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...

[mfluder]

Thanks bond, that helped me a lot.

@bobsc
Since you are encoding to 5.1 what is your opinion about this matter?

[bobsc]

mfluder,

Alot depends on your source and personal preferences, but I suggest download Azid 1.8 and read the Readme.

[Eric B]

The AC3 plugin by mausau seams to works correctly, actually on the same way that besweet ac3 to 5.1 aiff. It is pretty fast (7x on my XP1600+). But I have some problems:

Which bitrate do you use for 5.1 AAC? Whatever I choose, the file is encoded at 128k. It is because I have only the trial plugin (last version of nero 5.5)?

I sometimes used to keep the 5.1 AC3 for my 4.1 system. AC3 downscale to 4.1 in AC3filters works great.
On the contrary, my AAC or MP4 5.1 file has a lot of noise on my system.

Did anybody experiences the same problems?

Are the following screenshots the recommanded settings?
http://eric.ftp.free.fr/div/NeroAC3Plugin.PNG
http://eric.ftp.free.fr/div/NeroAACPlugin.PNG

[bond]

eric,
plz try latest version of nero (version 6) together with he-aac for multichannel encoding!

[bobsc]

New version out which clears up the downmixing issue. For 5.1 you still need to move up the center channel. It would be nice to have the -a option --maximize(normalize)added.

[bond]

ok, i checked all bugs i was aware of:

1) he-aac shut down issue - resolved for 2/0 output, not working for multichannel (-> "failure in aacenc32.dll")

2) encoding doesnt finish in nero mix - still there :(

3) cutted end with 2/0 output - resolved with ogg vorbis, lc-aac and wav (outputs exactly the same filesize as azid.exe), dont resolved with he-aac encoding (half a second is cut out)! didnt test anything else till now

4) ac3 playback in neromix works (choppy)

5) opening an .ac3 in wave editor works


anyone to confirm these results?

[mfluder]

Has anyone tried encoding an AC3 track that was more than 2 hours long? It seems that there is something wrong with AC3 decoder with those tracks because with 2 tracks that I have tried so far the first one came out only 12 minutes and the second one came out only 18 minutes after encoding. I came to this conclusion because I have also tried a track that was 1h 48m long and everything went fine. Could someone please try this and post the results?

Note: This happens with all releases of mausau's AC3 decoder so it isn't just the last one.

[mfluder]

Originally posted by bond
1) he-aac shut down issue - resolved for 2/0 output, not working for multichannel (-> "failure in aacenc32.dll")
Confirmed

2) encoding doesnt finish in nero mix - still there :(
Sorry, I haven't downloaded Nero Mix.

3) cutted end with 2/0 output - resolved with ogg vorbis, lc-aac and wav (outputs exactly the same filesize as azid.exe), dont resolved with he-aac encoding (half a second is cut out)! didnt test anything else till now
So far I have only been testing 5.1 streams but I will try this and let you know.

5) opening an .ac3 in wave editor works
Confirmed

[bond]

Originally posted by mfluder
Has anyone tried encoding an AC3 track that was more than 2 hours long? It seems that there is something wrong with AC3 decoder with those tracks because with 2 tracks that I have tried so far the first one came out only 12 minutes and the second one came out only 18 minutes after encoding. I came to this conclusion because I have also tried a track that was 1h 48m long and everything went fine. Could someone please try this and post the results?i cant confirm this:
i decoded two different ac3s, one longer than 2 hours and one shorter, both sound ok (i encoded to ogg vorbis)

or do you mean that you encoded to ac3?


btw. it just came to my mind that the possibility to add a delay/append silence would be really great!

[mfluder]

Originally posted by bond
i cant confirm this:
i decoded two different ac3s, one longer than 2 hours and one shorter, both sound ok (i encoded to ogg vorbis)

or do you mean that you encoded to ac3?
I don't use Nero for anything but AAC encoding (and for burning of course) so I haven't tried Vorbis encoding, I use BeSweet for that. What I wrote above is regarding 5.1 AC3 to 5.1 AAC encoding as that is what I'm currently testing.

Could you please look up for the exact length of that 2+ hours AC3 and post it here? It would be even better if you could try encoding it to a 5.1 AAC stream (if you have time of course) and see what happens.

Thanks,
mfluder

[bobsc]

New version out.

