Nero Releases FREE Reference Quality MPEG-4 Audio Command Line Encoder

[Ivan Dimkovic]

I am pleased to announce the launch of FREE Reference Quality MPEG-4 Audio solution from Nero, in the command line form!

* First in the world FREE 2-Pass MPEG-4 AAC Encoder
* Compression Ratios ranging from ultra high (58 CDs fit on one!) to High-End Audio (2.5:1), for absolutely perfect audiophile encodings
* Crystal Clear, Award Winning Sound Quality at every compression ratio and bit rate!
* Support for Embedded Album Art (Covers, Booklets, Lyrics!)
* Store Entire Audio Album in a Single .mp4 File with all the Features of an Audio CD embedded inside, but at a fraction of the space!
* Reference Quality MPEG-4 Audio Codec
* Fully Compatible with the Latest Version of the State-of-the-art MPEG-4 Audio Standard (LC-AAC, HE-AAC and HE-AAC v2)

http://www.nero.com/nerodigital/eng/Nero_Digital_Audio.html

[layer3maniac]

Right on Ivan!!! Very nice surprise indeed!

[buzzqw]

a very nice gift ! :thanks:

BHH

[Naito]

Thanks. Thanks. An one more ... thanks.

[layer3maniac]

I tested the 2 pass, using an old frontend I originally made for PsyTEL. Seems to work beautifully.

What a long, strange trip it's been.

[shon3i]

Encoder not working propertly i tried to encode and get fatality close app. cmd used

"neroAacEnc.exe" -he -br 96000 -if "proba.wav" -of "proba.mp4"


same thing i get when try to view help neroAacEnc.exe -help

[GmorG McRoth]

As I read on Hydrogenaudio forums there is some problems with AMD XP cpus (they don't support SSE2 or something like that)

I wonder do this support 5.1 sound encoding?

[shon3i]

As I read on Hydrogenaudio forums there is some problems with AMD XP cpus (they don't support SSE2 or something like that)New encoder and very big bug

I wonder do this support 5.1 sound encoding?
Yes support it, i tested with earlier version now can be good as Coding Technologies, maybe 2-pass can help something with better quality but will see

[GmorG McRoth]

Yes support it, i tested with earlier version now can be good as Coding Technologies, maybe 2-pass can help something with better quality but will see
hmm I think this is new encoder it's in exe state not in DLL like when it was part of nero package. I was wondering if they did not cripple it since it's free and all.

[shon3i]

hmm I think this is new encoder it's in exe state not in DLL like when it was part of nero package. I was wondering if they did not cripple it since it's free and all.
Yes i know what you think. This is only cli app for dll from nero 7.2.0.3b and worked with 5.1 via nero apps, so why work now. You think, is free app then have some limit. I don't think that

[GmorG McRoth]

I just have very little trust to ahead, I know I shuld have little more of it but after nero 7 I realy been unhappy with them.

[shon3i]

I just have very little trust to ahead, I know I shuld have little more of it but after nero 7 I realy been unhappy with them.
Totality i agree with you. I totaly disaponted with old AAC codec especialy 5.1 encoding. My favor is Coding Techologies. Whole Nero 7 package is so bugy.

[GmorG McRoth]

what was bad with 5.1 audio in nero 7? I did not notice anything horrifing at ~250 bitrates I was using for my encodes.

Off topic --- man I'm so happy I can uninstall nero, and never install it again (hopefully)

[shon3i]

Not only with HE and bitrates forom 96-128kbs, wihich is very terrible. You should be try Coding techologies for example @ 96kbs to see what is excelent quality for low rate comparing to old nero 7. Aslo CT is better at higer rates and LC fo 6ch for now. I not test this new encoder propertly becouse i don't have much wish to extract ac3->5.1wav and then covert it to aac using bugy NeroWave Editor. Now wait this cli app without sse2 support to see results.

btw. Encoding to AAC with high rate have no sense better keep ac3. Right

[GmorG McRoth]

I kinda like having mp4 file and this makes me use mp3 or AAC for audio and since i like to keep 5.1 audio directs me to AAC.

I never had chance to use CT. CT is included in pro version of winamp right? but i though it supports only stereo (I don't know I don't use winamp)

[shon3i]

I kinda like having mp4 file and this makes me use mp3 or AAC for audio and since i like to keep 5.1 audio directs me to AAC.Yes that's the mp4 limit.

I never had chance to use CT. CT is included in pro version of winamp right? but i though it supports only stereo (I don't know I don't use winamp)
No CT is completly free, and you can use it via BeHappy/BeLight/BeSweet, my favor is BeHappy. In Winamp is only stereo encoder becouse for audio cr rips only need stereo right, and winamp have little limited encoder, but when you register you get all acees to encoder and all options for stereo encoding, but you don't need register for normal rips

With BeLight/BeHappy/BeSweet and other third party apps, you have full support to encoder which including HE-AAC, HE-AACv2, LC-AAC, 5.1 encoding everything

[GmorG McRoth]

I noticed that nero encoder has problem with getting proper bitrates

I set it to encode at 192 kbit/s and file comes at 158

and the bigger the sampling rate the bigger the problem with 24bit 96KHz file reasult is at 115 (though at more silent parts avarage is about 80 kbit and sounds very smeared).

[shon3i]

Of curse first download BeHappy form here (http://www.mytempdir.com/642371), in this package have everything you need for CT encodig. You aslo need NicAudio.dll (http://www.mytempdir.com/642384)if you sorce will be ac3,dts etc you must put it into AviSynth directory. for more info about behappy search audio forum, Thanks

[Ivan Dimkovic]

hmm I think this is new encoder it's in exe state not in DLL like when it was part of nero package. I was wondering if they did not cripple it since it's free and all.

First of all, I would kindly ask you to stop this FUD - Nero Command Line encoder has absolutely no limits and it has highest possible quality - in fact, it has more options than Nero 7.2 plug-in, aimed for professionals.

Second, regarding the 5.1 encoding - shon3i, I would really like if you you test it before making any quality claims - I am quite sure you will be positively surprised.

Third, AMD bug has been fixed just now - new version will be online very soon.

And, last - but not least - may I also point out the three things:

- This is the first in the world, completely legal, and unlimited command line encoder (it is not a hack of third-party application) - with most powerful options you would expect from the command line app, e.g:

* Support for sampling rates of 8 kHz to 96 kHz
* Support for CBR, VBR, 2-PASS CBR and ABR
* Support for 32-bit floating point PCM input
* Full support for up to 7.1 channels
* Support for hinting for streaming

- HE-AAC encoder was proven to be the top performer in the 48 kbps area - ahead of Coding Technologies encoding solution. In fact, I have a very strong belief that you will find it the same, top-performing quality for 5.1

So, please - once again, do the tests and you will figure out why Nero is different.

[Ivan Dimkovic]

I noticed that nero encoder has problem with getting proper bitrates

I set it to encode at 192 kbit/s and file comes at 158


How about using -cbr 192000 switch? There is a readme.txt in the package.

[Skuto]

As I read on Hydrogenaudio forums there is some problems with AMD XP cpus (they don't support SSE2 or something like that)


The (currently available) encoder requires an SSE2 capable processor.

It has nothing to do with AMD, other than that the AMD Athlon XP doesn't have SSE2 capability (but neither does the Pentium III, whereas the Athlon64 or Opteron does and works with the published package).

[GmorG McRoth]

hard day Ivan Dimkovic huh, you sound stressed.

First of all I did not intended to spread any FUD if it sounds that way i'm sorry at the time it was just plain question for me.

my bad about bitrate thing, I typed just 192 without zeros no wonder bitrate was wird heh... I depend on GUIs too much lately.

[Skuto]

hmm I think this is new encoder it's in exe state not in DLL like when it was part of nero package. I was wondering if they did not cripple it since it's free and all.

The encoder is full featured and in fact newer than the one shipped in Nero 7.2.

[GmorG McRoth]

The encoder is full featured and in fact newer than the one shipped in Nero 7.2.
Yes, I understand.

[FFWD]

Just curious; has work started on integrating this encoder with Exact Audio Copy & MeGUI?

[Skuto]

Not only with HE and bitrates forom 96-128kbs, wihich is very terrible. You should be try Coding techologies for example @ 96kbs to see what is excelent quality for low rate comparing to old nero 7. Aslo CT is better at higer rates and LC fo 6ch for now.

I'm sorry, but this is just...not correct. Please see for example the latest test from the LAME authors which clearly show Nero to be equal or better than CT. I think I'll trust the results from a public, open test much more than numbers made up by some random person, thank you very much.

http://www.mp3-tech.org/tests/aac_48/results.html

There was also an 128kbps test, in which Nero scored equal with iTunes, LAME and Vorbis.

[Ivan Dimkovic]

hard day Ivan Dimkovic huh, you sound stressed.

Nothing beats travelling to three countries in 3 days :) But no, I am not stressed.

Point of my post is - there is a significant amount of FUD-like information here especially coming from few people, and I would kindly ask all people to test the software before making the quality claims and that's all ;)

When it comes to bit-rate, the point was not in 192 vs. 192000 but in -cbr switch. -br is ABR, -q is VBR and -cbr means CBR.

If you want perfectly CBR 192 kbps encodings for some streaming purpose - please use the -cbr switch ;)

[layer3maniac]

CBR works fine for me. Thanks Ivan, very gracious to give this away.

[GmorG McRoth]

When it comes to bit-rate, the point was not in 192 vs. 192000 but in -cbr switch. -br is ABR, -q is VBR and -cbr means CBR.

If you want perfectly CBR 192 kbps encodings for some streaming purpose - please use the -cbr switch ;)

from readme

-br <number> : Specifies "target bitrate" mode.
<number> is target bitrate in bits per second.
-cbr <number> : Specifies "target bitrate (streaming)" mode.
<number> is target bitrate in bits per second.
When neither of above quality/bitrate options is used,
the encoder defaults to equivalent of -q 0.5

this is little vague. I think people already know terminology like ABR and CBR, using it in readme whuld not be going too far.

[Ivan Dimkovic]

I think it is a good proposal, thanks - I will look into changing the terminology.

On the other hand, the bug was not AMD-related, but related to CPUs without SSE2 - so latest Intel and AMD chips were not affected by this bug.

[layer3maniac]

I think people already know terminology like ABR and CBR, using it in readme whuld not be going too far. -cbr <number> : Specifies "target bitrate (streaming)" mode. If people already know terminology like CBR then shouldn't they be able to comprehend that?

[GmorG McRoth]

If people already know terminology like CBR then shouldn't they be able to comprehend that?
Switch names can mean meny things I think i'm not alone who reads switch discription without much of thought about switch name.

[siddharthagandhi]

is there a GUI available I'm not good with command lines i hate writing it in and all that

[shon3i]

Second, regarding the 5.1 encoding - shon3i, I would really like if you you test it before making any quality claims - I am quite sure you will be positively surprised.I can't test becouse i have AMD machine. btw i do a little test with nero apps and i very happy with quality of 5.1 encoding but quality is not much better than CT, Sometimes i think is much wrost but when i test 2pass and other things from cli encoder i say my final word.
I'm sorry, but this is just...not correct. Please see for example the latest test from the LAME authors which clearly show Nero to be equal or better than CT. I think I'll trust the results from a public, open test much more than numbers made up by some random person, thank you very much.

http://www.mp3-tech.org/tests/aac_48/results.html

There was also an 128kbps test, in which Nero scored equal with iTunes, LAME and Vorbis.I am talking about 5.1 (6ch) encoding not stereo, so lastest Nero 7 encoders are very bad in 5.1 encoding, until this last which is now good, but from my some short test i think is not again so good like CT, but i don't try 2-pass mode becouse have trouble with encoder and wait new realise. btw about this test @48kbs, now when nero have CBR implementation we have to do a new test and see what is better when both codec have same settings and we have to say what is better.

[SeeMoreDigital]

Oh wow Ivan, I really was not expecting this... Many thanks :)

I hope to test the encoder as soon as somebody incorporates it into a GUI ;)


Cheers

[Skuto]

I can't test becouse i have AMD machine. btw i do a little test with nero apps and i very happy with quality of 5.1 encoding but quality is not much better than CT, Sometimes i think is much wrost but when i test 2pass and other things from cli encoder i say my final word.
I am talking about 5.1 (6ch) encoding not stereo, so lastest Nero 7 encoders are very bad in 5.1 encoding, until this last which is now good, but from my some short test i think is not again so good like CT, but i don't try 2-pass mode becouse have trouble with encoder and wait new realise. btw about this test @48kbs, now when nero have CBR implementation we have to do a new test and see what is better when both codec have same settings and we have to say what is better.

So, you can't test because you have a non-SSE2 machine, you tested and it's worse but you can't test it and you wait for a new release, you need a CBR implementation (which Nero has always had)...nothing what you say makes any sense whatsoever.

[Skuto]

Oh wow Ivan, I really was not expecting this... Many thanks :)

I hope to test the encoder as soon as somebody incorporates it into a GUI ;)


Cheers

Aren't they selling the GUI for 60 bucks on nero.com? :)

[GmorG McRoth]

So, you can't test because you have a non-SSE2 machine, you tested and it's worse but you can't test it and you wait for a new release, you need a CBR implementation (which Nero has always had)...nothing what you say makes any sense whatsoever.
He meant he tested with older version that was bundled with nero, If i'm not mistaken.

[GodofaGap]

First of all, I want so say: many thanks! This is the first time I can try AAC with the same ease and comparable quality as LAME and Vorbis. :)

About the bitrate, I encoded about half an hr of 5.1 audio with "-br 192000 -he". Afterwards, foobar2000 reported it to have 198 kbps. Is this an issue due to container overhead? Or did the encoder really miss the bitrate by 6 kbps (which wouldn't be that bad, just wondering)?

[Ivan Dimkovic]

If you used -br switch, you activated the ABR mode - which might have a miss in bit-rate within 5% range.

If you want to have 192 kbps constant bit-rate file, you could use -cbr instead of -br (however I would recommend this unless you have some very strict streaming requirements in mind)

[layer3maniac]

Just to clarify, 2pass is useless in cbr mode, no?

[Ivan Dimkovic]

Currently - 2pass should be used with -br command line switch

[GodofaGap]

If you used -br switch, you activated the ABR mode - which might have a miss in bit-rate within 5% range.

5 % deviation is acceptable to me. Thanks for the clarification!

[layer3maniac]

I just did the soundtrack for American Beauty using -br 192000, -2pass, in 6 channel. The quality is SUPERB!

[shon3i]

So, you can't test because you have a non-SSE2 machine, you tested and it's worse but you can't test it and you wait for a new release, you need a CBR implementation (which Nero has always had)...nothing what you say makes any sense whatsoever.
You not understand me, i doing test with new encoder via NeroWave Editor, but not with this cli. So this is same encoder from nero 7.2.03b but in exe file. So 5.1 encoding sounds good but not good as i like.

About CBR. I think at last public test @48 nero uses ABR vs CT CBR and prove that ABR is better than CBR, everyone know that ABR is much better in any case. So Nero now have real CBR then if we have a real winner of some test we need to compare codec with same settings. Nero CBR vs CT CBR

[layer3maniac]

You not understand me, i doing test with new encoder via NeroWave Editor, but not with this cli. So this is same encoder from nero 7.2.03b but in exe file. So 5.1 encoding sounds good but not good as i like. Once again...
This is NOT "the same encoder from nero 7.2.03b but in exe file".

[BigDid]

Oh wow Ivan, I really was not expecting this... Many thanks :)

I hope to test the encoder as soon as somebody incorporates it into a GUI ;)


Cheers
Hi SMD,

Check out Belight next rev, Kurtnoise is stating (in a french forum) that he is working on it :)

And my thanks to the devs and NERO for this gift.

Did

[Rockaria]

Finally, a great (official : legit) addition to the AAC (he) (multi channel) users community. Hope h/w vendors provide the fully supported(he/hev2, vbr, 6~ch) portable firmwares widely very soon (& cheap)!;)

I was able to integrate the cli encoder to foobar2k 0.83 for 1pass (vbr he) encoding and also have tested some 2 pass encoding with a foobar-decoded 6ch temp wav. Although it's somewhat earlier to talk about the encoding quality, I admit myself very much amused and encouraged by the quality service you provide in both ways. Thanks Nero...

By the way, maybe by the nature of the AAC codec, I found the AAC CBR bit rate fluctuates while decoding (with foobar faad & recent ffdshow with faad2 decoder) unlike other codecs, which was why I thought they were ABR or VBR before. Both CT & Nero version shows the fluctuations in CBR decodings. Any clarifications would be welcomed.

And while this version seems to support 4 bitrate modes : CBR(bitrate), VBR(quality/bitrate) and VBR(bitrate) 2pass with optional average segment period adjustment, I wonder if the last mode is anyhow related to the ABR or if the ABR mode is actually not released yet. Thanks again Ivan.

