Nero AAC Codec 1.3.3.0

[menno]

We just released a new version of our command line AAC encoder, decoder and MPEG-4 tagger applications on our website.

EDIT:
The webpage for the Nero AAC Encoder on the Nero website is currently unavailable. For a direct download of the package click here (ftp://ftp6.nero.com/tools/NeroDigitalAudio.zip).
A temporary information page can be found here (http://www.audiocoding.com/nero_aacenc.html).

Release notes for version 1.3.3.0:

neroAacEnc:
Improved multichannel encoding
Solved compatibility issues with some hardware devices
Improved CBR bitrate control
Ignore wave file length of 0 on -ignorelength
Solved issue with very long files and -2pass
Removed special SSE compile, all optimizations have been merged with the normal compile
Linux version speed optimized
Encoder exits gracefully and writes valid MP4 file on ctrl+c

neroAacDec:
Fixed incorrect channel output for 7.1 encodings

neroAacTag:
Linux version available

[sehgal.v7]

Cheers!!

[fibbingbear]

This is awesome :) ND-AAC is my favorite. Thanks for letting us know ahead of time :) :) :)

[buzzqw]

Fixed incorrect channel output for 7.1 encodings

does it mean that Nero AAC is able to encode to 7.1 aac ?

BHH

[menno]

does it mean that Nero AAC is able to encode to 7.1 aac ?


Yes it does

[Gusar]

Linux version speed optimizedThank you especially for that one.

[burfadel]

The title in the link still shows Nero 1.1.34.2, however the download is 1.3.0.0.

[menno]

The title in the link still shows Nero 1.1.34.2, however the download is 1.3.0.0.

Yeah this is known, that's why I mentioned it in the first post ;)

[lexor]

So what is the difference between this and the encoder in Nero suit?

[menno]

So what is the difference between this and the encoder in Nero suit?

Technically there is no difference between the command line encoder and the DLL included with the suite. In the suite the applications usually determine what options and possibilities are exposed to the user (that usually means less than is actually possible). The command line encoder gives more freedom in that sense.
The version currently in the suite is from the 1.1 version range, with some bug fixes compared to the previous command line encoder. The 1.3 versions will be in the next generation of Nero Suite.

[madshi]

Thank you for the new version! :)

A few questions, if you don't mind:

(1) Is 6.1 encoding supported? If so, in which order do you expect the channels?

(2) Did you find the time to analyze if 2pass encoding really produces superior results compared to 1pass encoding? If so, can you describe in which situation 2pass encoding has advantages and in which maybe not? And what are the exact advantages? (Still trying to decide whether it's worth it to add 2pass encoding support to eac3to. The main problem with 2pass encoding is that I can't pipe the data directly to the encoder via stdin).

Thanks a bunch!!

[cubanresourceful]

So, I wonder, what would be the difference between using NeroAAC or iTunes encoder? If you don't mind me asking, I can use both through CLI, (though, only NeroAAC works in foobar for me).

[tebasuna51]

Thanks for the new version!
Really the multichannel encode is improved.
The "Ignore wave file length of 0 on -ignorelength" is also welcome.

@madshi
1) Seems 6.1 isn't supported:
"ERROR: could not open AAC encoder"

2) I think the ABR encode (the only than need a 2pass method) have a main advantage over VBR: know the final file size.
But the criterion to encode must be the quality instead the file size.
Talking about movie tracks (eac3to scope), the differences are very big and the needed bitrate can be unpredictable.

[menno]

thanks for answering that tebusana51.

We currently support mono, stereo, 5.1 and 7.1 (and 2.1, but good luck decoding that properly). More general input configuration support is on the list for next version.

[LoRd_MuldeR]

If so, can you describe in which situation 2pass encoding has advantages and in which maybe not?

I'd guess that 2-Pass has a significant advantage for a source with a long "silent" part at the end. While 1-Pass cannot look ahead (at least not very far) and has to reserve bits for the upcoming parts, 2-Pass can move the bits to the parts where they are really needed and it doesn't need to reserve any bits for "unknown" (not yet processed) parts...

And what are the exact advantages?

2-Pass should deliver "constant quality" in ABR mode, while 1-Pass ABR can only optimize locally...

[hubblec4]

does it mean that Nero AAC is able to encode to 7.1 aac ?

BHH

How can I make a 7.1encode with neroaac.
I use eac3to. there i load a 7.1channel wav or a 7.1 dts hi-res file.
as output i use neroaac quallity 0.35 but the encoder shows the message neroaac can only convert to 6channel and downconvert the surround-channles to 5.1

maybe madshi can help??


hubble

[tebasuna51]

How can I make a 7.1encode with neroaac.

Until madshi modify eac3to you can use this command line:

NeroAacEnc -q 0.35 -if 8_Ch.wav -of 8_Ch.mp4

Warning: At least my ffdshow version and Bass libraries can't decode properly 7.1 mp4.
Faad, Foobar2000 and of course neroAacDec seems decode without problems.

[skromnibog]

I'd guess that 2-Pass has a significant advantage for a source with a long "silent" part at the end. While 1-Pass cannot look ahead (at least not very far) and has to reserve bits for the upcoming parts, 2-Pass can move the bits to the parts where they are really needed and it doesn't need to reserve any bits for "unknown" (not yet processed) parts...
2-Pass should deliver "constant quality" in ABR mode, while 1-Pass ABR can only optimize locally...
Great answer! Thank you!

