AC3 to AAC/MP4 with foobar2000

[tiki4]

Topic:

My friend tuning asked me about different ways to transcode AC3 to the AAC format. There are quite some options now that all make use of the Nero AAC encoder. Nero AAC is the only publically available codec that provides the 'High Efficiency' (HE-AAC) profile for low bitrate encoding which comes in very handy for DivX/XviD movies. Currently Nencode, Oagmachine (featuring BeSweet/bsn.dll) and foobar2000 make use of the Nero engine. However, foobar2000 is a little bit different it seems, as it doesn't use the Nero libraries directly, but the Nero API IMHO.

Aim:

Well, aim of this thread should be to compile a guide how to trancode AC3 to AAC/MP4 in foobar2000. fb2k isn't the easiest program to use IMHO and people who did it could share their experience here.

Differences to Oagmachine:

THE FOLLOWING STATEMENT IS NO LONGER VALID, RESAMPLING ISSUE IS SOLVED WITH BeSweet 1.5b25, THANKS DSPGuru

Well, Nencode and BeSweet share a common problem which has to do with resampling. Choosing a certain bitrate or VBR profile the Nero encoder expects certain sampling rates to be provided. Now all DVD audio is 48 kHz and the current solution is to resample in BeSweet/Oagmachine to 44.1 kHz, otherwise the output file gets broken in certain scenarios. fb2k on the other hand seems to manage this 'more correctly'. In what sense is to be found out.

Preliminary how-to (for 5.1 output):

In the first place you need foobar2000 from Case's site (http://www.saunalahti.fi/~cse/html/foobar.html). It is labeled 'special installer'. During installation make sure that AC3 is selected in the optional components/Input section and also the disk writer component. Also you need the Nero encoder/decoder from the same site. Copy the included foo_nero.dll to your foobar2000/components folder. Finally you need a full version of Nero 6 installed.

Now here begins the fun: Add an AC3 file to the fb2k playlist. Now in the preferences look for components/diskwriter. Select 'Nero encoder' as output format. Check that DSP and replaygain isn't used. Now check for 'Nero encoder' component and select "MPEG4 Advanced Audio Coding" as output format. Configure the Nero plugin to your likings. Don't forget to enable 'dynamic range reduction' in the AC3 component settings. Now close preferences again and right click your AC3 in the fb2k playlist window. Choose convert. That's it in principle. You may change the output naming and the output directory.

Feeback:

Now, please provide your feedback and corrections! Tell us your findings and remember that the AC3 decoding part in foobar isn't azid, but liba52 like Valex's AC3filter uses it. Thus different results are absolutely possible.

Thanks to tuning for the idea.

[bobsc]

Your preliminary how-to is for 5.1 output.
If you want 2 channel output you would enable DSP and go to DSP Manager and move "Convert 5.1 to stereo" to Active DSPs.

[mikeson]

I use this method some time and found only one problem till now:
- when source is 2ch AC3, result is always 5.1 AAC/MP4, there seems to be bug in foo_ac3.dll

[bobsc]

@mikeson
I never tried a 2ch AC3, but try "Convert 5.1 to stereo".

[Tuning]

Thanks tiki4.:D

@bobsc,
Do the option: convert stereo to 4 channel (in DSP manager)affect AAC encoding? :)
Thanks

[bobsc]

Originally posted by Tuning
@bobsc,
Do the option: convert stereo to 4 channel (in DSP manager)affect AAC encoding? :)
Thanks
Never tried it.

[tiki4]

Originally posted by bobsc
Your preliminary how-to is for 5.1 output.
If you want 2 channel output you would enable DSP and go to DSP Manager and move "Convert 5.1 to stereo" to Active DSPs.

That's right, I edited that right away. Any more comments in comparison to Oagmachine? I guess foobar2000 is a versatile tool and since it can also play FLAC/Cue/MKA I use it again a little more often.

Thanks for your answers so far, but I still hope more people try that.