[bond]

ok, here is my 032 updated "buglist":

resolved:

1) encoding doesnt finish in nero mix (but see bug 8)

2) cutted end with 2/0 output and he-aac - resolved in normal nero (also see bug 8)

3) "rear channel filtering" is now shown completely


not resolved:

4) he-aac shut down issue - not working with multichannel (but working with 2/0 output) -> "failure in aacenc32.dll"
perhaps a he-aac plugin bug?

5) nero plugin 032 doesnt output the same stereo .wav (didnt test multichannel) as azid.exe with the same settings chosen (this wasnt the case in 031)

6) also the output is different if there is directly transcoded from ac3 to lossy format, or if there is ac3 -> wav -> lossy
i tested vorbis and he-aac (ac3 is 1min):
he-aac: 141 bytes difference
vorbis: 1 byte difference

7) while encoding a 5.1 aac (both he-aac and lc-aac) from a 5.1 ac3 source it can happen that not the whole file is getting encoded, and only the beginning will be output (btw. no problems like this when the file gets downmixed to 2/0)
it seems that this happens only with ac3s longer than 2 hours

now two bugs only occuring in neromix:
8) cutted end with 2/0 output and he-aac - not resolved in neromix (half a second is cut out) but resolved in normal nero

9) i noticed that neromix (not nero) produces he-aac output files which have ugly, clearly hearable artifacts every second or so (i used 10db output gain, dialog norm and light dyn compr)

[bond]

Originally posted by mfluder
Has anyone tried encoding an AC3 track that was more than 2 hours long? It seems that there is something wrong with AC3 decoder with those tracks because with 2 tracks that I have tried so far the first one came out only 12 minutes and the second one came out only 18 minutes after encoding. I came to this conclusion because I have also tried a track that was 1h 48m long and everything went fine.
---
What I wrote above is regarding 5.1 AC3 to 5.1 AAC encoding as that is what I'm currently testing.yup, i can confirm that (updated bug list :D)
i encoded a 126:57 min long 5.1 ac3 file and nero outputted a 2:41 min long 5.1 mp4 (he-aac) file!

btw. did you try both he-aac and lc-aac?

[mfluder]

Originally posted by bond
yup, i can confirm that (updated bug list :D)
i encoded a 126:57 min long 5.1 ac3 file and nero outputted a 2:41 min long 5.1 mp4 (he-aac) file!

btw. did you try both he-aac and lc-aac?
Yes, I tried both HE AAC and LC AAC and the same thing happens. I also tried downmixing that very same 2+ hours stream to 2 channels in AC3 decoder and everything works as expected, the encoded file length is exactly the same as the source one. So it's definately a bug in AC3 decoder.

Thank you very much for your tests and that bug list bond, it is greatly appreciated :)

[bobsc]

Originally posted by mfluder
Has anyone tried encoding an AC3 track that was more than 2 hours long? It seems that there is something wrong with AC3 decoder with those tracks because with 2 tracks that I have tried so far the first one came out only 12 minutes and the second one came out only 18 minutes after encoding. I came to this conclusion because I have also tried a track that was 1h 48m long and everything went fine.

Has anyone tried the NeroWavReplace.dll

[bond]

Originally posted by bobsc
Has anyone tried the NeroWavReplace.dll exactly the same (i think that the new wav.dll from nero6 already has all multichannel capabilities included which were in the wav replacement dll)

[bobsc]

How about DeXT's version?

[bond]

do you have a link to dext's version? didnt find one!

[bobsc]

Also, I noticed LFE is being downmixed to L,R when doing 5.1?

[mfluder]

Just tried DeXT's version. Same thing. IMHO this problem isn't at all connected to 'wav.dll' plugin, I think it is solely AC3 decoder problem.

