Nero AC3 Plugin (based on azid and ac3enc)

[bond]

As most of you guys probably know nero (yes the burning software ;) ) offers one of the best possiblities to encode in aac with it's native mp4-plugin at the moment!
the problem for us ripping people is that, when we want to use nero's audio plugins with an ac3 source, we have to decode the ac3 to wav first (which causes huge filesizes in many cases) and only then we can encode it with nero
with an ac3 plugin it would be possible to transcode directly in nero which would save time and space!

mausau (author of many nero plugins) writes on his page (http://neroplugins.cd-rw.org/) that "if you'd like to develop plugin for Nero, you have to get here (ftp://ftp6.nero.com/NeroSDK-v1.02.zip) the latest Nero Plugin SDK from Nero (http://www.nero.com/en/content/c1002825712128.html) website"

as an encoder there exists free ac3enc (http://dspguru.notrace.dk/BS_ac3enc.zip) (sources are included) which is also used in besweet
as an decoder (which will be the most important i think) you could use the source from valex' ac3filter (http://sourceforge.net/projects/ac3filter) or the free tool azid (http://www.doom9.org/Soft21/Audio/azid-1.8.zip) (although i dont know where you can get the sources for the last one)!


i hope anybody has the skills and the will to write such a plugin (i only have the will :( )? it would really help using the nero audio-encoding features (especially mp4/aac)

edit: just changed subject title

[S_O]

That would really be great. Have you ask mausau to develope an ac3 plug-in for Nero? Normally a ac3 plug-in for Nero doesn´t make much ense, because you nearly never use ac3, expecially not for music, but with that it makes sense. Point him to these sources:

Decoder (GPL) (it the decoder that ac3filter uses): http://liba52.sourceforge.net/

Encoder (L-GPL): ac3enc of ffmpeg (http://ffmpeg.sourceforge.net/). He only needs these files from CVS for ac3enc:
avcodec.h
common.c
common.h
ac3enc.c
ac3.h
ac3tab.h

The interface is quite simple it seems, if he wants to know how it works, he can take a look into the ac3enc.dll-source of besweet, which can be found here: http://dspguru.notrace.dk/plugins.htm

Edit: The file is now named ac3.h, not ac3enc.h.

[bond]

yeah i already wrote him but he didnt replied :(

perhaps you can also send him a mail: mausau at yahoo dot com

[Defiler]

Just replying to add my support for such a thing. I have yet to see the 5.1 WAV -> AAC process in Nero work, though I greatly desire it to.

[Animaniac]

I would love to see this. Nero crashes whenever I do anything with a 5.1 WAV. -_-;;

[bond]

A work-around for the wav-problems is to set '5.1 AIF' instead of '5.1 WAV' in the BeSweet output options! Nero will read it nicely, even 10 GB files (thanks to ChristianHJW for this tip!)

but an ac3 plugin would of course still save time and space...

[Animaniac]

Originally posted by bond
A work-around for the wav-problems is to set '5.1 AIF' instead of '5.1 WAV' in the BeSweet output options! Nero will read it nicely, even 10 GB files (thanks to ChristianHJW for this tip!)

but an ac3 plugin would of course still save time and space...

I get a Visual C++ Runtime Library appliation abnormally terminated when I try to open a 5.1 AIFF. This is the CLI line I am using.

BeSweet -core( -input "project AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3" -output "project AC3 T01 3_2ch 448Kbps DELAY 0ms.aif" -6chaiff -logfilea "BeSweet.log" ) -azid( --maximize ) -ota( -d 0 ) -split( -start 0 -end 300 )


EDIT: Apparently I needed VB6 Runtime libraries. It's ironic how I needed to downgrade.

EDIT #2: Scratch that, now I can open the 5.1 AIFF, but AAC encoding gives me the 3 errors I mentioned before, in the other thread.

[Defiler]

Wow. Nero works with 5.1 AIF. That's neat. Heh.

[bond]

I got a mail from mausau: he says he will see what he can do...

perhaps someone can write him a mail too to speed the development a little bit up :D

[symonjfox]

Yes, it's a good idea create such plugin! Good idea :)

For use of 5.1 wavs you must use the updated wavplugin else you'll get an error.

[Animaniac]

There's no more need for it, since PsyTel was proven better than Nero (at least at 128 Kbps stereo, but most likely that extrapolates to a general claim).

[bond]

Originally posted by Animaniac
There's no more need for it, since PsyTel was proven better than Nero (at least at 128 Kbps stereo, but most likely that extrapolates to a general claim). hm at least it doesnt harm and in contrast to nero psytel isnt getting better in the future (aac+ etc...)

but huh so i would say that dspguru's good old :D aacmachine (which already supports psytel) is back in the race...