[tebasuna51]

We currently support mono, stereo, 5.1 and 7.1 (and 2.1, but good luck decoding that properly). More general input configuration support is on the list for next version.

Good to know, thanks.

There are different bitrate range for 1, 2, 6 or 8 channels?
Valid values for -br and -cbr parameters?

[hubblec4]

Until madshi modify eac3to you can use this command line:

NeroAacEnc -q 0.35 -if 8_Ch.wav -of 8_Ch.mp4

Warning: At least my ffdshow version and Bass libraries can't decode properly 7.1 mp4.
Faad, Foobar2000 and of course neroAacDec seems decode without problems.

thanks for helping.

right, i can encode it but i cant hear it. i guess the coreaac-decoder filter dont support 7.1

hubble

[Kurtnoise]

Linux version speed optimized

http://pix.nofrag.com/d/3/6/f04d84c7c8d1708b163e70df8f535.png (http://pix.nofrag.com/d/3/6/f04d84c7c8d1708b163e70df8f535.html)

http://pix.nofrag.com/c/7/5/8b288473f3a1305c40769f67d984c.png (http://pix.nofrag.com/c/7/5/8b288473f3a1305c40769f67d984c.html)

nice...:)

w/ Core2 Duo T550 @1.83GHz

[menno]

Good to know, thanks.

There are different bitrate range for 1, 2, 6 or 8 channels?
Valid values for -br and -cbr parameters?

I won't post the exact ranges here, but in general, 3 times more channels will have 3 times higher bitrate limits, etc :)

[shon3i]

Good to know, thanks.

There are different bitrate range for 1, 2, 6 or 8 channels?
Valid values for -br and -cbr parameters?

For 6 channels:

HE-AAC 192-256kbps (128kbps in some cases)
LC-AAC everything above.

If you want good/transparent quality.

for 8 channels maybe HE-AAC @320 and above.

[tomos]

hi, is this multithreaded?

also, can anyone recommend a good GUI for this (in windows i mean)?

thanks guys :)

[LoRd_MuldeR]

also, can anyone recommend a good GUI for this (in windows i mean)?

I might be prejudiced, but:
http://forum.doom9.org/showpost.php?p=920811&postcount=1

And this one is multi-threaded, independent from the encoder in use ;)

[tomos]

ahh, nice.

will check it out. thanks mate :)

[KornX]

Hey menno,

are there any plans for that?
http://en.wikipedia.org/wiki/MPEG-4_SLS

Just curious!

KornX

[menno]

Hey menno,

are there any plans for that?
http://en.wikipedia.org/wiki/MPEG-4_SLS


Plans yes (for a lot of things, BTW ;) ), but nothing has been decided yet. The technology is certainly very interesting.

[menno]

New link: http://www.nero.com/eng/downloads-nerodigital-nero-aac-codec.php

[LoRd_MuldeR]

New link: http://www.nero.com/eng/downloads-nerodigital-nero-aac-codec.php

Damn, this breaks the suggested download link in my applications :D

Why did you guys not add a redirect at least? :p

[madshi]

This page is still not updated:

http://www.nero.com/eng/down-ndaudio.php

Will that be updated sooner or alter? I'm asking because eac3to polls this page to find out whether there's a new Nero AAC version available. Or is there a better page I should check? It would be nice if you could keep the URL stable. Thanks! :)

[menno]

I think that page will be removed, but I'm not sure. The web team has been notified that this page is still there with the old version numbers.

[madshi]

Will the new URL stay constant for future updates?

I'm wondering whether it makes any sense to keep an "is a new version available?" check in my tool if Nero keeps changing the URLs with every new version! ;)

Thanks!

[LoRd_MuldeR]

I think that page will be removed, but I'm not sure. The web team has been notified that this page is still there with the old version numbers.

They could add a simple redirect instead of removing it...

And they should remember the existence of this URL too:
http://www.nero.com/enu/down-ndaudio.php


Will the new URL stay constant for future updates?

I hope so, too :)

Of course one could check for "http://www.my-server.com/check-nero.php" and redirect to the actual Nero site from there.
This way you could keep URL changes transparent to the application, but it adds dependency to another server...

[Kurtnoise]

why not using the ftp instead ?

ftp://ftp6.nero.com/tools/NeroDigitalAudio.zip

[menno]

Yes, I think the ftp location will remain constant. The current change in web address is mostly due to the total redesign that was launched today, I don't expect this to happen often. The old address has been constant for at least a couple of years.

[LoRd_MuldeR]

why not using the ftp instead ?

ftp://ftp6.nero.com/tools/NeroDigitalAudio.zip

I don't think the Nero AG will be happy if I deeplink to files on their FTP server, skipping the license agreement :p

[madshi]

I don't think the Nero AG will be happy if I deeplink to files on their FTP server, skipping the license agreement :p
Also I'd have to completely download the file to check if it's new or not. What I want is just notifying the user that there's a new version available.

Anyway, if there's hope that the new web URL will stay the same for a while then that's good enough for me. Thanks!