[bond]

seems that dexts and rjamorims wavreplaces are the same (at least they have the same version number and filesize)

btw. i came to the conclusion that the official wav.dll (2006) still doesnt support opening 5.1 .wav files (in contrary to the replacement)

[bond]

more specific results on the bug list:

ad 4)
a) works with vorbis
b) doesnt work with the wav replacement and he-aac
c) doesnt work with a 6ch wav i had (nero also quits)!!! -> seems to be a he-aac plugin bug!
[d) i also just want to mention (so that i dont forget) that first both 2.0 and 5.1 he-aac encoding didnt work, and now only 5.1 doesnt work (with the same he-aac plugin of course) so i think it can also be possible that the ac3 plugin has/had something to do with it]

ad 5) has nothing to do with encoding -> ac3 plugin issue!
also in 031 this problem wasnt there as the same output was produced as with azid.exe

ad 6) ac3 plugin issue as it happens with vorbis too!
(btw i also tested vorbis -> (wav ->) wma to be sure and both results had the same correct filesize!)

ad 7)
a) works with vorbis
b) doesnt work with wav replace
perhaps this is also caused by nero as i read somewhere that nero has problems with handling huge files (although that's why it was recommended to use the wav replace, which also doesnt work)?
bobsc or mfluder can you plz search some info on the "nero and too huge 5.1 wav files bug" as i am so tired that i am going to fall asleep right now (party before a working day isnt a good combination :D ) - what caused this problem?
i remember that some people started to use aiff instead of wav because of this problem...


ad 8) same problem with 2/0 vorbis input -> seems to be a neromix or he-aac plugin bug
[just to not forget: the same problem happened in nero before and now it works. it also worked with neromix before but encoding didnt finish
thats why i think that perhaps the ac3 plugin can also have something to do with it?]
btw. neromix also uses higher bitrates than nero with the same settings -> seems to be a neromix bug!?

ad 9) i am not sure anymore about this bug as i now downloaded the new in_mp4 and this problem doesnt seem to be there anymore

[bond]

Originally posted by bobsc
Also, I noticed LFE is being downmixed to L,R when doing 5.1?what options did you use for
lrlfe and lfe?

plz also search the forum to find out what db values are the default ones in azid which avoid any mixing of the lfe channel?
according to the azid doc 0db is the default value for lrlfe (there is nothing said about that for lfe :( )

good night :D

[bobsc]

Originally posted by bobsc
LFE is being downmixed to L,R when doing 5.1?
Someone said this on another forum. Can anybody confirm this?

[calinb]

Originally posted by bobsc
LFE is being downmixed to L,R when doing 5.1?

Someone said this on another forum. Can anybody confirm this?

That doesn't seem to be the case, presently. As discussed elsewhere in the forums, you have to swap the L&C channels in your AC3 plugin configuration to get correct channel order.

Here's a VOB that should help:
http://web.rekvizit.ru/drivers/DVD/ac3test/ac3test.zip

Here's the thread where I found it:
http://zebra.fh-weingarten.de/~maxi/html/mplayer-dev-eng/2001-11/msg00605.html

I extracted the AC3 with DVD2AVI and use it to test Nero.

[bobsc]

Nice sample, it's better than the one I have.

[calinb]

Originally posted by calinb
That doesn't seem to be the case, presently.

On second thought, bobsc's concern about the LFE ch. being downmixed to L and R need still needs to be checked. I just now remembered that my subwoofer amp has both an LFE input and high-pass passthrough inputs for the front L&R speakers. (It "siphons off" all the low freq. program material.) Although my front speakers are full range, I'd never hear it if the LFE material was being downmixed to them.

My surround speakers don't have any low-end, so I'd never hear any LFE downmix from them either.

I could check it out by pulling the L&R cables off the back of my subwoofer and jumpering them directly to my L&R power amp, but I've crammed the sub amp into a very inaccessible place -- maybe I can do it this weekend!

Anyone else checked this out?

[bobsc]

Originally posted by calinb
On second thought, bobsc's concern about the LFE ch. being downmixed to L and R still needs to be checked.
Seems OK to me.

calinb, please do your test anyway just to be sure.

[frauz]

I probably didn't focused the point. Can you definitely explain if it's possible and how to get Ac3 5.1 -> he-aac 5.1 with mousua Ac3 and MP4 Nero plug-in? I tried with theese ones but I always get nero 6.0 crashed, even using the Ac3 configuration "workaround" mentioned some post above. :confused:

[Fr4nz]

Is it possibile to add a Dolby Surround 2 downmix?