[JohnV]

Originally posted by Animaniac
There's no more need for it, since PsyTel was proven better than Nero (at least at 128 Kbps stereo, but most likely that extrapolates to a general claim).
?
Maybe you understood the results wrong. Psytel 128cbr was definitely not proven better than Nero 128cbr. Infact they were in practise very very close to each others. The difference is totally insignificant and in no way say which is better.
I suggest you read the analysis page again.

[bond]

great news!!!

thanks to mausau there now is an .ac3 decoder plugin for nero available (based on azid)!

grab it on mausau's nero plugins page: http://neroplugins.cd-rw.org!
direct link to the plugin: nxmyac3.zip (http://neroplugins.cd-rw.org/files/nxmyac3.zip)


now it should be possible to directly transcode from .ac3 to .mp4 in nero without a wave in between

enjoy :)

[rjamorim]

Great news! No more crappy FAAC quality transcoding with foobar :D

[bobsc]

Surround channels are not being decoded.

[S_O]

For me it doesn´t work (like several other plug-ins, too), opened ac3 in WaveEdit, no problem, the waveform is loaded correctly. But when I press play I get only silence, also when I want to save, the output file is only silence. It seems decoding works one time, and this one time was the waveform display :(

[bond]

Mausau writes in a mail:

Right now the default settings are being used - Dolby Surround compatible downmix, no dynamic compression, 2 front channels, 0 rear channels.

When I find the time, I may release a newer version which allows decoding options.

[bobsc]

All is needed is the 3/2 option.

[bobsc]

New version out.

[bond]

seems like my bugging helped :D

ps:
view the screenshots on mausau's page to see how you can tweak the azid settings!

[bond]

hm, it doesnt seem to work with he-aac (nero just quits)
with lc-aac it seems to work fine...

anyone too having problems with he-aac?

[bobsc]

Same here, but shouldn't be plugins fault.

[mfluder]

The same thing happens here. But it happens only if you press "Settings" and even if you don't change anything it will crash. And this only happens if HE AAC is selected, with LC AAC it works fine and it doesn't matter how much you mess with the settings, it will never crash.

But there is a workaround though. You can configure the codec from Nero's "Audio plugin manager settings" - File -> Preferences... -> General Tab -> Plug-in Lookup... -> Encoders Tab and then of course you configure the codec the way you want and proceed with the usual encoding but without configuring the codec from "Encode Files" dialog because this will lead to a crash.

Hope this helps,
mfluder

[bond]

great, thanks a lot for this workaround!

[bobsc]

I just tried NeroMIX transcoding AC-3 to LC and HE AAC, but that doesn't work at all.(Starts but never finishes)

[bond]

Originally posted by bobsc
I just tried NeroMIX transcoding AC-3 to LC and HE AAC, but that doesn't work at all.(Starts but never finishes) same here, i also tried it with the ogg vorbis plugin too

[bobsc]

Originally posted by mfluder
But there is a workaround though. You can configure the codec from Nero's "Audio plugin manager settings" - File -> Preferences... -> General Tab -> Plug-in Lookup... -> Encoders Tab and then of course you configure the codec the way you want and proceed with the usual encoding but without configuring the codec from "Encode Files" dialog because this will lead to a crash.
Another workaround is just use the default settings which are HE AAC

[bond]

hm, it seems that the last second of the ac3 isnt getting decoded by the ac3 plugin (try ac3->wav in besweet and nero and compare the ends)

[mfluder]

Originally posted by bobsc
Another workaround is just use the default settings which are HE AAC
Of course it is. But what if you want to change the bitrate or you want to export to AAC stream so you can mux it. What I wrote is exactly the same thing, AAC encoder is using default settings which you set by configuring the codec from Nero's plugin dialog.

[bobsc]

Originally posted by mfluder
But it happens only if you press "Settings" and even if you don't change anything it will crash.

Not true. You would have to change something. Your post wasn't clear to me. Like I said you could just open Nero add files and hit go if the default settings are to your liking, which are VBR 50-70 HE.

[mfluder]

Originally posted by bobsc
Not true. You would have to change something.
I have just checked this again and if you hit 'Settings' and then 'Cancel' it won't crash, but if you hit 'OK' and even if you don't change anything it will crash. So it looks like we are both right :)

[bobsc]

mfluder,

Thanks for the clarification, I'm wondering why there are so many quirks.