Is there a way to normalize the file without replaygaining? No direct show filter can make use of the replaygain tags so far and possibly foobar2000 uses Ape2 tags for that, please correct me.

tiki4

[bond]

nice guide!

i would also mention that next to nero, people can also use the faac codec, which is getting constantly better (but shouldnt be used for multichannel as it doesnt provide he-aac)

Originally posted by tiki4
Is there a way to normalize the file without replaygaining? No direct show filter can make use of the replaygain tags so far and possibly foobar2000 uses Ape2 tags for that, please correct me.the official way would be to adjust the "volume element" in MP4 (other ways by using tags or so, will probably never be supported by standalones etc.), but no tool can do that till now (was discussed here (http://forum.doom9.org/showthread.php?s=&threadid=62848))...

[Tuning]

Originally posted by bond
the official way would be to adjust the "volume element" in MP4 (other ways by using tags or so, will probably never be supported by standalones etc.), but no tool can do that till now (was discussed here (http://forum.doom9.org/showthread.php?s=&threadid=62848))...

@bond

how this volume element works. i.e do mp4 container gives infos to decoder to increase volume or similar. Can a brief explanation be done?

Thanks

[bond]

Originally posted by Tuning
how this volume element works. i.e do mp4 container gives infos to decoder to increase volume or similar. Can a brief explanation be done?you will find everything that is know about the volume element in the thread i pointed to
afaik the splitter needs to pass that info to the decoder (from what shitowax said the 3ivx splitter already would handle it)

it seems that apple uses an own tag to store postgain info in mp4, but i dont know exactly how it works...

[Tuning]

afaik the splitter needs to pass that info to the decoder (from what shitowax said the 3ivx splitter already would handle it)

Thanks for the info!:)

[tiki4]

Actually I found out, how to maximize the sound when transcoding AC3 to AAC. It's quite easy, as you just have to replaygain the original AC3 and enable 'use replaygain' in the Diskwriter options. This won't work for 5.1 -> 2.0 so far, but I'll work on that, too. I start to like that little app, although the GUI could really need some improvement.

tiki4

[bond]

Originally posted by tiki4
Differences to Oagmachine:
Well, Nencode and BeSweet share a common problem which has to do with resampling. Choosing a certain bitrate or VBR profile the Nero encoder expects certain sampling rates to be provided. Now all DVD audio is 48 kHz and the current solution is to resample in BeSweet/Oagmachine to 44.1 kHz, otherwise the output file gets broken in certain scenarios. fb2k on the other hand seems to manage this 'more correctly'. In what sense is to be found out.its not the fault of the nero encoder but of the nencoder tool, which besweet is based on:

Originally written by menno
besweet and nencoder don't respect the input request of the encoder
basically you provide some info to the encoder config, then the encoder config tells what kind of input it needs
besweet and nencoder completely ignore that and just provide data as is

foobar2000 does this correctly. it doesnt use nencoder, it has it's own foo_nero component that handles the input request of the nero encoder correctlymaybe it would be a good idea to base bsn.dll on the source of this foo_nero plugin than on nencoder...

[bobsc]

Originally posted by bond
maybe it would be a good idea to base bsn.dll on the source of this foo_nero plugin than on nencoder...
I think that would be a good idea also.

[bond]

Originally written by menno
basically you provide some info to the encoder config, then the encoder config tells what kind of input it needsto make clear what this means menno told me the following:

1) the nero encoder dlls themselves dont do any resampling
2) the resampling is done by an own nero resampling plugin
3) the user chooses a bitrate and the encoder requests a specific sampling rate as input
4) now the resampling plugin would normally resample to this specific sampling rate
5) but nencoder and besweet ignore this request, therefore the encoder dlls dont get the requested input sampling rate

example:
a user chooses 80kbps as output in the nero encoder. the encoder requests 32khz for that bitrate, but besweet gives him 48khz or 44.1khz (or whatever else the user choose to resample to in besweet)
->
the output is borked

you have to feed the nero encoder with the sampling rate he needs for a specific bitrate, otherwise you will get a wrong output

in besweet you have to do this manually by using ssrc, foobar can do this automatically...