[bond]

Originally posted by frauz
I probably didn't focused the point. Can you definitely explain if it's possible and how to get Ac3 5.1 -> he-aac 5.1 with mousua Ac3 and MP4 Nero plug-in? I tried with theese ones but I always get nero 6.0 crashed, even using the Ac3 configuration "workaround" mentioned some post above. yup it works
1) make sure you use the correct channel order mentioned above in ac3 plugin
2) configure the aac/mp4 plugin under file->options->plugins... and dont push "settings" in the encoding dialog
3) there seems to be also a bug in aac plugin with file longer than 2 hours

Originally posted by Fr4nz
Is it possibile to add a Dolby Surround 2 downmix?sure, if mausau implements it :p

[frauz]

Originally posted by bond
yup it works
...configure the aac/mp4 plugin under file->options->plugins...
mmm... it doesn't work... configuration not available from there... must be some system mess up :(

[bond]

open nero burning rom -> file -> preferences -> general -> plugin lookup -> decoders -> dolby digital (ac3) -> configure -> choose your settings (right channel order!) -> ok -> encoders -> mpeg-4 advanced ... -> configure -> choose your settings -> ok -> close -> ok -> extras -> encode files -> add -> choose your file -> mpeg-4 adv... -> go

should work ;)

[Fr4nz]

Originally posted by bond
sure, if mausau implements it :p [/B]

Anyone could ask him??

[bond]

is already on my list i will tell him

[frauz]

All clear! Finally worked. I have wrongly been using Nero wave editor instead of the Nero Encode tool :p )
Unfortunately found a problem; there's a channel (usually the third, can also see it through Nero Wave Editor) which is encoded as a one-second-based ticking. Any suggestion?

I also occasionally noticed the crash problem mentioned in this 3d; I think it could be addressed to ac3 plug-in configuration, when tryin to set 5 channels instead of 6 (for exemple without lfe). In that case I noticed aac plug-in crashing if i didn't set -silence- as sixth output channel (but setting the only 5 "base" channels as output). This way I could easily manage he-aac settings within the encode box (no need to switch to file->preferences->plug-in...),

I'd be also really gladder (quite difficult thx to your help giving) if you told me how to listen (:p) to 5.1: needed filters, players used, and so on. Assume I already use the winamp plug-in but I'd like a 5.1 output and it doesn't seem automatically set.

Completely grateful.

[bond]

Originally posted by frauz
Unfortunately found a problem; there's a channel (usually the third, can also see it through Nero Wave Editor) which is encoded as a one-second-based ticking. Any suggestion?dont get this bug when encoding 5.1 ac3 to 5.1 ogg so if it is a bug than its a aac plugin one

I also occasionally noticed the crash problem mentioned in this 3d; I think it could be addressed to ac3 plug-in configuration, when tryin to set 5 channels instead of 6 (for exemple without lfe). In that case I noticed aac plug-in crashing if i didn't set -silence- as sixth output channel (but setting the only 5 "base" channels as output). This way I could easily manage he-aac settings within the encode box (no need to switch to file->preferences->plug-in...)i am too lazy to test that myself :p

I'd be also really gladder (quite difficult thx to your help giving) if you told me how to listen (:p) to 5.1: needed filters, players used, and so on. Assume I already use the winamp plug-in but I'd like a 5.1 output and it doesn't seem automatically set.i guess you want to transcode a movie soundtrack, if so you first need a directshow based player like bsplayer, media player classic or zoomplayer (windows media player is crap)
than you need coreaac directshow filter
than you need a container format which is able to handle aac together with video (for example ogm, mkv or mp4 - not avi)

if you only want to listen to 5.1 aac/mp4 sound use winamp together with the latest (!) in_mp4 plugin, which you can download from rarewares
make sure you dont tick "downmix to stereo" in the plugin settings!

[calinb]

Originally posted by calinb
On second thought, bobsc's concern about the LFE ch. being downmixed to L and R need still needs to be checked.

Finally got around to doing this--seems to be okay (not downmixed).

[melvinvc]

Excuse my English, please.
I have Win98Se and WinMe in the same computer. No matter what way I use to get the aac file (Aac machine, Nero), I have a result file without the SR sound . I muxed the file in a mkv container and played wiht several DS players and the matrix mixer showed this thing. I must select in the Ac3 plugin l,r,c,ls,rs,lfe to get the center channel in the correct order, is different as I read in this 3d.
Btw., as you know, Win98SE does not have multichannel support, but I can play in Winamp 2.91 the ogg vorbis 5.1 files with the correct order, the 5.1 wav files with the center channel in the rigth
speaker.
Any ideas?

