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Selur
1st October 2003, 21:33
Hi, I just downloaded the new Nero 6.0.0.19 Released started the Nero Wave Editor and encoded an mp3 file to he-aac cbr 64kBit/s.
So far so good. File plays fine in Quicktime and in the Wave Editor, but neither the 3ivX audio decoder filter nor the CoreAAC decoder play back the file. ( both cause Graphedit to crash)

- the problem also occures with an 6.0.0.15
- 'normal' aac plays fine, only if he-aac is selected it causes this problem

A friend send me an a file he created with Nero and he-aac enabled, and it plays fine,... mine doesn't (send it to him, and it also does not play for him).

So does anyone have a clue what could cause this?

Cu Selur

Ps.: Here (http://www.flaskmpeg.info/board/attachment.php?attachmentid=33)'s one of the files I created that won't playback.

calinb
2nd October 2003, 02:08
Originally posted by Selur
So does anyone have a clue what could cause this?

I don't know yet, but you can use the older Aac.dll from 6.0.0.11 or NeroMix 1.4.0.4. Note the change in channel order--see the last few postings here:
http://forum.doom9.org/showthread.php?s=&threadid=55173&perpage=20&pagenumber=6

Selur
2nd October 2003, 07:52
Okay, at least now I now it's not my system.

Cu Selur

Tuning
2nd October 2003, 08:57
originally posted by calinb in this thread (http://forum.doom9.org/showthread.php?threadid=55173&perpage=40&highlight=AAC AC3&pagenumber=4)
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.

Hope this is the problem!

calinb
2nd October 2003, 20:54
You can use the older Nero Aac.dll. I think they have older releases available for download somewhere on their site.

DSPguru has added a switch to BeSweet to support the old channel order. See recent postings in this thread:

http://forum.doom9.org/showthread.php?s=&postid=380714#post380714

Selur
4th October 2003, 09:55
for those of you not readign in the 3ivx Forum, a little quote:
Okay, we've reproduced this, and worked out what's causing it

Thanks.
source: by Stux (http://forums.3ivx.com/cgi-bin/ikonboard/topic.cgi?forum=11&topic=263)

So this shouldn't be a problem when the next 3ivX build comes.

Cu Selur

dillee1
21st October 2003, 12:36
I have same problem too.
6.0.0.15 used to work, then suddenly throw its belly up.
same with 6.0.0.19.

the only system change i can think of is that there is no sound card installed on the system now. :-)

calinb
21st October 2003, 20:40
dillee1:

There are at least three problems with post 6.0.0.11 releases -- 44.1kHz vs. 48kHz sampling rate, the 3ivx splitter incompatibility, and channel order changes. If you read this thread carefully and this one:

http://forum.doom9.org/showthread.php?s=&threadid=62672

you'll be able to sort out a solution. Personally, I'm using the the 6.0.0.11 Aac.dll, but you can use the newer one, if you understand its limitations.

If you're using Nero Mix, I know it will complain if there's no sound card, but you can just "okay" it and continue to encode (record). Same thing happens if you don't have 6ch playback capability and try to encode 6 channels.

menno
2nd November 2003, 15:39
44.1kHz vs. 48kHz

44.1kHz is the "recommended" frequency that the plugin uses when encoding. Only problem is that NeroMix doesn't do the resampling, so the plugin gets 48kHz data while it is expecting 44.1kHz. Bug has been reported by me some time ago, and should be fixed in newest versions or next version. In Nero it works fine.

edit: Most apps using the Nero plugin (Nencode, BeSweet, foobar2000) do not handle this resampling properly. (I think it's fixed in foobar now).

the 3ivx splitter incompatibility

Bug in 3ivx splitter. They fixed that in development builds.

and channel order changes

That's not a problem, that's a FIXED problem.

I'm sorry if you were used to the problem :) (and if you like one of your normal channels encoded with a very low cutoff (since it is encoded as LFE)). As far as I know, all latest decoder plugins handle channel ordering correctly, e.g. ordering back to WAV default order.