[edit] mistyping : CBR(quality/bitrate)->CBR(bitrate)

[CruNcher]

Hi Ivan :)
neroAacEnc.exe -br 160000 -2pass -if test.wav -of test

*************************************************************
* *
* Nero Digital Audio Reference MPEG-4 & 3GPP Audio Encoder *
* Copyright (C) 2006 Nero AG *
* All Rights Reserved Worldwide *
* *
* Package build date: May 1 2006 *
* *
* See -help for a complete list of available parameters. *
* *
*************************************************************
_


no current status indicator ?
at least something like 1pass running please wait... would be nice ;)

[setarip_old]

@Ivan Dimkovic

My compliments to both you and NERO ;>}

Is v.050106 the version that's been adjusted to deal properly with AMD systems?

[layer3maniac]

Negative, he said he hopes to get the fixed version uploaded tomorrow. BTW, my understanding is that it isn't an AMD problem, it's an SSE2 (or lack thereof, actually) problem. It affects PIII's too, but not AMD's which do have SSE2 support (like Athlon64's).

The (currently available) encoder requires an SSE2 capable processor. It has nothing to do with AMD, other than that the AMD Athlon XP doesn't have SSE2 capability (but neither does the Pentium III, whereas the Athlon64 or Opteron does and works with the published package).

[Teegedeck]

I couldn't believe my eyes when I saw that - three cheers and everything! ^_^

A full-blown CLI encoder. This makes things so much easier. Man... Thanks!:thanks:

[Latexxx]

Just curious; has work started on integrating this encoder with Exact Audio Copy & MeGUI?
No special integration is required with eac. Just select command line encoder, pick nero's encoder, and use -if %s -of %d as parameters.

[dimzon]

First of all, I would kindly ask you to stop this FUD - Nero Command Line encoder has absolutely no limits and it has highest possible quality - in fact, it has more options than Nero 7.2 plug-in, aimed for professionals.

- This is the first in the world, completely legal, and unlimited command line encoder (it is not a hack of third-party application)
Thanx a lot!!!
Still waiting for AMD XP fixed encoder (maybe you can temporaly upload it @ mytempdir.com or rapidshare.de ?)
Now some technical questions:

How does You detect source length ? Does You use appropriate field from WAV header structure or just read source until EOF? If You are using field from WAV header it means there are possible problems with long streams (WAV field is 32bit only ) Ups, there are ignorelength option, sorry
Is it possible to make special 2-pass mode for STDIN? In this case caller must provide double-length stream as source (repeat original source twice) and provide samplecount in original via special command-line parameter
Any chance to be able encode raw pcm ( headerless wav)? In this case caller must provide bitspersample/channelcount/samplerate and optional samplecount (for 2-pass from STDIN) via command-line.
Any chance to be able to encode to ADTS AAC (some muxers reque AAC without container).

BTW/Offtopic.
As I said before I'm even ready to paid for full Nero package to be able to encode without GUI (via command-line) so thanx You a lot for free solution
And I want same for AVC (I'm NOT asking for FREE solution, I'm ready to paid for full Nero package) but I want to be able to encode WITHOUT Your GUI too. Please, forward this statement to appropriate ppls from Nero team. I believe, development of the small cli wrapper around AVC DS encoding filter will be cost mostly nothing for You but will get great satisfaction for expirienced users!

BTW/Offtopic. I'm still banned @ HA forum - NOBODY contacted me Yet :scared:

is there a GUI available I'm not good with command lines i hate writing it in and all that
It is possible to write BeHappy extension for this encoder. I will write it right after AMD XP fix will be avaluable.

[shon3i]

Once again...
This is NOT "the same encoder from nero 7.2.03b but in exe file".
What is the different, this encoder have only 2pass. Or if this completly new codec comparing to 7.2.0.3b then congratulate to Ivan for writing totality new codec from scratch in this short period about 3 days.

[Ivan Dimkovic]

shon3i
About CBR. I think at last public test @48 nero uses ABR vs CT CBR and prove that ABR is better than CBR, everyone know that ABR is much better in any case. So Nero now have real CBR then if we have a real winner of some test we need to compare codec with same settings. Nero CBR vs CT CBR

Your claim 'ABR is much better in any case' is dead flat wrong. There was a listening test, comparing Nero ABR, Nero CBR and Nero VBR here:

http://www.hydrogenaudio.org/forums/index.php?showtopic=41191&hl=

And you can see that ABR is certainly not "much better in any case" but "just a little bit better" at least when it comes to 48 kbps bit rate.



What is the different, this encoder have only 2pass. Or if this completly new codec comparing to 7.2.0.3b then congratulate to Ivan for writing totality new codec from scratch in this short period about 3 days.

May 1st version contains:

- Slight improvement in Mid/Side coding
- Fixed LFE channel handling for 5.1 material (important!)
- Fixed HE-AAC v2 bug (small quality improvement)

So, for 5.1 content it should be a bit better ;)

[SeeMoreDigital]

Hi SMD,

Check out Belight next rev, Kurtnoise is stating (in a french forum) that he is working on it :)Thanks for the heads up Did.... I'll keep a watch out ;)

[shon3i]

Your claim 'ABR is much better in any case' is dead flat wrong. There was a listening test, comparing Nero ABR, Nero CBR and Nero VBR here:

http://www.hydrogenaudio.org/forums/...opic=41191&hl=

And you can see that ABR is certainly not "much better in any case" but "just a little bit better" at least when it comes to 48 kbps bit rate.ABR is always good than CBR, and i see that graphs milion times, You are use one sample to test differents, but different sample different case but always good for ABR ;

So, for 5.1 content it should be a bit betterI am wroting this post from my friend. And i now tested this cli and 5.1 CBR comparing to 5.1 CT CBR - not sounds good, 2pass 5.1 comaring to 5.1 CT CBR - sounds good but i think to hear some artefatcs in center channel when come some speech.

[Ivan Dimkovic]

ABR is always good than CBR, and i see that graphs milion times, You are use one sample to test differents, but different sample different case but always good for ABR ;

If you saw the graphs million times, I am somehow surprised that you haven't seen the test conditions at least half million times ;)

http://www.hydrogenaudio.org/forums/index.php?showtopic=40858



ftp://ftp.morbo.org/pub/Nero_48_Test/test1.rar -- Test 1 SETUP
ftp://ftp.morbo.org/pub/Nero_48_Test/test1_atrain.rar -- Test 1, Sample 1
ftp://ftp.morbo.org/pub/Nero_48_Test/test1_biniou.rar -- Test 1, Sample 2
ftp://ftp.morbo.org/pub/Nero_48_Test/test1_clapton_44k.rar -- Test 1, Sample 3
ftp://ftp.morbo.org/pub/Nero_48_Test/test1_es02.rar -- Test 1, Sample 4
ftp://ftp.morbo.org/pub/Nero_48_Test/test1_french_ad.rar -- Test 1, Sample 5
ftp://ftp.morbo.org/pub/Nero_48_Test/test1_harp40_1.rar -- Test 1, Sample 6
ftp://ftp.morbo.org/pub/Nero_48_Test/test1_mahler.rar -- Test 1, Sample 7
ftp://ftp.morbo.org/pub/Nero_48_Test/test1_sandman.rar -- Test 1, Sample 8
ftp://ftp.morbo.org/pub/Nero_48_Test/test1_si02.rar -- Test 1, Sample 9
ftp://ftp.morbo.org/pub/Nero_48_Test/test1_sm01.rar -- Test 1, Sample 10
ftp://ftp.morbo.org/pub/Nero_48_Test/test1_sympathy.rar -- Test 1, Sample 11
ftp://ftp.morbo.org/pub/Nero_48_Test/test1_te09_44k.rar -- Test 1, Sample 12
ftp://ftp.morbo.org/pub/Nero_48_Test/test1_yello.rar -- Test 1, Sample 13
ftp://ftp.morbo.org/pub/Nero_48_Test/test1_youcantdothat.rar -- Test 1, Sample 14

So it is 14 various samples, with different kinds of artifacts not a single sample.


I am wroting this post from my friend. And i now tested this cli and 5.1 CBR comparing to 5.1 CT CBR - not sounds good, 2pass 5.1 comaring to 5.1 CT CBR - sounds good but i think to hear some artefatcs in center channel when come some speech.

Would be good to have a listening test actually :)

[Ivan Dimkovic]

New Encoder is Uploaded!

[shon3i]

So it is 14 various samples, with different kinds of artifacts not a single sample.
Yes but ABR is always better, so we then use CBR instead right.


Would be good to have a listening test actually I am always on

[Ivan Dimkovic]

Yes but ABR is always better, so we then use CBR instead right.

It is always better, but if you take a look at the test results of 48 kbps tests, you would find out that the relative difference was smaller than relative difference between Nero ABR and CT CBR.

This does not mean too much, except the fact that Nero ABR scored little bit better than CT CBR, but not better enough to be called statistically significant.

If you wish to test Nero CBR vs CT CBR - please go on :) But I personally do not see too much point in such test - I dunno who would use inferior mode (CBR) when there is better mode available (ABR) - except for streaming, of course.

[shon3i]

This does not mean too much, except the fact that Nero ABR scored little bit better than CT CBR, but not better enough to be called statistically significant.No is much better in scores, but in practice it is. I am not participate in this test, but i listen all of this samples and i can only say that nero sounds better than CT in all cases. Sound like encoded in much bigger rate about 56,64kbs.

If you wish to test Nero CBR vs CT CBR - please go on But I personally do not see too much point in such test - I dunno who would use inferior mode (CBR) when there is better mode available (ABR) - except for streaming, of course.I think to new public test, if you want to find right winner. in other case like you say this no have sense. I think that will be Orban very fast add answer for this nero step

[shon3i]

@Ivan Dimkovic why i always get oversize when use boost. Bigger boost bigger oversize. In CT i use doble bigger bost newer get oversize.

[GmorG McRoth]

Hi I just downloaded updated encoder, could you explain why in package are 5 exes? 3 in win32 folder and 2 in root. what is diffrent in those 2 in root? are they there by accident?

[Ivan Dimkovic]

@Ivan Dimkovic why i always get oversize when use boost. Bigger boost bigger oversize. In CT i use doble bigger bost newer get oversize.

I don't understand what exactly do you mean:

- What bit-rate mode are you using? (abr/cbr/vbr/2-pass)

If you are using anything else than CBR expect the bit rate to change with the psychoacoustic properties of the signal - if your 'boost' operation increased the volume of the file, it could also mean than some parts that were judged to be below the threshold of hearing (and, hence, not coded) are now coded.

- CT is a Constant Bit Rate encoder - you can encode digital silence and you will still get the same size - if you want to use the same mode in Nero - use the -cbr switch, and the results are going to be exactly the same.

@GmorG McRoth, it seems exes were copied to two locations, they are the same - we'll replace this asap.

[shon3i]

- CT is a Constant Bit Rate encoder - you can encode digital silence and you will still get the same size - if you want to use the same mode in Nero - use the -cbr switch, and the results are going to be exactly the same.I am now encoded ac3(6ch) to mp4(6ch) via CT @ 96kbs HE and i get 88MB of final filesize, which is calculated by calculator from megui for that bitrate.

I do a same thing with nero using -cbr switch as you recommend and i get 97MB.

Boost applyed is +4.5dB.

I aslo tryed with 2pass mode. his downed file size to 93MB

[Ivan Dimkovic]

MPEG-4 file format can bring additional overhead to the file size. We calculate the bit rate of the elementary stream (AAC) inside the file, and excluding the anything else in the MPEG-4 container.

[dimzon]

New Encoder is Uploaded!
BeHappy extension to support this perfect cli encoder is uploaded @ BeHappy Workspace @ gotDotNet (http://gotdotnet.com/Workspaces/Workspace.aspx?id=1bb59ddf-901b-43a5-bd54-b0999e8e223e)

[shon3i]

Thanks A Lot dimzon, and Ivan of course for encoder

MPEG-4 file format can bring additional overhead to the file size. We calculate the bit rate of the elementary stream (AAC) inside the file, and excluding the anything else in the MPEG-4 container.You think when i extract mp4 file to aac i get correct size.

[Ivan Dimkovic]

Yes - elementary stream should be fully compatible with the definition of 'constant bit rate' in ISO/14496-3 criteria.

BUT - you have to extract to headerless or ADIF aac elementary stream - if you extract it to ADTS AAC, extraction tool with add additional overhead of 7 bytes to each frame - which can be quite a lot if we are talking about a movie - since there can be e.g. 40 frames per second :)

[shon3i]

BUT - you have to extract to headerless or ADIF aac elementary stream - if you extract it to ADTS AAC, extraction tool with add additional overhead of 7 bytes to each frame - which can be quite a lot if we are talking about a movie - since there can be e.g. 40 frames per second OMG, so what to do then, im just extracted from mp4 to ADTS AAC i get for 97 to 95, 93 to 90MB

[Ivan Dimkovic]

I think that mp4creator from MPEG4IP has the option to extract to headerless AAC tracks - there you can get the exact bit-rate of AAC stream, without any overhead.

But I am not sure why do you need that for? Headerless AAC is almost useless.

[dimzon]

@Ivan Dimkovic
Any response on this post?
http://forum.doom9.org/showthread.php?p=823740#post823740

Thanx


Add:
Can You post here link to NeroDigitalAudio logo? Thanx!

[Ivan Dimkovic]

@Dimzon,

I just forwarded this message to the developers working on the frontend.

I can only answer for ADTS AAC - and the answer is, no. We do not plan to support ADTS AAC as many of the encoder features are impossible with this outdated transport multiplex format.

[SeeMoreDigital]

But I am not sure why do you need that for? Headerless AAC is almost useless.Indeed...

I can't ever remember having sync issues with my AAC streams... and I've de-muxed them from MP4 files containing ASP video and re-muxed them into MP4 files containg AVC video.

I've even generated 6Ch AAC streams (from the same AC3 source) using different encoders (Nero, FAAC, WinAMP) and muxed them into the same MP4 file (with video), so I can flick thru' them to compare any differences...

I do it all now with YAMB (before that it was mp4UI) and they are always in sync ;)

[shon3i]

I think that mp4creator from MPEG4IP has the option to extract to headerless AAC tracks - there you can get the exact bit-rate of AAC stream, without any overhead.

But I am not sure why do you need that for? Headerless AAC is almost useless.I must down file size to mux vith video stream right. I get final filesize about 10mb larger instead 695mb i get 707MB which can fit on standard CD.

btw. Ivan why 2pass not work with stdin. This is powerfull option for long encodings can't be used for movies. I am not crazy to convert ac3->5.1wav

btw. i do some other test with 5.1 CBR encoding en i get good quality, with 2pass i get litle bit better quality than CT

[Ivan Dimkovic]

I must down file size to mux vith video stream right. I get final filesize about 10mb larger instead 695mb i get 707MB which can fit on standard CD.


If you are muxing from MP4 file, you can be sure that only elementary AAC packets are being muxed - not the whole MP4 file, so you do not to worry if you selected -cbr switch.

btw. Ivan why 2pass not work with stdin. This is powerfull option for long encodings can't be used for movies. I am not crazy to convert ac3->5.1wav

It does not work ATM as it would be somewhat complicated to support that - but we will see if there is a way to do it.

[shon3i]

If you are muxing from MP4 file, you can be sure that only elementary AAC packets are being muxed - not the whole MP4 file, so you do not to worry if you selected -cbr switch.Yes i use cbr switch but get oversize instead 88 i get 97 which in final file result 707MB.

It does not work ATM as it would be somewhat complicated to support that - but we will see if there is a way to do it.I hope is that soon, but VBR mode can change becouse is very good. Is there any chance to integrate ReplyGain, and M/S on/of

[killerhex]

anyone have a GUI for this or can make one hving trouble wih comandline

[dimzon]

It does not work ATM as it would be somewhat complicated to support that - but we will see if there is a way to do it.
As I said before caller can send double-sized original to encoder's stdin (repeat original twice) and provide valid samplecount via command line. So You can write something like this


readWavHeaderFromStdIn();
int64 sampleCount = getInt64ArgumentFromCommandLine("sampleCount")
Initialize1stPassEncoder();
for(i=0;i<sampleCount;i++)
{
readSampleFromStdInAndFeedItTo1stPassEncoder()
}
Finalize1stPassEncoder();
Initialize2ndPassEncoder();
for(i=0;i<sampleCount;i++)
{
readSampleFromStdInAndFeedItTo2ndPassEncoder()
}
Finalize2ndPassEncoder();

[dimzon]

anyone have a GUI for this or can make one hving trouble wih comandline
BeHappy ;)

[GmorG McRoth]

I have problem with behappy when I try to encode ac3 file (using nicaudio plugin from behappy site) to ND AAC CBR @ 224 kbps it fails on me with this log
nero encoder and decoder are in same dir as BeHappy
Starting job King Kong T01 3_2ch 448Kbps DELAY 0ms.ac3->audio.m4a
Found Audio Stream
Channels=6, BitsPerSample=16, SampleRate=48000Hz
neroAacEnc.exe -cbr 224000 -ignorelength -if - -of "D:\Moje wideo\DVD\King Kong\audio.m4a"
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Error: System.IO.IOException: Potok został zakończony.

at System.IO.__Error.WinIOError(Int32 errorCode, String maybeFullPath)
at System.IO.FileStream.WriteCore(Byte[] buffer, Int32 offset, Int32 count)
at System.IO.FileStream.Write(Byte[] array, Int32 offset, Int32 count)
at BeHappy.Encoder.encode()
#### Encoder StdOut ####
ERROR: could not parse WAV file

#### Encoder StdErr ####
*************************************************************
* *
* Nero Digital Audio Reference MPEG-4 & 3GPP Audio Encoder *
* Copyright (C) 2006 Nero AG *
* All Rights Reserved Worldwide *
* *
* Package build date: May 1 2006 *
* *
* See -help for a complete list of available parameters. *
* *
*************************************************************

[Sirber]

woot woot!

thanks! :D

[dimzon]

ERROR: could not parse WAV file

Hmmm... Seems like something is wrong in Nero encoder itself! Because BeHappy produce 100% valid WAV HEADER (and it was tested multiple times!)