[calinb]

Originally posted by melvinvc
Excuse my English, please.
I have Win98Se and WinMe in the same computer. No matter what way I use to get the aac file (Aac machine, Nero), I have a result file without the SR sound .

Somewhere here in the forums, ChristianHJW explains why Matroska does not yet work correctly with HE-AAC, but I can't find the posting ATM.

Use graphedit /3ivx splitter /oggmux or use mp4creator60, but I haven't learn how to do it with mp4creator yet. Use CoreAAC for playback.

Originally posted by melvinvc
I must select in the Ac3 plugin l,r,c,ls,rs,lfe to get the center channel in the correct order, is different as I read in this 3d.

That's not the order we've found for the Nero AC3 plugin to Nero AAC directly, but if you go through some other processing, you might have to swizzle the order again. For example, I've found that it needs to change if I use BeSweetGUI and aif intermediate files. Whatever order works!

[Teegedeck]

Originally posted by calinb
Somewhere here in the forums, ChristianHJW explains why Matroska does not yet work correctly with HE-AAC, but I can't find the posting ATM.
AFAIK there already is a new tag, 'aac-is-sbr' or something. I haven't tested that, though.

[bond]

mosu is going to release a new mkvmerge version with mp4 parser soon (which automatically detects he-aac in .mp4)

[Doom9]

am I correct to assume that the only way to configure the decoder is to go to Nero's plugin settings and that in order to have a reasonable volume, the max gain has to be found manually and entered into the decoder?

[bond]

yup, you have to change the value of every channel
besweet's hybridgain uses for example a value of 10db for gain (output!)

[DSPguru]

Originally posted by bond
besweet's hybridgain uses for example a value of 10db for gain (output!) that's the pregain (10db), the postgain level is determined at the end of the process (and that's why we call it "postgain" ;)).

[bond]

Originally posted by DSPguru
that's the pregain (10db), the postgain level is determined at the end of the process (and that's why we call it "postgain" ;)). i didnt mean that it is postgain, i just meant the output option in the plugin (which refers to the -o option in azid (input would be the -ch option) afaik)

[Doom9]

me thinks DG is our best option to integrate the aac encoder mentioned here (http://forum.doom9.org/showthread.php?s=&threadid=62202) into besweet <wink> Manually finding gain is a pain in the butt.. I was thinking about updating the AAC guide, but I'm not quite happy with the process as it is right now.

[bond]

Originally posted by Doom9
me thinks DG is our best option to integrate the aac encoder mentioned here into besweetyup, that would be definitely the best

Manually finding gain is a pain in the butt.. I was thinking about updating the AAC guide, but I'm not quite happy with the process as it is right now.hm, you dont need to manually find the gain value every time, just use always 10db
besweet's hybridgain (which dspguru recommends to use) also uses always (!) 10db for all 5.1 ac3 sources + dialog normalization, which can also be ticked in the ac3 plugin (never had a problem with these settings). the only thing the nero plugin cant handle is postgain, but i dont know of any aacgain implementation available atm :( dspguru?

btw. i heared rumours that new nero digital will allow direct ac3 to aac/mp4 transcoding for free...

[DSPguru]

i once wrote an aacgain tool, but never fully debugged it to release a stable build.


currently, AACMachine supports pregain normalize, which is good as well, but takes longer time.
if there's any (serious!) vb programmer around willing to update AACMachine to support the two alternative encoders (nero+quicktime), i'll share AACMachine's source-code with him.

later on, i'll try to release a stable aacgain, and so AacMachine will be a complete gui for ac3/dts/vob/avi.. --> aac transcoding with hybridgain support.

[Wilbert]

It would be nice if it supports faac (= free) instead of psytel, cause the latter has this channel mapping bug.

[DSPguru]

Originally posted by Wilbert
It would be nice if it supports faac (= free) instead of psytel, cause the latter has this channel mapping bug. so how about someone takes the quicktime-cli source-code and the nero-cli source-code and faac's source-code and builds a single executable that does it all ?

i would have done so, but have 0 free time.

come on, there must be another developer around who could contribute !
Nic can do this (actually, he can do ANYTHING), but he's been really busy lately.
perhaps valex has some free time.. ?