[LoRd_MuldeR]

Also I'd have to completely download the file to check if it's new or not. What I want is just notifying the user that there's a new version available.

You could check the file date without downloading it...

[madshi]

You could check the file date without downloading it...
True.

[plonk420]

this encoder blew my mind ... after looking at multiple multi-channel encoders, this one worked on my first try...! (source was 24-bit 5.1 48khz audio from DVD-A discs) no issues with winamp, MPC, or even quicktime that somehow got loaded on one of the computers here at work...

i really didn't want to have to do a 24bit-to-16bit conversion for each mono file for BeSweet or the old DTS encoder. thanks guys!

[JustABDFan]

Is there any way to tell neroaacenc to output a raw aac stream instead of an .mp4?

[menno]

Is there any way to tell neroaacenc to output a raw aac stream instead of an .mp4?

No, only mp4 is supported

[Wedgedkc]

Great job! Any chance of an osx version?

[LoRd_MuldeR]

Is there any way to tell neroaacenc to output a raw aac stream instead of an .mp4?No, only mp4 is supported

MP4Box can be used to extract raw AAC from MP4 container...

[menno]

Great job! Any chance of an osx version?

It's on the list and we understand that many people would like this, but no guarantees that it will ever come.

[lexor]

Shouldn't the Linux version run in command prompt on OS X? I recall running a bunch of linux apps under osx back in university, hell I even ran Octave (matlab clone), and that baby is anything but a simple app (it did require installation of X11 though).

[LoRd_MuldeR]

If possible at all, then a Linux-x86 build would run on OS X on Intel-x86 CPU only. Running on OS X on PowerPC CPU is impossible, except for using an emulator (and loosing a LOT of speed) maybe...

[Selur]

Would be nice if one could set the channel order in neroaacenc so when piping multichannel ac3/dts/.. from ffmpeg to neroaacenc one would not have a wrong channel mapping,...

[menno]

Why doesn't ffmpeg output MS channel order?

[madshi]

Why doesn't ffmpeg output MS channel order?
I think they're working on these things at the moment. Right now every ffmpeg decoder outputs in its own native channel order. But that's probably about to change soon, as far as I've read on the ffmpeg news list...

[menno]

Ok, good to know. This channel reordering option will not be added to NeroAACEnc. When accepting WAV as input I don't think it's unreasonable to just expect MS channel ordering.

[madshi]

Actually a WAV file with incorrect channel ordering is plain broken IMHO. Things may be different with raw/PCM data. But with WAV the channel order is clearly specified.

[LigH]

There is a difference between the usual channel order in AC3 and the one in WAV and AAC. I guess ffmpeg just keeps the order from the AC3 source and passes that into the WAV file. If I remember right, the MUX file wizard in BeLight uses the same order for WAV and AAC, but a different for AC3 (in case someone uses BeSweet to encode AC3). This information might help constructing a "mixer" parameter for SoX to get a reordering in the middle of the pipe...

[tebasuna51]

Seems sox don't accept the incomplete wav header send by ffmpeg (worse than the old neroaacenc):
ffmpeg -i test.ac3 -acodec pcm_s16le -ac 6 -f wav -ar 48000 - | sox -t wav - -t wav - remix 2 4 3 1 5 6 | neroAacEnc -ignorelength -if - -of audio.mp4
...
\sox wav: Length in output .wav header will be wrong since can't seek to fix it

av_interleaved_write_frame(): Error while opening file

We need a intermediate wav file to do the conversion properly:
ffmpeg -i test.ac3 -acodec pcm_s16le -ac 6 -f wav -ar 48000 z1.wav
sox -t wav z1.wav -t wav - remix 2 4 3 1 5 6 | neroAacEnc -ignorelength -if - -of audio.mp4

BTW, Windows users only need:

eac3to test.ac3 audio.mp4

(neroaacenc at same folder than eac3to, libav decoder used like ffmpeg).

[Kurtnoise]

for *nix users, there is still the mkfifo trick that should work too...

[Selur]

@tebasuna51: eac3to depends on some directshow filters, so at least for me it's not an alternative to use ffmpeg+neroaacenc
@Kurtnoue13: care the explain further? (link? details?)

sox wav: Length in output .wav header will be wrong since can't seek to fix it
seems like we are back at the problem, that ffmpeg outputs 'broken' headers :[

[madshi]

@tebasuna51: eac3to depends on some directshow filters, so at least for me it's not an alternative to use ffmpeg+neroaacenc
eac3to doesn't depend on DirectShow filters. eac3to uses some DirectShow filters if they are available, because some of them provide better quality compared to libav/ffmpeg. But if those DirectShow filters are not available, eac3to can get along just fine, too, by using the libav library distributed with eac3to.

[tebasuna51]

@tebasuna51: eac3to depends on some directshow filters, so at least for me it's not an alternative to use ffmpeg+neroaacenc

Like madshi say you, don't need directshow filters, the -libav (avcodec.dll, avutil-49.dll) included is the same than use ffmpeg but better:
- Correct channel mapping
- Don't lose precission (32/24 bitdepth instead 16)
- Don't apply Dialog Normalization
- Don't apply Dynamic Range Compression (BTW the -drc_scale parameter in ffmpeg don't work for me in Sherpya-r14277)

seems like we are back at the problem, that ffmpeg outputs 'broken' headers :[

There are two fields in wav header (RiffLength = FileLength - 8, and DataLength) that can't be know when begin the decode of VBR streams, then the problem can be at the soft than accept STDIN but abort if these fields aren't correct.