Ivan, can You check it or at least provide binary wich produce additional diagnostic output (not only "could not parse WAV file")

Add: this is some code wich I'm using to create WAV header: http://forum.doom9.org/showthread.php?p=758659#post758659
take look @ FAAD_MAGIC_VALUE usage. Maybe it's a reason (according source filename "King Kong T01 3_2ch 448Kbps DELAY 0ms.ac3" I think it produce WAV > 4GB so 0xFFFFFF00 is used instead actual data size).

[Ivan Dimkovic]

Dimzon,

We will look into the issue - please send us the wave file itself that causes problems (you can dump it to the file) and we will check

[Sirber]

Added on digg:
http://www.digg.com/technology/Nero_Releases_FREE_Reference_Quality_MPEG-4_Audio_Command_Line_Encoder

@Ivan Dimkovic
I used a bit of your first post :D

[dimzon]

Dimzon,

We will look into the issue - please send us the wave file itself that causes problems (you can dump it to the file) and we will check
please read this post again (it was updated) http://forum.doom9.org/showthread.php?p=823850#post823850
I beleive reason is bcz uncompressed ac3 file is bigger than 4GB!
This is the problem for the wav limit 4 GB. And 2 GB limit is for soft using 'signed long int' for these fields.
The faad decoder (aac) can generate 'invalid' wav's > 4 GB, and fill these fields (RiffLength and DataLength) with 0xFFFFFF00.
Foobar also make 'invalid' wav's > 4 GB, and fill these fields with the 4 low order bytes.
I prefer the faad method (0xFFFFFF00) because Foobar method can be confused with valid fields.

[GmorG McRoth]

please read this post again (it was updated) http://forum.doom9.org/showthread.php?p=823850#post823850
I beleive reason is bcz uncompressed ac3 file is bigger than 4GB!
This is the problem for the wav limit 4 GB. And 2 GB limit is for soft using 'signed long int' for these fields.
The faad decoder (aac) can generate 'invalid' wav's > 4 GB, and fill these fields (RiffLength and DataLength) with 0xFFFFFF00.
Foobar also make 'invalid' wav's > 4 GB, and fill these fields with the 4 low order bytes.
I prefer the faad method (0xFFFFFF00) because Foobar method can be confused with valid fields.
yes I decompressed it to wav and file has over 6GB size and is not properly played by foobar (shows only 2 hours run time and ac3 is 3 hours).

[Ivan Dimkovic]

Are you using -ignorelength for setting up the AAC encoder commandline in case of this file?

[GmorG McRoth]

Are you using -ignorelength for setting up the AAC encoder commandline in case of this file?
No actually did not used that switch when I was experimenting with that file, but it's not the case since I probably won't use direct from wav encoding since it's not convenient. I'm more interested in making BeHappy work (and BeHappy uses this switch).

I just made a test just for fun with this command line

"C:\Program Files\Moje Programy\AAC nero\neroAacEnc.exe" -br 224000 -2pass -ignorelength -if "D:\Moje wideo\DVD\King Kong\King Kong T01 3_2ch 448Kbps DELAY 0ms.wav" -of "D:\Moje wideo\DVD\King Kong\King Kong T01 3_2ch 448Kbps DELAY 0ms.m4v"

and there is "could not parse the wav file" error

sidenote (I renamed sse2 version to not sse2 version name to not need to change all settings I made in foobar2000)

[Ivan Dimkovic]

@Dimzon,

Could you please confirm that you set -ignorelength in case of this file? Because if the -ignorelength is specified, encoder ignores the WAV headers with regards to length.

[buzzqw]

could be possible to implement direct ac3 input support ?

:thanks:

BHH

[dimzon]

@Dimzon,

Could you please confirm that you set -ignorelength in case of this file?
Yes. As You can see above in BeHappy log actual command line is
neroAacEnc.exe -cbr 224000 -ignorelength -if - -of "D:\Moje wideo\DVD\King Kong\audio.m4a"

[Ivan Dimkovic]

could be possible to implement direct ac3 input support ?

No this is not possible - it is very hard to provide free codecs when there are royalties that have to be paid - please take into consideration that Nero Command Line AAC encoder is licensed AAC distribution - we do pay for that.

@Dimzon, thx - I informed the engineers about this problem.

[Sharktooth]

Great work Ivan!
Nice to see Nero is still a friendly company:)

[dimzon]

could be possible to implement direct ac3 input support ?
Use foobar2000 or behappy to transcode ac3
or (if You need 100% CLI solution) pipeline decoding via ffmpeg to neroaacenc ;)

[buzzqw]

thanks to both Ivan and Dimzon

(if You need 100% CLI solution) pipeline decoding via ffmpeg to neroaacenc

nice tips (but is ffmpeg supporting 5ch to stdout?, i will try)

BHH

[Rockaria]

btw. Ivan why 2pass not work with stdin. This is powerfull option for long encodings can't be used for movies. I am not crazy to convert ac3->5.1wav
It does not work ATM as it would be somewhat complicated to support that - but we will see if there is a way to do it.
The ideal way would be 1st pass analysis-> rewind -> 2nd pass encoding like what avisynth is doing with 2 pass volume gain.
For a temporary gui integration work around, foobar2k's %s(instead of -) input file definition for the cli encoding(-br 128000 -2pass -if %s -of %d) will use a temporary wav in the destination folder, which would be a lot faster if you can use the ram disk large enough depending on the available extra ram(the virtual memory use would be a lot smoother though).

- What bit-rate mode are you using? (abr/cbr/vbr/2-pass)
-br/-cbr but no -abr/-vbr
-2pass/-2passperiod but why no -1passperiod for -br?
-segment/period size for -cbr(the segment seems not be 1 sample)/-abr/-vbr(entire length segmented abr?) The reason why I asked for the clarification of the bit rate options in my previous post is that we seem to use different terms for the unclarified similar phenomena.
As both quality based & bitrate based options are widely used for VBR(ie. vorbis, lame...). The difference between the ABR and VBR for the bitrate based option seems to be lying in the segment/period size(samples/ms).
Also check, the CBR for AAC seems to be natively ABR(nSamples for the unit segment), which can be verified watching the dynamic bit rate status within the foobar2k or ffdshow. Any inputs?

[dimzon]

The ideal way would be 1st pass analysis-> rewind -> 2nd pass encoding like what avisynth is doing with 2 pass volume gain.
It's impossible for STDIN...

[Rockaria]

It's impossible for STDIN...That seems to be true without any callbacks to the middleware or server...

[MatMaul]

or like xvid for example, two switch (-pass1 and -pass2 for example) : the first pass creates a stats file and the 2nd pass reads it

[tebasuna51]

This is the problem for the wav limit 4 GB.
The faad decoder (aac) can generate 'invalid' wav's > 4 GB, and fill these fields (RiffLength and DataLength) with 0xFFFFFF00.
Foobar also make 'invalid' wav's > 4 GB, and fill these fields with the 4 low order bytes.
I prefer the faad method (0xFFFFFF00) because Foobar method can be confused with valid fields.
New Foobar 0.9 fill these fields with:
RiffLength = 0x7FFFFFEC
DataLength = 0x7FFFFFC8
Both invalids for wav's > 2GB, but work ok with neroAacEnc -ignorelength because RiffLength - DataLength must be equal to 36 (for this kind of wav header).

I check wav's > 4GB with any value in DataLength and +36 in RiffLength and always work with neroAacEnc -ignorelength, then the problem is use same value for the two fields.

For BeHappy the solution is fill DataLength with 0xFFFFFEDC (0xFFFFFF00 - 36).

For neroAacEnc the solution, if want support wav's > 4GB in Faad style, is ignore also the coherence between RiffLength and DataLength not only the real length.

EDIT: NeroAacDec is also capable to create wav's > 4GB, send a "overflow" message but continue to write the full wav (at last with a 130 min. m4a 5.1). And fill RiffLength and DataLength with =0x00000000 (invalid for neroAacEnc).

[Rockaria]

or like xvid for example, two switch (-pass1 and -pass2 for example) : the first pass creates a stats file and the 2nd pass reads it
My previous 2 pass gui server integration suggestion for the cli encoder actually performs 3i & 1.5o HD access(1i & 1o from the server and 2i & 0.5o by the encoder).
There are some possible performance improvements I can think of for now :

1. if the cli encoder manages the temporary wav, it will be 2i & 1.5o, reducing 1i.

2. if the server & encoder can handle 2 pass as an option(stdout 2 times in a session), it becomes 2i & 0.5o, no temp wav access, same effect as callback-rewind, but a bit lighter than the above 2 session method.

3. using the ram disk will result in 1i & 0.5o effect but would be subject to the short clip(music) use only, depending on the extra available ram size.

4. if the encoder decides to support 1 pass vbr/abr segment size adjustment, it would be the normal 1i & 0.5o transcoding but under the constraints of system resources(virtual memory) and encoding quality by the segment size & bit allocation algorithm.

[dimzon]

@Ivan
How to map old vbr modes (tape/radio/internet/etc) to new Q parameter (best fit)
I'm planning to write drop-in bsn.dll replacement to be able to use new Nero encoder via BeSweet :)

[shon3i]

| Old Presets | Quality Levels |
| Tape | -q=0.0 |
| Radio | -q=0.1 |
| Internet | -q=0.2 |
| Streaming | -q=0.3 |
| Normal | -q=0.5 |
| Extreme | -q=0.6 |
| Audiophile | -q=0.8 |
| Transcoding | -q=1.0 | Something like this

[Kurtnoise]

no...according to Garf, this is completely useless now.

[shon3i]

Now is just beter add slider with quality numbers instead these profiles.

[imcold]

Ivan & Nero: Thank you very much! ;)

[Oline 61]

Thanks Nero. I have been using Nero 6, and I really wanted the Nero 7 AAC codec, but didn't want to go whole hog and buy the package.

[buzzqw]

@tebasuna51
so... if i use azid.exe (or using besweet ) to demux a 5.1ch ac3 to 5.1 wav and this wav is over 4gb , i will however able to encode this wav with neroaacenc without problem ?

thanks for answer !

BHH

[tebasuna51]

@buzzqw
Sorry, I don't test if azid can make wav > 4GB, and can't test now.

If you have this wav > 4GB try with neroAacEnc -ignorelength, the message:
"ERROR: could not parse WAV file"
is instantaneous.

[Inc]

@tebasuna51
so... if i use azid.exe (or using besweet ) to demux a 5.1ch ac3 to 5.1 wav and this wav is over 4gb , i will however able to encode this wav with neroaacenc without problem ?

thanks for answer !

BHH
Did you try using Azid.exe serving the data via stdout (not shure if its possible)?
Also what about the channel order? Do ffmpeg.exe/Azid.exe realign the channelorder to 5.1 Wav specs or are these channels just passed through in their given source order?

[jjseth]

Originally Posted by Kurtnoise from HA forum
| Old Presets | Quality Levels |
| Tape | -q=0.0 |
| Radio | -q=0.1 |
| Internet | -q=0.2 |
| Streaming | -q=0.3 |
| Normal | -q=0.5 |
| Extreme | -q=0.6 |
| Audiophile | -q=0.8 |
| Transcoding | -q=1.0 |

New settings posted by Garf:

Average bitrate <-> Quality table

Bitrate->Quality
15----->0.05
32----->0.15
63----->0.25
99----->0.35
146---->0.45
197---->0.55
248---->0.65
299---->0.75
350---->0.85
401---->0.95


Using VBR mode (-q) will give the best quality. The default setting is -q 0.5, which will give about 160-170kbps on average. For ABR, use the -br options. For CBR, use -cbr. Note that the bitrate is in bits per second, not kilobits.

Note that using 2-pass encoding makes no sense in VBR mode!

When encoding from the commandline, sending the input over stdin is supported, but then you should also use -ignorelength. This is not needed when using temporary files.

Settings:

foobar2000 0.9.x (recommended over 0.8.3):

Extension: MP4 (or M4A, if you prefer)
Parameters: -ignorelength -q 0.5 -if - -of %d
Highest BPS mode supported: 32

EAC:

-if %s -of %d

[castellanos]

Hi guys!
Sorry for the question but, Can somebody provide a command line example, so I can see how it works?
I have been trying:
"D:\Nero\Win32\neroAacEnc.exe" -he -cbr 160000 -if "D:\Work\The Swan.wav" -of "D:\Work\The Swan.mp4"
But no luck.
Greetings!

[Drachir]

Do ffmpeg.exe/Azid.exe realign the channelorder to 5.1 Wav specs or are these channels just passed through in their given source order?

At Linux it seems to work this way:

>mkfifo audio.wav
>wine ./neroAacEnc.exe -ignorelength -q 0.3 -if audio.wav -of ger.mp4 & mplayer dvd:// -alang de -vc null -vo null -af channels=6:6:0:0:1:1:2:4:3:5:4:2:5:3 -ao pcm:waveheader:file=audio.wav -channels 6

Not sure if the channel order is right, I have only stereo speaker.

[buzzqw]

@Inc

azid.exe piping is know to be broken :(

and about channel order... i truly don't know !

BHH

[Rockaria]

Note that using 2-pass encoding makes no sense in VBR mode!
The keyword seems to be variable bit rate per second.

http://en.wikipedia.org/w/index.php?title=Average_bit_rate&oldid=44638008
Average bit rate refers to the average amount of data transferred per second. This is commonly referred to for digital music or video. An mp3 file, for example, that has an average bit rate of 128 kbit/s transfers, on average, 128,000 bits every second. It can have higher bit rate and lower bit rate parts, and the average bit rate is obtained by dividing the sum of the bit rate of each sample by the number of samples. Bit rate is not the only measure of audio/video quality, as some formats such as wma and Vorbis produce higher sound quality than the standard mp3 format at the same bit rate.
http://en.wikipedia.org/w/index.php?title=Variable_bit_rate&oldid=47326883
Variable bit rate (VBR) is a term used in telecommunications and computing that relates to sound or video encoding. As opposed to constant bit rate (CBR), VBR files vary the amount of output data per time segment. VBR allows a higher bit rate (and therefore more storage space) to be allocated to the more complex segments of sound files while less space is allocated to less complex segments. The average of these rates is calculated to produce an average bit rate for the file that will represent its overall sound quality.

MP3, WMA, Vorbis, and AAC files can optionally be encoded with VBR.
An interpretation identical to some recent definitions or trends..
http://www.hydrogenaudio.org/forums/index.php?showtopic=30553&st=0&p=264475&#
Nature of modes:
CBR - gives you constant bitrate through stream and predictable size. Quality can vary through the stream.
VBR - gives you constant quality through the stream (is controled via number of parameters) and completely unpredictable size.
ABR - gives you more constant quality than CBR while maintain some control over bitrate.
Note: actually, many new standarts like for example AAC have no CBR mode at all. They always working in ABR, using bit reservoir. If bit reservour depth is no more than defined in standart for used mode, then such a mode is called "CBR" while technically it is not.
My understanding based on the more formal definition so far is that :

. The ABR keeps the equal bit rate per second time frame, but allocates bits variably within the frame based on the complexity. It allocates equal bits to each second.

. So using the ABR for streams having variable bit rates over each second frame sounds quite confusing.

. If allocating even bits(ABR) on each unit(period/segment/frame/reservoir : sample<second<file) but distributing variably depending on the complexity within is the goal, we can safely say ABR is a subset of bitrate based VBR which has both controls over size(overall) and even quality(inside).

Although we are given the freedom of creative expression, if anybody wanna influence the majority reasonably with the new definitions, please update the Wikipedia (http://en.wikipedia.org/w/index.php?title=Average_bit_rate&oldid=44638008).

[Pasqui]

Hi guys!
Sorry for the question but, Can somebody provide a command line example, so I can see how it works?
I have been trying:
"D:\Nero\Win32\neroAacEnc.exe" -he -cbr 160000 -if "D:\Work\The Swan.wav" -of "D:\Work\The Swan.mp4"
But no luck.
Greetings!

It makes no sense to activate HE at such a high bitrate (unless your WAV file is 5.1). Remove "-he" and you will get a perfect CBR AAC LC 160kbps file.

[castellanos]

Thanks Pasqui!
I got it! :cool:

[NWDaniels91]

I ran some tests and I have a question. Here's what I tried:

"-q - 0.4 -2pass" and it returned a 126kbps LC-AAC file
"-q - 0.3 -2pass" and it returned a 83kbps HE-AAC file
"-q - 0.3 -lc -2pass" and it returned a 131kbps LC-AAC file

I realize that forcing LC or HE is not recommended, but in this test, setting the quality to 0.3 gave a higher bitrate than 0.4. Can anyone give a reason why this would be?