[DSPguru]

Originally posted by Wilbert
It would be nice if it supports faac (= free) instead of psytel, cause the latter has this channel mapping bug. so how about someone takes the quicktime-cli source-code and the nero-cli source-code and faac's source-code and builds a single executable that does it all ?

then, AacMachine will support this exectuable.

another approach is supporting BeSweet.dll in this unified executable and so we have a complete solution in a single tool.

i would have done so, but have 0 free time.

come on, there must be another developer around who could contribute !
Nic can do this (actually, he can do ANYTHING), but he's been really busy lately.
perhaps valex has some free time.. ?

[ChristianHJW]

Originally posted by DSPguru i would have done so, but have 0 free time.

DG, sorry for jumping in so late here, and its well possible i completely miss the point now as i cant read the whole thread, but wouldnt it be a much easier and better way if you could simply change the channel order in BeSweet if you are writing AIFF files ? This is working perfectly for me until today, and will allow all the normalizing and sampling rate conversions of BeSweet, and without the stupid WAV file size limitations.

menno told me that he thinks that you are using a WAV channel order for 5.1 files when writing AIFF files, and that you obviously dont knwo that there is a standard 5.1 channel order for those.

Wouldnt this be the easiest solution ?

[DSPguru]

the current 6chaiff feature is used in tools like dvd2svcd to encode to 5.1mp2 with the phillips encoder. the channel ordering is compatible with their tools.

i can add another switch for creating an alternative 6chaiff with different mapping. what is that mapping that nero needs ?

[Wilbert]

AFAIK the standard aiff channel order is:

FL
RL
C
FR
RR
LFE

I guess Nero6 does it correctly now [in Nero5 you had to give aiff/wav the same ordening as the aac], and wants this ordening.

[calinb]

Originally posted by Wilbert

I guess Nero6 does it correctly now [in Nero5 you had to give aiff/wav the same ordening as the aac], and wants this ordening.
I have the latest Nero6 (NeroMix) and, if I use BeSweet to mux 6 wav files, I've found the correct order to be:
C
FL
FR
SL
SR
LFE

At least that's the order of the files in my list.mux file that I use to generate an aif file. This is my workaround (6 posts down--about 1/2 way through the post):
http://forum.doom9.org/showthread.php?s=&postid=376653#post376653

[Wilbert]

But you are creating an AC3 with this order, don't you (cause that's the channel ordening in an AC3 file)?

edit: it's indeed the AAC ordening, like said below.

[calinb]

Originally posted by Wilbert
But you are creating an AC3 with this order, don't you (cause that's the channel ordening in an AC3 file)?
No, I'm creating an aac/mp4 file. That channel order works in Nero6 "Mix" for both AC3 file input (using the AC3 plugin) or 6 wav files muxed with BeSweet to an aif input file. I used the AC3 stream in this file to check it:

http://web.rekvizit.ru/drivers/DVD/ac3test/ac3test.zip

I listened to the AC3 file. I listened to each of the (mono) wav files that I extracted from the AC3 file with BeSweet (6wav output). I listened to the final mp4 file. All six channels were correct in all cases and each of the 6 wav files contained the correct content. I have no idea if they are "correct" in the intermediate aif file because I don't have anything that will play 6ch. aif, but I assume that they are correct because Nero produces the correct output. If this is not the case, is the Nero aif plugin assuming an incorrect order (AC3 ordering rather than aif ordering) for 6ch aif input?

[bond]

C, FL, FR, SL, SR, LFE - is the official aac channel order
FL, C, FR, SL, SR, LFE - is the official ac3 channel order

it is possible to change the channel order in the nero ac3 plugin...

[calinb]

Originally posted by bond
it is possible to change the channel order in the nero ac3 plugin...
Right, but because the nero ac3 plugin can't do much processing other than preset gain and normalize, we've been trying to do some pre-processing with BeSweet (like downsampling to 44.1 and two-pass gain/normalize, etc. Unfortunately, the only way I've found to do this and get the correct channel order in the final Nero aac/mp4 file, is to go AC3 >> 6 wavs >> list.mux file >> aif >> aac/mp4. BeSweet is used for all but the last step.

Bond, thanks for the "spec." Looks like Nero aac/mp4 plugin always wants the input in AAC order, right?

