav
C:\Documents and Settings\userx\My Documents\For encoding tests\audio\Test_SR.wav
C:\Documents and Settings\userx\My Documents\For encoding tests\audio\Test_LFE.wav

Note that the files do not have to be in any specific order as long as you have the correct indicator in the file name. You also do not have to have all 6 channels if you only need FL, FR, LFE for example, you only need to use those files and so on.

For some reason, I couldn't upload the file to mytempdir this time as it kept failing. I uploaded it to rapidshare which I hate because of the waiting crap they do. If someone else can grab this from rapidshare and see if you can put it on mytempdir and share that link, it would be appreciated.

DiGiT@LON€

4th January 2008, 11:10

shon3i

4th January 2008, 12:00

@Chumbo, nice stuff, you should enable to use all multichannel encoders like AAC, and aslo WAV.

Is it possible to make reverse process one multichannel wave to 6 waves?

tebasuna51

4th January 2008, 13:20

@Chumbo, nice stuff, you should enable to use all multichannel encoders like AAC, and aslo WAV.

I can't understand this new stuff.

For what use BeHappy to encode six monowavs to ac3/aac out off AviSynth environment? There are others GUI's to do this.

You always can use BeHappy to do this inside AviSynth with DSP's actives and with any output: ac3, aac, ogg, wav
Instead your .src, .txt or .lst file you only need an .avs file like:

fl = WavSource("X:\Path\Test_FL.wav")
fr = WavSource("X:\Path\Test_FR.wav")
fc = WavSource("X:\Path\Test_C.wav")
lf = WavSource("X:\Path\Test_LFE.wav")
sl = WavSource("X:\Path\Test_SL.wav")
sr = WavSource("X:\Path\Test_SR.wav")
MergeChannels(fl, fr, fc, lf, sl, sr)

And open this .avs in BeHappy with all AviSynth DSP's and features available: delay, timestretch, resample, ...

Maybe we can construct a utility to write this .avs with a graphical interface like BeLight or Wisodev WavtoAc3, but the Chumbo mod I think is out off the BeHappy scope.

Is it possible to make reverse process one multichannel wave to 6 waves?

I don't understand your request. WavSplit can output six monowavs or three stereo wav's from a multichannel input.

tebasuna51

4th January 2008, 14:18

Wrong File ID. Can somebody upload it?
Thanks.

BeHappy_20071016 (http://files.filefront.com/BeHappy+200710167z/;9377564;/fileinfo.html)

In the second post (http://forum.doom9.org/showthread.php?p=758076#post758076) of this thread I try to maintain the last useful links.

DiGiT@LON€

4th January 2008, 16:19

BeHappy_20071016 (http://files.filefront.com/BeHappy+200710167z/;9377564;/fileinfo.html)

In the second post (http://forum.doom9.org/showthread.php?p=758076#post758076) of this thread I try to maintain the last useful links.
Thank you.

Chumbo

4th January 2008, 16:40

...but the Chumbo mod I think is out off the BeHappy scope.
...
Believe it or not, I agree with you regarding the scope. I was just tinkering around to see if it can be done, i.e., without avs and using aften directly. I had some down time and this is what I wasted it on. ;) I may scrap this altogether, but I wanted to get the input/output redirection working just so I learn how that stuff works too.

Personally, I use WAV to AC3 Encoder when my source is multiple wav files. It's quick and I don't have to build a batch file. :)

shon3i

5th January 2008, 00:22

well tebasuna51, you right, i totaly forgot metods :).

Chumbo

6th January 2008, 22:54

Okay, here (http://www.filefactory.com/file/5de667/) is the link to my previous post (http://forum.doom9.org/showthread.php?p=1082618#post1082618).

Like tebasuna51 said, this feature is really something that's out of the scope of BeHappy, but I just wanted to tinker and see if it can be done. It's there if you want to use it, otherwise, it can be ignored.

Included is wisodev's 703 build of aften.

[EDIT]FYI, I just went to wisodev's site (http://win32builds.sourceforge.net/aften/index.html) and there's a newer 723 build available.

Chumbo

6th January 2008, 23:00

...fl = WavSource("X:\Path\Test_FL.wav")
fr = WavSource("X:\Path\Test_FR.wav")
fc = WavSource("X:\Path\Test_C.wav")
lf = WavSource("X:\Path\Test_LFE.wav")
sl = WavSource("X:\Path\Test_SL.wav")
sr = WavSource("X:\Path\Test_SR.wav")
MergeChannels(fl, fr, fc, lf, sl, sr)
...
Maybe we can construct a utility to write this .avs with a graphical interface like BeLight or Wisodev WavtoAc3...
tebasuna, what do you think about adding a gui button to BeHappy that brings up a new UI that allows the user to select the wav files, assign them to the channel and then it creates the avs script for them and puts it directly into BeHappy to encode? It would be a friendly way to have the user construct it without having to manually put it into an avs file.

tebasuna51

6th January 2008, 23:26

tebasuna, what do you think about adding a gui button to BeHappy that brings up a new UI that allows the user to select the wav files, assign them to the channel and then it creates the avs script for them and puts it directly into BeHappy to encode? It would be a friendly way to have the user construct it without having to manually put it into an avs file.

Can be helpful for many users yes.

dimzon

24th January 2008, 19:52

Hi!
I'm moving BeHappy to CodePlex.
http://www.codeplex.com/BeHappy
I will create separate project's for AvisynthWrapper, WinampAAC etc...

dimzon

24th January 2008, 20:08

WARNING 2 ALL BeHappy "moders"
Dear friends! I decide to continue BeHappy development.
Current CodePlex project (http://www.codeplex.com/BeHappy) contains latest official build (from 2006-07-19). So welcome to join this project and upload your improvements ;) Otherwise You must start Your own BeHappy fork with different project name (to avoid version mishmash).

If You want to participate post Your CodePlex nick's here, I will add You to this project.

2 tebasuna51 I trust You, so do You want to be BeHappy project coordinator?

PS. Sorry for my really crappy english, I have no practics for last 2 years :mad:

shon3i

24th January 2008, 22:03

I can't belive, Dimzon you are still alive :D, where are you been man. I miss you :D:D WELCOME BACK!!!!

tebasuna51

25th January 2008, 05:02

Dear friends! I decide to continue BeHappy development.
Hi, dimzon!
You are welcome back. You need explain us your 'holidays' :rolleyes:

If You want to participate post Your CodePlex nick's here, I will add You to this project.
2 tebasuna51 I trust You, so do You want to be BeHappy project coordinator?
Thanks for your trust, of course if can help to BeHappy project you can add my CodePlex nick (always tebasuna51).

And, I think, the project coordinator job is not so hard like you can see by the thread activity. Or you have new ideas about?
For me BeHappy is near perfect and only need some update when AviSynth/decoders/encoders change.

Actually the free soft audio development seems focused to decode new audio formats of HD/BD specs. Maybe (also for Megui) we need new AviSynth decoders for eac3, TrueHD, ...

And about 'mods' don't worry, I think the more important is only delete some code in AviSynthWrapper to take advantage of the new AviSinth 2.5.7 about 32 bit float output.

The last sources are in the second post of this thread. Do you want talk about these finish sources or one per one with all the 'moders'?

patul

25th January 2008, 05:50

OMG Dimzon, I am very glad you're back... where have you been? :D :D Nice to see you again, buddy...

You know, meanwhile you were off, tebasuna51, Chumbo and shon3i were doing very good job to add quite a lot of updates, kudos for them.

I agree with tebasuna51 that BeHappy is nearly perfect and my wish is just keep it as simple as it can be, it will help n00b like me a LOT...

I'm looking forward for updates :D :D

dimzon

25th January 2008, 15:01

Hi 2 all!
I was extremely busy in my real life. Unfortunately I still doesn't have much free time now too...
I'm very surprised (really, I was shocked after quick look at this thread after 2 years) for Your updates e.t.c. But Why You doesn't create project hosting on some service like CodePlex / SourceForge / GoogleCode? File exchange via FileFactory is a really ugly solution for project management...

now talking about "mods"
first of all I have commited latest (2007-10-16) BeHappy sources onto CodePlex. it doesn't include custom avisynth filters etc.
I think we must create different project for AviSynth filters (one for all or one project per filter) bcz they can be used without BeHappy itself.