This last NeroAacEnc version work fine now with the parameter -ignorelength. Is a Sox problem.

[LiFe]

If you want to convert AC3 to AAC, and you're having trouble with directshow filters, or you have to use an intermediate wav anyway, just use Azid. I convert all my AC3 to wav using Azid set to Maximum (peak volume set to 100%, no DRC or normalisation), then I convert all the wav to AAC using SNG front end. Let me know if you want more details.

[Selur]

can one pipe into and outoff azid ?

[buzzqw]

azid don't accept or output to pipe :(

BHH

[tebasuna51]

To use azid and neroaacenc you can use BeSweet/BeLight.

If you have:
NeroAacEnc.exe at BeSweet folder and
BeSweet v1.5b31 by DSPguru.
Using azid.dll v1.9 (b922) by Midas (midas@egon.gyaloglo.hu).
Using bsn.dll replacement by Dimzon & Kurtnoise, (http://kurtnoise.free.fr/index.php?dir=BeLight/&file=bsn_20070513.zip) Build May 14 2007, 00:00:26

the command line needed is:

BeSweet -core( -input 6a321.ac3 -output 6a321.m4a ) -azid( --maximize ) -bsn( -vbr 0.35 -6chnew )

[Selur]

@tebasuna: want to use the method under Linux&Windows so besweet isn't really an option,.. :) (or is besweet also available under Linux)

[tebasuna51]

I only found windows version of azid, there are also linux versions?

Maybe you need wait to ffmpeg mods.
I try one year ago (http://forum.doom9.org/showthread.php?p=1043590#post1043590) but without luck.

[hubblec4]

Hello

I found a small mistake.
I had encoded an AC3-file with neroaacenc q=0.37

Format : MPEG-4
Format profile : Base Media / Version 2
Codec ID : mp42
Dateigröße : 245 MiB
Duration : 1h 55min
Gesamte Bitrate : 297 Kbps
Kodierungsdatum : UTC 2008-10-12 10:18:53
Tagging-Datum : UTC 2008-10-12 10:31:58
Verwendetes Programm : Nero AAC codec / 1.3.3.0
cdec : ndaudio 1.3.3.0 / -q 0.37

Audio
Format : AAC
Format/Info : Advanced Audio Codec
Format version : Version 4
Format profile : LC
Format settings, SBR : Nein
Codec ID : 40
Duration : 1h 55min
Bitraten-Modus : Variable
Bitrate : 295 Kbps
Maximum bit rate : 354 Kbps
Kanäle : 6 Kanäle
Channel positions : Front: L C R, Rear: L R, LFE
Samplingrate : 48,0 KHz
Auflösung : 16 bits
Stream size : 244 MiB (99%)
Kodierungsdatum : UTC 2008-10-12 10:18:53
Tagging-Datum : UTC 2008-10-12 10:31:58


and the same AC3 wit neroaacenc q=0.35

Format : MPEG-4
Format profile : Base Media / Version 2
Codec ID : mp42
Dateigröße : 248 MiB
Duration : 1h 55min
Gesamte Bitrate : 301 Kbps
Kodierungsdatum : UTC 2008-10-12 10:37:29
Tagging-Datum : UTC 2008-10-12 10:50:29
Verwendetes Programm : Nero AAC codec / 1.3.3.0
cdec : ndaudio 1.3.3.0 / -q 0.35

Audio
Format : AAC
Format/Info : Advanced Audio Codec
Format version : Version 4
Format profile : LC
Format settings, SBR : Nein
Codec ID : 40
Duration : 1h 55min
Bitraten-Modus : Variable
Bitrate : 299 Kbps
Maximum bit rate : 358 Kbps
Kanäle : 6 Kanäle
Channel positions : Front: L C R, Rear: L R, LFE
Samplingrate : 48,0 KHz
Auflösung : 16 bits
Stream size : 247 MiB (99%)
Kodierungsdatum : UTC 2008-10-12 10:37:29
Tagging-Datum : UTC 2008-10-12 10:50:29

Why is the 0.37.mp4 smaller as the 0.35.mp4?

hubbble

[LigH]

I only found windows version of azid, there are also linux versions?

I am quite sure -- no. Both EXE and DLL were closed-source. Midas seems to be M.I.A.

[skromnibog]

I found a small mistake.
I had encoded an AC3-file with neroaacenc q=0.37
and the same AC3 wit neroaacenc q=0.35
Why is the 0.37.mp4 smaller as the 0.35.mp4?

This is normal behavior. It happens because different psychoacoustics is used for q0.35 and q0.37. Some jumps like this might occur at some q values for some files. When decoding big set of files, average bitrate should increase, but is not guaranteed.

[hubblec4]

This is normal behavior. It happens because different psychoacoustics is used for q0.35 and q0.37. Some jumps like this might occur at some q values for some files. When decoding big set of files, average bitrate should increase, but is not guaranteed.

ok. and which mp4-file sounds better? (0.35 or 0.37)

i thought a higher -q makes a bigger file with better sound??

hubble

[nautilus7]

He didn't say this. He said that because of the way the aac compression works and because the q values your are comparing are so close, this can happen. It's in the range of "statistical error". If you encode a lot of files, thoses with q 0,37 would have bigger size summed together than those with q 0,35.