[bond]

Originally posted by calinb
Unfortunately, the only way I've found to do this and get the correct channel order in the final Nero aac/mp4 file, is to go AC3 >> 6 wavs >> list.mux file >> aif >> aac/mp4. BeSweet is used for all but the last step.it is also possible to change the channel order already in besweet under azid (the same way the plugin can change it)

[calinb]

Originally posted by bond
it is also possible to change the channel order already in besweet under azid (the same way the plugin can change it)

Well, have you tried this? Neither ChristianHJW nor I could get that to work. It doesn't matter what order you specify in the azid command, it never changes the output order.

Bond, P.S. Isn't that a new icon/badge you're sporting in the forums? Congrats!

[mfluder]

Here is what Menno said about the input channel order of the latest AAC plugin:


This channel ordering has already been solved. And the next version of the plugin will assume l/r/c/lfe/sl/sr (standard WAV ordering) input.

Mind you that he posted this on August 17, so that means that the latest Nero 6.0.0.15 comes with AAC plugin with that input order. You can find the whole thread here (http://www.hydrogenaudio.org/index.php?showtopic=12034) (which was btw started by our very own bond).

I have also verified this by changing the output order of the AC3 plugin to the one from Menno's post and just to make sure by decoding AC3 to WAV using BeSweet. Everything sounds as it suppose to so this is the channel order you have to use to get correct results.

Hope this helps,
mfluder

Edit: I almost forgot to congratulate bond on his new status, so congratulations bond :)

[calinb]

Hmmm, this shouldn't be so hard, right? I have the latest NeroMIX (1.4.0.4) but I only have Nero Ultra Edition 6.0.0.11. I guess I'll install 6.0.0.15 and see if anything's changed.

[bond]

Originally posted by calinb
Well, have you tried this? Neither ChristianHJW nor I could get that to work. It doesn't matter what order you specify in the azid command, it never changes the output order. :eek: i have to admit i never tried it (dont have a 5.1 system)

Originally posted by mfluder
I almost forgot to congratulate bond on his new status, so congratulations bond :)Bond, P.S. Isn't that a new icon/badge you're sporting in the forums? Congrats!thanks for the congratulations :)

which was btw started by our very own bond :D

[DSPguru]

Originally posted by calinb
I have the latest Nero6 (NeroMix) and, if I use BeSweet to mux 6 wav files, I've found the correct order to be:
C
FL
FR
SL
SR
LFE Originally posted by bond
C, FL, FR, SL, SR, LFE - is the official aac channel order a match !

[ChristianHJW]

You guys are great :D !!! So i need to update Nero from .11 to .15 and all is fine ?

So, BeSweet [ azid -ota --> ssrc 48->44.1 32 FP --> AIFF ] --> Nero 6.x.x.15 [ AIFF --> MP4 HE-AAC ] --> mkvmerge

is my way now :D !!!! Cheers !

[DSPguru]

i really don't know if "all is fine". i didn't test nero..

all i can say is that if you guys want, i can add a new switch to create a 6chaiff with channels mapping stated by calinb & bond.

[mfluder]

It's true, the channel order stated by calinb and bond was the correct one in previous releases but starting the latest Nero 6.0.0.15 release this channel order has been changed to the one from Menno's post: l/r/c/lfe/sl/sr (standard WAV ordering). This also aplies to AIFF.

So DSPguru you don't have to change anything, calinb just needs to download the latest release of Nero and all is fine. But I wouldn't mind if you make direct AC3 -> AAC transcoding possible in your great tool using nencode code :D

mfluder

[bobsc]

Originally posted by mfluder
So DSPguru you don't have to change anything, calinb just needs to download the latest release of Nero and all is fine. But I wouldn't mind if you make direct AC3 -> AAC transcoding possible in your great tool using nencode code :D
I wouldn't mind either :D

[calinb]

Originally posted by mfluder
calinb just needs to download the latest release of Nero and all is fine.

I wish :( I downloaded Nero 6.0.0.15 (the latest) and updated. It installed Aac.dll dated 8/20/2003. (The latest NeroMix, 1.4.0.4, still contains Aac.dll dated 7/21/2003.)

Unfortunately, the newest dll doesn't produce mp4 files that playback with 3ivx splitter >> CoreAAC. I checked with graphedit too and it just crashes like wmp8 and mpc. Files I encoded with the older dll still play fine and new files play okay (2 ch. only) with NeroMix.