Now about float processing into AviSynth wrapper ( http://www.codeplex.com/AvisynthWrapper ). I don't like current solution bcz it can broke compatibility. I think we must keep dimzon_avs_init as is and add new dimzon_avs_init_2 and use it. Don't take me wrong, I really want to avoid all possible Version/Fork Hell.

Now about binary releases. I think somebody can combine full package with installer and upload it onto CodePlex.

2 tebasuna51 - You are BeHappy/AvisynthWrapper coordinator now, welcome! Feel free to add new developers to this projects.

tebasuna51

25th January 2008, 21:23

first of all I have commited latest (2007-10-16) BeHappy sources onto CodePlex. it doesn't include custom avisynth filters etc.
Thanks, I see.

I think we must create different project for AviSynth filters (one for all or one project per filter) bcz they can be used without BeHappy itself.
Is true, there are not only for BeHappy, maybe:
- AviSynthWrapper (MeGUI, ...)
- enc_aacPlus (WinampAAC) (command-line or with Foobar, eac3to, ...)
- enc_AudX_CLI (command-line or with Foobar, eac3to, ...)
- Bepipe (command-line used to automate jobs)
- WavSplit (can work in comand-line or with Foobar)
- BassAudio (AviSynth input filter)
- RaWav (AviSynth input filter)

Now about float processing into AviSynth wrapper ( http://www.codeplex.com/AvisynthWrapper ). I don't like current solution bcz it can broke compatibility. I think we must keep dimzon_avs_init as is and add new dimzon_avs_init_2 and use it.
No problem, we can use dimzon_avs_init_2() in AvisynthWrapper.cs and the AvisynthWrapper.dll can remain unique

Now about binary releases. I think somebody can combine full package with installer and upload it onto CodePlex.
Yes, Shon3i was make this job until now.
I think we can make a release after the AvisynthWrapper change and enc_aacPlus solution (actually there are a little bug, but maybe there are also other problems).
Please check the enc_aacPlus thread (http://forum.doom9.org/showthread.php?p=977901#post977901) and open the WinampAAC project with a unified code.

You are BeHappy/AvisynthWrapper coordinator now, welcome! Feel free to add new developers to this projects.
Thanks, first I must learn how CodePlex work.

Chumbo

26th January 2008, 02:59

tebasuna51

26th January 2008, 14:13

Hey guys,
What's the best way to get this mod into the project? I think this is a useful UI "fix" for the DSP stuff. I had added it to the last test mod I did.

+ Updated the DSP Move buttons to behave according to selection and position.

I'll have to check how CodePlex works too.

First you need register in CodePlex, after you can say me the nick (if other than Chumbo) and I can add you to the project like developer.

BTW, I don't know what do your recent update (Updated the DSP Move buttons ...) for me work fine with old version.

Chumbo

26th January 2008, 15:13

First you need register in CodePlex, after you can say me the nick (if other than Chumbo) and I can add you to the project like developer.

BTW, I don't know what do your recent update (Updated the DSP Move buttons ...) for me work fine with old version.
Okay, sounds good. Yes it worked fine, but things like when you select the first item the Move Up shouldn't be active, likewise, when you select the last item the Move Down shouldn't be active. And if you have none selected, then none of the Move buttons should be active and so on. Just a way to keep the user from getting in trouble and more intuitive. :)

[EDIT]Okay, I registered over there using Chumbo.

tebasuna51

26th January 2008, 16:17

Okay, I registered over there using Chumbo.

Now you are BeHappy project developer.
At Source Code tab visit the CodePlex Client project

Chumbo

27th January 2008, 00:56

Now you are BeHappy project developer.
At Source Code tab visit the CodePlex Client project
Okay, I've checked out everything using TortoiseSVN and made the modifications. How do you want me to check it back in? You want me to commit the changes or branch it to a new repository? Or something else? I may have a few more questions until we solidfy developer protocol. Thanks much.

[EDIT] Okay, it was easy enough to figure out. The CodePlex Client will be easy enough to use, so I won't use the SVN stuff. I checked in the update and created a hidden maintenance mod under the Releases tab with a status of Planned. It only contains my latest .exe compiled release build. You can decide on making it an official release or not. I included the changes in the Description too.

shon3i

27th January 2008, 23:17

Yes, Shon3i was make this job until now.
I think we can make a release after the AvisynthWrapper change and enc_aacPlus solution (actually there are a little bug, but maybe there are also other problems).
Please check the enc_aacPlus thread and open the WinampAAC project with a unified code.i will continue to release packages, and aslo to update enc_aacplus. Currently i am busy until next month, when i will release new package and fixed enc_aacplus like you suggest

Chumbo

28th January 2008, 01:41

tebasuna,
When you have a minute. I moved my last build to the Planned stage. If you can check it out and then decide if you want to make it a release so when shon3i is ready the new package can include it. :) Thanks much.

tebasuna51

28th January 2008, 13:27

I wait until AviSynthWrapper and enc_aacplus are modified to do a new release. Thanks.

grennis

29th January 2008, 20:46

What happened to BePipe?

tebasuna51

1st February 2008, 13:54

@Dimzon
I know you are busy, I try to do the job I can:
- There are a new AviSynthWrapper release (init_2 and strncpy_s), BeHappy work like with the old_one (only 16 bit output) then don't be used yet until the mod to call init_2 (I'm over this).

But we need you about the enc_aacplus project and, maybe, with Bepipe project:
- The last bepipe binary have little bugs and can be improved but the sources in old GotDotNet are from a pre-alpha build (.NET v1.1). Do you have the last sources?

dimzon

1st February 2008, 15:35

@Dimzon
I know you are busy, I try to do the job I can:
- There are a new AviSynthWrapper release (init_2 and strncpy_s), BeHappy work like with the old_one (only 16 bit output) then don't be used yet until the mod to call init_2 (I'm over this).
fine

But we need you about the enc_aacplus project
It's easy, I will create it @ CodePlex and make You coordinator
http://www.codeplex.com/aacPlusCLI

The last bepipe binary have little bugs and can be improved but the sources in old GotDotNet are from a pre-alpha build (.NET v1.1). Do you have the last sources?
No. Actually BePipe was very easy application, it's possible to recreate it by striping down BeHappy ;) Does anybody really need BePipe?

tebasuna51

5th February 2008, 18:10

Well, I hope all sources and binaries needed for BeHappy can be obtained from:

BeHappy - Home (http://www.codeplex.com/BeHappy)
AvisynthWrapper - Home (http://www.codeplex.com/AvisynthWrapper)
aacPlusCLI - Home (http://www.codeplex.com/aacPlusCLI)

Shon3i and Alwa, like recent collaborators, are explicitly invited to work like developers but any suggestion are welcome.

@Chumbo, your mod are included (read my message in your Planned build)

Chumbo

5th February 2008, 19:44

alwa

6th February 2008, 17:46

Add me, name is alwa. Thanks. :)

shon3i

6th February 2008, 20:10

Add me, name is alwa. Thanks. :)
Me to, as shon3i :)

@tebasuna51, did you compile last enc_aacplus?

tebasuna51

6th February 2008, 20:49

Add me, name is alwa. Thanks. :)

Done

@shon3i, first you must register your name in CodePlex.

And yes, I make a compile from the actual source (many warnings about deprecated functions but seems work). There are any problem?
I'm not a expert C programmer and I need your help.

NorthPole

8th February 2008, 00:12

No. Actually BePipe was very easy application, it's possible to recreate it by striping down BeHappy ;) Does anybody really need BePipe?

Actually, I still use Bepipe...(Probably the only one). Although I did made a couple of minor, local mods to the code which I can't seem to remember right now. Works good for my needs.

Dimzon, Glad to have your back!

NorthPole

8th February 2008, 00:19

The last bepipe binary have little bugs and can be improved but the sources in old GotDotNet are from a pre-alpha build (.NET v1.1). Do you have the last sources?