[GmorG McRoth]

2-pass mode is only for -br switch not -q

[NWDaniels91]

2-pass mode is only for -br switch not -q

Ah, thanks.

New question: Why can't Nero ShowTime 2 play back these files?

[GmorG McRoth]

what extansion you gave them? m4a or mp4? booth shuld work, I had problem becouse accidentaly I was naming them m4v (I made a typo) and foobar2000 didn't want to play them. I had to rename extansion. though nero show time is video player more than audio player so mayby it can't play audio. I can't check it out myself becouse im not using showtime.

[NWDaniels91]

I used the .mp4 file extension (of course). Nero ShowTime can play .mp3 and .mpa audio files, and it can play AAC .mp4 audio files when encoded with Nero's GUI, but it can't play AAC .mp4 files encoded by the Nero CLI. It says:

Cannot play this media file. The file is either corrupt or the application does not support the format you are trying to play.

[SeeMoreDigital]

I used the .mp4 file extension (of course). Nero ShowTime can play .mp3 and .mpa audio files, and it can play AAC .mp4 audio files when encoded with Nero's GUI, but it can't play AAC .mp4 files encoded by the Nero CLI. It says:

Cannot play this media file. The file is either corrupt or the application does not support the format you are trying to play.Come on guys...

You can't simply "re-name" the file extension from .AAC to .MP4 and expect these files to play in ShowTime player.

You must correctly mux the .AAC streams into the .MP4 container....


Cheers

[shon3i]

the file extension from .AAC to .MP4hmm, i think is that not a good idea. AAC is raw, so better use yamb/mp4box to mux this aac into mp4

[GmorG McRoth]

If I'm not mistaken nero AAC encoder creates mp4 files, so renaming them to m4a is not that dreadfull.

[tebasuna51]

@tebasuna51
so... if i use azid.exe (or using besweet ) to demux a 5.1ch ac3 to 5.1 wav and this wav is over 4gb , i will however able to encode this wav with neroaacenc without problem ?
I make a test decoding with Azid a 130 min ac3, the wav output is > 4GB and with coherent RiffLenth-DataLengh then is accepted by neroAacEnc -ignorelength.

The command used is:
azid -d3/2 -L0 -l1 -ol,r,c,lfe,sl,sr test.ac3 test.wav

you can see also how select the correct channel order for the wav6 with the -o parameter.

Other useful, but optional, parameters are:
-c normal (to apply DRC)
--maximize (to maximize the output)

[NWDaniels91]

Come on guys...

You can't simply "re-name" the file extension from .AAC to .MP4 and expect these files to play in ShowTime player.

You must correctly mux the .AAC streams into the .MP4 container....


Cheers

If I have this output a .aac file, it will play in MPC but will not play in foobar2000 or ShowTime. If I try to mux it into .mp4 with YAMB, it just creates an empty 8.7 KB file.

If I have this output a .mp4 file, it will play in MPC and foobar2000, but not ShowTime. If I try to mux it into .mp4 with YAMB, it correctly creates an .mp4 file that can be played in ShowTime.

What's up with this weird behavior? I'm no expert when it comes to YAMB, but says it can mux raw AAC files, so why does it have problems with these?

I have re-read this thread, and it doesn't say anywhere whether this is in fact encodes to raw AAC file or if it writes into the MP4 container. According to Nero, this program can "Store Entire Audio Album in a Single .mp4 File." So which is it?

[DeathTheSheep]

I was amazed with the quality your encoder offers at VBR q 0.15, but disappointed with bitrates only slightly lower due to the incredibly aggressive lowpass. Since I'm a user of mobile devices which respond very badly to HE-AAC, I was actually wondering if there is/can be a way for the advanced user to specify the desired lowpass frequency in the CLI for low-bitrate LC-AAC encodings--I've found a lack of annoying artifacts at bitrates as low as those attained by the use of q 0.15, much to my pleasant surprise.

I'm hoping, however, that your superb LC-AAC codec is capable of competing with Vorbis at 64kbps with high lowpass frequencies. I don't mind the introduction of a few minor encoding artifacts as long as the high-end of my audio is more faithfully preserved (I also happen to listen to various pop music which responds very well to a heightened or non-existent lowpass, even at what are often considered "insanely" low bitrates for other types of music).

Thanks for considering my proposal! (I also posted over at HA in less detail).

[killerhex]

is there a cdripper that supports this encoder

[DeathTheSheep]

Sure. Any CD ripper that supports encoding to custom CLI with "stdout" will have a blast with this thing. :P Heck, even good ol' fb2k.91 can use it... ;)

[DeathTheSheep]

AAC is raw, so better use yamb/mp4box to mux this aac into mp4
It already is. The encoder produces standard-complient MP4 AAC files.
I realize that forcing LC or HE is not recommended, but in this test, setting the quality to 0.3 gave a higher bitrate than 0.4. Can anyone give a reason why this would be?
Once a specific mode is forced, the quality scale is altered to better suit the quality of the mode. -q 0 for HE-AACv2 is obviously going to produce lower bitrate files than -q 0 for LC-AAC, etc.

[chipzoller]

But this encoder doesn't accept AC3 as a valid input file, so would this be a future addition, or is there something else to be done to have DVD audio tracks transcoded to AAC using this new app.?

[layer3maniac]

@chipzoller
Convert ac3 to wav, feed wav into encoder, perhaps?

[buzzqw]

But this encoder doesn't accept AC3 as a valid input file, so would this be a future addition, or is there something else to be done to have DVD audio tracks transcoded to AAC using this new app.?

i asked the same question few posts before. Read. (but as short answer NO, since for decoding ac3 royalties but be paid)

About wav file >4GB+ , i decoded "My fair lady" to 5.1 wav file with BeSweet and get 5.34 GB file.

NeroAACEnc encoded this file perfectly. :cool:

thanks to all (an now wait for Besweer support with some Dimzon Plugins !!!)

BHH

[GmorG McRoth]

check run time, when I was encoding big wav, output had 2 hours runtime (one hour missing).

[Elias]

I am pleased to announce the launch of FREE Reference Quality MPEG-4 Audio solution from Nero, in the command line form!

* First in the world FREE 2-Pass MPEG-4 AAC Encoder
* Compression Ratios ranging from ultra high (58 CDs fit on one!) to High-End Audio (2.5:1), for absolutely perfect audiophile encodings
* Crystal Clear, Award Winning Sound Quality at every compression ratio and bit rate!
* Support for Embedded Album Art (Covers, Booklets, Lyrics!)
* Store Entire Audio Album in a Single .mp4 File with all the Features of an Audio CD embedded inside, but at a fraction of the space!
* Reference Quality MPEG-4 Audio Codec
* Fully Compatible with the Latest Version of the State-of-the-art MPEG-4 Audio Standard (LC-AAC, HE-AAC and HE-AAC v2)

http://www.nero.com/nerodigital/eng/Nero_Digital_Audio.html
Now this was pretty awesome. Thanks a lot, Nero!

[GmorG McRoth]

I especially like this part:

FREE 2-Pass MPEG-4 AAC Encoder

[buzzqw]

@GmorG McRoth

:eek: :eek: wav audio was truncate !!!

original movie lenght ac3: 2h 46min 10sec
wav audio lenght: 2.04.17
mp4 lenght : 2.04.17 (fully playable and syncronized with wav)

the problem is on besweet !!! (i used the stable belight packace for 5.1 ac3 -> 5.1 wav)

:mad:

BHH

[GmorG McRoth]

I say, wav was fine just most readers can't get past 4GB point with reading.

[tebasuna51]

i decoded "My fair lady" to 5.1 wav file with BeSweet and get 5.34 GB file.
...
original movie lenght ac3: 2h 46min 10sec
wav audio lenght: 2.04.17
mp4 lenght : 2.04.17 (fully playable and syncronized with wav)

the problem is on besweet !!! (i used the stable belight packace for 5.1 ac3 -> 5.1 wav)
Your 5.34 GB wav file have about 2 h. 46 m.

I test BeSweet v1.5b31 with azid.dll v1.9 (b922) by Midas and work for wav > 4GB, like azid.exe v1.9 tested before http://forum.doom9.org/showthread.php?p=824344#post824344

With BeSweet-azid DataLength is fixed to 2:04:16.54 (0xFFFFFFD3) then if you use neroAacEnc without -ignorelength your mp4 must be 2:04:16.54

But, if you use -ignorelength, the mp4 must have the full length (works for me).

[buzzqw]

i reencoded the wav with ignorelegth options.

:thanks: i got the right lenght, even if wav is reported as 2.04.17 the mp4 is of 2.46.10 , the reight length !


i have to say that for an original ac3 (5.1ch 384kbs) of 467351kb ( 456mb) i got with -q 0.4 (not specifing lc or he) an mp4 with size bigger than orginal ! (468 mb, average bitrate -with foobar - of 392) :confused:

now trying with -he options (and -q 0.4)

i will report

BHH

[chipzoller]

@ layer3maniac

Convert ac3 to wav, feed wav into encoder, perhaps?

Surely you know that every time you convert to different formats you ineviteibly loose some detail. What you suggest is not an ideal process.

@ buzzqw

i asked the same question few posts before. Read. (but as short answer NO, since for decoding ac3 royalties but be paid)

I did search for 'AC3' using the 'seach this thread' option but found no results (and still do not). Maybe something is up here.

Thanks, I'll stick with the encoder in belight for now.

[Ivan Dimkovic]

Surely you know that every time you convert to different formats you ineviteibly loose some detail. What you suggest is not an ideal process.

You do not lose any detail by converting lossy source to a lossless format, such as linear PCM.

AAC encoder anyway deals with linear PCM input - so there should be no difference whether you feed it frame-by-frame with some decoded data (that has to be in the PCM format anyway) - or you decode to PCM and then encode all that data separately.

[chipzoller]

You do not lose any detail by converting lossy source to a lossless format, such as linear PCM.

O...I did not consider that aspect. I thus stand corrected :)

[fakey]

Thank you for releasing this handy but powerful encoder.:thanks:

I've got two questions while playing with this.

New settings posted by Garf:
Highest BPS mode supported: 32


1. Would it be a better idea to feed the encoder with 32bit dithered one(by foobar)?

2. When I set 2pass -q mode, what happens inside the encoder logic?
Fallback to corresponding 2pass -br mode?, 1pass -q mode? or maybe something hybrid?
I've read that 2pass with q setting doesn't make sense, but just outa curiosity.:D

[buzzqw]

just for sake purpose i encoded the same wav to -q 0.4 HE

got 357mb file, against a 468mb (-q 0.4 LC) and an original 456mb ac3

BHH

[Inc]

i asked the same question few posts before. Read. (but as short answer NO, since for decoding ac3 royalties but be paid)

About wav file >4GB+ , i decoded "My fair lady" to 5.1 wav file with BeSweet and get 5.34 GB file.

NeroAACEnc encoded this file perfectly. :cool:

thanks to all (an now wait for Besweer support with some Dimzon Plugins !!!)

BHHI just compiled the a52dec.exe binary. It seems that it comes as a playback binary only?!
Well anyhow ... maybe someone could modify the code so the decoded pcm data will be served incl. wav header via stdout.
This would make ac3->aac transcoding much more easier without big temp wav file generations.

EDIT:

Seems it depends on the output mode setting ......

The help offers this:
usage: d:\a52dec-0.7.4\a52dec-0.7.4\src\a52dec.exe [-o <mode>] [-s [<track>]] t <pid>] [-c] [-r] [-a] \
[-g <gain>] <file>
-s use program stream demultiplexer, track 0-7 or 0x80-0x87
-t use transport stream demultiplexer, pid 0x10-0x1ffe
-c use c implementation, disables all accelerations
-r disable dynamic range compression
-a disable level adjustment based on output mode
-g add specified gain in decibels, -96.0 to +96.0
-o audio output mode
win
windolby
wav
wavdolby
aif
aifdolby
peak
peakdolby
null
null4
null6
float

The a52dec.exe binary: http://www.mytempdir.com/650299

So if -o wav is given then maybe we also stdout could work ... actualy I have no time this night, so maybe someone likes to try

yepp ....:

a52dec.exe -o wavdolby "X:\myAudio.ac3" > X:/myAudio.wav

but still downmixed to 2ch .....

[drew~y]

Hi, thanks for the encoder!

does it make use of dual core cpu's ?

[Inc]

The binary above imho only makes sense if we can make it work outputting 5.1 channels also for serving to the Nero aac binary.
No matter if using -o wav or -o wavdolby .... both do result in a wav of the same final size and channels (2).
PS: Yep, the Source was 5.1. ;)

[dohcmark8]

I am having trouble getting true CBR files out of the encoder.. I am using "-ignorelength -cbr 160000 -lc -if - -of %d" and when I play in Showtime it shows the bitrate constantly changing...

[Ivan Dimkovic]

AAC has bit-reservoir which is up to 12 kbits large for stereo files.

So even 'CBR' files can fluctate as long as fluctation does not violate the bit-buffer limits.

[fight2win]

what all files should i have in belight folder so that i can encode ac3 to aac using this new nero encoder?

[Kurtnoise]

BeSweet/BeLight doesn't support this encoder yet.

[fight2win]

ok, thanks.

[NWDaniels91]

I don't know who is correct, but at least there is some confusion that someone should clear up. From the Doom9 front page:

Nero has launched a free AAC audio encoder that does two pass encoding and supports all AAC extensions (HE, PS). The encoder is already supported in BeLight and will soon be integrated into MeGUI.

[FFWD]

Kurtnoise13 makes BeLight, not Doom9...

[Kurtnoise]

I think Doom9 has mixed up BeLight with BeHappy...;)

[cgrey]

* Support for Embedded Album Art (Covers, Booklets, Lyrics!)
* Store Entire Audio Album in a Single .mp4 File with all the Features of an Audio CD embedded inside, but at a fraction of the space!



Any documentation on these two features, and which player(s) support the second?

[zombi55]

I would like to know if there is a way to see the transcoding progress (the percentage or something as that) ?

[Rockaria]

@zombi55, if you integrate the encoder to foobar2k, as are quoted many times even in this thread, you will have a progress bar for 1(st) pass processing.

[Rockaria]

This earlier quoted new quality mode(-q) guideline :
Bitrate->Quality
15----->0.05
32----->0.15
63----->0.25
99----->0.35
146---->0.45
197---->0.55
248---->0.65
299---->0.75
350---->0.85
401---->0.95
is not representing the Pure Quality Grade which even a novice knows by instinct and won't expect it any longer.

The pure(100%) even quality mode should be unpredictable of the resultng bit rate(ABR == total bits/nSamples) or size at all, but can only be expressed as a 'ratio == target quality bitrate / perfect quality bitrate' when the perfect quality bit rate totally depends on the complexity of the stream.
For an example, over 90% of the fidelity of the voice clip can be attained with far less bit rate than that from the orchestra.
So my conclusion about the -q mode in this encoder is that it is a 'compromise' between the pure quality based VBR mode and pure ABR mode or in a word, bitrate based VBR mode.
I guess it's implemented as an extention of the pure ABR mode by extending the one second time frame(where we can apply the pure quality based VBR mode inside for the bit efficiency) to some bigger period(and we know this option/-period is touchable only by the coder).

If we extend the time frame, as an extreme, to the entire length of the clip, I guess that's what the encoder call -2pass -br.
And we also know the 2 pass mode is inevitable when the frame length is bigger than the buffer available for the smooth OS operation.
This is sort of a VBR mode deciding the target entire bit rate(ABR) first(bitrated based), then distributing the bits effectively for even quality inside.

So whatever reversed dialects we use, if the scenarios(policies) are similar to above, I guess the bit efficiency would look like : '-cbr<-br<(-q<)-br-longer frame<-br-2pass' for the similar bitrate. And I have no idea what the -2passperiod (http://www.hydrogenaudio.org/forums/index.php?showtopic=44275&st=225#entry390430) is used for.

[edit] related links
http://www.hydrogenaudio.org/forums/index.php?showtopic=44275&st=25#entry389216
http://forum.doom9.org/showpost.php?p=824210&postcount=114
http://www.hydrogenaudio.org/forums/index.php?showtopic=44310
http://www.hydrogenaudio.org/forums/index.php?showtopic=44275&st=275#entry390684
http://forum.doom9.org/showthread.php?p=825314#post825314
http://forum.doom9.org/showpost.php?p=825612&postcount=193
http://www.hydrogenaudio.org/forums/index.php?showtopic=44275&st=300#entry391037

http://forum.doom9.org/showpost.php?p=823499&postcount=19
http://www.hydrogenaudio.org/forums/index.php?showtopic=44275&st=50#entry389236
http://www.hydrogenaudio.org/forums/index.php?showtopic=44275&st=75#entry389337
http://www.hydrogenaudio.org/forums/index.php?showtopic=44275&st=175#entry389869
http://www.hydrogenaudio.org/forums/index.php?showtopic=44275&st=225#entry390426

http://forum.doom9.org/showpost.php?p=823648&postcount=48
http://forum.doom9.org/showpost.php?p=823923&postcount=99
http://forum.doom9.org/showpost.php?p=823989&postcount=104
http://forum.doom9.org/showpost.php?p=824236&postcount=118

http://www.hydrogenaudio.org/forums/index.php?showtopic=44275&st=100#entry389380

[menno]

Those bitrates were just given based on some tests over a set of files, to give an idea what kind of bitrate you can expect, on average.
The bitrate you will get on 1 file you encode depends only on the complexity of that 1 file. It IS pure quality based VBR.