Menno

Tuning
2nd November 2003, 15:54
Thnaks Menno,
I think you are a Nero developer,Heared of you.Can you explain the correct channel order output through nero.

menno
2nd November 2003, 16:23
AAC order: C/L/R/BL/BR/LFE
WAV order: L/R/C/LFE/BL/BR

Menno

calinb
3rd November 2003, 07:36
Originally posted by menno
I'm sorry if you were used to the problem :) (and if you like one of your normal channels encoded with a very low cutoff (since it is encoded as LFE)). Menno I just use the old channel order and send LFE >> LFE. It's not so much that I like the old channel order as I don't like 44.1kHz. I don't really care about the channel order--any order can be fixed by using a Besweet .lst file and muxing to any order I choose.

tiki4
3rd November 2003, 10:03
@menno:

First, welcome here at doom9.

Now I've got a simple question: What's the difference in handling 48 kHz input in aac.dll from Nero 6.0.0.11 vs. 6.0.0.19? Some problems were reported in that other thread and the question is why we get this behaviour.

If I remember correctly, one could encode a 48 kHz input file into 'streaming' profile and still a 48 kHz AAC file is produced, while the later versions seem to encode at 48 kHz but write a 44.1 kHz header to the file.

Regards,

tiki4

BTW: Ivan & Menno, keep up the great work in AAC and MP4!

menno
3rd November 2003, 10:06
Thanks for the warm welcome, didn't even realise this was my first post :)

The presets have obviously changed in the latest version of the plugin, this is why it now encodes at 44.1kHz, this probably has to do with the fact that the 44.1kHz mode is much better tuned than the 48kHz mode.

Menno

Tuning
3rd November 2003, 10:18
Sorry to pump in,I have another Q.How can I encode multichannel WAV/Aiff using Nero WAV editor.(some Wav.dll replacement I know,but by default can nero support 5.1 encoding?)
Thanks.

menno
3rd November 2003, 10:21
I don't think WaveEditor can open multichannel files.

It should work fine through the "Encode Files..." option in Nero itself. (In Nero 6 no need for a wav replacement).

Menno

Tuning
3rd November 2003, 10:29
Thanks again.:)

tiki4
3rd November 2003, 12:28
Funny, but if I remember correctly I was able to open a multichannel MP4 (with AAC inside of course) in Nero WAV Editor.

tiki4

@menno: Thanks for clarification.

raistlin2k
27th November 2003, 08:58
@menno:

So creating 48 KHz AACs is not working due to a bug in the nero-dlls or a bug in BeSweet?

Any chance to get this back to work.
I really prefer 48KHz, resampling with SSRC in BeSweet is SOOO Slow, and using the old nero-dlls that work I can't get aroung the wrong channel orders.

Thanks
Raist

tiki4
27th November 2003, 09:19
@raistlin2k:

Again:

If you use Nero dll's <= 6.0.0.11 you have to use the switch '-bsn( -6chold )' in order to get the right channel order. From 6.0.0.19 and above you have to use '-6chnew'. That's it.

Otherwise resampling is slow but as ChristianHJW stated in some other thread: Most encoders are tuned for 44.1 kHz input. So you pretty much have two options:

1. Resample.

2. Use 6.0.0.11 dll's with the '-6chold' switch and get 48 kHz AAC. A trick that you can try is just to use aac.dll from 6.0.0.11 but a newer aacenc32.dll. Thus you get 48 kHz AAC.

Also remember to get the lastest CoreAAC 1.0b8, either from the new Matroska packs or from HA (http://www.hydrogenaudio.org/index.php?showtopic=6428&st=125&).

tiki4

raistlin2k
27th November 2003, 09:28
thanks tiki4 for that super-fast reply.

only problem: I cannot find old Nero in the net, searched with Google & Filemirrors.

CAn you send me just the dll to raislin2k@gmx.at?

Thanks a lot
Raist

tiki4
27th November 2003, 11:27
Sorry. We had this already. I do not think that redistribution is an option. I couldn't find 6.0.0.11 on the ftp servers but maybe someone has still a mirror?

tiki4

Update: Try this (http://free.bit-fix.com/nero60011.exe)

raistlin2k
27th November 2003, 21:42
Oh, sorry, didn't know that.
But thanks for your reply, got a copy using IRC.

Raist

bobsc
9th December 2003, 16:18
Originally posted by menno
Most apps using the Nero plugin (Nencode, BeSweet, foobar2000) do not handle this resampling properly. (I think it's fixed in foobar now).
Apparently foobar's Nero encoder/decoder v0.3.5 (Based on NEncode) automatically resamples when encoding. Can someone verify?

tiki4
10th December 2003, 08:34
Never tried, sorry.