@tebasuna51

I have the original code from November 2005 if you are interested. (Probably the same as your).

tebasuna51

8th February 2008, 03:21

I have the original code from November 2005 if you are interested. (Probably the same as your).

Yes from 2005-11-25, but I think is not the last.

Bepipe is useful for command line usage and I see some GUI's than use Bepipe also. BTW we can use also Wavi and SoundOut for avs script's.

NorthPole

15th February 2008, 20:43

Has anyone had success using the TTA encoder with BeHappy/Bepipe. http://tta.sourceforge.net/

I can't seem to get the pipe to work. I can get it to encode correctly using the command line with a wav file but not with the pipe.

Thanks.

NorthPole

15th February 2008, 22:26

Update:

Found a working patched version here http://etree.org/shnutils/shntool/

Seems to work OK with BeHappy.

tebasuna51

16th February 2008, 01:02

Found a working patched version here http://etree.org/shnutils/shntool/

Seems to work OK with BeHappy.

Yep, with this version and a ttaenc.extension file like:
<?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 Name="TTAenc 3.4.1" UniqueID="e482a270-dc16-11dc-95ff-0800200c9a66">
<ExecutableFileName>ttaenc.exe</ExecutableFileName>
<Script>32==Audiobits(last)?ConvertAudioTo24bit(last):last</Script>
<ExecutableArguments>-e -o "{0}" -</ExecutableArguments>
<SupportedFileExtension>tta</SupportedFileExtension>
</AudioEncoder>
</BeHappy.Extension>

we can support output lossless tta files.

NorthPole

16th February 2008, 22:07

@tebasuna51,

Thanks for the extension... saved me the work!

tebasuna51

16th April 2008, 03:16

New BeHappy 0.1.9.50201 (http://www.codeplex.com/BeHappy) release, include:

- NicAudio v2.0.1 (http://www.codeplex.com/NicAudio)
- BassAudio for Bass 2.4 libraries. Now seems work with multichannel aac and others formats.
- enc_aacPlus (http://www.codeplex.com/aacPlusCLI)
- TTA support and other minor changes.

Geleodor

16th April 2008, 10:33

BassAudio seems to make incorrect channel mapping, when I decode .ac3 to mono .wavs. I have just got FL, but it is central channel; BR, but it is LFE....and so on
NicAudio make it correctly...

tebasuna51

16th April 2008, 11:50

BassAudio seems to make incorrect channel mapping, when I decode .ac3 to mono .wavs. I have just got FL, but it is central channel; BR, but it is LFE....and so on
NicAudio make it correctly...

You are right, like you can read in the plugins/readme.txt I don't recommend bass_ac3 for ac3 decode because there are problems with channelmapping for some acmod-lfeon options.

Always use NicAudio v2 (with this acmod-lfeon options revised) for ac3, dts, mpeg, lpcm and other uncompresed formats.

For aac, ogg, wma, flac, and others you can use BassAudio.

mob

17th April 2008, 02:07

tebasuna51

17th April 2008, 04:15

Hey guys,

I got BeHappy from http://www.codeplex.com/BeHappy/ and it is very good!

So I have a question about how it is adding a switch to the command line of LAME.

When I set it up to do VBR 5 it give this command:

lame.exe -v -V 5 --nohist --vbr-new -S --silent - "C:\test.mp3"

So why is it -v -V 5? Instead it should be only -V 5 right?

And also, another question. In the LAME encoder configuration, what is that setting: comply as much as possible to ISO MPEG spec?

Thanks! :)
Yes, maybe we need some update :

-v -V 5 --vbr-new

actually is enough use only -V 5 because the default is --vbr-new and -v is superfluous.

'comply as much as possible to ISO MPEG spec'

do nothing because the corresponding parameter:
--strictly-enforce-ISO
is not included, BTW I don't know if anybody need this.

mob

17th April 2008, 06:04

Yes, maybe we need some update :

-v -V 5 --vbr-new

actually is enough use only -V 5 because the default is --vbr-new and -v is superfluous.

'comply as much as possible to ISO MPEG spec'

do nothing because the corresponding parameter:
--strictly-enforce-ISO
is not included, BTW I don't know if anybody need this.

Yeah I knew something had to be wrong because I always check the hydrogenaudio for the best settings for LAME: http://wiki.hydrogenaudio.org/index.php?title=LAME#Recommended_encoder_settings

and they never use that -v switch. Only -V as in -V 4 --vbr-new

Well, perhaps you can add a feature so we can use our own commandline switches. Sometimes I like to just use my own commandline switches without using the dialog with all the options. It gives more control if you let the user make his own commandline options. Can this be added maybe?

Geleodor

17th April 2008, 11:06

What settings should I use to encode multichannel AAC from multichannel.wav? Or multichannel AAC can be just as input, not output ?

tebasuna51

17th April 2008, 13:08

and they never use that -v switch. Only -V as in -V 4 --vbr-new
Don't worry about this. Using more parameters we guarantee compatibility with old and beta versions because the defaults can change. See that:
VBR options: 3.97
-v use variable bitrate (VBR) (--vbr-old)
--vbr-old use old variable bitrate (VBR) routine
--vbr-new use new variable bitrate (VBR) routine
-V n quality setting for VBR. default n=4
0=high quality,bigger files. 9=smaller files

VBR options: 3.98 beta 7
-V n quality setting for VBR. default n=4
0=high quality,bigger files. 9=smaller files
-v the same as -V 4
--vbr-old use old variable bitrate (VBR) routine
--vbr-new use new variable bitrate (VBR) routine (default)
The syntax:
-v -V 5 --vbr-new
work fine with 3.96, 3.97 and 3.98 beta Lame versions.

Well, perhaps you can add a feature so we can use our own commandline switches. Sometimes I like to just use my own commandline switches without using the dialog with all the options. It gives more control if you let the user make his own commandline options. Can this be added maybe?
What is your BeHappy version?
Only the bitrate, with the allowed values, are forced in the dialog.
There are a window to add your commandline switches, I always add at least -h.

tebasuna51

17th April 2008, 13:41

...multichannel AAC can be just as input, not output ?
Maybe a input if you select BassAudioSource (BassAudio.dll, Bass.dll and bass_aac.dll required in ...AviSynth 2.5\plugins folder) or DirectShowSource with some DS filter to decode aac well configured.

And can be a output if you have the encoders. BeHappy can work with:
- Nero Digitall AAC (NeroAacEnc.exe in BeHappy folder or encoder subfolder)
- Coding Technologies aacPlus (enc_aacPlus.exe, enc_aacplus.dll and nscrt.dll in BeHappy folder or encoder subfolder)

What settings should I use to encode multichannel AAC from multichannel.wav?

1) With NeroAacEnc I recommend you use:
- VariableBitRate with quality between 2.5 and 4 (is your choice)
- Aac Profile: Automatic
- The rest unchecked
The output is always in m4a/mp4 container.

2) With CT only CBR is allowed and for multichannel:
- Profile LC-AAC with bitrates between 224-320 Kb/s
- Or Profile HE-AAC for bitrates between 192-223 Kb/s
- Ignore the Channel Mode valid only for stereo.
- The rest unchecked
The output is .aac if you need m4a/mp4 container you need also Mp4Box.exe in BeHappy folder or encoder subfolder.

mob

17th April 2008, 22:38

Hey tebasuna51, try to test it out like I did and see how it happens:

I used any MP3 file for testing. I input the MP3 file to BeHappy and I want to transcode it with -V 2 --vbr-new so I set it up like this with the dialog:

http://img378.imageshack.us/img378/5493/95187853mk0.th.jpg (http://img378.imageshack.us/my.php?image=95187853mk0.jpg)

But then it sends the MP3 to LAME with this command instead: -v -V 2 --nohist --vbr-new -S --silent

Now, you said this going to be working fine for all the versions of LAME, but I find that this is not true.