[Ivan Dimkovic]

I am quite surprised by the ammount of wild speculations people could make, especially with regards to VBR :)

-q is a pure quality VBR mode - whoever doubts that, try compressing few tracks of different music nature with the same -q parameter, and you would notice that the bit-rate is quite different.

As Menno said - "average" bit rate per -q parameter is just a rough guideliness - measured on very large set of music, and the same guideliness exist for other quality-based VBR encoders, such as Ogg Vorbis and MPC.

Individual bit-rate could vary quite a lot, and in -q mode depends solely on psymodel

[FFWD]

Ivan, would you recommend 2-pass ABR or 1-pass VBR for feature length movie content? I am asking this because movies can contain large parts of speech and silence.

[Ivan Dimkovic]

I think this needs lots of testing - I would say that if movie contains lot of different passages, 2-pass would be definitely better than CBR/ABR.

Regarding the quality VBR vs. 2-pass - this will need some testing.

[zombi55]

@zombi55, if you integrate the encoder to foobar2k, as are quoted many times even in this thread, you will have a progress bar for 1(st) pass processing.

I don't use foobar2k. Anyway, it isn't well thinking from nero.

[shon3i]

Regarding the quality VBR vs. 2-pass - this will need some testing.Yes but we need test some real stuff like sound for movie not some short samples.

[Rockaria]

Glad Ivan is not using the ambiguous AVERAGES for the -2pass mode any more in this forum.:)

The reason for the 'speculations', 'FUD' or 'surprise' is quite obvious,
Although some knows(but obviously FUD in programming) want the majority to follow them without any doubt, blindedly, we cannot stop thinking and evaluating to find the best way to meet the individual needs with the tools where belongs the neroaacencoder.exe just as a candidate.

So the -q mode in this new encoder is same as old versions, then what is your feeling about the other stupid user speculations or Doubt on other options such as -br -2pass -2passperiod...

I couldn't find anywhere explaining them reasonably...

[Ivan Dimkovic]

So the -q mode in this new encoder is same as old versions, then what is your feeling about the other stupid user speculations or Doubt on other options such as -br -2pass -2passperiod...

I don't get :)

Quality-based VBR mode (-q) is of course not the same as in old AAC encoder, as now people have full freedom to specify any quality within the acceptable range, and not just few predefined presets (tape, radio, etc...) like in the old encoder.

Regarding -2passperiod - what is so confiusing there? -2passperiod tells encoder to average bit-rate in the user-specified time period, when the encoder runs in the 2-pass encoding mode

This option does not have any impact on single-pass quality-based VBR, as you can probably figure out by trying to enable it and observing the results.

Single-pass quality VBR (-q X) just allocates bits according to the psymodel - there is no "averaging" whatsoever - what was "averaged" in the HA guidelines is bit-rate for the large set of music material, just as the guideline how much you can expect a Q number to take bits, ON average (and certainly not per specific track).

Double-pass VBR (-br X -2pass) will average the bit-rate according to the user demand, specified by the -br switch

Double-pass VBR + -2passperiod (-br X -2pass -2passperiod Y) will do the same as above, but it will average the bit-rate in the specific period of time, that user can select

So encoding modes are:

1. CBR (or constant bit rate): -cbr XXXXXX bits/second
2. ABR (or average bit rate): -br XXXXXXX bits/second
3. Quality Based VBR (single-pass): -q X.XX (where X is the quality)
4. 2-Pass VBR which averages to user-specified bitrate: -br XXXXXX -2pass
5. 2-pass VBR which averages to user-specifiec bitrate over custom interval: -br XXXXXX -2pass -2passperiod YYYYYYY

As you can see - there are total 5 encoding modes of the new encoder - any other combination of the switches makes absolutely no sense - and if it works (e.g. -q and -2pass) it is a result of unpredictable behavior and will be disabled in the next release.

[Rockaria]

Surely you don't get.;)
The 'same' I meant for the -q is the basic mechanism not the more granular trivials.

.ABR (or average bit rate): -br XXXXXXX bits/second : it's a good progress, very clear.
.-2passperiod YYYYYYY : do you expect any benifit on this? and what's your guidelines?

Also don't you expect any benefit of the -period adjustment on the 1 pass -br? I have never been interested in -q -2pass, to be clear.

[Ivan Dimkovic]

As I said:


Regarding -2passperiod - what is so confiusing there? -2passperiod tells encoder to average bit-rate in the user-specified time period, when the encoder runs in the 2-pass encoding mode

So, -2passperiod has obviously no impact on any single-pass encoding mode.

Modes that work in the single encoding pass:

-cbr, -br, -q

Modes that work in the two-passes (1: analysis, 2: coding):

-br -2pass and -br -2pass -2passperiod

[Rockaria]

Hmm, thanks anyway. More user feedbacks will tell...
As I said earlier, I appreciated your contributions already.

[fakey]

what is so confiusing there?

...

So encoding modes are:

1. CBR (or constant bit rate): -cbr XXXXXX bits/second
2. ABR (or average bit rate): -br XXXXXXX bits/second
3. Quality Based VBR (single-pass): -q X.XX (where X is the quality)
4. 2-Pass VBR which averages to user-specified bitrate: -br XXXXXX -2pass
5. 2-pass VBR which averages to user-specifiec bitrate over custom interval: -br XXXXXX -2pass -2passperiod YYYYYYY

As you can see - there are total 5 encoding modes of the new encoder - any other combination of the switches makes absolutely no sense - and if it works (e.g. -q and -2pass) it is a result of unpredictable behavior and will be disabled in the next release.

I think that's the part which caused this confusion.

Let me try to clear things up in more easy-to-understand terms,

[-cbr] Constant BitRate mode with bit reservoir

[-br] ABR mode

[-q] quality based VBR mode
(something behaves like --crf mode of x264)

[-2pass -br] Bitrate based VBR mode
(VBR mode with target bitrate, just as 2pass mode in DivX, XviD or x264)

[-2passperiod] ABR mode with user specific averaging period

In this way, "-2pass -q mode doesn't make sense" statement makes sense, which is meant to do quality based 2pass encodings.

Is that right?

[Ivan Dimkovic]

Yes, correct and thanks for the clarifications :)

When you set encoder to -q mode (quality based VBR) you basically force the encoder just to look at psymodel and nothing else.

So, -q mode in conjuction with 2-pass does not make too much sense.

[obieobieobie]

Thank you, Ivan and Nero for this encoder. I just encoded a song for my mobile from a 4 MB MP3 to a 2.5 MB MP4.

Yeah, yeah, I know.. Lossy-lossy bad, but this is a mobile phone and it still sounded good. :)

[buzzqw]

@Ivan Dimkovic

i need (:thanks:) for a gui the max and min bitrate for cbr/br in combination with with lc/he/hev2 and for 2ch and 5ch.

THANKS !

BHH

[buzzqw]

@Inc

Will be possible to correct a52dec.exe or azid.exe to give a correct 5.1 stdout output ?

since lame/oggenc/neroaacenc support stdin this could be a quite fast way of encoding.

BHH

[Naito]

As the first I apologize for my bad English.

I made a Performance-Test aacenc32.dll 3.2.0.30 vs. CLI-Encoder 01.05.2006 (http://forum.gleitz.info/showpost.php?p=268361&postcount=34) in the Doom9 | Gleitz-Forum (Sorry, only into German available).

My English is too bad, but I already ask someone to the translating the results into English (http://forum.doom9.org/showthread.php?p=826682#post826682). Up to then I hope that the results also understandable. Please to pay attention to the diagramms (completely diagramms available in the attachment).

Particularly interesting is the q-Setting, that lies between the q-Setting with HE force and the q-Setting with LC force. Maybe, the Setting did somewhat to merge HE and LC (I like to call it "Hybrid").

With misunderstandings, ask please. I hope to answer then.

EDIT: False statement about the Hybrid! See below.

[Ivan Dimkovic]

I am sorry, but there is no "Hybrid HE+LC" mode - encoder either does LC, or it does HE - and there is nothing in between :)

[Skuto]

Particularly interesting is the q-Setting, that lies between the q-Setting with HE force and the q-Setting with LC force. Maybe, the Setting did somewhat to merge HE and LC (I like to call it "Hybrid").


You seem to seriously misunderstand something.

HE-AAC is an extension on top of LC-AAC. Every HE-AAC is hence also an LC-AAC encode. Every LC-AAC decoder (including the ones who have never heard of HE-AAC) can play back HE-AAC for this reason, although they will be missing the high frequency part if they don't understand the HE-AAC extension (if they fail, they are by definition broken and not AAC compatible).

There is no such thing as a "hybrid" mode. It's either LC-AAC, or LC-AAC+SBR (called HE-AAC).


Wenn die Datei als HE-AAC erkannt wird, müssten ständig die SBR-Frequenzen reproduziert werden, obwohl die im LC-Abschnitt schon vorhanden sind.


This is incorrect. The LC mode will never contain any frequencies which are reproduced by SBR. That would be entirely redundant and hence useless.


Ich hoffe, das Nero den Decoder als OpenSource freigibt.


Like this http://www.audiocoding.com/ ?

[Naito]

@ Ivan Dimkovic
Can you explain please how the reseluts "VBR (free)" in the diagram comes? I don't understand it.

[Ivan Dimkovic]

As I don't really undestand German that well - I don't really know what is questionable here.

In the current encoder (and that might change):

For q modes < 0.31 - we use HE-AAC
For q modes >= 0.31 - we use LC-AAC

There is nothing in between really :)

[Dark-Cracker]

is there a way to made it work with an avs input ? (pipe or somethink ?)

++

[Naito]

@ Ivan Dimkovic

Ok, I will correct my statements.

EDIT: Q: If the Encoder switch q > 0.31 from HE-AAC to LC-AAC, the "VBR (free)" and "LC (force)" must be identical. But it isn't. (Not on the same line, it It approaches on only.) Why?

[Skuto]

-q scale is different if you force the encoder to some mode.

If you look at your graph, it has to be, or there would be no way to encode at 200-400kbps....

[Naito]

So -q use encodertweaks, they are not used with force LC (-q -lc) or force HE (-q -he)? That could explain something.

[LoRd_MuldeR]

My first attempt to make a simple GUI for neroAacEnc.exe is here:
http://mulder.dummwiedeutsch.de/pub/downloads/testing/neroAacEncGUI.2006-05-10.7z

:)


http://img489.imageshack.us/img489/3484/neroaaacenc5xj.png

[tebasuna51]

is there a way to made it work with an avs input ? (pipe or somethink ?)
Maybe:

bepipe --script "Import(^Audio.avs^)" | neroAacEnc.exe -cbr 224000 -if - -of "output.m4a"

EDIT: Work also with 5.1 48 KHz 130 min. (wav equivalent > 4 GB), using -ignorelength switch.

[[)370|\|470!2]

Getting overbitrate when encoding 5.1 wav -cbr 448000
i get approx 490 kbps avg in resulting file.

[Rockaria]

I think that's the part which caused this confusion.

Let me try to clear things up in more easy-to-understand terms,

[-cbr] Constant BitRate mode with bit reservoir

[-br] ABR mode

[-q] quality based VBR mode
(something behaves like --crf mode of x264)

[-2pass -br] Bitrate based VBR mode
(VBR mode with target bitrate, just as 2pass mode in DivX, XviD or x264)

[-2passperiod] ABR mode with user specific averaging period

In this way, "-2pass -q mode doesn't make sense" statement makes sense, which is meant to do quality based 2pass encodings.

Is that right?
Finally, my familiar words...:goodpost:

I agree in general but with just some reminds :

. in order to be called as ABR the interval should be 1 sec, equal amount of allocated bits to distribute inside of each unit, by current definition, otherwise, I'd call it as VBR like CBR when it is technically ABR.

. when the -2passperiod is possible, so should be -1passperiod logically for the -br, which is just adding/adjusting some more intervals on the units possibly to improve the distribution quality.

. the averaging(distributing) with the target bitrate actually happens on all the modes except the -q mode theoretically, which of course, does not fully qualify the mode to be ABR.

[SeeMoreDigital]

My first attempt to make a simple GUI for neroAacEnc.exe

http://img489.imageshack.us/img489/3484/neroaaacenc5xj.pngHey... It's not bad :)

How about starting a separate thread for it?


Cheers

[LoRd_MuldeR]

imcold already startet a thread for his tool.
Do we really need another one for mine?

// EDIT

Update:
http://mulder.dummwiedeutsch.de/pub/downloads/testing/neroAacEncGUI.2006-05-10.7z

[fakey]

. in order to be called as ABR the interval should be 1 sec, equal amount of allocated bits to distribute inside of each unit, by current definition, otherwise, I'd call it as VBR like CBR when it is technically ABR.

. when the -2passperiod is possible, so should be -1passperiod logically for the -br, which is just adding/adjusting some more intervals on the units possibly to improve the distribution quality.

. the averaging(distributing) with the target bitrate actually happens on all the modes except the -q mode theoretically, which of course, does not fully qualify the mode to be ABR.



Yeah, that's why I thought we would need some clarification in familiar words.

Here or at HA forum, nero guys seems like saying 2pass bitrate-based VBR as ABR*.

Actually Garf is referring to DivX/XviD 2pass thingie also as ABR.
http://www.hydrogenaudio.org/forums/index.php?showtopic=44275&st=50&p=389236&#entry389236

This can be right in a way that average bitrate converges around some target bitrate, but does not agree to our convention or definition of what is VBR/ABR.

I'm not willing to claim they are totally wrong, but quite confusing at least.:confused:

I think this is just a small mistake in selecting the word.

OTOH I agree with you in technical definition.

But in AAC, even CBR has a bit reservoir bigger than MP3's thus quite often said to be a kind of ABR (in conventional point of view).

So, maybe it's time we need to expand or revise the definition of ABR (averaging) interval (1 second).

The encoder has already functions there.

And it's compliant to the MPEG Audio specification.

We can adjust the interval to be other than 1 sec and I think what we need is to understand and make decent use of the feature.

Anyway ABR was just compromise between VBR and CBR.

Things are that we can have more flexibility in AAC ABR(or neroAAC ABR).


In conclusion, VBR can distribute bits across the whole length, and ABR needs to average at some points.


*1pass ABR is no problem here.

[deets]

hmm, any wav i extract from my DVD (converted 5.1 ac3) say they cannot be parsed :(

wav extracted from audio cd works just fine

edit: seem to work if i use belight to convert the ac3 to wav and not virtualdub :)

[Rockaria]

I'm not willing to claim they are totally wrong, but quite confusing at least.
So, maybe it's time we need to expand or revise the definition of ABR (averaging) interval (1 second).
*1pass ABR is no problem here.I agree.:cool:

But the current modes are measured by bps. So as far as the Second(possibly streaming?) is engaged, unfortunately we are not gonna be free from this dialects confusions.

Anyway, I have no problem wearing a mapping table(interpreter). But watching themselves using different norms for the same thing had been really confusing (for the INTERESTS in the tiny/big tool)...

The trend I am reading is that the guys are trying to make it INTELLIGENT : Black Boxed : you choose the tool and we will move your focus entirely out of it, no worry...no AAC-HE-VBR but just ND.

I have been a person who failed to adapt myself to the adaptive programming..
Which is so sur/sub-realistic that awakening myself through the understanding the whole mechanism based on the five generic factors could have been far easier.

I don't believe the intelligence of codes too much as it will save us time significantly or free us from the worry at all.
Maybe it can help us, if it ever happens, eventually become totally neutralized : disarmed for being equipped, no toys.

My belief is : I plan, orgarnize and play with all the flexibility, the tools(components) do the simple things reliably, because it's simple, no big brother.

[Naito]

lousy translation made by Redfox (no, I don't have an english spell-cecker installed on my machine)
Last weekend I (Naito) showed my AMD-Duron 800 how it feels to be a Calculator. Alltogetter, it had to encode 32h and 15min Audio into AAC. After Neros release of the AAC-CLI-Encoder I decided to make some exausive testing. The following Spreedsheds containing the results.