[Tuning]

Finally I could play with foobar2000 (..not forgetting I was winampfan till yesterday...). I have some newbie doubts, need to be clarified by someone.

What is replaygain?
How to use it?

I found it can also be used while transcoding. But i did my best to find out what is it....:( no luck. The final mp4 file didn't had any replaygain info eventhough i used it while encoding.:(

In short:
I did scan replay gain for full album. Then selected nero encoder and HE-AAC in diskwriter options. Also ticked "use replay gain' in processing. Nothing special was found in final mp4 file.

Another Q: Is it possible to transcode wma to blah format using foobar2k?
I always get foo_wma.dll failed to load error. I think this plugin is missing in installation. Where can I get it.

Thanks. :)

[bobsc]

http://replaygain.hydrogenaudio.org/
http://www.saunalahti.fi/cse/foobar2000/

[tiki4]

Originally posted by bond
to make clear what this means menno told me the following:

1) the nero encoder dlls themselves dont do any resampling
2) the resampling is done by an own nero resampling plugin
3) the user chooses a bitrate and the encoder requests a specific sampling rate as input
4) now the resampling plugin would normally resample to this specific sampling rate
5) but nencoder and besweet ignore this request, therefore the encoder dlls dont get the requested input sampling rate

example:
a user chooses 80kbps as output in the nero encoder. the encoder requests 32khz for that bitrate, but besweet gives him 48khz or 44.1khz (or whatever else the user choose to resample to in besweet)
->
the output is borked

you have to feed the nero encoder with the sampling rate he needs for a specific bitrate, otherwise you will get a wrong output

in besweet you have to do this manually by using ssrc, foobar can do this automatically...

Sorry, bond, but I think it isn't that easy.

An example:

1. I use BeSweet with latest Nero dlls. I choose '-6chnew', and 'streaming' + HE-AAC. Output is borked, as expected.

2. I decode to 6ch AIFF in BeSweet without resampling and use this as input in the Nero 'encode files'. Nero resamples that internally to 44.1 kHz (b.t.w, I chose 'sine' 100 for playback and other purposes in the resampler setup as recommended by Ivan). Same settings 'streaming' and High efficiency. The output file is 44.1 kHz but has nasty clicks.

3. I use foobar2000 with foo_nero.dll. Same settings as above, no resampling. The output file is definitely 48 kHz and plays flawless. Strange, isn't it? I rather expected to get the same results as with direct encoding in Nero.

tiki4

[tiki4]

Originally posted by Tuning
Finally I could play with foobar2000 (..not forgetting I was winampfan till yesterday...). I have some newbie doubts, need to be clarified by someone.

What is replaygain?
How to use it?

I found it can also be used while transcoding. But i did my best to find out what is it....:( no luck. The final mp4 file didn't had any replaygain info eventhough i used it while encoding.:(

In short:
I did scan replay gain for full album. Then selected nero encoder and HE-AAC in diskwriter options. Also ticked "use replay gain' in processing. Nothing special was found in final mp4 file.

Another Q: Is it possible to transcode wma to blah format using foobar2k?
I always get foo_wma.dll failed to load error. I think this plugin is missing in installation. Where can I get it.

Thanks. :)

1. You may use replaygain to find the max volume without clipping. The thing is you can't do something like two-pass normalize in foobar2000. However, replaygaining the input AC3 files and using that info during transcoding to AAC should yield the same effect.