I am using LAME 3.97 from: http://rarewares.org/dancer/dancer.php?f=lame-current

And when I pass my test MP3 file to the LAME commandline encoder (without using BeHappy) then I get a totally different file. So BeHappy is adding unwanted command options and it's causing the encode to be different. Take a look at my findings:

_____________________________________

test.mp3 -> BeHappy -> output.mp3 with these settings:

http://img378.imageshack.us/img378/5493/95187853mk0.th.jpg (http://img378.imageshack.us/my.php?image=95187853mk0.jpg)

Gives me: output.mp3 (5.94 MB) CRC: d4502ba8

_____________________________________

But if I only use LAME encoder in the commandline:

test.mp3 -> LAME.exe -> output.mp3 with these settings:

-V 2 --vbr-new

Gives me: output.mp3 (6.21 MB) CRC: d4238540

_____________________________________

So you can see, using BeHappy is not accurate because you are adding switches that are confusing it.

That is why I need an advanced option so I can disable all the things that BeHappy is adding to the commandline and then I'll use my own commandline switches.

btw, I'm using the same version/build of LAME for BeHappy in my test. So the output should come out as the same file. But it is different as you can see. Try to do the test also and see how it happens. This is why you should allow the people to make their own switches. If you add switches to their commands then it will mess up the output :(

tebasuna51

18th April 2008, 01:20

Believe me! :)
The parameters are ok.

Your test is not correct, you can't compare this:

1.mp3 -> Lame Decoder -> Lame Encoder -> 11.mp3

with:

1.mp3 -> NicMPG123/BassAudio decoder (32 float) -> 32 int conversion -> Lame Encoder -> 12.mp3

When BeHappy decode a file use the max precision (32 bits float here) to allow operations like stretch, resample, ... without lose quality.

Unfortunately Lame don't accept 32 bits float and we need convert to 32 int before send the data to Lame.

Like you see the process is different. But you can do the test using a uncompressed wav file. I get a wav file (ripped from CD) and I make the test (with -V 2):

1.wav -> Lame.exe -> 11.mp3 4.196.754 bytes
1.wav -> BeHappy -> 12.mp3 4.196.754 bytes

And bitidentical (Total Commander) of course.
Make the test yourself. :rolleyes:

mob

18th April 2008, 01:58

LOL tebasuna51 :)

So maybe I don't know much about all of this stuff. But if you say it is good then I believe you because you know a lot more about this kind of stuff then me :)

Anyway, see my whole problem is that I am going to convert a PAL mp3 to a NTSC mp3 and I found BeHappy to be the best one to do the job.

I need to do this PAL -> NTSC conversion because the PAL audio is messed up with a high pitch sound. I need to get it back to 23.976 and get it back to the original sound.

So I do a timestretch with BeHappy and I select the option Rate for the Rate Control:

http://img443.imageshack.us/img443/8636/sdf1ie2.th.jpg (http://img443.imageshack.us/my.php?image=sdf1ie2.jpg)

This gives me the exact kind of conversion that I need. And BeHappy really makes me Happy :)

Except now I found this is what they say about Timestretch on the AviSynth website:

Timestretch() by the nature of the algorithm used causes noticeable distortion in the result, generally use it for speech only, never for high quality music.

So is it true? Is it really causing "noticeable distortion" and bad artifacts? Maybe there is a method of doing the conversion that will not cause noticeable distortion?

tebasuna51

18th April 2008, 03:19

Anyway, see my whole problem is that I am going to convert a PAL mp3 to a NTSC mp3 and I found BeHappy to be the best one to do the job.

I need to do this PAL -> NTSC conversion because the PAL audio is messed up with a high pitch sound. I need to get it back to 23.976 and get it back to the original sound.

So I do a timestretch with BeHappy and I select the option Rate for the Rate Control

Please don't use with audio:
"convert a PAL mp3 to a NTSC mp3"
because the audio is the same in PAL or NTSC systems.

The correct way to do a PAL -> NTSC conversion is modify the video and let untouched the audio.

Seems you have a 25 fps video and want play this video at 23.976, without conversion the duration is greater and you need enlarge also the audio to match the new video duration. Like the initial audio have high pitch (maybe by the inverse conversion) the desired method is Rate.

Then you are lucky because this method don't produce "noticeable distortion or bad artifacts". Is a simple change of samplerate.

Only when we need change the duration preserving the pitch (or change the pitch without change the duration) the conversion can't be perfect.

mob

18th April 2008, 03:35

Seems you have a 25 fps video and want play this video at 23.976
Yep that's exactly what I have to do. I already got the video from 25fps -> 23.976. My only trouble now is to get the audio to match it.

And the only reason I need to do it is because when they made the PAL version of my DVD, they speed up the audio and it sounds too high pitch. If I slow it back down to the normal 23.976fps then it will sound normal again. They make the PAL DVDs so bad like this :(

Like the initial audio have high pitch (maybe by the inverse conversion) the desired method is Rate.

Then you are lucky because this method don't produce "noticeable distortion or bad artifacts". Is a simple change of samplerate.

Yep! I used the Rate method like you see in this screenshot:

http://img443.imageshack.us/img443/8636/sdf1ie2.th.jpg (http://img443.imageshack.us/my.php?image=sdf1ie2.jpg)

So you mean, with this method I can be safe because it doesn't have the distortion problem?

If yes, than that's good. And it fixes all my problems :)

tebasuna51

18th April 2008, 04:20

So you mean, with this method I can be safe because it doesn't have the distortion problem?

Yep, the process is change the samplerate from 48000 Hz to 50050 Hz then there are more samples, and after play the audio at 48000 Hz then the duration grow and the pitch decrease.

mob

18th April 2008, 07:21

Yep, the process is change the samplerate from 48000 Hz to 50050 Hz then there are more samples, and after play the audio at 48000 Hz then the duration grow and the pitch decrease.

WOW :)

This is perfect!

BeHappy really makes me Happy hehe

...but tebasuna51 makes me smart. :)

Thanks for helping me and explaining everything to me. I really appreciate it man :)

:thanks:

adrianmak

29th April 2008, 14:52

the gui looks complex.

How do I use it to convert a 5.1 ac3 to 6 individual wave file ?

tebasuna51

29th April 2008, 20:32

How do I use it to convert a 5.1 ac3 to 6 individual wave file ?
System requirements
* Microsoft .NET Framework Version 2.0
* Avisynth v2.57

Install
1) From the BeHappy package uncompress the following files/subfolders in a folder at your choice:
AvisynthWrapper.dll
BeHappy.exe
encoder
extensions
plugins

2) Copy nicaudio.dll from plugins subfolder to your AviSynth 2.5/plugins subfolder.

Process
Run BeHappy.exe and in New Job Tab:

1) Select the [1] Source method NicAc3Source and [...] Configure. I suggest you don't use DRC.
Select [...] your source ac3 file.

2) Select the desired [2] Tweak functions.
Maybe you need include a Delay detected in your source.

3) Select the desired [3] Digital Signal Processing functions. Each function can be Configured and Moved Up/Down because are executed in descending order.
The NicAudio decoder output have 32 bit float samples. Maybe you want Convert Sample To 16 bit (or 24 int ...)

4) Select the [4] Destination format WavSplit and [...] Configure like Mono Wav's.
Select [...] your desired output filename. This filename is used like prefix and each channel is suffixed with _FL, _FR, ... and so on.

5) Enqueue. Go to Queue Tab and Start the job.

cweb

30th April 2008, 12:19

When I quite BeHappy v0.19.50201 I get an error "Unhandled exception has occurred in your application. If you click Continue the application will ignore this error and attempt to continue. If you click Quit, the application will close immediately.