[skromnibog]

and which mp4-file sounds better? (0.35 or 0.37)
That depends on a person. At q values where jump occurs and different psychoacoustic parameters are used, it is up to a listener to determine which q value is better for him/her. Listen to some of your favorite music at 0.35 and 0.37 and choose the one that sounds better to you. Of course don't do it just on one sample, because results may vary depending on a sample used.
In general, every user is encouraged to use the q value that suits him/her and if quality is good enough is to be determined only by listening of the person choosing q.

[Gusar]

Ok, this is weird...$ time a52dec -o wav *T80* 2> /dev/null | neroAacEnc -ignorelength -q 0.25 -if - -of linux.mp4
*************************************************************
* *
* Nero AAC Encoder *
* Copyright 2008 Nero AG *
* All Rights Reserved Worldwide *
* *
* Package build date: Sep 17 2008 *
* Package version: 1.3.3.0 *
* *
* See -help for a complete list of available parameters. *
* *
*************************************************************

Processed 2551 seconds...

real 3m49.771s
user 3m42.114s
sys 0m2.325s$ time a52dec -o wav *T80* 2> /dev/null | wine neroAacEnc.exe -ignorelength -q 0.25 -if - -of win.mp4
*************************************************************
* *
* Nero AAC Encoder *
* Copyright 2008 Nero AG *
* All Rights Reserved Worldwide *
* *
* Package build date: Sep 17 2008 *
* Package version: 1.3.3.0 *
* *
* See -help for a complete list of available parameters. *
* *
*************************************************************

Processed 2551 seconds...

real 2m47.440s
user 2m39.143s
sys 0m2.799s
The above just shows that the Linux version is still slower than the Windows version. But now comes the weird part:$ ls *.mp4
-rw-r--r-- 1 21M okt 15 12:11 linux.mp4
-rw-r--r-- 1 22M okt 15 12:14 win.mp4
The Linux version produces a different encode!

[menno]

The Linux version produces a different encode!

I'm guessing this is because of the 1000's of floating point comparisons the encoder makes each frame. With a different compiler you will get different results. But I will check this more thoroughly.

[madshi]

Would you Nero guys mind adding support for IEEE float style WAV files (wave format extensible GUID "{00000003-0000-0010-8000-00AA00389B71}") in the native bitdepth of your encoder (32bit or 64bit)?

The reason I'm asking is this: In my audio processing tool (eac3to) I often end up with floating point data. Currently I have to downconvert that to 24bit PCM because the Nero AAC Encoder doesn't accept floating point input. But internally AAC encoding is done in floating point, right? So having direct floating point input support would improve both performance and audio quality (slightly).

Thanks! :)

[menno]

Would you Nero guys mind adding support for IEEE float style WAV files (wave format extensible GUID "{00000003-0000-0010-8000-00AA00389B71}") in the native bitdepth of your encoder (32bit or 64bit)?

The reason I'm asking is this: In my audio processing tool (eac3to) I often end up with floating point data. Currently I have to downconvert that to 24bit PCM because the Nero AAC Encoder doesn't accept floating point input. But internally AAC encoding is done in floating point, right? So having direct floating point input support would improve both performance and audio quality (slightly).

Thanks! :)

Hmm, I just double checked this and we already support this input format, although only in 32 bit. It works fine on the test samples I have. Maybe you can send me a short sample that doesn't work on our encoder and I can have a closer look at it.

[madshi]

I'm sorry, that was my fault. I thought I had tested 32bit float, but it seems I did not. It does work fine - thanks! :)

[madshi]

Happy New Year to you Nero people!

One little request: Would you be willing to add stdout support to both your command line AAC encoder and decoder? That would be nice!

[menno]

Happy New Year to you Nero people!

One little request: Would you be willing to add stdout support to both your command line AAC encoder and decoder? That would be nice!

Thanks! Same wishes to all our users, and those we still need to convince :)

For the decoder stdout support should be possible, but with the encoder we might run into some problems with the mp4 format, but I'll look into it.

[Dark Shikari]

For the decoder stdout support should be possible, but with the encoder we might run into some problems with the mp4 format, but I'll look into it.You should be able to do it if you put the header at the end instead of the start. That's the method x264 uses for outputting an mp4 to stdout.

[madshi]

For the decoder stdout support should be possible, but with the encoder we might run into some problems with the mp4 format, but I'll look into it.
Maybe for the encoder you could output a raw AAC stream, if stdout is activated? Or leave the header "size" field empty (or set to highest possible value)? Or Dark Shikari's suggestion sounds good to me, too.

Thanks in any case for looking into this!

[lexor]

You should be able to do it if you put the header at the end instead of the start. That's the method x264 uses for outputting an mp4 to stdout.

hmm, I wonder if that's the reason psp chocks on mp4 made by x264, but plays same raw streams muxed with mp4box fine.

[Dark Shikari]

hmm, I wonder if that's the reason psp chocks on mp4 made by x264, but plays same raw streams muxed with mp4box fine.Yes, you have to remux or use qt-faststart for many devices and for web playback when using either x264's or ffmpeg's muxer. This is by design.