2. foo_wma.dll just loads correctly if you have the WM9 Codecs installed (either WMP 9, WME 9 or just the codec pack, it's on microsoft.com [bäh]).

tiki4

P.S. I tried now two times to send you PM, but it seems your mailbox is full :(

[mikeson]

@tiki4:
Actually I found out, how to maximize the sound when transcoding AC3 to AAC
Thanks a lot for this tip, tiki4. I've just transcoded 2 5.1 AC3 files to AAC (replaygained AC3 first), then tried to replaygain transcoded AAC files again and replaygain_track_gain was about 0.1 dB, so replaygain was really used while transcoding. ;)

[KpeX]

Originally posted by tiki4
An example:

1. I use BeSweet with latest Nero dlls. I choose '-6chnew', and 'streaming' + HE-AAC. Output is borked, as expected.

2. I decode to 6ch AIFF in BeSweet without resampling and use this as input in the Nero 'encode files'. Nero resamples that internally to 44.1 kHz (b.t.w, I chose 'sine' 100 for playback and other purposes in the resampler setup as recommended by Ivan). Same settings 'streaming' and High efficiency. The output file is 44.1 kHz but has nasty clicks.

3. I use foobar2000 with foo_nero.dll. Same settings as above, no resampling. The output file is definitely 48 kHz and plays flawless. Strange, isn't it? I rather expected to get the same results as with direct encoding in Nero.

tiki4

Interesting. #2 is the one that is really strange. Any chance the Nero resampler isn't very good? I'm unable to test right now but what if you decode and resample to a 44.1 KHz aiff and then feed that to Nero?

[bond]

Originally posted by KpeX
Interesting. #2 is the one that is really strange. Any chance the Nero resampler isn't very good? I'm unable to test right now but what if you decode and resample to a 44.1 KHz aiff and then feed that to Nero?yes maybe there goes something wrong with neros resample plugin or with their aiff decoder??

i got the infos i posted from menno (ahead) so i guess they are true ;)

[bobsc]

Originally posted by tiki4
3. I use foobar2000 with foo_nero.dll. Same settings as above, no resampling. The output file is definitely 48 kHz and plays flawless. Strange, isn't it? I rather expected to get the same results as with direct encoding in Nero.
This is the one I find "interesting". First foobar downsamples then encodes and finally upsamples.

[mikeson]

Originally posted by bobsc
This is the one I find "interesting". First foobar downsamples then encodes and finally upsamples.
Are you sure about this?

[KpeX]

Originally posted by bobsc
This is the one I find "interesting". First foobar downsamples then encodes and finally upsamples.

Erm...so foobar upsamples an already encoded AAC stream...I think not..don't post BS.

[bobsc]

Originally posted by mikeson
Are you sure about this?
No, but I wonder how you get 48Khz output in this case.

[bond]

according to menno it is a bug in neros aiff decoder
should be fixed in the next release...

[bobsc]

Originally posted by KpeX
Erm...so foobar upsamples an already encoded AAC stream...I think not..don't post BS.
Maybe you took my remark out of context, but your statement was uncalled-for.
If you know how the example produces 48Khz output then please explain.

[tiki4]

I also don't know, why foobar2000 is cabable to provide 48 kHz output. But the file definitely shows 48 kHz in foobar2000 and with CoreAAC and also plays at correct speed with correct length. So there is something strange going on.

About AIFF encoding in Nero: Good to hear that menno looks into that. I guess it's really a bug in resampler or AIFF decoder, AFAIK it was O.K. when encoding from 44.1 kHz file.

tiki4

[KpeX]

Foobar must have some different interaction with the Nero encoder and is able to correctly indicate that the file is 48khz rather than ignoring the resampling request (pure speculation).

[The Belgain]

Just to confirm here too that encoding from 5.1 ac3 --> 5.1 HE-AAC at 48kHz works fine with foobar (better quality and much faster than using BeSweet with resampling). Still quite slow though. What kind of speeds are people getting? I get about 3x realtime on my XP2000+.

A couple of questions though:

1. Does the channel mapping work correctly (I seem to recall that there was a problem with Nero expecting non-standard channel input?)?