Item has already been added. Key in dictionary:
'58ab9132-50c8-11dc-8314-0800200c9a66' Key being added:
'58ab9132-50c8-11dc-8314-0800200c9a66'
"

Any ideas on what could be wrong?

tebasuna51

30th April 2008, 14:29

Item has already been added. Key in dictionary:
'58ab9132-50c8-11dc-8314-0800200c9a66' Key being added:
'58ab9132-50c8-11dc-8314-0800200c9a66'
"

Any ideas on what could be wrong?
Seems the NicAudio ID (see Nicaudio.extension) is duplicated in your machine:
<AudioSource UniqueID="58ab9132-50c8-11dc-8314-0800200c9a66">

If you don't know what is the problem try edit Nicaudio.extension and change the line by other one, for instance:
<AudioSource UniqueID="58ab9132-50c8-11dc-8314-0800200c9b66">

tebasuna51

30th April 2008, 19:08

EpheMeroN

5th May 2008, 06:52

BeHappy v0.1.9.50201 is reporting the following error when trying to do AC3 > mp3 transcode:

Starting job VTS_01_1 T80 3_2ch 384Kbps DELAY 0ms.ac3->dvd-xvid-audio.mp3
Error: BeHappy.AviSynthException: Script error: there is no function named "NicAc3Source"
at BeHappy.AviSynthClip..ctor(String func, String arg, AviSynthColorspace forceColorspace, AviSynthScriptEnvironment env)
at BeHappy.Encoder.encode()

I have the latest NicAudio.dll in both BeHappy\Plugins dir and AviSynth\Plugins dir. What could be causing this?

cweb

5th May 2008, 09:52

@cweb

Maybe you have the old RaWav.extension in BeHappy folder.
Now is inside NicAudio.extension (same ID), and you need delete RaWav.extension and also RaWav.dll in AviSynth 2.5\plugins folder if exist.
For the time being I have downgraded to the previous version.

But thanks, I will reinstall the latest version and try that.

tebasuna51

5th May 2008, 10:41

BeHappy v0.1.9.59201 is reporting the following error when trying to do AC3 > mp3 transcode:
Maybe BeHappy v0.1.9.50201?

I have the latest NicAudio.dll in both BeHappy\Plugins dir and AviSynth\Plugins dir. What could be causing this?

Check your registry. A correct AviSynth install make the keys (translate to your language):
[HKEY_LOCAL_MACHINE\SOFTWARE\AviSynth]
@="C:\\Archivos de programa\\AviSynth 2.5"
"plugindir2_5"="C:\\Archivos de programa\\AviSynth 2.5\\plugins"

NicAudio.dll must be at "plugindir2_5" folder

EpheMeroN

5th May 2008, 11:00

@tebasuna51: You were correct about the version number. I corrected it on my previous post!

Does BeHappy look for the NicAudio.dll in just the default install location? Because I have all my a/v apps installed to "C:\Multimedia" so AviSynth is in "C:\Multimedia\AviSynth 2.5\Plugins". Every other program I have that uses AviSynth functions fine.

tebasuna51

5th May 2008, 14:16

Does BeHappy look for the NicAudio.dll in just the default install location? Because I have all my a/v apps installed to "C:\Multimedia" so AviSynth is in "C:\Multimedia\AviSynth 2.5\Plugins". Every other program I have that uses AviSynth functions fine.

Well, BeHappy call AviSynth, and AviSynth load the plugins in the folder indicated by the "plugindir2_5" registry value.

If, in your machine, is "C:\Multimedia\AviSynth 2.5\Plugins", then NicAudio.dll, BassAudio.dll, bass_*.dll, sox.dll ... must be there.

EpheMeroN

5th May 2008, 21:26

@tebasuna51: All the files are in the plugins directory so I have no clue why BeHappy is giving me that error.

Anyone have ideas?

tebasuna51

5th May 2008, 23:17

EpheMeroN

6th May 2008, 00:19

@EpheMeroN
To discard others problems try this:

1) Instead 'Enqueue' you job, use 'Export AviSynth Script'

2) Edit the .avs created with Notepad and insert like first line:
LoadPlugin("C:\Multimedia\AviSynth 2.5\Plugins\NicAudio.dll")
Save the file.

3) Open the .avs file with BeHappy instead your source. You don't need add DSP functions, only encode with Lame.
Starting job audio-tst.avs->audio-tst.mp3
Found Audio Stream
Channels=6, BitsPerSample=32 int, SampleRate=48000Hz
encoder\lame.exe --abr 128 --nohist -h -S --silent - "C:\Users\xxxxxxx\Desktop\audio-tst.mp3"
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Error: System.IO.IOException: The pipe has been ended.

at System.IO.__Error.WinIOError(Int32 errorCode, String maybeFullPath)
at System.IO.FileStream.WriteCore(Byte[] buffer, Int32 offset, Int32 count)
at System.IO.FileStream.Write(Byte[] array, Int32 offset, Int32 count)
at BeHappy.Encoder.encode()

tebasuna51

6th May 2008, 02:03

Ok. One problem solved.
The actual problem is evident.
You try encode 'Channels=6' with Lame.
Lame only support mono/stereo.
You need select a downmix DSP function, I recommend you use DPL II

EpheMeroN

6th May 2008, 04:39

Ok. One problem solved.
The actual problem is evident.
You try encode 'Channels=6' with Lame.
Lame only support mono/stereo.
You need select a downmix DSP function, I recommend you use DPL II
Okay. Doing that worked just fine!
Starting job test.avs->test.mp3
Found Audio Stream
Channels=2, BitsPerSample=32 int, SampleRate=48000Hz
encoder\lame.exe --abr 128 --nohist -h -S --silent - "C:\Users\xxxxxxx\Desktop\test.mp3"
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Finalizing encoder

So based on what we've figured out, BeHappy looks (by default) for the NicAudio.dll in the "C:\Program Files\AviSynth 2.5\Plugins" directory regardless of where you really have AviSynth installed. Could this be manually corrected in that NicAudio.extension file? Or what could be done?

Btw, for the purpose of diagnosing this issue, I have been using the latest BeHappy (stated in previous post), and using Windows Vista Home Premium as my operating system.

tebasuna51

6th May 2008, 11:08

So based on what we've figured out, BeHappy looks (by default) for the NicAudio.dll in the "C:\Program Files\AviSynth 2.5\Plugins" directory regardless of where you really have AviSynth installed.
BeHappy don't load AviSynth plugins at all.
I think is a AviSynth issue.
Could this be manually corrected in that NicAudio.extension file? Or what could be done?
Yep, you can patch NicAudio.extension changing each line:
<Value></Value>
For instance:
<Value>NicAc3Source("{0}")</Value>
must be
<Value>LoadPlugin("C:\Multimedia\AviSynth 2.5\Plugins\NicAudio.dll")
NicAc3Source("{0}")</Value>

Btw, for the purpose of diagnosing this issue, I have been using the latest BeHappy (stated in previous post), and using Windows Vista Home Premium as my operating system.
And can you confirm your registry is correct?
Must be like this:
[HKEY_LOCAL_MACHINE\SOFTWARE\AviSynth]
@="C:\\Multimedia\\AviSynth 2.5"
"plugindir2_5"="C:\\Multimedia\\AviSynth 2.5\\plugins"

EpheMeroN

6th May 2008, 19:42

And can you confirm your registry is correct?
Must be like this:
[HKEY_LOCAL_MACHINE\SOFTWARE\AviSynth]
@="C:\\Multimedia\\AviSynth 2.5"
"plugindir2_5"="C:\\Multimedia\\AviSynth 2.5\\plugins"
Confirmed!
http://img225.imageshack.us/img225/714/90001324wu4.jpg

tebasuna51

7th May 2008, 00:38

Confirmed!

Thanks. Then I don't know what is the problem.

EpheMeroN

7th May 2008, 01:45

I don't either! Maybe the author of BeHappy can take a look.

EpheMeroN

7th May 2008, 08:56

1) More options for DRC from NicAudio.dll please? So instead of just DRC you can choose Light, Normal, Heavy DRC in the option box.

I assume the default DRC option uses Normal DRC correct?

2) My previous issue (last 2 pages within this thread) with not being able to load NicAudio.dll without first having to save as .avs and then manually type in plugin location.

tebasuna51

7th May 2008, 13:58

1) More options for DRC from NicAudio.dll please? So instead of just DRC you can choose Light, Normal, Heavy DRC in the option box.