[sl1pkn07]

thanks for the new version!

[fachman]

This is normal behavior. It happens because different psychoacoustics is used for q0.35 and q0.37. Some jumps like this might occur at some q values for some files. When decoding big set of files, average bitrate should increase, but is not guaranteed.

Hi

Can you give us the q factor, bitrate numbers where your encoder changes its scheme of work, by adjusting psychoacoustics or other tricks???
For example at which bitrate your encoder with Auto setting changes AAC LC to AAC HE with 5.1 sound??? 160?? 128?

I also noticed some strange things regarding your encoder.
When we set q=0,2 and encode the same file with AAC HE and AAC LC, the LC version is much smaller.
With q=0,21 HE is smaller than LC. Glitch???
I am talking about 5.1 audio, because I have tested it on several files. I am not sure about stereo.

I thought that HE is the right way for smaller bitrates, so with the same quality it should take less bits???



Thanx for the good work

[madshi]

One more request: Could you please allow 24bit output for the Nero AAC decoder? The Nero DirectShow AAC decoder outputs 24bit, but the command line decoder only 16bit.

Thanks!

[burfadel]

I thought for very low bitrates HE is better, for high bitrates LC is better? :) ...

[menno]

fachman, that seems like a glitch indeed, do you get that consistently over multiple files? I will look into it.

madshi, noted.

[nautilus7]

menno, when encoding with NeroAacEnc, is there any delay value added by default to the audio data? If yes, could you tell me the exact amount?

[menno]

Yes, depends on profile (LC, HE, HEv2) and samplerate. However, the encoder writes gapless info to the mp4 file (using chapter list) and a capable decoder will compensate for this so the result will be 0 samples delay. For LC the delay is 2048 samples IIRC, for HE/HEv2 a little bit more (2500 or something).

[nautilus7]

OK, thanks, but could you explain it a little further?

For example in LC profile and 48KHz how much is the delay in ms?

What you say about a capable decoder... Is such decoder the neroaacdec included in the encoder package? And nero 7 dshow decoder is not such? (see this (http://forum.doom9.org/showthread.php?p=1238245#post1238245) post)

[menno]

Divide the nr of samples by the samplerate (in kHz) and you will have milliseconds.

[nautilus7]

Thanks again.

[microchip8]

Does neroAacEnc implement TNS and/or backward prediction? If not, any plans to add these features?

[menno]

Does neroAacEnc implement TNS and/or backward prediction? If not, any plans to add these features?

TNS: yes
prediction: No, and it wouldn't make sense because there are hardly any players (hardware and software) that can decode that. The gain is also said to be minimal.

[microchip8]

Thanks for the info, menno :)

So I guess TNS is automatically turned on? Why not a switch to let the user decide if he wants it or not?

EDIT: also is there any benefit in using 2pass mode?

[menno]

So I guess TNS is automatically turned on? Why not a switch to let the user decide if he wants it or not?


TNS is not something that is just on or off over the whole file. Usage of TNS and determining it's parameters is done by the encoder based on the audio content. We believe our encoder is good enough at doing this to assume that disabling TNS will almost always deteriorate quality. If TNS doesn't benefit on a particular frame, the encoder will turn it off itself.


EDIT: also is there any benefit in using 2pass mode?

2pass is useful when you want constant quality with an exact predetermined output size.

[microchip8]

TNS is not something that is just on or off over the whole file. Usage of TNS and determining it's parameters is done by the encoder based on the audio content. We believe our encoder is good enough at doing this to assume that disabling TNS will almost always deteriorate quality. If TNS doesn't benefit on a particular frame, the encoder will turn it off itself.



2pass is useful when you want constant quality with an exact predetermined output size.

thanks, that clears it up. One final question, though. Which will sound better? ABR 120 kbps 2pass or VBR quality mode with roughly the same bitrate as ABR. Sorry for stupid question, but I'm trying to chose the "best" mode so I can rip my albums here :)

Also, is there an easy way of translating quality values to average bitrates?

[menno]

Have a look here: http://www.audiocoding.com/nero_aacenc.html
It's only a general idea, on average it should be fairly correct, but on individual files you will get differences.

[microchip8]

Have a look here: http://www.audiocoding.com/nero_aacenc.html
It's only a general idea, on average it should be fairly correct, but on individual files you will get differences.

Thanks for that. Currently, when encoding stuff with neroAacEnc, it only prints processed seconds. Is it possible to add in a future version things like bitrate information similar to how LAME does it?

[jeffy]

@menno: The links in the first post are not valid anymore. What happened? The search engine of nero.com doesn't find anything related. Thank you.

[menno]

Yeah, right now the pages are unavailable, they're working on some new ones. I updated the links in the first post.

[audyovydeo]

Yeah, right now the pages are unavailable, they're working on some new ones. I updated the links in the first post.

Pages are back ... to 1.3.3.0 !!!
Somehow I was expecting a new version, naive me...

anything new forthcoming ?

cheers
a/v

[b66pak]

hi, first i want to thank you for this great tool...

i want to know in terms of quality which is better:


neroaacenc.exe -2pass -br 96000 -if audio.wav -of audio.2pass.m4a

neroaacenc.exe -q 0.36 -if audio.wav -of audio.q.m4a


the output files (audio.2pass.m4a and audio.q.m4a) have the same size!
_

L.E. how can i obtain the best quality for a given size?
_

[nurbs]

2pass and the -q switch should give the same quality for the same size.