2. If there any way to do delay correction automatically with foobar?

3. What's the best way to do normalisation? Replaygain?

[mikeson]

@The Belgain:

1. Yes, it has been solved long time ago AFAIK.
2. No, you have to do it yourself in muxer.
3. I use replaygain without any troubles (replaygain original AC3, check Use Replaygain in Diskwritter and transcode).

[The Belgain]

Ok, cheers. What's the best muxer to use to mux multiple aac tracks into mp4? 3ivx muxer doesn't seem to have any options for delay that I can make out.

[Stux]

3ivx Muxer of course ;)

You can use the Stream Shifter filter to add delay to various streams prior to muxing

StreamShifter + 3ivx Muxer Guide
http://www.3ivx.com/support/windows/encoding/ge_shift.html

[The Belgain]

Right, thanks for that reply, but as far as I can make out streamshifter works by delaying the video a certain amount either way and doing nothing to the audio. I'm trying to mux 2 audio tracks into the file, both of which have different delays. Streamshifter doesn't seem to do for this.

Also, is there any simpler way to use 3ivx muxer than manually doing it with graphedit for each file? I'm encoding a whole season of Alias, and manually doing this for 22 files would be a major pain in the a**.

Apart from that, since we have a member of the 3ivx team on this thread (and well done for all your work), I'd like to repoprt that the PAR settings only seem to work correctly (both in avi and mp4) if 3ivx is used as video decoder. Doesn't work using ffdshow, VideoLAN, or Nero.

[shitowax]

Our PAR in .mp4 is completly ISO standard. If a player supports MPEG-4 PAR, it should support it (last time I checked, ND showtime supported it)... There is no standard way to support PAR in .avi. We choose to use the EXACT same method than in .mov/.mp4, that way our directshow decoder performs exactly the same way if the bitstream comes from a .mp4 or a .avi. After that, there is no standard way to send the MPEG-4 DecoderSpecificInfo in directshow. We choose (and it's not a secret) to put it at the end of the VIH structure the splitter sends... Some "unsupported decoders" versions supports it, some don't. Feel free to bug the different developpers to support it.

Concerning your muxing problem, we may have something for you in the next versions, but for the moment, there is no way to add precise delay in graphedit except this streamshifter (that seems fairly broken). You have to know that if you manage to do your audio encodings, video encoding and muxing from a single muxed source in a single directshow graph, you won't have any synchronization problem...

Originally posted by The Belgain
Right, thanks for that reply, but as far as I can make out streamshifter works by delaying the video a certain amount either way and doing nothing to the audio. I'm trying to mux 2 audio tracks into the file, both of which have different delays. Streamshifter doesn't seem to do for this.

Also, is there any simpler way to use 3ivx muxer than manually doing it with graphedit for each file? I'm encoding a whole season of Alias, and manually doing this for 22 files would be a major pain in the a**.

Apart from that, since we have a member of the 3ivx team on this thread (and well done for all your work), I'd like to repoprt that the PAR settings only seem to work correctly (both in avi and mp4) if 3ivx is used as video decoder. Doesn't work using ffdshow, VideoLAN, or Nero.

[The Belgain]

Right, thanks for the quick reply...really nice to have a dev here in the forums to help people out.

So am I correct in thinking that if a 3ivx-encoded video clip is muxed into mp4, then the AR on playback will be correct regardless of decoder and player (provided it's directshow) used as long as 3ivx mp4 splitter is used?

As there's no way of doing what I need in mp4 at the moment, I think I may just use Matroska instead as a container. It should handle this ok. Keep up the good work on the mp4 front though...

And sorry about slightly highjacking this thread, I realise this is slightly OT.

[shitowax]

We can make sure our .mp4 files are 100% ISO compliant. After that, if you use a non-directshow .mp4 player not supporting the AR infos (quicktime or philips), or if you use a directshow decoder not supporting AR (DivX or some XviD or ffdshow versions), or if you use a directshow player not supporting custom AR settings (wmp8), it will not work, and there is nothing we can do.