I assume the default DRC option uses Normal DRC correct?
Inside the ac3 stream can have a unique attenuation value for each block (256 samples) of audio data. Some decoders offer the option of modify this internal value by half or double (Azid, PowerDVD, ...). Other decoders can ignore the internal DRC value (Ac3Filter) and offer drc methods with continuous adjust.

The liba52 decoder offer the option to write external routines to manage DRC but the spirit of the AviSynth decoders I think must be:
- The decoder just decode the internal format to uncompressed audio data with the minimum process.
- Any data transformation must be done out of decoders to be generic. For instance liba52 have downmix functions, but I think this process must be done with more flexible AviSynth functions and independent of source format.

Then the object of DRC option in ac3 (and dts) decoder is recover the unique value present in ac3 stream recommended by the source author.

If you want some other distinct adjust you can write your filters using the compand function of sox filter. You have some samples in this post (http://forum.doom9.org/showthread.php?p=779165#post779165)

2) My previous issue (last 2 pages within this thread) with not being able to load NicAudio.dll without first having to save as .avs and then manually type in plugin location.

To know more about this problem, I suggest another test with bepipe:

- In last BeHappy_r50201.7z release there are a encoder\Bepipe.7z, please decompress the file and you have a bepipe folder with:
Bepipe.bat
Bepipe.exe
WavFix.exe
Edit with Notepad the Bepipe.bat:
@echo off
rem Sample of use to write correct wav header with Bepipe (if < 4GB). Not neded with encoders.
Bepipe --script "BassAudioSource(^D:\Musica\Ejemplos\High.wma^)" | Wavfix - output.wav
pause
and replace "BassAudioSource" with "NicAc3Source", and the example wma file with an ac3 in your system (let the necessary ^)

Run (Double Click) Bepipe.bat and let me know the result.

Chumbo

7th May 2008, 16:05

I don't either! Maybe the author of BeHappy can take a look.
Like tebasuna said, it's not a BeHappy issue. What you may consider doing is getting the latest install package from Shon3i on the 2nd post here (http://forum.doom9.org/showthread.php?p=758076#post758076) and then update the latest BeHappy and plug-ins after that. The install package has everything including avisynth. I'd remove your existing avisynth and BeHappy installations first and then install the Shon3i package.

EpheMeroN

8th May 2008, 19:14

Like tebasuna said, it's not a BeHappy issue. What you may consider doing is getting the latest install package from Shon3i on the 2nd post here (http://forum.doom9.org/showthread.php?p=758076#post758076) and then update the latest BeHappy and plug-ins after that. The install package has everything including avisynth. I'd remove your existing avisynth and BeHappy installations first and then install the Shon3i package.
Is there something different in the older Shon3i package than w/ the official latest BeHappy that might make a difference?

tebasuna51

9th May 2008, 01:01

Is there something different in the older Shon3i package than w/ the official latest BeHappy that might make a difference?

Is a installer (also AviSynth) to be sure. Of course with a official installation all must work, like in many others PC's, but here the interest is know for what don't work with your custom install.

You say:
"Every other program I have that uses AviSynth functions fine."
Please, what others programs?
You need load plugins?
And the Bepipe test I suggest?

masscamp24

9th May 2008, 20:21

Can Behappy transcode 2 stereo wav to a 5.1 ac3 stereo

tebasuna51

9th May 2008, 20:52

Can Behappy transcode 2 stereo wav to a 5.1 ac3 stereo

Please clarify your question.
'2 stereo wav' can be 2 wav files with 2 channel each, but what is '5.1 ac3 stereo'?

masscamp24

9th May 2008, 21:45

Please clarify your question.
'2 stereo wav' can be 2 wav files with 2 channel each, but what is '5.1 ac3 stereo'

I mean wav 2 channel to ac3 5.1 channel

tebasuna51

10th May 2008, 01:01

I mean wav 2 channel to ac3 5.1 channel

Of course, there are 6 profiles to choose in 5.1 Upmix DSP function.

jangai

11th May 2008, 20:17

With BeHappy to convert an AAC 6 ch file to AC3 5.1 with following script :
########################################
#Created by BeHappy v0.1.9.50201
#Creation timestamp: 11/05/2008 18:30:53
########################################
#Source FileName:C:\Documents and Settings\Jangai\Mes documents\Mes videos\Audio.aac
#Target FileName:C:\Documents and Settings\Jangai\Mes documents\Mes videos\Audio.ac3
########################################

########################################
# [Source: AviSynth]
########################################
Import("C:\Documents and Settings\Jangai\Mes documents\Mes videos\Audio.aac")

########################################
# [BeHappy: Delay Audio by -10 ms ]
DelayAudio( -10.0/1000.0)

########################################
# [Encoder: Aften AC3 CBR @ 448 kbps, L+R, ReadToEof, DRC: None]
########################################

I get following error at the beginning of process :

Starting job Audio.aac->Audio.ac3
Error: BeHappy.AviSynthException: unexpected character "ÿ"
à BeHappy.AviSynthClip..ctor(String func, String arg, AviSynthColorspace forceColorspace, AviSynthScriptEnvironment env)
à BeHappy.Encoder.encode()

Same error with ffmpeg AC3 encoder...

Anybody has a small idea to steer my searches ?
Thanks in advance :thanks:

tebasuna51

11th May 2008, 23:55

With BeHappy to convert an AAC 6 ch file to AC3 5.1 with following script :
...
# [Source: AviSynth]
########################################
Import("C:\Documents and Settings\Jangai\Mes documents\Mes videos\Audio.aac")
...

You must select the appropriate method (decoder) to open your source file.
The AviSynth method Import() is for .avs files
To decode aac you need BassAudio() method.

When you select an .aac like input file the BassAudio method is automatically selected.

jangai

12th May 2008, 10:24

You must select the appropriate method (decoder) to open your source file.
The AviSynth method Import() is for .avs files
To decode aac you need BassAudio() method.

When you select an .aac like input file the BassAudio method is automatically selected.
OK ! I was far of the right way...
Unfortunately, BassAudio method seems hard to activate for me :

Command panel with (good ?) input method :
http://i41.servimg.com/u/f41/11/71/09/36/behapp11.jpg (http://www.servimg.com/image_preview.php?i=59&u=11710936)

Preview:
http://i41.servimg.com/u/f41/11/71/09/36/behapp13.jpg (http://www.servimg.com/image_preview.php?i=61&u=11710936)

Log :
http://i41.servimg.com/u/f41/11/71/09/36/behapp12.jpg (http://www.servimg.com/image_preview.php?i=60&u=11710936)

Checked my BassAudio installation into AviSynth/plugin folder...
All seems right... But... is not ! :-)
A missing DLL somewhere ?

tebasuna51

12th May 2008, 10:54

OK ! I was far of the right way...
Unfortunately, BassAudio method seems hard to activate for me :

...
Checked my BassAudio installation into AviSynth/plugin folder...
All seems right... But... is not ! :-)
A missing DLL somewhere ?

1) Check your registry. A correct AviSynth install make the keys (translate to your language):
[HKEY_LOCAL_MACHINE\SOFTWARE\AviSynth]
@="C:\\Archivos de programa\\AviSynth 2.5"
"plugindir2_5"="C:\\Archivos de programa\\AviSynth 2.5\\plugins"

2) Check the needed .dll at "plugindir2_5" folder:
BassAudio.dll
Bass.dll
Bass_aac.dll

3) If don't work, you can patch (Notepad) BassAudio.extension replacing the black part with your "plugindir2_5" folder:
<?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="BassAudio" UniqueID="368e0760-ae08-11da-a746-0800200c9a66">
<Script>LoadPlugin("C:\Archivos de programa\AviSynth 2.5\Plugins\BassAudio.dll")
bassAudioSource("{0}")</Script>
<SupportedFileExtension>aac</SupportedFileExtension>
<SupportedFileExtension>aiff</SupportedFileExtension>
<SupportedFileExtension>ape</SupportedFileExtension>
<SupportedFileExtension>cda</SupportedFileExtension>
<SupportedFileExtension>flac</SupportedFileExtension>
<SupportedFileExtension>m4a</SupportedFileExtension>
<SupportedFileExtension>mp1</SupportedFileExtension>
<SupportedFileExtension>mp2</SupportedFileExtension>
<SupportedFileExtension>mp3</SupportedFileExtension>
<SupportedFileExtension>mp4</SupportedFileExtension>
<SupportedFileExtension>mpc</SupportedFileExtension>
<SupportedFileExtension>ogg</SupportedFileExtension>
<SupportedFileExtension>tta</SupportedFileExtension>
<SupportedFileExtension>wav</SupportedFileExtension>
<SupportedFileExtension>wma</SupportedFileExtension>
<SupportedFileExtension>wv</SupportedFileExtension>
</AudioSource>
</BeHappy.Extension>

jangai

12th May 2008, 15:12

Many thanks... I had to follow the 1-2-3 parts of your advices.