[bond]

Originally posted by bond
hm, it seems that the last second of the ac3 isnt getting decoded by the ac3 plugin (try ac3->wav in besweet and nero and compare the ends) i can specify this now:
the cutted end just occurs if only ch_l and ch_r are chosen under "output channel sequence"
if all 6 channels are chosen (even if the 5.1 ac3 is decoded to 2 channels) the failure doesnt occur

this is definitely a bug in the plugin because when i use azid.exe in besweet to decode the .ac3 by using -ol,r --ch0=none,0db --ch1=none,0db -d 2/0 which should be the same commandline as used in the plugin, the end isnt cut off!


does anyone have similar results?

[bobsc]

bond,

What your saying might be true, but why would you want to use -o l,r -d 2/0 with a 5.1 AC-3.

[bond]

Originally posted by bobsc
What your saying might be true, but why would you want to use -o l,r -d 2/0 with a 5.1 AC-3hm, did i miss something?
is there another way to downmix to stereo in azid?

[bobsc]

-ol,c,r,sl,sr,lfe -d 2/0 will downmix all input to stereo.

[bond]

-d 2/0 is the option to choose for downmixing to stereo

-ol,c,r... is the channel order (called "output channel sequence" in the plugin)
in azid the -o option can be skipped, i just used it because the plugin seems to use it too (as it cant be disabled) and i wanted to achieve exactly the same results with the exe and the plugin!

[bobsc]

New version out which includes AC-3 encoder.

[bond]

ok, small update on the end cut off issue:

in nero mix it doesnt cut off the end but it doesnt stop decoding like bobsc already reported!

[bond]

seems like the he-aac bug is resolved with new plugin (tested only stereo file)

edit: no, it doesnt work with multichannel input

anyone to confirm this? bobsc? :D

[bobsc]

Same here, I'm only testing multichannel. I realy don't think its the plugins fault. I wonder if Nero 6.0.0.11 makes any difference.

[mfluder]

I don't know if you guys are already aware of this but anyways here it goes. If you are encoding 5.1 AC3 to 5.1 AAC you have to change output channel sequence in mausau's AC3 decoder dialog to this order:

1 - Center Front
2 - Left
3 - Right
4 - Left Surround (Rear)
5 - Right Surround (Rear)
6 - LFE

You can find out more about this channel ordering here (http://www.hydrogenaudio.org/index.php?showtopic=10986). If you don't do this not only you'll get wrong channel order but you'll also get higher bitrates. If you currently don't have any 5.1 system (like I don't) you can easily spot if the channel order is wrong by downmixing the encoded stream to stereo. You will hear center channel only on one speaker. When you change the channel order to a correct one center channel should be downmixed properly and you should hear it on both speakers.

[bobsc]

mfluder,

You are correct and I was aware. Thanks for the clarification.

[bond]

Originally posted by mfluder
If you don't do this not only you'll get wrong channel order but you'll also get higher bitrates. If you currently don't have any 5.1 system (like I don't) you can easily spot if the channel order is wrong by downmixing the encoded stream to stereo.hm, if you want to downmix to stereo you have to choose only left and right anyways (otherwise you will get a 6ch file with only sound in ch_l and ch_r)

[bobsc]

bond,

mfluder is correct if you want a 2 channel file just select 2/0.

[bond]

Originally posted by bobsc
mfluder is correct if you want a 2 channel file just select 2/0.nah!
if i check 2/0 and "output channel sequence" c, l, r, sl, sr, lfe i get a 6ch output file with sound only in 2 channels (l, r) and 4 silent channels (c, sl, sr, lfe) i checked that with matrix mixer

am i the only one with these results :confused:

[bobsc]

bond,

What you say is correct select l,r 2/0 for stereo, but if want a 5.1 file with the proper channel order select c,l,r sl,sr,lfe 3/2.

[mfluder]

What about dynamic compression? Do you guys leave it alone or do you set it to another value? In BeSweet I'm always using normal dynamic compression but then again I was never using BeSweet for 5.1 encoding. I either downmix to DS or DS2 or I keep the original AC3 file. And what about this 'Input' 'Output' thing, what about that? And also, do you use any kind of gain?

[bond]

Originally posted by bobsc
if want a 5.1 file with the proper channel order select c,l,r sl,sr,lfe 3/2.yup

Originally posted by mfluder
What about dynamic compression? Do you guys leave it alone or do you set it to another value? In BeSweet I'm always using normal dynamic compression but then again I was never using BeSweet for 5.1 encoding.i use surround together with light or normal for 2ch output (still have to decide :D but most of the time i use light - dunno why :D)

And what about this 'Input' 'Output' thing, what about that?in besweet gui, under azid on the corner up right, it says overall output settings or so (in the left corner you have the input settings) dunno what the difference is but in besweet i always used to set the output settings, so i set the output settings in the plugin too ;)

dont forget to choose a setting always for each channel seperately !!!

And also, do you use any kind of gain? hybridgain uses 10db gain (output) and dialog normalization, so i use that in the plugin too...