[Stux]

But on the other hand, if you use a player which *does* support MPEG-4 PAR, then it *will* work :)

VLC, Mplayer, 3ivx, Nero and I believe the latest versions of XviD *do* support MPEG-4 PAR (and thus 3ivx's MPEG-4 PAR)

[The Belgain]

VLC doesn't seem to work by the looks of things (dunno what it was decoding with).

Are there any plans to develep the muxer into a tool one can use without Graphedit (ie something like mkvmerge gui, or avimux)? This would make the process much more hassle-free.

[Stux]

A tool which uses the muxer is in development

[bond]

now thats more than great news! ;)

[The Belgain]

Great news indeed.

[Malow]

Tnks tiki4 for the mini-guide. now is easy to make good 5.1 encoding.

[tiki4]

Huh, holy sh**,

this guide is somewhat outdated and IMHO no longer necessary. Of course, if you like to use foobar2000 then do so, but latest BeSweet/bsn.dll resolve the resampling issues, because it deals with it automatically. The only thing that comes to my mind - where you can't use BeSweet - is, if you try to encode to 128 kBit CBR 5.1 HE-AAC. Well, this isn't a good quality anyway, but Nero codec then resets profile to LC-AAC and your results are even worse.

Cheers, happy encoding!

tiki4

[mikeson]

And what about downsampling? foobar2k doesn't need to downsample 48KHz audio to 44KHz for transcoding to AAC, BeSweet did. But maybe that changed from that times... :confused:

[tiki4]

Yep. I think that is still true. Well, one might argue that downsampling is a bad idea, but most modern codecs (LAME for example) downsample if the bitrate gets too small for 48 or 44.1 kHz. BeSweet takes care for this automatically, while foo_nero does not. Actually BeSweet behaves like the Nero frontend to the codec that also downsamples.

Now, while downsampling is surely a slow process that also means that the AAC codec gets what it was trained for. The problem with resampling is, that most modern soundcards only can work with 48 kHz and it depends about the built-in resampler of the sound card if you get good quality or bad quality. Anyhow there were many discussions about this on HA and the conclusion always seemed to be that you only can hear the difference with artificial signals (sine wave) and not with real music.

So, decide for yourself now.

tiki4

P.S.: One reason that speaks for BeSweet and against foobar is the fact that BeSweet uses azid.dll while fb2k uses liba52 for decoding AC3. The latter should have slightly worse quality than azid. I think that was stated by KpeX some time ago.

[bond]

Originally posted by tiki4
Actually BeSweet behaves like the Nero frontend to the codec that also downsamples.hm i am not so sure about that as last time i tested it nero didnt downsample a 48khz 2.0 input wav in nero burning rom, when encoding to aac (even he-aac)

[bobsc]

Originally posted by tiki4
P.S.: One reason that speaks for BeSweet and against foobar is the fact that BeSweet uses azid.dll while fb2k uses liba52 for decoding AC3. The latter should have slightly worse quality than azid. I think that was stated by KpeX some time ago.
Azid and liba52 decoding (http://forum.doom9.org/showthread.php?threadid=68147&highlight=ac3+decoder)

[tiki4]

Originally posted by bond
hm i am not so sure about that as last time i tested it nero didnt downsample a 48khz 2.0 input wav in nero burning rom, when encoding to aac (even he-aac)

Hm. You may be right. I think this behaviour changed several times in the Nero 6.x series.

@bobsc:

Thanks for the link. I think one can conclude from KpeX post that one won't hear the difference between azid.dll and liba52. But it is also true that azid offers much more options.

tiki4

[KpeX]

Originally posted by tiki4
I think one can conclude from KpeX post that one won't hear the difference between azid.dll and liba52. But it is also true that azid offers much more options. These are my feelings on the issue. I doubt that one could ABX a liba52 decoded wav versus an azid decoded wav....but there's only one way to know for sure. IMO azid's impressive amount of options is one of the main reasons besweet should be used to transcode to Nero 5.1 AAC, the other being that it correctly resamples as the encoder requests.