1. Testsystem

AMD Duron @ 832MHz with 512MB SD-RAM
Windows XP Professional incl. SP 2 (no Patches)
Nero aacenc32.dll 3.2.0.30 (from Nero 6.6.1.4)
Nero AAC-CLI-Encoder 01.05.2006
BeSweet 1.51 Beta31
foobar2000 0.9.1 with AC3-Plugin 0.7 (foo_ac3.dll)


2. Test Conditions

I used BeSweet to transcode with aaenc32.dll.
The Commandline locked like that:
"BeSweet.exe -core( -input AC3.ac3 -output Art-Profil.mp4 -logfile Art-Profil.log ) -bsn( -6chnew -vbr_Profil -codecquality_high -aacprofile_Art)"
'Art' means LC or HE. Profil reachs from Tape till Transcoding.
For transcoding of VBR with the AAC-CLI-Encoder I useed foobar2000.
- The commandline for 'forsed' LC or HE locked like this:
HE (force): "-ignorelength -he -q 0.xx -if - -of %d"
LC (force): "-ignorelength -lc -q 0.xx -if - -of %d"
- The commandline for 'free' VBR locked like this:
VBR (free): "-ignorelength -q 0.xx -if - -of %d"
Replace 'xx' with the certain values. The "-" tells foobar2000 to use a pipe.
For the ABR-Transcoding I converted AC3 with BeSweet into DD-WAV and feeded the AAC-CLI-Encoder with a Batch-File.
For 5.1@2.0 I used the DSP in foobar2000 activated and the same Commandlineparameter like for 5.1
The sourcefile was the german soundtrack of „Ghost in the Shell ~ Stand Alone Complex – Episode 01“.
bitrate = 448
codec = ATSC A/52
channels = 6
samplerate = 48000
----------
72388608 samples @ 48000Hz = 25:08.096 (Length)
File size: 84 453 376 bytes = 80,5 MB

3.1 Results: 5.1-Test (the interesting stuff is at the bottom)

| aacenc32.dll 3.2.0.30 | CLI (free) : CLI (force) | CLI |
| Profil HE LC LC/HE | VBR Hybrid : VBR HE LC LC/HE | ABR 2pass Abw. |
----------------|---------------------------------------|-----------------------:-------------------------------|-------------------------------|
kbps | Tape 109 155 1,42 | 0.10 not : 0.10 81 165 2,04 | 100 104 1,04% |
% | 24,3% 34,6% | possib. : 18,1% 36,8% | HE 23,2% |
Size (MB) | 19,74 28,23 | : 14,68 29,98 | 18,80 |
Speed | 0,85 0,84 | : 1,41 0,88 | 0,82 |
----------------|---------------------------------------|-----------------------:-------------------------------|-------------------------------|
kbps | Radio 129 203 1,57 | 0.15 101 : 0.15 117 224 1,91 | 125 126 1,01% |
% | 28,8% 45,3% | HE 22,5% : 26,1% 50,0% | HE 28,1% |
Size (MB) | 23,33 36,80 | 18,33 : 21,16 40,52 | 22,82 |
Speed | 0,85 0,80 | 1,13 : 1,23 0,87 | 0,81 |
----------------|---------------------------------------|-----------------------:-------------------------------|-------------------------------|
kbps | Internet 140 307 2,19 | 0.20 145 : 0.20 153 286 1,87 | 150 150 1,00% |
% | 31,3% 68,5% | HE 32,4% : 34,2% 63,8% | HE 33,5% |
Size (MB) | 25,34 55,44 | 26,19 : 27,69 51,78 | 27,10 |
Speed | 0,85 0,72 | 1,18 : 1,19 0,82 | 0,74 |
----------------|---------------------------------------|-----------------------:-------------------------------|-------------------------------|
kbps | Streaming 172 381 2,22 | 0.25 190 : 0.25 191 347 1,82 | 175 176 1,01% |
% | 38,4% 85,0% | HE 42,4% : 42,6% 77,5% | HE 39,3% |
Size (MB) | 31,10 68,75 | 34,25 : 34,52 62,57 | 31,82 |
Speed | 0,81 0,68 | 1,25 : 1,16 0,82 | 0,73 |
----------------|---------------------------------------|-----------------------:-------------------------------|-------------------------------|
kbps | Normal 190 485 2,55 | 0.30 234 : 0.30 228 405 1,78 | 200 198 0,99% |
% | 42,4% 108,3% | HE 52,2% : 50,9% 90,4% | HE 44,2% |
Size (MB) | 34,31 87,51 | 42,17 : 41,10 73,17 | 35,78 |
Speed | 0,79 0,58 | 1,19 : 1,16 0,84 | 0,73 |
----------------|---------------------------------------|-----------------------:-------------------------------|-------------------------------|
kbps | Extreme 273 625 2,29 | 0.31 263 : 0.31 235 417 1,77 | 225 219 0,97% |
% | 60,9% 139,5% | LC 58,7% : 52,5% 93,1% | HE 48,9% |
Size (MB) | 49,26 112,61 | 47,53 : 42,43 75,27 | 0,32 |
Speed | 0,76 0,44 | 0,73 : 1,12 0,81 | |
----------------|---------------------------------------|-----------------------:-------------------------------|-------------------------------|
kbps | Audiophile 308 646 2,10 | 0.35 320 : 0.35 264 470 1,78 | 250 242 0,97% |
% | 68,8% 144,2% | LC 71,4% : 58,9% 104,9% | HE 54,0% |
Size (MB) | 55,58 116,36 | 57,82 : 47,64 84,70 | 43,71 |
Speed | 0,79 0,46 | 0,82 : 1,32 0,83 | 0,74 |
----------------|---------------------------------------|-----------------------:-------------------------------|-------------------------------|
kbps | Transcoding 339 744 2,19 | 0.40 390 : 0.40 299 528 1,77 | 275 274 1,00% |
% | 75,7% 166,1% | LC 87,1% : 66,7% 117,9% | LC 61,2% |
Size (MB) | 61,00 134,09 | 70,42 : 53,92 95,19 | 49,61 |
Speed | 0,77 0,55 | 0,79 : 1,23 0,84 | 0,50 |
----------------|---------------------------------------|-----------------------:-------------------------------|-------------------------------|
kbps | | 0.45 465 : 0.45 334 586 1,75 | 300 293 0,98% |
% | | LC 103,8% : 74,6% 130,8% | LC 65,4% |
Size (MB) | | 83,93 : 60,21 105,57 | 52,91 |
Speed | | 0,81 : 1,26 0,85 | 0,51 |
----------------|---------------------------------------|-----------------------:-------------------------------|-------------------------------|
kbps | | 0.49 521 : 0.49 362 633 1,75 | 325 315 0,97% |
% | | LC 116,3% : 80,8% 141,3% | LC 70,3% |
Size (MB) | | 93,92 : 65,26 114,08 | 56,83 |
Speed | | 0,84 : 1,28 0,81 | 0,51 |
----------------|---------------------------------------|-----------------------:-------------------------------|-------------------------------|
kbps | | 0.50 535 : 0.50 not 644 | 350 337 0,96% |
% | | LC 119,4% : possib. 143,8% | LC 75,2% |
Size (MB) | | 96,39 : 116,14 | 0,31 |
Speed | | 0,84 : 0,84 | |
----------------|---------------------------------------|-------------------------------------------------------|-------------------------------|
kbps | ; . | 375 348 0,93% |
% | ; . | LC 77,7% |
Size (MB) | ; . | 0,31 |
Speed | ; . | |
----------------|-----------------------------------------------------------------------------------------------|-------------------------------|
kbps | ; . | 400 367 0,92% |
% | ; . | LC 81,9% |
Size (MB) | ; . | 0,31 |
Speed | ; . | |
----------------|-----------------------------------------------------------------------------------------------|-------------------------------|
kbps | ; . | 425 389 0,92% |
% | ; . | LC 86,8% |
Size (MB) | ; . | 0,31 |
Speed | ; . | |
----------------|-----------------------------------------------------------------------------------------------|-------------------------------|
kbps | ; . | 450 410 0,91% |
% | ; . | LC 91,5% |
Size (MB) | ; . | 0,31 |
Speed | ; . | |
----------------|-----------------------------------------------------------------------------------------------|-------------------------------|
. . . .
. . . .
aprox. Speed/ kbps 0,81 0,63 2,07 . 0,96 . 1,24 0,84 1,85 . 0,59 .
standartdevation 0,04 0,15 0,38 . 0,20 . 0,09 0,02 0,10 . 0,20 .
aprox. overhead 0,19% 0,35% . 0,26% . 0,20% 0,35% . 0,28% 0,97% .




The Deviation of the Bitrate is around 2%.
0.00 <= 0.15 is HEv2; 0.16 <= 0.30 is HE; 0.31 <= 1.00 is LC
ABR: 100kbps = 100.000bps

Note that the standartdevation is the root from the vary, but Excel calculates them wrong. So better be carefull with those values. I didn't corected those results because they don't matter statisticly. The are just for your orientation. The avertage Overhad is calculated by the percentage part of the filesize minus the percentage of the bitrate(You can find the values in an attachment at the original posting)

3.2 Results: 5.1@2.0

| CLI (free) | CLI (force) |
| VBR Hybrid 5.1/2.0 | VBR HE 5.1/2.0 : LC 5.1/2.0 ' LC/HE |
----------------|-----------------------|-------------------------------------------------------|
kbps | 0.10 24 not | 0.10 22 3,68 : 40 4,13 ' 1,82 |
Size (MB) | 4,50 possible| 4,03 3,64 : 7,46 4,02 ' |
Speed | 4,51 | 3,41 : 2,54 ' |
----------------|-----------------------|-------------------------------------------------------|
kbps | 0.20 42 3,45 | 0.20 44 3,48 : 79 3,62 ' 1,80 |
Size (MB) | 7,64 3,43 | 8,00 3,46 : 14,40 3,60 ' |
Speed | 2,86 | 2,87 : 2,11 ' |
----------------|-----------------------|-------------------------------------------------------|
kbps | 0.30 69 3,39 | 0.30 68 3,35 : 113 3,58 ' 1,66 |
Size (MB) | 12,60 3,77 | 12,33 3,33 : 20,61 3,55 ' |
Speed | 2,80 | 2,89 : 2,12 ' |
----------------|-----------------------|-------------------------------------------------------|
kbps | 0.40 109 3,58 | 0.40 86 3,48 : 140 3,77 ' 1,63 |
Size (MB) | 19,87 3,54 | 15,59 3,46 : 25,41 3,75 ' |
Speed | 2,16 | 2,97 : 2,12 ' |
----------------|-----------------------|-------------------------------------------------------|
kbps | 0.50 141 3,79 | 0.50 102 not : 186 3,46 ' 1,82 |
Size (MB) | 25,71 3,75 | 18,51 possible : 33,69 3,45 ' |
Speed | 2,07 | 2,83 : 2,10 ' |
----------------|-----------------------|-------------------------------------------------------|
. . ' .
. . ' .
average (kbps) 3,55 . 3,50 . 3,71 ' 1,75 .
standartdeviation (kbps) 0,18 . 0,14 . 0,26 ' 0,09 .
average (MB) 3,62 . 3,47 . 3,67 ' .
standartdeviation (MB) 0,17 . 0,13 . 0,22 ' .

4.Appraisal of Results

The optinization is most noticeable with LC at multichannel. While 3.2.0.0.30 LC is aprox. 2,1-times bigger then HE, the improvement with the CLI-Encoder is stronger noticeable (aprox. 1,85).
It is interesting that an 5.1 encode is aprox. 3,5 times bigger then an similer 2.0. This might indicate an insufficient channelcuppling.
HE is only posible till „-q 0.49“, with values above Nero displays an error and refuses to encode.
The 'free'-VBR-Modus (-q 0.xx without -lc or -he switch)is quite interesting. The bitrates here are very differend then with forced switchs (internal encoder-optimasions around the psychacusic-model, please note this! (http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=44275&view=findpost&p=391037)) We have to use liesening-tests to figure out what modus offers the best Audio quality) But you shuldnt force the switchs without an good reason.


http://img183.imageshack.us/img183/1640/neroaaccliencoder01052006gerad.th.png
http://img169.imageshack.us/img169/3734/neroaacenc32dll320303mm.th.png

Bysides, if somebody asks hiself „Why is he doing this?“. Plain simple, its fun for me.

The attachment (in german) contains screenshots, grafics, spreadsheets, batch-files and logs. (foobar2000 was 'handmade')
Attachment (http://forum.gleitz.info/attachment.php?attachmentid=78390&d=1147181169)

_____

Many thanks to Redfox for translation.

[elmimmo]

* Support for Embedded Album Art (Covers, Booklets, Lyrics!)
* Store Entire Audio Album in a Single .mp4 File with all the Features of an Audio CD embedded inside, but at a fraction of the space!Any documentation on these two features, and which player(s) support the second?I concur. Ivan, could you please expand on how that works?

[shon3i]

@Naito i don't know how you compare new nero cli encoder and old nero 6 encoder which is very terrible in quality especialy for HE and multichannel encoding because he have some bug which is corrected in this new CLI.

[Rockaria]

...because he have some bug which is corrected in this new CLI.
He himself admitted the violation of their own TOS#8 : proofs!


The 'free'-VBR-Modus (-q 0.xx without -lc or -he switch)is quite interesting. The bitrates here are very differend then with forced switchs (internal encoder-optimasions around the psychacusic-model, please note this!) We have to use liesening-tests to figure out what modus offers the best Audio quality) But you shuldnt force the switchs without an good reason.

The 'psycho..' seems to belong somewhere around 'arbitrary or delicate' or even particularly intelligent.

Because there is absolutely no formal dictionary yet, I agree anybody who is interested in, may have to approach in both ways to figure out the features(target/current) and pros & cons for own use or customized use for target group of users : top-down and bottom-up way.

http://en.wikipedia.org/wiki/Audio_data_compression : Lossy Compression

1st step : perceptual redundancy reduction using psychoacoustics model : Lets regards it as the perfect quality bitrate that human organ can perceive without the fidelity loss.
The innovation of lossy audio compression was to use psychoacoustics to recognize that not all data in an audio stream can be perceived by the human auditory system. Most lossy compression reduces perceptual redundancy by first identifying sounds which are considered perceptually irrelevant, that is, sounds that are very hard to hear. Typical examples include high frequencies, or sounds that occur at the same time as other louder sounds. Those sounds are coded with decreased accuracy or not coded at all.
2nd step : further lossy compression using target bit rate : Let's call it averaging or equal quality loss distributing
If reducing perceptual redundancy does not achieve sufficient compression for a particular application, it may require further lossy compression with a difference in quality that can be more readily perceived by a user. Most lossy compression schemes allow compression parameters to be adjusted to achieve a target rate of data, usually expressed as a bit rate. Again, the data reduction will be guided by some model of how important the sound is as perceived by the human ear, with the goal of efficiency and optimized quality for the target data rate. (There are many different models used for this perceptual analysis, some better suited to different types of audio than others.) Hence, depending on the bandwidth and storage requirements, the use of lossy compression may result in a perceived reduction of the audio quality that ranges from none to severe. Of course, that trade-off is usually intentional
Then, where belongs the -q mode averaging?
I guess it's in the 2nd step possibly using a psymodel again to achieve the target quality by further distributing the loss evenly.
If the bit rate is given as a input, norlmally it will be CBR, ABR or VBR mode, bit rate based.
If the target quality ratio is given, then the target bitrate for the segment will be dynamically calculated(or referenced) by the target quality ratio and actual perfect quality bitrate from the 1st step, and be called as quality based.

And as you see, the 1st step is(can be) used regardless of the modes..
But as for the -q mode, I guess the two steps can be processed in one step if both steps are particularly using the same psymodel

[shon3i]

@Ivan Dimkovic is there any chance in near future that this decoder neroAACDec support stdout, or did you have some other decoder for that

[dimzon]

@Ivan Dimkovic is there any chance in near future that this decoder neroAACDec support stdout, or did you have some other decoder for that
Actually I don't understand why just do not use FREE GPL-ed FAAD2 for it...

[shon3i]

I tryed but, i don't have lucky

comandline i use

"faad.exe" -w "silent_hill_h720p.aac" > "silent_hill_h720p.wav" | "neroAacEnc.exe" -if - -of "silent_hill_h720p.mp4"

and nero encoder say can't parase wav file

what i doing wrong, please tell me

[Rockaria]

I may have to put some more words because it has something in it.

4.Appraisal of Results
The optinization is most noticeable with LC at multichannel. While 3.2.0.0.30 LC is aprox. 2,1-times bigger then HE, the improvement with the CLI-Encoder is stronger noticeable (aprox. 1,85).

: It could simply mean the compression(averaging) models(& parameters) are changed in the new CLI version.

It is interesting that an 5.1 encode is aprox. 3,5 times bigger then an similer 2.0. This might indicate an insufficient channelcuppling.

: Normally 2.0~2.5x is expected compared to the stereo, but also it depends on the degree of the original clip channel seperations. I believe it must not exceed 3.0 even if the LFE has the full bandwidth usage.

HE is only posible till „-q 0.49“, with values above Nero displays an error and refuses to encode.

: ...

The 'free'-VBR-Modus (-q 0.xx without -lc or -he switch)is quite interesting. The bitrates here are very differend then with forced switchs (internal encoder-optimasions around the psychacusic-model, please note this!) We have to use liesening-tests to figure out what modus offers the best Audio quality) But you shuldnt force the switchs without an good reason.

: The CT version of AAC(CBR) comparison is showing that the HE quality becomes similar less than around @80kbps for the stereo and @200kbps(if we consider 80*2.5) for 6ch.
This is quite reasonable when we consider the difference is only the SBR band processing, which would be of smaller effect(difference) as the bit rate goes higher.