-Create a "plugindir2_5" folder and fill it with right dll collection :
http://i41.servimg.com/u/f41/11/71/09/36/behapp14.jpg (http://www.servimg.com/image_preview.php?i=62&u=11710936)

-Check registry to see plugindir2_5 folder access key was right (it was) :http://i41.servimg.com/u/f41/11/71/09/36/behapp15.jpg (http://www.servimg.com/image_preview.php?i=63&u=11710936)

-Patch the xml Behappy BassAudio.extension file with correct order and path :http://i41.servimg.com/u/f41/11/71/09/36/behapp16.jpg (http://www.servimg.com/image_preview.php?i=64&u=11710936)

And : it would have been working... No ! Arrgggggghhhhhhh! I check and check again my syntaxes... The DLL doesn't load ...
http://i41.servimg.com/u/f41/11/71/09/36/behapp17.jpg (http://www.servimg.com/image_preview.php?i=65&u=11710936)

Is it a bad DLL release or a so big mistake that I can see it ???

tebasuna51

12th May 2008, 21:11

Must be
C:\Program Files\AviSynth 2.5\plugins

BassAudio.dll 69.632 14/03/2008 09:08
bass.dll 97.336 08/04/2008 15:07
bass_aac.dll 150.904 27/02/2008 17:10
bass_ac3.dll 16.736 27/02/2008 17:10
bass_alac.dll 12.144 27/02/2008 20:03
bass_ape.dll 33.112 27/02/2008 17:10
bass_cda.dll 16.952 02/04/2008 12:28
bass_flac.dll 25.152 02/04/2008 12:26
bass_midi.dll 25.152 24/03/2008 15:07
bass_mpc.dll 16.200 27/02/2008 21:46
bass_ofr.dll 5.960 27/02/2008 17:11
bass_spx.dll 47.936 27/02/2008 18:25
bass_tta.dll 8.536 27/02/2008 21:49
bass_wma.dll 15.416 02/04/2008 12:32
bass_wv.dll 27.704 02/04/2008 12:36
The 'plugindir2_5' is the name in regedit.
And check the dates, seems you have v2.3 bass files.

jangai

12th May 2008, 23:54

And check the dates, seems you have v2.3 bass files.

OK I understand my mistakes, but I have also a release problem...
For Bass files, I have only BassAudio.dll from 2008 and bass.dll, bass_aac.dll, bass_xxx.dll are all from 2007...

Now, I think file linkage is good, but I have serious releases mismatch into all BassAudio DLLs from my AviSynth plugins folder...

My sources for these files (BeHappy, BassAudio and updates) are :
-BeHappy_20070324.exe for main package and
-BassAudio23.7z

My question is : Are theses references OK and if not, what are the good ones and where to find the good ones ?

Thank you for your patience, but the actual "packaging" of BeHappy with BassAudio extensions is not a very intuitive cruise... ;-)

tebasuna51

13th May 2008, 00:00

Thank you for your patience, but the actual "packaging" of BeHappy with BassAudio extensions is not a very intuitive cruise... ;-)
In ...\plugins\readme.txt:

(1) The oficial releases v2.4 are in http://www.un4seen.com/bass.html

EpheMeroN

16th May 2008, 12:15

Hey guys! Sorry I didn't run the BeHappy tests for my current issues. I had to leave for a week on short notice.

I'll run the tests come Saturday morning. Thanks for the help!

cweb

18th May 2008, 16:05

@cweb

Maybe you have the old RaWav.extension in BeHappy folder.
Now is inside NicAudio.extension (same ID), and you need delete RaWav.extension and also RaWav.dll in AviSynth 2.5\plugins folder if exist.

I just reinstalled the new version. I followed your tip to delete the old RaWav extension and dll in behappy's plugins directory. All seems to be working fine.

Something I noticed also, I might be doing something wrong, I have a demuxed ac3, trying to convert it to an ogg vorbis file, and I choose to DownMix it, well the end result contains a lot of hissing. Not sure what's the problem, but I'm using the latest oggenc2 from rarewares (P3/amd build) - oggenc2.85-aoTuVb5.5-P3.zip

tebasuna51

18th May 2008, 18:13

Something I noticed also, I might be doing something wrong, I have a demuxed ac3, trying to convert it to an ogg vorbis file, and I choose to DownMix it, well the end result contains a lot of hissing. Not sure what's the problem, but I'm using the latest oggenc2 from rarewares (P3/amd build) - oggenc2.85-aoTuVb5.5-P3.zip

Sorry, my fault.

I forget use the parameter
--raw-format=3
when send float samples in last BeHappy revision, must be corrected in the next revision.

To work until next revision you can use, like last DSP, a Convert Sample To 32 bit int.

Adub

18th May 2008, 22:43

Oh, good. I was just about to post about the ogg problem, and look! Here it is. Good to know that it wasn't something I was doing. Can't wait for the next version!! This is THE audio conversion software now, so keep going tebasuna51!! Your work is very appreciated.

cweb

19th May 2008, 07:39

Sorry, my fault.

I forget use the parameter
--raw-format=3
when send float samples in last BeHappy revision, must be corrected in the next revision.

To work until next revision you can use, like last DSP, a Convert Sample To 32 bit int.
thanks! I too wanted to say you are doing a great job!

tebasuna51

21st May 2008, 21:13

New BeHappy release BeHappy 0.1.9.50202 (https://www.codeplex.com/Release/ProjectReleases.aspx?ProjectName=BeHappy&ReleaseId=13626)

+ Solved bug when sending float samples to oggenc2 encoder.

+ Added new experimental features related with headers for multichannel audio data:

-ChannelMask: CMask in [2] Tweak. When checked the header is WAVE_FORMAT_EXTENSIBLE.
Selecting 0 the default for the current NumChannels is applied.

-HeaderType: Head. in [2] Tweak. 0 for WAV, 1 for W64, 2 for RF64, 3 for W64 if >4GB, 4 for RF64 if >4GB
Use only for Wav Writer or encoders with support for these headers.

We can output WAVE_FORMAT_EXTENSIBLE, w64 and rf64 uncompressed files.
Is recommended, but not mandatory, rename the W64 files like .w64

At least Flac seems go support these formats >4GB in next version.

Adub

26th May 2008, 04:51

Great!! Thanks a lot tebasuna51!! I'll test it out soon.

cweb

26th May 2008, 11:00

Great!! Thanks a lot tebasuna51!! I'll test it out soon.
I'll be doing the same myself.. thanks for this update!

DaniH

1st June 2008, 23:17

Hi!
What about 5.1 AC3 to 5.1 HE-AAC (Nero), is it supported?
I tried it and got 2 channel AAC-s every time...

tebasuna51

2nd June 2008, 02:31

Hi!
What about 5.1 AC3 to 5.1 HE-AAC (Nero), is it supported?
I tried it and got 2 channel AAC-s every time...
Maybe use you DirectShowSource to open the ac3?

Use NicAc3Source or configure properly your DirectShow decoder.

If still obtain stereo from 5.1 please put the log (copy the output window and paste here), and the avs generated by 'Export AviSynth script' button.

DaniH

2nd June 2008, 09:07

Maybe use you DirectShowSource to open the ac3?

Use NicAc3Source or configure properly your DirectShow decoder.