But lately I find myself using FAAC more often than Nero. Hopefully this will be supported in BeSweet one day soon :)

[bobsc]

Originally posted by KpeX
But lately I find myself using FAAC more often than Nero. Hopefully this will be supported in BeSweet one day soon :)
FAAC's development seems to be at a standstill since knik's departure, but hopefully someone will pick up the torch.

[KpeX]

Originally posted by bobsc
FAAC's development seems to be at a standstill since knik's departure, but hopefully someone will pick up the torch. Agreed, I'd hate to see the project die, especially with the quality improvements as of late. It is likely that the project will attract more developers however, since it is the major open source aac project.

[jimmy basushi]

i tested besweet to foobar alot for ogg and aac. foobar won on ability to use different versions for ogg and aac and simplicity of batch encoding. does anyone know a site like rarewares that release new ogg libvorbis file? cli encoder gives you the ability to use floating files also, but they cant be longer then 1.5 hours for some reason.

i get lot of stereo collase if i use besweet/nero also for some reason yet when i use foobar it comes out perfect for low bitrate he-aac? kpex do you use the foobar input for faac? i gave it a few try quality was good.

[KpeX]

Originally posted by jimmy basushi
i tested besweet to foobar alot for ogg and aac. foobar won on ability to use different versions for ogg and aac and simplicity of batch encoding. does anyone know a site like rarewares that release new ogg libvorbis file? cli encoder gives you the ability to use floating files also, but they cant be longer then 1.5 hours for some reason. Libvorbis is available on rarewares in several versions, and also on the xiph/vorbis homepages.Originally posted by jimmy basushi
i get lot of stereo collase if i use besweet/nero also for some reason yet when i use foobar it comes out perfect for low bitrate he-aac? I don't fully understand what problem you're having with bsn. Please restate and/or post logfiles. Originally posted by jimmy basushi
kpex do you use the foobar input for faac? i gave it a few try quality was good. I usually decode with BeSweet to six channel wav (due to besweet's numerous decoding options) and encode six channel wav to MP4 with FAAC. I haven't tested yet how well FAAC handles 24/32 bit/floating point wavs.

[jimmy basushi]

i was talking about playback i used to be able to hear a slight difference but i havent tried it again for a while. stereo collapse = when you hear both channels sounding like mono for a tiny second.
i look at rarewares, i cant get one like aotuv. only gt3b2 and 1.0.1

[P0l1m0rph1c]

Originally posted by jimmy basushi
i was talking about playback i used to be able to hear a slight difference but i havent tried it again for a while. stereo collapse = when you hear both channels sounding like mono for a tiny second.
i look at rarewares, i cant get one like aotuv. only gt3b2 and 1.0.1

Sorry, but what does this have to do with AAC? Both aotuv, gt3b2 and 1.0.1 are Vorbis encoders, i don't get what do they have to do with this subject...

Anyway, you can find aoTuv here (http://www.geocities.jp/aoyoume/aotuv/)

[mikeson]

So may I ask what more experienced users that I am recommend?

Transcoding via BeSweet AND downsampling or transcoding via fb2k and NOT downsampling? What do you think is more effective and in which way can we achieve better results?

[KpeX]

I wouldn't say one is clearly better than the other - however, when using BeSweet you have more options and you are using the Nero AAC encoder as its developers intended.

[jimmy basushi]

Originally posted by P0l1m0rph1c
Sorry, but what does this have to do with AAC? Both aotuv, gt3b2 and 1.0.1 are Vorbis encoders, i don't get what do they have to do with this subject...

Anyway, you can find aoTuv here (http://www.geocities.jp/aoyoume/aotuv/)

in pervious post, we talked about getting different ogg versions look back. and that link doesnt have libvorbis dll's that besweet uses? i try downloading them all.

and on audio test, i work out why aac sound strange.. it was a dshow problem.