The two Nero graphs seem to say that the two modes are simply using different lossy compression models(& parameters), no resonable relationship to each other.
The -q auto mode(without LC/HE switch) could be said a bit intelligent, of course depending on the point of views.

I believe in the competetions for the end-users and tech itself.

http://www.codingtechnologies.com/products/assets/3GPP_mushra_18_48kbps_2.gif

[Edit] My remarks are after ':'s, no bold

[dimzon]

MeGUI since 0.2.3.2138 support NeroDigital CLI encoder ;)
http://img66.imageshack.us/img66/9431/untitledrrr4us.png
http://img66.imageshack.us/img66/1689/untitledfff7yz.png

[Rockaria]

MeGUI since 0.2.3.2138 support NeroDigital CLI encoder ;)

hmm, I see two things :rolleyes: :
. Adaptive Bitrate : for -br (-2pass)?
. CT AAC encoder : dropped?

[shon3i]

hmm, I see two things :rolleyes: :
. Adaptive Bitrate : for -br (-2pass)?
. CT AAC encoder : dropped?
Agree,

1. and is there any chance to somehow integrate replygain from foobar, or something to first scan audio file and find higesest possible gain like foobar and than apply with amplifydb in avs script, because in that way we get very loud file without clipping/distorsion.

2. Can some help me with this. I try to re-encode some aac to mp4, but i can't to make stdout(faad)-stdin(nero).

I tryed but, i don't have lucky

comandline i use


"faad.exe" -w "silent_hill_h720p.aac" > "silent_hill_h720p.wav" | "neroAacEnc.exe" -if - -of "silent_hill_h720p.mp4"


and nero encoder say can't parase wav file

what i doing wrong, please tell me

[Rockaria]

Isn't the foobar2k(v0.9~) standard input plugin good at AAC multi channel decoding? I think just adding some ND cli encoder profiles to it will solve the second issue.
------------------------------------------------------
ND 128k -2pass auto : -br 128000 -2pass -if %s -of %d -ignorelength
ND 128k auto : -br 128000 -if - -of %d -ignorelength
...
------------------------------------------------------

Using bepipe/avs2wav with an avs file processing AmplifyDb(foobar scanned replaygain + ~3.0db) helped me, manual though, but faster than avisynth 2 pass volume gain. You may have to adjust the increment(3db) depending on the # of channels and/or foobar2k versions. I tested it with v0.83 long ago.
------------------------------------------------------
bepipe --script "Import(^Audio.avs^)" | neroAacEnc.exe -cbr 224000 -if - -of "output.m4a"

[shon3i]

OK Rockaria but you little missunderstand me.

First about ReplayGain:

I founded that in foobar 0.9 have excelent ReplyGain. For example, i used 6ch ac3 input and scan. ReaplyGain found maximum value for gain for example +10.54dB, i apply gain and then encode for example to Nero(MP4), and i get very loud file without any clipping/distording.

Same thing happens when for example not process that file in foobar, than only scan with RG and than this 10.54dB apply in avs script with amplyfydB function.

My question is how we can use that RG from foobar and implement in MeGUI/BeHappy or there is similar function/program which can do that job.

Second:

I wan't use bepipe or directshowsource, because i have more than one filter for aac input. I must all it to shutdown to process that file normaly. So why then i need AAC decoder.
btw foobar 0.9 can't handle aac files, only mp4. or i miss some plugin. Thanks

[dimzon]

Actually it's possible to write AviSynth AAC source filter/plugin with ReplayGain support. I can use FAAD2 as starting point for it...
Unfortunally it request some amount of time, so, maybe, it will be implemented later...

[shon3i]

Actually it's possible to write AviSynth AAC source filter/plugin with ReplayGain support. I can use FAAD2 as starting point for it...
Unfortunally it request some amount of time, so, maybe, it will be implemented later...
In BeHappy/MeGUI we need AAC/MP4 input, possible with (FAAD)

This ReplayGain filter scaner if possible, for all input files.

[Rockaria]

I also thought that the answer couldn't be that simple when you ask..:D

That's true, the foobar2k generic Replaygain meta tag is not standard widely accepted yet(so manual setting is inevitable with other decoders). But it is great to hear that the v0.9 version is excellent at 6ch replaygain scanning, my experience with v0.83 before was that it is set somewhat lower than max gain, it's also possibly because of the wrong source channel mapping(6ch aac/mp4).

I have no problem playing the aac he cbr(aacplus : winamp bundled 'Kaje - Hey Buddy..") with old foobar2k imbeded standard input plugin, which is I think mpeg4 aac. Even if you have a raw mpeg2 aac, I guess foobar2k mp4 mux tool can wrap it to mp4? I could be wrong somewhere here.

The avisynth(bepipe) directshowsource uses system default filter merit(priority). so you must either set the merits correctly, have only a correct filter installed, or use other aac supported avisynth plugin.

[Latexxx]

In BeHappy/MeGUI we need AAC/MP4 input, possible with (FAAD)

This ReplayGain filter scaner if possible, for all input files.
Adaptive bitrate? Shouldn't that that be average bitrate.

[shon3i]

I also thought that the answer couldn't be that simple when you ask.. Of course is not, but must asked.

That's true, the foobar2k generic Replaygain meta tag is not standard widely accepted yet(so manual setting is inevitable with other decoders). But it is great to hear that the v0.9 version is excellent at 6ch replaygain scanning, my experience with v0.83 before was that it is set somewhat lower than max gain, it's also possibly because of the wrong source channel mapping(6ch aac/mp4).
You should try this foobar 0.92.2 beta 2 is amazing, support this new Nero CLI, ReplayGain is aslo amazing, newermid on source, works perfectly with 6ch,stereo,mono, everything.

I ask Ivan about nero ReplyGain and he say that, this encoder support RG, but how i don't know. Aslo see this on HA forums.

I have no problem playing the aac he cbr(aacplus : winamp bundled 'Kaje - Hey Buddy..") with old foobar2k imbeded standard input plugin, which is I think mpeg4 aac. Even if you have a raw mpeg2 aac, I guess foobar2k mp4 mux tool can wrap it to mp4? I could be wrong somewhere here.Aslo no problems with 0.83 but with 0.9 aac files not supported anymore, except only in mp4. I use YAMB to mux it.

The avisynth(bepipe) directshowsource uses system default filter merit(priority). so you must either set the merits correctly, have only a correct filter installed, or use other aac supported avisynth plugin.Yes i know, in my system to filter merit looks like this

source->CoreAAC->AC3Filter, so everytime i must off ac3filter, or in CoreAAC switch off dominix.

[Rockaria]

Of course is not, but must asked.
...
Yes i know, in my system to filter merit looks like this

source->CoreAAC->AC3Filter, so everytime i must off ac3filter, or in CoreAAC switch off dominix.Pls, don't get me wrong...;)
I meant, u never say useless(simple) thing, although it is not sometimes expressed clearly(in a sentence).

The priority(merit) setting is useful when there are more than one filters for the same functions.
in this case, the aac source(spliter) filters and decoders are related.

[shon3i]

Pls, don't get me wrong...
I meant, u never say useless(simple) thing, although it is not sometimes expressed clearly(in a sentence).sorry but my english is bad, so sometimes i can't express correctly. I am not get you wrong.

The priority(merit) setting is useful when there are more than one filters for the same functions.
in this case, the aac source(spliter) filters and decoders are related.
That is ok, but i want clear decoder->encoder.

[Rockaria]

If I userstand you correctly, the language doesn't matter much, unless you are intentionally abstracting the words to tease me...

I meant 'get me wrong' as 'misunderstand me'. Pls corrrect me if I got you worng.

[SeeMoreDigital]

Hi Dimzon,

Any possibility your NeroDigital GUI can be made available outside of MeGUI?


Cheers

[dimzon]

Hi Dimzon,

Any possibility your NeroDigital GUI can be made available outside of MeGUI?


Cheers
Any reason agains MeGUI?
You doesn't need any other stuff (like x264 etc) to encode AAC - just MeGUI itself + Nero CLI encoder.

[GodofaGap]

The 'free'-VBR-Modus (-q 0.xx without -lc or -he switch)is quite interesting. The bitrates here are very differend then with forced switchs (internal encoder-optimasions around the psychacusic-model, please note this!) We have to use liesening-tests to figure out what modus offers the best Audio quality) But you shuldnt force the switchs without an good reason.
The quality scale is different with forced switches. That's why you cannot compare it with normal mode. One is probably not better than the other, but you will need to use a different q value to achieve the same bitrate/quality.

But forcing LC isn't good for any reason but compatability.

[Rockaria]

That's not mine! My remarks are after ':'s, no BOLD

Or quoting randomly with different ids is a new trend?;)
The 'free'-VBR-Modus (-q 0.xx without -lc or -he switch)is quite interesting. The bitrates here are very differend then with forced switchs (internal encoder-optimasions around the psychacusic-model, please note this!) We have to use liesening-tests to figure out what modus offers the best Audio quality) But you shuldnt force the switchs without an good reason.

The CT version of AAC(CBR) comparison is showing that the HE quality becomes similar less than around @80kbps for the stereo and @200kbps(if we consider 80*2.5) for 6ch. This is quite reasonable when we consider the difference is only the SBR band processing, which would be of smaller effect(difference) as the bit rate goes higher.

The two Nero graphs seem to say that the two modes are simply using different lossy compression models(& parameters), no resonable relationship to each other.
The -q auto mode(without LC/HE switch) could be said a bit intelligent, of course depending on the point of views....

[Naito]

Please note, that my test is only for discussion about the performance, not for listening quality. Thanks.
I compared the CLI with the aacen32.dll to show, how good or bad the new ist tuned. (To show a difference, I have to make a comparison)

How I said, the VBR(free) ist quite interesting. But I can't say that "-q 0.40 -he", "-q 0.21 -lc" and "-q 0.34" (all around @300kbps) gives the same sound-qaulity.
Because ...
The two Nero graphs seem to say that the two modes are simply using different lossy compression models(& parameters), no resonable relationship to each other.
The -q auto mode(without LC/HE switch) could be said a bit intelligent, of course depending on the point of views....
... I think the same. But only Nero has the answer.

I hope Listening Test will follow soon to see how good the CLI is.

Up to then
Naito

[GodofaGap]

@Rockaria:

if you are quoting someone please use quote tags. Otherwise I get horribly confused. ;)

(please no bold (http://forum.doom9.org/showpost.php?p=827290&postcount=206))

[Rockaria]

@GodofaGap, a good point, but when you get into the middle of the thread, firstly check where the chunk of issue is started. Which way, you can easily avoid the terrible confusion.

Also the reason why I do not use too many quotes onto a person's post is that it may bring up the feeling of too much differences which obviously causes unfomfortable surround, I've been feeling through lots of forums engagement. Does it make sense?

[Adub]

I found a bug! I am not going to be sent to jail, am I? Hahah!

Anyway on to the bug. When encoding with Nero Free AAC in VBR mode, I tried to go below Q=.12, and it started giving be errors like ERROR: no valid SBR configuration found

I got the above error when I went to Q=.11. No such problems with Q=.12.

Just though I would let you know.
ps. I tried both of the encoders, sse2 and normal. Both crashed after the first pass. Great encoder guys, by the way! just a little problem.

Thanks!

Edit: I was using a source with these specs Channels=6, BitsPerSample=16, SampleRate=48000Hz

It might help. Sharktooth said that the quality might be too low for 6 channel encoding, but wrong error message anyway.

[DnA]

I have a question about the encoding quality: when I executed the following command:

neroaacenc_sse2 -2pass -br 128000 -if "Input.wav" -of "Output.mp4"

I got worse results (artifacts added) than without "-2pass" option

neroaacenc_sse2 -br 128000 -if "Input.wav" -of "Output.mp4".

For testers: The sample was the 20th Century Fox intro in "Fight Club" movie.

[GodofaGap]

Also the reason why I do not use too many quotes onto a person's post is that it may bring up the feeling of too much differences which obviously causes unfomfortable surround, I've been feeling through lots of forums engagement. Does it make sense?
Perhaps, although I personally do not feel this way as long as each quote is justified and does not result in too much hair-splitting on unimportant points. I agree it's often not necessary to quote the whole OP posts to pieces in order to reply in an understandable manner.

Personally, I often read this forum early in the morning and late at night so it does happen I am not fully concentrated on deciding/looking back if something is a quote or not. I DO rely on the quote tags for that, or at least some quotation marks. And I think these tags were brought to life to avoid the confusion I have had here now. In any case, I certainly do not automatically associate bold text with a quote, but as text that is emphasized, considered important by the poster.

My apologies again for the misquote and for the off-topic.

[dimzon]

I'm looking for a good and clean NeroDigitalAudio logo

this is NeroDigitalAudio logo, but quality is wrost and I need monocrome logo
http://www.nero.com/nerodigital/img/ND_Audio.jpg

this is NeroDigital logo, everything is fine but I need NeroDigitalAudio logo
http://www.nero.com/nerodigital/img/ND_Logo_Thumb.gif

[_ck_]

Please correct me if I am misunderstanding this but there is no stand-alone hardware in existance or under development right now that will support this standard yet? Right?

Is there an ACM and DirectShow codec available at this time?
Wikipedia doesn't seem to have anything (yet) about the audio side of Nero's codecs so I am fairly lost without a FAQ...

[futurex]

I'm looking for a good and clean NeroDigitalAudio logo

this is NeroDigitalAudio logo, but quality is wrost and I need monocrome logo
http://www.nero.com/nerodigital/img/ND_Audio.jpg

this is NeroDigital logo, everything is fine but I need NeroDigitalAudio logo
http://www.nero.com/nerodigital/img/ND_Logo_Thumb.gif


incase nobody is able to find one, here's one i just slapped together :)

http://img64.imageshack.us/img64/7382/nerologonew4du.png

[shon3i]

Is there an ACM and DirectShow codec available at this time?I think that ACM codec newer been maked, because ACM for AAC is same as WFV for H264. DirectShow can be only decoder and you can use ffdshow,CoreAAC.

[Rockaria]

... I think the same. But only Nero has the answer. ..and don't want it dosclosed for some reason..perhaps yet...

@GodofaGap,:D
OK, I knew you didn't have any bad will judging by the content. So perhaps I just wanted it closed by biginning with my apology causing your mistake anyhow(as you know). But my final adjustment was in the beliefs that what is most important in the conversation is listening(reading) and correcting(own, ASAP) before any preference of the styles if we really want to continue the conversation smooth, reasonable and useful.

I have also seen so many times the quotes being abused, especially in debates(no discussion or conversation), with their mighty scissors, creating totally different content..

[Kurtnoise]

I got the above error when I went to Q=.11. No such problems with Q=.12.
It's not really a bug but more a warning I think...and I'm pretty sure that you don't perceive any differences between 0.11 and 0.12 for multichannel streams.

[Kurtnoise]

I think that ACM codec newer been maked, because ACM for AAC is same as WFV for H264.
huh, no...:rolleyes:

DirectShow can be only decoder.
Wrong...Fortunately we can have some DShow encoders filters. :rolleyes:

[Kurtnoise]

Is there an ACM and DirectShow codec available at this time?...
For DirectShow codec, install Nero Recode but it uses the old AAC encoder engine.

[AVmaniac]

Any idea how the 2pass mode of the nero cli works?
Because I have really high bitrates in zones of no noise in my encoded material.
for example:
The intro of Starshiptroopers(blank screen not a sound until the drums appear), ther ffdshow shows bitrates of 200kbps and above. really weird

[SeeMoreDigital]

I'm looking for a good and clean NeroDigitalAudio logo

Okay... here's my 800x400 pixel effort: -

http://img377.imageshack.us/img377/1784/digitalaudiologo4cl.th.png (http://img377.imageshack.us/my.php?image=digitalaudiologo4cl.png)


Cheers

[GodofaGap]

I don't want to be rude, but shouldn't Nero take care of its own logo's?

[Ivan Dimkovic]

Nero Digital logo exists - Nero Digital Audio is coming soon.

Now, I must remind concerned people that the usage of Nero Digital / Nero Digital Audio logos in third-party products is not allowed without certification.

[siddharthagandhi]

Yes, but he can use the logos that SeeMoreDigital and the other guy created...right?....

those weren't created by Nero so they should be legal...

[Ivan Dimkovic]

Hate to be the nitpick :) But they do incorporate offical Nero logo in their design - which might violate copyright.

[Kostarum Rex Persia]

Ivan, when we will see updated version of Nero Burning Rom package, and Recode 2.

Excuse me, but programs that I use, do not use official Nero logo in their design.

[Ivan Dimkovic]

Ivan, when we will see updated version of Nero Burning Rom package, and Recode 2.

NBR and Recode are updated in regular scheduled Nero updates.


Excuse me, but programs that I use, do not use official Nero logo in their design.

I was refering to an idea posted here to design custom Nero Digital Audio logos, that incorporate official Nero branding, and nothing else.

[dimzon]

Now, I must remind concerned people that the usage of Nero Digital / Nero Digital Audio logos in third-party products is not allowed without certification.
Does it mean I can't use ND logo in GUI for your CLI encoder? I'm using your own (written by yourselfes) encoder, so why not?

[GodofaGap]

Because you will have to be granted permission to use it first. Otherwise you would be in violation of copyright law.