If still obtain stereo from 5.1 please put the log (copy the output window and paste here), and the avs generated by 'Export AviSynth script' button.

Dear tebasuna51,

I used NicAC3Source of course. Didn't try DirectShowSource. I'll do some more tests and post the log.
I checked the channel count in SMPlayer's information window. Is there a dedicated program that analyzes sound streams, maybe even channel mapping? I would like to check whether channels gets mixed up after encoding or not (many people had similar problems)

UPDATE: sorry it seems SMplayer info is b0rked, downloaded GSpot, nice prog, but there is no channel mapping info :(
Now I get another error:
Error: System.IO.IOException: Pipe ended (or something) when using NicAC3Audio. Tried to convert to WAV, then to AAC, same IO problem...
Here is the output
-------------------------------------------------------------------------------
Starting job test_Track1.wav->test_Track1_b9624cce58bf4e8a8f05db6536ef8296.mp4
Found Audio Stream
Channels=6, BitsPerSample=32 float, SampleRate=48000Hz
encoder\neroAacEnc.exe -ignorelength -he -q 0.6 -if - -of "D:\Test\test_Track1_b9624cce58bf4e8a8f05db6536ef8296.mp4"
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Error: System.IO.IOException: A pipe használata befejeződött. //(Meaning pipe ended - my addition)

at System.IO.__Error.WinIOError(Int32 errorCode, String maybeFullPath)
at System.IO.FileStream.WriteCore(Byte[] buffer, Int32 offset, Int32 count)
at System.IO.FileStream.Write(Byte[] array, Int32 offset, Int32 count)
at BeHappy.Encoder.encode()
#### Encoder StdOut ####
ERROR: no valid SBR configuration found
-------------------------------------------------------------------------------
This is the wav input - with AC3 input I get the same error.

DaniH

2nd June 2008, 09:58

Okay, me again, sorry :)
So I found out about this IO problem: I get this when i choose HE-AAC with VBR (quality)...
LC-AAC with VBR, or HE-AAC with CBR works... So why no VBR???

The only question that remains is how I could check the channel mappings of the AC3 input and the AAC output (recommend a program please)!

P.S. I did some research on the forums, found this (http://forum.doom9.org/showthread.php?t=120192&highlight=channel):
Skelsgard @ 4th Jan. 2007 wrote
"Nero AAC encoder CLI has proper mapping. As long as your WAV is correctly mapped as a multichannel WAV, you will have no problem."
So i presume if a correctly mapped AC3 (and I see no reason for it to be not correct - it's the DVD source itself) gets converted to AAC via NicAC3Decoder and NeroAacEnc CLI (latest), the channel mapping will be correct (and I can sleep calmly :))?
Tebasuna51, you too recommended the CLI encoder in the same topic.

tebasuna51

2nd June 2008, 11:12

Okay, me again, sorry :)
So I found out about this IO problem: I get this when i choose HE-AAC with VBR (quality)...
LC-AAC with VBR, or HE-AAC with CBR works... So why no VBR???
When use VBR quality use the 'Automatic' set. Only in the limits you can select the mode HE or LC.
The HE-AAC option is for low quality and q=0.6 is high quality then Neroaacenc reject the petition. I don't remember now, but the limit is near q=0.3

The only question that remains is how I could check the channel mappings of the AC3 input and the AAC output (recommend a program please)!

I don't know what you want check. An ac3 5.1 only have one possible channels configuration: LFE on, acmod=7 (mean left, right and center front channels and left, right surround channels)

You can see the acmod/lfeon possible values is this post (http://forum.doom9.org/showthread.php?p=1134394#post1134394)

About aac I don't know other info than the number of channels.

You can use MediaInfo to know data about media streams.

DaniH

2nd June 2008, 12:12

I don't know what you want check. An ac3 5.1 only have one possible channels configuration: LFE on, acmod=7 (mean left, right and center front channels and left, right surround channels)
Thank you for your post!
What I've meant was that the AV forums of the net are full of topics that discuss the mixed-up channel order that results from an 5.1AC3->5.1AAC encode. Left gets mixed up with Right etc... But none of the topics came to a conclusion, and that confused me a bit, that's why I asked.

tebasuna51

2nd June 2008, 13:25

Thank you for your post!
What I've meant was that the AV forums of the net are full of topics that discuss the mixed-up channel order that results from an 5.1AC3->5.1AAC encode. Left gets mixed up with Right etc... But none of the topics came to a conclusion, and that confused me a bit, that's why I asked.

Don't worry about this. BeHappy always produce correct channel mapping output.

Only old style decoders/encoders can have channel mapping problems. Between the decoders/encoders used by BeHappy, only ffmpeg(for ac3) and oggenc2(ogg) need a remapping done automatically.

masscamp24

3rd June 2008, 01:50

How does one install ffmpeg programme for ac3 in behappy? I install it in the encoder folder. try to convert a wav file to ac3 (upmix to 5.1 channels) got a error indicating that encoder not found.

tebasuna51

3rd June 2008, 02:42

How does one install ffmpeg programme for ac3 in behappy? I install it in the encoder folder. try to convert a wav file to ac3 (upmix to 5.1 channels) got a error indicating that encoder not found.

Maybe is a ffmpeg without the ac3 encoder. There are many ffmpeg binary and you must select the appropriate.
Not problem with my old version:
Starting job Jap6.wav->zzJap6.ac3
Found Audio Stream
Channels=6, BitsPerSample=16 int, SampleRate=48000Hz
encoder\ffmpeg.exe -i - -y -acodec ac3 -ab 384 "E:\zzJap6.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: 4608 kb/s
Stream #0.0: Audio: pcm_s16le, 48000 Hz, 5:1, 4608 kb/s
Output #0, ac3, to 'E:\zzJap6.ac3':
Stream #0.0: Audio: ac3, 48000 Hz, 5:1, 384 kb/s
Stream mapping:
Stream #0.0 -> #0.0

video:0kB audio:938kB global headers:0kB muxing overhead 0.000000%

SpAwN_gUy

3rd June 2008, 10:33

Hi guys ... it's me :) ..again.. i've moved to Vista x64 ... and.. i'm having all king of troubles :) ..

Starting job Rus_25fps.ac3->Rus_25fps_a2f77829a9834bdba1e700f216907c71.ac3
Error: System.BadImageFormatException: An attempt was made to load a program with an incorrect format. (Exception from HRESULT: 0x8007000B)
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.AviSynthScriptEnvironment.ParseScript(String script)
at BeHappy.Encoder.encode()
and this happens all the time.. it seems like i'm doing something wrong.. :( (latest BeHappy.. previous has the same problem)

any solution, please?

(well, it may be due to x86 and x64 bitness of various software.. i'll try BePipe now...)
UPD: BePipe decodes.. but ffmpeg - does not encode (something like "Can't open file for writing")
btw.. same stuff happens to allmost latest megui ... (i'll move there)

masscamp24

3rd June 2008, 21:01

[QUOTE=tebasuna51;1145102]Maybe is a ffmpeg without the ac3 encoder. There are many ffmpeg binary and you must select the appropriate.

Where can I find the correct binary? :stupid:

tebasuna51

4th June 2008, 00:54

Maybe is a ffmpeg without the ac3 encoder. There are many ffmpeg binary and you must select the appropriate.

Where can I find the correct binary? :stupid:

I can't recommend ffmpeg like ac3 encoder, use aften (http://win32builds.sourceforge.net/aften/index.html) instead.

dimzon

4th June 2008, 12:38

Hi guys ... it's me :) ..again.. i've moved to Vista x64 ... and.. i'm having all king of troubles :) ..

Starting job Rus_25fps.ac3->Rus_25fps_a2f77829a9834bdba1e700f216907c71.ac3
Error: System.BadImageFormatException: An attempt was made to load a program with an incorrect format. (Exception from HRESULT: 0x8007000B)
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.AviSynthScriptEnvironment.ParseScript(String script)
at BeHappy.Encoder.encode()
and this happens all the time.. it seems like i'm doing something wrong.. :( (latest BeHappy.. previous has the same problem)

any solution, please?

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