[porcinemodjo]

i realize this post will be breaking a couple of rules here...but im really on the edge.
Apart from the fact that tiki says this guide is no longer valid, i used foobar to encode an ac3 file to an HE-aac with the nero plugin. However, the only option for HE-aac caused the output to be in the form of an mp4 file. I tried demuxing the aac from the mp4 file with ivan and menno + mp4creator, but it creates a batch file which doesnt work.

I also saw the option of muxing the mp4 file itself with my xvid encoded avi file, but i want my final file to be an avi and not an mp4.

The thing is that ive done this before, but i cant remember what i did to do it. Something to do with matroska. Im sure the answer to my prob is littered all over the forum, but i dont now WHAT to look for exactly.

to summarize:
1)can i get the nero plugin to give me an aac file instead of an mp4? how?
2)or how else can i demux my mp4 to its aac file
3)or how can i mux an mp4 file into an avi

I would appreciate any help.

PS im a real noob, so be gentle :(

[SeeMoreDigital]

Originally posted by porcinemodjo
...to summarize:
1)can i get the nero plugin to give me an aac file instead of an mp4? how?
2)or how else can i demux my mp4 to its aac file
3)or how can i mux an mp4 file into an avi There's a great "easy-to-use-tool" called mp4UI, you could try!


Cheers

[pogo stick]

Originally posted by porcinemodjo
1)can i get the nero plugin to give me an aac file instead of an mp4? how?
2)or how else can i demux my mp4 to its aac file
3)or how can i mux an mp4 file into an avi
Actually all your questions would be solved if you would read AAC FAQ (http://forum.doom9.org/showthread.php?s=&threadid=68300#post424070) and MP4 FAQ (http://forum.doom9.org/showthread.php?threadid=62723).
It's very popular and interesting to read!

1) There is Export ISO 13818-7 in Nero encoder.
2) Foobar2000 can do it too. I don't remember how this option called. Just click right button on mp4 file and you'll figure it out easily.
3) Both mp4 and avi are container formats and you can't put one container to another.

[JnZ]

Originally posted by porcinemodjo

to summarize:
1)can i get the nero plugin to give me an aac file instead of an mp4? how?
2)or how else can i demux my mp4 to its aac file
3)or how can i mux an mp4 file into an avi

I would appreciate any help.

PS im a real noob, so be gentle :(

1) In Nero encoder settings set Export ISO 13818-7. This will give you mp4 and aac.
2) AVImux can mux AAC's to AVI.

[porcinemodjo]

thanks to all for the help...the Export ISO 13818-7 did the job. I had tried it earlier, but when i checked the o/p directory while it was encoding, i only saw the mp4 file, so i cancelled the conversion. I didnt kow it created the aac file after completion of the mp4.

Thanks to all once again.

[SeeMoreDigital]

It should look something like this: -
http://img105.exs.cx/img105/3600/NeroDigital_GUI_in_Foobar2000.png


Cheers

[neo_anderson]

i have an ac3 file which has a delay of -49ms, so how to set delay in foobar2000?

[JnZ]

Originally posted by neo_anderson
i have an ac3 file which has a delay of -49ms, so how to set delay in foobar2000?
You don't need set delay in foobar2000. You set delay in mux program (VDM, Avimux, etc.). I think that every mux program has this feature.

[killerhex]

can foobar encode AAC 5.1 back to AC3 5.1
for ac3 is easier to decode when playing back

[shon3i]

can foobar encode AAC 5.1 back to AC3 5.1
for ac3 is easier to decode when playing back
Yes, of course, but you need free ac3 cli encoder called Aften.

[DAKnn]

As it is possible convert to MP4/AAC(Multichannel) if foo_nero.dll any more does not work with the new version Foobar2000 0.9.x

[tebasuna51]

As it is possible convert to MP4/AAC(Multichannel) if foo_nero.dll any more does not work with the new version Foobar2000 0.9.x

Use NeroAacEnc free and better than old dll method.