View Full Version : Nero AC3 Plugin (based on azid and ac3enc)
mfluder
3rd August 2003, 18:14
Thanks bond, that helped me a lot.
@bobsc
Since you are encoding to 5.1 what is your opinion about this matter?
bobsc
3rd August 2003, 20:14
mfluder,
Alot depends on your source and personal preferences, but I suggest download Azid 1.8 and read the Readme.
Eric B
3rd August 2003, 20:36
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
3rd August 2003, 20:49
eric,
plz try latest version of nero (version 6) together with he-aac for multichannel encoding!
bobsc
4th August 2003, 11:57
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
4th August 2003, 18:08
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
4th August 2003, 18:44
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
4th August 2003, 19:22
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
4th August 2003, 21:22
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
4th August 2003, 23:22
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
6th August 2003, 11:31
New version out.
bond
6th August 2003, 16:48
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
6th August 2003, 16:50
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
6th August 2003, 20:14
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
6th August 2003, 21:20
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
6th August 2003, 21:46
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
6th August 2003, 21:53
How about DeXT's version?
bond
7th August 2003, 06:34
do you have a link to dext's version? didnt find one!
bobsc
7th August 2003, 11:46
Also, I noticed LFE is being downmixed to L,R when doing 5.1?
mfluder
7th August 2003, 17:42
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
7th August 2003, 18:28
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
7th August 2003, 18:49
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
7th August 2003, 18:56
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
8th August 2003, 08:38
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
8th August 2003, 19:20
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
8th August 2003, 21:05
Nice sample, it's better than the one I have.
calinb
9th August 2003, 00:40
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
9th August 2003, 03:32
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
10th August 2003, 17:22
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
10th August 2003, 17:24
Is it possibile to add a Dolby Surround 2 downmix?
bond
10th August 2003, 17:41
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
10th August 2003, 19:00
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
10th August 2003, 19:06
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
10th August 2003, 20:02
Originally posted by bond
sure, if mausau implements it :p [/B]
Anyone could ask him??
bond
10th August 2003, 20:13
is already on my list i will tell him
frauz
10th August 2003, 20:35
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
10th August 2003, 20:59
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
30th August 2003, 09:38
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
6th September 2003, 20:21
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
9th September 2003, 19:08
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
15th September 2003, 20:05
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
15th September 2003, 20:13
mosu is going to release a new mkvmerge version with mp4 parser soon (which automatically detects he-aac in .mp4)
Doom9
28th September 2003, 21:10
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
28th September 2003, 21:13
yup, you have to change the value of every channel
besweet's hybridgain uses for example a value of 10db for gain (output!)
DSPguru
29th September 2003, 04:09
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
29th September 2003, 06:26
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
29th September 2003, 10:19
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
29th September 2003, 10:41
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
29th September 2003, 15:13
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
29th September 2003, 15:25
It would be nice if it supports faac (= free) instead of psytel, cause the latter has this channel mapping bug.
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