[skromnibog]

To achieve best quality with desired size for a single file use 2pass abr.
To achieve best quality with desired size for your complete music collection use vbr.
More about q parameter here: http://www.audiocoding.com/nero_aacenc.html

[b66pak]

thanks for the answers...
_

L.E. what are the conditions that trigger aac-he and aac-hev2 encoding...i need to do a 64kbt aac-he (or hev2) encoding and the help file states: "Note that the above switches (-lc, -he, -hev2) should not be used; optimal AAC profile is automatically determined from quality/bitrate settings when no override is specified."
_

[b66pak]

nobody?
_

[nurbs]

HE is used for q < 0.31. I think HEv2 is used for q < 0.26 (maybe q < 0.21; you'll have to test). I don't know about abr because I hardly ever use it, but HE becomes better below about 80kbps for stereo and HEv2 below about 56.
Since you don't care if it uses HE or HEv2 just use 64kbps and you'll see what comes out.

[b66pak]

is this meaning that he is only for 64k and below? 2pass 96k he would be worse than 2pass 96k lc?
_

[nurbs]

is this meaning that he is only for 64k and below? 2pass 96k he would be worse than 2pass 96k lc? Yes, HE shouldn't be used at ~80 and above because it will sound worse than LC. Seriously if you don't care which profile is used just pick a bitrate and let the encoder decide.

[Umamio]

@menno Is there a method or a switch that can be used to disable the writing of gapless playback info? If not, is there a way to strip this data post-encode?

[menno]

@menno Is there a method or a switch that can be used to disable the writing of gapless playback info? If not, is there a way to strip this data post-encode?

I think mp4box can remove the chapter list.

[Seraphic-]

Hi menno,

Does NeroAAC 1.3.3.0 support 7.1 encoding?
In my below topic, I was having some issues with 7.1 encoding/decoding

http://forum.doom9.org/showthread.php?t=146005

[skromnibog]

Hi menno,
Does NeroAAC 1.3.3.0 support 7.1 encoding?
In my below topic, I was having some issues with 7.1 encoding/decoding
http://forum.doom9.org/showthread.php?t=146005
I believe that all your questions are answered in that thread. Short: Yes, it supports.

[Eric B]

Concerning multi tracking encoding or transcoding from an AC3 or DTS source (5.1), could AAC 5.1 achieve transparency? Which bit rate is required? Is there any public comparison between 256k LC and 192k HE (multichannel)?

[Sharktooth]

your question is meaningless.
it is like asking at what bitrate you should encode a video to reach transparency.
it depends on the source and, ofcourse, LC mode is better for high quality encodes (but requires more bitrate).

[shon3i]

ofcourse, LC mode is better for high quality encodesThis is absolutley not true for multichannel encodings, and HE-AAC 160kbps, LC-AAC 320, DD 448, DD+ 384, DTS 1.5 have almost tied quality

[Eric B]

basically, I've ripped Star wars episode 4 to 6, which contains "Dolby Digital Surround EX" tracks, and I hesitate between muxing these ac3 tracks (2*400MB!) or the aac-he @ 192k with the avc video track (~1 GB).

[Blue_MiSfit]

Well, if you've dropped the video size so much, it'd be a real pity to waste so much space on AC3 :)

I'd go for LC-AAC @ 320, or HE-AAC at ~192 or whatever. Do some listening tests, and see what you think!

Nero's encoder is really freaking good :D

~MiSfit

[Jeroi]

Menno, is nero planning to release multithreaded version of aac codec?

[Selur]

Got a problem and I'm not sure if it's a nero bug/missing feature or if I'm doing something wrong. :)
I normally use ffmpeg to decoder raw audio streams and pipe the output to neroAacEnc for reencoding without a problem,..

Now I got this file:
General
Complete name : d:\old.dts
Format : DTS
Format/Info : Digital Theater Systems
File size : 1.27 GiB
Duration : 2h 0mn
Overall bit rate : 1 510 Kbps

Audio
Format : DTS
Format/Info : Digital Theater Systems
Duration : 2h 0mn
Bit rate mode : Constant
Bit rate : 1 510 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 24 bits
Compression mode : Lossy
Stream size : 1.27 GiB (100%)
which has a length > 2h (normally I reencode on files < 2h)
when I now call: ffmpeg.exe -threads 2 -v -10 -y -i "D:\old.dts" -acodec pcm_s16le -f wav - | neroAacEnc.exe -if - -br 448000 -ignorelength -of "new.aac"
neroAacEnc outputs ERROR: error while encoding frame and stops the encoding.

To be sure this is not a ffmpeg problem I called: ffmpeg.exe -threads 2 -v -10 -y -i "D:\old.dts" -acodec pcm_s16le -f wav new.wav
and get the following file:
General
Complete name : d:\new.wav
Format : Wave
File size : 3.88 GiB
Duration : 2h 0mn
Overall bit rate : 4 608 Kbps

Audio
ID : 0
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Signed
Codec ID : 00001000-0000-0100-8000-00AA00389B71
Codec ID/Hint : Microsoft
Duration : 2h 0mn
Bit rate mode : Constant
Bit rate : 4 608 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Stream size : 3.88 GiB (100%) which, like the dts plays fine (in mpc-hc).
When I now feed neroAacEnc with the wav I get the same error as before. :(

Is this a neroAacEnc limitation/bug or did I do something wrong?