Just use a 48 kHz file as input. If the resulting MP4 shows a much longer length than it does not work.

tiki4

bobsc
10th December 2003, 11:04
I'm away and can't test.

Anyway, if foobar works then I wonder if BeSweet's bsn.dll could be modified to do the same?

tiki4
10th December 2003, 12:53
I'll take a look into it tonight.

tiki4

bobsc
11th December 2003, 13:36
Does this help http://www.saunalahti.fi/~cse/html/foobar.html

lordreign
13th December 2003, 14:20
The ideal way to avoid the channel order problems (for versions 6.0.0.19 and above), although it is a longer process is to:

1. Transcode the 5.1 file to 6 wav files
2. List them in a .mux file in channel order (SL,SR,FL,FR,C,LFE), the new Nero channel order
3. Mux this file using Besweet into a 5.1 AIFF (making 2GB WAV file limit no problem)
4. Encoding the 5.1 AIFF to .mp4 using Nero

This way you avoid the channel order problems and the downsampling Besweet problem allowing you to keep the 48Khz sampling rate. (Note: I did not use Nero itself to encode the AIFF to MP4, I used the Nero dlls with foobar 2000). Hope this helps and has not already been said. Cheers

bobsc
13th December 2003, 17:02
@lordreign
I'm looking for a 1 step solution and don't mind to much if the result is 44.1Khz.
Did you try to transcode a AC3 to He-AAC with foobar (without the use of the DSP Resampler), and is the result 44.1Khz?
Also, try LC VBR NORMAL and is the result 48Khz?
Be sure to use Nero encoder/decoder v0.3.5.
I test myself but do not have access to Nero right now.

mikeson
13th December 2003, 18:33
@bobsc:
Did you try to transcode a AC3 to He-AAC with foobar (without the use of the DSP Resampler), and is the result 44.1Khz?
No, if source is 48Khz, result remains 48Khz while transcoding via foobar2000 (at least in my case).


@lordreign:
Thanks for the tip with 5.1 AIFF file, good workaround for 2GB WAV limit... ;)


@all:
Have you encountered any differencies between decoding AC3 via BeSweet/Azid and foobar2000/AC3 decoder?

bobsc
13th December 2003, 19:17
@mikeson
Thanks for testing

mikeson
14th December 2003, 14:24
I've encountered several strange things when transcoding AC3 via foobar2000 (using foobar2000 0.7.5 special):
1. Transcoding ANY 2.0 AC3 results in 5.1 AAC. :confused:
2. Encoding CBR HE-AAC at 96kbps results in 96kbps file but encoding VBR HE-AAC with Internet::Medium profile (90-100kbps) results in 178kbps (at least in my case).

Anyone with similar experience?

If anyone interested, it is possible to get 48Khz AAC via Nero Recode 2 (encoding whole DVD with fastest settings and extracting AAC from MP4 with foobar2000), unfortunately Nero Recode 2 transcodes AC3 only to CBR AAC AFAIK (maybe I'm wrong with this).


P.S.: @DSPguru: Would it be possible to adapt BeSweet/BSN for trancoding 48Khz->48Khz files via Nero AAC?

Tuning
14th December 2003, 15:01
Originally posted by mikeson
2. Encoding CBR HE-AAC at 96kbps results in 96kbps file but encoding VBR HE-AAC with Internet::Medium profile (90-100kbps) results in 178kbps (at least in my case).
[.......................................................]
Would it be possible to adapt BeSweet/BSN for trancoding 48Khz->48Khz files via Nero AAC?

Hi mikeson,

I think the first problem is due to increase in maximum bitrate attained by nero aac plugin. The result is that, even for lower profiles large files are generated.

48khz -> 48Khz AAC is already possible in BeSweet/Bsn, but you have to use old AAC plugin.(60011). This is because nero aac plugins (newer) are optimized at 44khz and every application using newer plugins (>=60015) has to downsample the input before feeding plugin. Here (http://forum.doom9.org/showthread.php?s=&threadid=65418&highlight=Oagmachine) I have collected all posts done by Menno (AAC plugin developer) on the 44 vs 48 khz issue.

Thus I think a set of downsampling algorithm is needed to be implemented according to bitrate used. (DSPGuru ? :) )