So I copied Aac.dll 7/21/2003 back to my system and all is well again--except I'm back to the terrible channel order workaround!

Anyone run into this?

DSPguru, if you add the switch to create a 6chaiff with the C,L,R,SL,SR,LFE order, it appears to still be useful :). Yeah, nencode capability would be even better, but I think we'll still have to deal with the channel order or Aac.dll problem.

[calinb]

Looks like Nero 6.0.0.19 is out, with the same problem!
http://forum.doom9.org/showthread.php?s=&threadid=62570&highlight=3ivx+AND+nero

WRT aac encodings made with the new Aac.dll:

1. I tried muxing .mp4 to ogm using the 3ivx splitter--same crash! (Oggmuxer/Nic's Oggmux would probably still work for LC-AAC, but without the 3ivx splitter workaround for the SBR tag, HE-AAC is not possible.)

2. I tried muxing .mp4 to mkv using mkvmerge 7.0--success!

3. I tried muxing .aac file to mp4 with mp4creator60--success!

So it appears that 3ivx splitter is incomaptible with the newest Nero AAC.

All of the above was done with 5.1 HE-AAC audio and yes, the new Aac.dll produces the new channel order--at least I believe so. I'll know after I listen on my 5.1 HTPC. I just have headphones now.

[DSPguru]

currently, there are no plans for integrating Nero's commercial tools with BeSweet. sorry..

i have added a new switch for azid, already supported by BeSweetGUI :
-azid( -oc,l,r,sl,sr,lfe )

i'll upload a new v1.5b21 binary later today.
you'll find the new switch in the "changes.txt" file.

[sillKotscha]

Originally posted by DSPguru
currently, there are no plans for integrating Nero's commercial tools with BeSweet. sorry..



hi Dg,

not the comercial tools but what about Nencode (as already mentioned...)

A great tool is now being hosted at RareWares: NEncoder, by darp, is a command line utility that uses the Nero AAC DLLs to encode wave files from the command line. It's available at the AAC section.

about the new build - very nice, thank you Sir!

cheers Sill

[gubbah]

I had the 3ivx problem too when I installed Nero 6.0.0.19...I just unregistered NeVideo.ax in the "C:\Program Files\Common Files\Ahead\DSFilter" directory and re-installed 3ivx and everything worked OK.

[calinb]

Originally posted by gubbah
I had the 3ivx problem too when I installed Nero 6.0.0.19...I just unregistered NeVideo.ax in the "C:\Program Files\Common Files\Ahead\DSFilter" directory and re-installed 3ivx and everything worked OK.
I think this is a different problem. Yes, I had to re-install 3ivx too (or at least re-register the splitter), but the problem that's more significant is the latest Aac.dll doesn't work with 3ivxDSMediaSplitter.ax. NeVideo.ax doesn't have anything to do with the DirectShow filter graph that I'm talking about (rendering an mp4 file with AAC audio only.)

DSPguru: we can never say thanks too many times! The azid switch really helps--given the 3ivx splitter/new Aac.dll incompatibility. I'll just stick with the old Aac.dll.

[shitowax]

The problem is not Nero's channel order at all ... It's the fact that the latest Nero produces audio frames with a duration == 0. OpenQuicktime used in the 3ivx splitter doesn't appreciate that at all ;)
We're working on solving this problem ASAP.

[calinb]

Originally posted by shitowax
The problem is not Nero's channel order at all ...
Yes, I/we realize that. It just happens that Nero changed the channel order in the same release that broke the compatibility with 3ivx. So you have a choice of the old channel order (AAC order -- until DSPguru's latest build, it was not well-supported by BeSweet/Azid), or the new channel order and the 3ivx incompatibility :(

Whatever--we're happy to hear you are working on it and now we know the root of the problem!

[tiki4]

Originally posted by shitowax
The problem is not Nero's channel order at all ... It's the fact that the latest Nero produces audio frames with a duration == 0. OpenQuicktime used in the 3ivx splitter doesn't appreciate that at all ;)
We're working on solving this problem ASAP.

Just to throw in some speculation: Is this the new feature of Nero AAC of gapless playback, e.g. in Winamp or foobar2000? I thought about this before...

tiki4