Cu Selur

[Midzuki]

Maybe neroaacenc expects a "Wave Format Extensible" flag,
which ffmpeg is not supplying ? :confused: :confused: :confused:

EDIT:

Try eac3to instead of ffmpeg.

[Selur]

Try eac3to instead of ffmpeg.
will look into eac3to, but it really isn't a solution since I can't use it under linux without wine ;)
(I try to keep my command line calls cross platform compatible)
-> eac3to works, but if someone knows a way to do this with ffmpeg please post,..

[Midzuki]

^ sox, compiled with ffmpeg support :)

[b66pak]

i have no problem with this ffmpeg build:

the line:
ffmpeg -i audio.dts -acodec pcm_s16le -f wav - | neroaacenc -if - -br 448000 -ignorelength -of audio.mp4

FFmpeg version SVN-r26071, Copyright (c) 2000-2010 the FFmpeg developers
built on Dec 22 2010 04:07:12 with gcc 4.4.2
configuration: --enable-gpl --enable-version3 --enable-libgsm --enable-libvorb
is --enable-libtheora --enable-libspeex --enable-libmp3lame --enable-libopenjpeg
--enable-libschroedinger --enable-libopencore_amrwb --enable-libopencore_amrnb
--enable-libvpx --disable-decoder=libvpx --arch=x86 --enable-runtime-cpudetect -
-enable-libxvid --enable-libx264 --enable-librtmp --extra-libs='-lrtmp -lpolarss
l -lws2_32 -lwinmm' --target-os=mingw32 --enable-avisynth --enable-w32threads --
cross-prefix=i686-mingw32- --cc='ccache i686-mingw32-gcc' --enable-memalign-hack

libavutil 50.35. 0 / 50.35. 0
libavcore 0.16. 0 / 0.16. 0
libavcodec 52.100. 0 / 52.100. 0
libavformat 52.88. 0 / 52.88. 0
libavdevice 52. 2. 2 / 52. 2. 2
libavfilter 1.69. 0 / 1.69. 0
libswscale 0.12. 0 / 0.12. 0
_

N.B. neroaacenc output aac ONLY in .mp4! by naming your output .aac you will only get a .mp4 with .aac extension...
_

[Selur]

@b66pak: thanks, so it seems the dts stream is b0rked in some way. :(

N.B. neroaacenc output aac ONLY in .mp4! by naming your output .aac you will only get a .mp4 with .aac extension...
I know :)

[manolito]

As I am quite new to the AAC audio format, please forgive me if this topic has been discussed and maybe solved before...:p

The AAC file was created by StaxRip (which uses NeroAAC) using default parameters (Profile Automatic, Quality 0.30). It was muxed into an MKV using MKVMerge. This MKV plays flawlessly in every software player I fed it to.

I then used MKVExtract to demux the audio track. Also this AAC file plays without problems. MediaInfo has this about the file:
Allgemein
Vollständiger Name : I:\de.aac
Format : ADTS
Format/Info : Audio Data Transport Stream
Dateigröße : 71,0 MiB

Audio
Format : AAC
Format/Info : Advanced Audio Codec
Format-Version : Version 4
Format-Profil : HE-AAC / LC
Bitraten-Modus : variabel
Kanäle : 2 Kanäle
Kanal-Positionen : Front: L R
Samplingrate : 48,0 KHz / 24,0 KHz
Stream-Größe : 71,0 MiB (100%)

When I try to decode this file using NeroAACDec.exe ít gives me this error message:
moov box not found
and quits.

I googled for this error message, but I only found out that this mostly applies to MOV files, and that some required header information is missing.

BTW I am using the current version of NeroAAC (1.5.1). And I also should mention that FAAD.exe decodes this file nicely.


Any idea what is going wrong here?


Cheers
manolito

[LoRd_MuldeR]

I think the Nero AAC encoder only outputs AAC streams in a MP4 container (.mp4), not as ADTS file (.aac). And, consequently, their decoder only supports MP4 files as input :eek:

The solution is quite simple: Use FAAD (link (http://www.rarewares.org/aac-decoders.php)) instead of the Nero AAC decoder. In my experience FAAD supports ADTS files just fine ;)

[LigH]

And don't be afraid about quality. Even though FAAC (the free encoder) is not able to encode HE-AAC, FAAD (the free decoder) can decode all relevant kinds of AAC.

[Selur]

just wondering http://listening-tests.hydrogenaudio.org/igorc/aac-96-a/ mentions NeroAacEnc 1.5.4 but the download over at http://www.nero.com/eng/technologies-aac-codec.html only offers 1.5.1 anyone know where to get 1.5.4 ? Okay, forget that, the 1.5.1.zip includes 1.5.4

[LigH]

The title on the download page says "Version 1.5.1"; after you downloaded, unpacked, and ran the executable, it reports being "Version 1.5.4" instead.