I'm sure you understand that Nero wants to do some quality assurance on products/software that use their certification logos.

[stax76]

Next time don't ask and nobody will complain...

BeLight uses the logo as well...

[DnA]

@Nero engineers

Do you have any idea, why -2pass -br sounds worse than just -br? I really like the 2pass encoding and do not have any slight idea, why it sounds worse than one pass encoding.

[Ivan Dimkovic]

Does it mean I can't use ND logo in GUI for your CLI encoder? I'm using your own (written by yourselfes) encoder, so why not?

Because having a Nero Digital logo on the device/product means that the device is:

- Certified under strict requirements and QA verification
- Nero logo really stands for the quality of audio of this product

And all above means - for the entire product.

Free AAC encoder we are talking about here, is free for personal and noncommercial use - and this is the license condition one has to accept in order to use this free product.

So this license does not include bundling/redistribution with third party applications, also it does not include right to use Nero copyrighted intellectual property - such as logos.

Also, in addition to this - even if we wanted to give free right to bundle Nero AAC encoder with 3-party applications, or give right for commercial use, this would be in violation of the AAC patent license agreement "annual cap" rule, and we would certainly not be in position to provide encoder free of charge anymore.

So, I think the solution as it is now is the best for the users and community - encoding software is free of charge, and available by download from Nero - for non-commercial and personal use.

Everything else (distribution, bundling, etc...), including the logo usage is not granted with this free license - people willing to gain such rights should contact Nero directly.

[Ivan Dimkovic]

@Nero engineers

Do you have any idea, why -2pass -br sounds worse than just -br? I really like the 2pass encoding and do not have any slight idea, why it sounds worse than one pass encoding.

It could theoretically happen as ABR (-br) could allocate more bits than limited by 2-pass encoding. Please send us the samples and we will check.

[DnA]

Please send us the samples and we will check.

The audio sample that I used was the sound of the 20th Century Fox intro in "Fight Club" movie (DTS converted via BeHappy (NicDtsSource.dll) to 6 channel WAV). I suppose it will happen also on other 20th CF movies intros. The beginning has a lot of artifacts.

I hope that this will help to find the problem.

Otherwise, congratulations for the perfect encoder. :)

[Kurtnoise]

About ND logos and graphics :

http://www.nero.com/nerodigital/eng/Nero_Digital.html
http://www.nero.com/nerodigital/eng/Main_profiles.html
http://www.nero.com/nerodigital/eng/copyright.html

Documents available on this web site

Permission to use documents and related graphics available from this server ("Server") is granted, provided that:

* The copyright notice below appears in all copies and that both the copyright notice and this permission notice appear.
© 2004 Nero AG, Karlsbad, Germany. All rights reserved.
* Use of documents and related graphics available from this Server is for informational and non-commercial or personal use only.
* No documents and related graphics available from this Server are modified in any way.
* No graphics available from this Server are used separately from accompanying text.
.....

[GodofaGap]

* No graphics available from this Server are used separately from accompanying text.

Including the accompanying text with the certfication logo's is a bit troublesome for a GUI... :)

[dimzon]

- Certified under strict requirements and QA verification
- Nero logo really stands for the quality of audio of this product

Does'nt Your own CLI encoder meet this requrements.
I just want to show ND logo in configuration form to show encoder's orign and make GUI eye-candy look...


Samples:
http://img147.imageshack.us/img147/1221/untitled6pg.png

http://img226.imageshack.us/img226/222/20mg.png

http://img477.imageshack.us/img477/9522/untitled3qx.png

http://img297.imageshack.us/img297/5789/untitled6ur.jpg

[Ivan Dimkovic]

No, just having our encoder supported does not mean granting the right to logo use.

I am, however, not the person to grant you right to use Nero Digital / Nero Digitial Audio logos in your software - I just want to keep people informed with the right information.

For example, other companies have rules as well (e.g. Dolby) with regards to usage of their Logo, too.

[SeeMoreDigital]

Hi Ivan,

Logo permissions permitting... You've got to admit Dimzon's proposal sure looks cool though ;)


Cheers

[Carpo]

sure looks pukka :)

[Romario]

I don't know why you guys arguing about logo, if Ivan says it's illegal to use it, then don't put logo in your GUI. Simple as that.

[siddharthagandhi]

Hey dimzon that last screenshot is from what software?


And he didnt say it was illegal. He said you need permission.

[dimzon]

Hey dimzon that last screenshot is from what software?
BeHappy ;)

And he didnt say it was illegal. He said you need permission.
It has been removed today. New form looks:
http://img240.imageshack.us/img240/4835/untitled0lc1.png

[SeeMoreDigital]

I wonder something like this would be allowable: -

http://img57.imageshack.us/img57/6182/freereference9ts.th.png (http://img57.imageshack.us/my.php?image=freereference9ts.png)


Cheers

[dimzon]

I wonder something like this would be allowable: -

http://img57.imageshack.us/img57/6182/freereference9ts.th.png (http://img57.imageshack.us/my.php?image=freereference9ts.png)


Cheers
Ivan Dimkovich ?

[dimzon]

variations:
http://img105.imageshack.us/img105/7582/dnaac04kk.pnghttp://img115.imageshack.us/img115/4757/dnaac14kv.png
http://img139.imageshack.us/img139/4532/dnaac20ui.pnghttp://img123.imageshack.us/img123/6416/dnaac32ch.png
http://img110.imageshack.us/img110/3168/dnaac43dw.pnghttp://img110.imageshack.us/img110/2144/dnaac56fr.png

[SeeMoreDigital]

Hey... they look cool :D

[imcold]

I use this: http://img220.imageshack.us/img220/9720/sngicon648im.png
:D

[Sirber]

General #0
Count : 167
Count of stream of t : 1
Kind of stream : General
StreamKindID : 0
Count of video strea : 0
Count of audio strea : 1
Count of text stream : 0
Count of chapter str : 0
Complete name : C:\Projets\RealAnime5\dist\temp\audio.wav
Folder name : C:\Projets\RealAnime5\dist\temp
File name : audio
File extension : wav
File size : 4594340
File size : 4.38 MiB
File size : 4 MiB
File size : 4.4 MiB
File size : 4.38 MiB
File size : 4.382 MiB
Format : WAV
Format : Wave
Extensions : WAV
Overal BitRate : 1536054
Overal BitRate : 1536 Kbps
PlayTime : 23928
PlayTime : 23s 928ms
PlayTime : 23s 928ms
PlayTime : 23s 928ms
PlayTime : 00:00:23.928
Audio codecs : PCM

Audio #0
Count : 35
Count of stream of t : 1
Kind of stream : Audio
StreamKindID : 0
Codec : 1
Codec : PCM
Info : Microsoft PCM
Url : http://www.microsoft.com/windows/
Bit rate : 1536000
Bit rate : 1536 Kbps
Bit rate mode : CBR
Channel(s) : 2
Channel(s) : 2 channels
Sampling rate : 48000
Sampling rate : 48 KHz
SamplingCount : 1148544This file was made from avs2wav, play #1 in MPC, but nero tells me "Error: cannor parse WAV".

[Sharktooth]

coz avs2wav produces weird wav headers..i tould you that in msn too... that's why i suggested using bepipe in sng...:P

[Rockaria]

The MPC handles the seemingly corresponding nero encoder option -ignorelength internally for decoding(playing).
I also experienced a similar bug(pending) when using avs2wav on 44.1k(6ch) sources. Using last.SSRC(48000).SSRC(44100) was my temporary fix.
Also redirecting the bepipe stdout to a wav may solve the issue : bepipe --script "import(^xx.avs^)" > xxx.wav

[Peter1234]

Sirber

This file was made from avs2wav, play #1 in MPC, but nero tells me "Error: cannor parse WAV".

When using avs2wav, I have been using BeSweet to re-encode the avs2wav output wave file before using Nero encoder. This may not be the best solution, but it works.

[otherdarom]

@Ivan Dimkovic:

When will the Linux version come ?
Will it be able to convert from other formats not only wav ?

[StratosFeaR]

I searched this thread for an answer but couldn't find one.

Can anyone explain to me why this codec sounds really good through windows (Foobar, Media Player Classic) yet when I try to play the same file on my OSX box (iTunes, Quicktime) or on my iPod it sounds awful?

I encoded a few wav's using the '-q 0.2' setting and I was amazed at the quality on my windows box but as soon as I tried the file on the iPod, iTunes etc it sounds like a really poorly encoded track.

Cheers.

[imcold]

If I had to guess, i'd say that iTunes & Quicktime doesn't support SBR, so it replays only low frequency signal of the file. Can anybody confirm this?

[SeeMoreDigital]

Indeed, neither player supports the SBR element of an AAC-HE stream...

Shame really... I would have thought in this day-and-age, support for SBR would be a basic option.


Cheers

[Rockaria]

Without a doubt, with those price range, the user-requirments of the he-vbr, hrtf,,, and even the spdif-out(2~6~ch) are feasible.

Maybe the 'Apple-AAC' only is regarded as the 'true(==authorized==useful :rolleyes: ) AAC' to some interest group?

[StratosFeaR]

Indeed, neither player supports the SBR element of an AAC-HE stream...

Shame really... I would have thought in this day-and-age, support for SBR would be a basic option.


Cheers

Thanks, that is a damn shame. 29kbps was quite listenable and it would have allowed me to squeeze a huge amount more onto the iPod.

[JimmyZ]

great, it would be sweet if a avc cmd line encoder comes out too:D

[Stian]

great, it would be sweet if a avc cmd line encoder comes out too:D

You are kidding right? x264!

[Oline 61]

Jimmy, I'm really hoping that you are asking for a Nero CLI AVC encoder!

[shon3i]

Jimmy, I'm really hoping that you are asking for a Nero CLI AVC encoder!

Of course he think, but i think that is impossible becaose ateme make encoder, not Nero.

[Eric B]

Thanks for this new CLI encoder, I can finally use besweet to encode my 5.1 AC3.

I wonder how this new encoder is comparable with the encoders in Nero7

I've tested to reencode a 2 channels 192k AC3 to AAC.
1) This new free encoder via Besweet/BSN in streaming profile -> got a 102k AAC-LC file.
2) idem with internet profile -> got a 66k AAC-HE file.
3) Under Nero Burning Rom 7.0 (AAC encoder 4.2.0.27), with HE streaming high quality profile -> got a 57k AAC-HE file.

I wonder which AAC file from these 3 should theoricaly give the best quality. I suppose the new free encoder should be better than the old one of Nero 7.0, but which such a big difference of bitrates?

[williedigital]

"Store Entire Audio Album in a Single .mp4 File with all the Features of an Audio CD embedded inside, but at a fraction of the space!"

This sounds like a really cool feature. Has anyone tried it out?
Few questions I wondered about--

1. How does the tagging work?
2. How do these files play in both software (winamp, itunes, etc) and hardware (ipod, etc) players? Can you skip tracks, etc?
3. Is it possible to make an mp4 file of this sort from already existing audio files, assuming they are part of the .mp4 spec (aac, mp3, etc).
4. Is it possible to "access" each of the individual tracks, in case you wanted to make a mix cd?

[dimzon]

1. How does the tagging work?
2. How do these files play in both software (winamp, itunes, etc) and hardware (ipod, etc) players? Can you skip tracks, etc?
3. Is it possible to make an mp4 file of this sort from already existing audio files, assuming they are part of the .mp4 spec (aac, mp3, etc).
4. Is it possible to "access" each of the individual tracks, in case you wanted to make a mix cd?


AFAIK free CLI encoder from Nero doesn't perform any tagging itself BUT you can use other 3-rd party tools (foobar2000 or Tag&Rename)
Foobar2000 plays it fine (it shows every mp4 chapter as separate item in playlist). And You can use ANY DirectShowPlayer in combination with Haali Media Splitter
Yes, it's easy, use MP4BOX for it
Via Foobar2000 you can decode separate chapters into WAV files. Or you can split one MP4 to separate MP4 using MP4BOX

[fanbanlo]

sorry, a little OT.

How can I tell if my mobile phone is playing HE correctly, or just playing as LC?

Thanks.

[Oline 61]

sorry, a little OT.

How can I tell if my mobile phone is playing HE correctly, or just playing as LC?

Thanks.
You should be able to hear the difference.

[Kostarum Rex Persia]

What about Intesivity Stereo for LC-AAC in the updated codec, can Nero support this in new update?

[fanbanlo]

I found 64kbps unforced yeild HE (reported as SBR by foobar), yet LC sounds better...

Any chance Nero CLI will support Mono AAC encode?
I tried a mono Wave file as input and it still produces a 2-channel AAC (reported by foobar)?

[kaabob]

Hi,

I'm planning to encode Memento in 6-channel HE-AAC

This is a 1hr 53min movie that I plan to backup along with x264 and HE-AAC in a mp4 container to fit on a 700 MB cd...

Any recommendations as to how I should encode the multichannel audio? I see some using ABR 128, and others using VBR 0.3...

[shon3i]

Any recommendations as to how I should encode the multichannel audio? I see some using ABR 128, and others using VBR 0.3...I encode with 112 CBR and everything just fine, btw dont go below 96 and everything will be fine

[SeeMoreDigital]

Hi,

I'm planning to encode Memento in 6-channel HE-AAC

This is a 1hr 53min movie that I plan to backup along with x264 and HE-AAC in a mp4 container to fit on a 700 MB cd...

Any recommendations as to how I should encode the multichannel audio? I see some using ABR 128, and others using VBR 0.3...I wonder if you can do me a favour?

When you've finished generating your 6Ch AAC-HE audio file with Nero's AAC encoder. As a cross reference, could you generate another 6Ch AAC-HE file using Coding Technologies (CT) AAC encoder, at the same bit-rate?

PM me if you need information about CT encoding applications and settings.


Many thanks

[MatMaul]

I wonder if you can do me a favour?

When you've finished generating your 6Ch AAC-HE audio file with Nero's AAC encoder. As a cross reference, could you generate another 6Ch AAC-HE file using Coding Technologies (CT) AAC encoder, at the same bit-rate?

PM me if you need information about CT encoding applications and settings.


Many thanks

I can do that if you want, I have an english he 5.1 aac encode of star wars 4 at 192 kbps (q=0.25) with nero I can encode also with ct if you want (I use foobar for transcoding).

[SeeMoreDigital]

Thanks Mat,

Given that we now have a whole load of people thinking of generating 6Ch AAC-HE streams with Nero's new encoder, I think now is as good a time as any to start comparing Nero's "free" encoder with other encoders.

However, it would be rude to do it within this thread ;)

[MatMaul]

You think we must do that in doom9 forum or in hydrogenaudio ?
I can encode different samples with different encoders but I can't make abx test, I haven't 6ch equipment.
Can you create the thread, here or in hydrogenaudio?

[williedigital]

dizmon, thanks for the reply

a few questions about using mp4box.

1. What command do I use to combine existing mp3's? Am I muxing them or joining them?
2. Where do you add artwork with mp4box?

[dimzon]

dizmon, thanks for the reply

a few questions about using mp4box.

1. What command do I use to combine existing mp3's? Am I muxing them or joining them?
2. Where do you add artwork with mp4box?


If You want to create one mp4 file (with) from multiple mp3 files you must

create chapter file
choose one of

join mp3 via any tool, mux resulting big mp3 into mp4 via mp4box
mux every mp3 into separate mp4 via mp4box, join them together into one big mp4 via mp4box

import prepared chapters file into mp4 using mp4box

Refer to mp4box documentation or/and use mp4box GUI like YAMB (http://forum.doom9.org/showthread.php?t=93927).


Anycase mp4 format itself is not this thread discussion, refer to this forum (http://forum.doom9.org/forumdisplay.php?f=74)

Cheers

[fanbanlo]

Doesn't XP just uses '>' instead of '|' for piping? Do STDIN and STDOUT work under XP at all?

I wanted to wma -> wav using ffmpeg, and feed Nero CLI as STDIN.

I tried:
C:\ffmpeg>ffmpeg -i INPUT.wma -acodec pcm_s16le - > neroaacenc_sse -q 0.28 -if - -of OUTPUT.mp4

FFmpeg version CVS, Copyright (c) 2000-2004 Fabrice Bellard
configuration: --enable-theora --enable-mp3lame --enable-libogg --enable-vorb
is --enable-faad --enable-faac --enable-xvid --enable-x264 --enable-mingw32 --en
able-a52 --enable-dts --enable-pp --enable-gpl --enable-memalign-hack --enable-a
mr_nb --enable-amr_wb
libavutil version: 49.0.0
libavcodec version: 51.1.0
libavformat version: 50.0.0
built on Jan 29 2006 03:34:22, gcc: 4.0.2
Input #0, asf, from 'ga.wma':
Duration: 00:41:10.4, start: 0.608000, bitrate: 20 kb/s
Stream #0.0: Audio: wmav2, 22050 Hz, mono, 20 kb/s
Unable for find a suitable output format for 'pipe:'

What can I do to make it work?

Thank you!

[tebasuna51]

Doesn't XP just uses '>' instead of '|' for piping? Do STDIN and STDOUT work under XP at all?
Nope.

Program1 > Program2
write the Program1 output over a file named Program2, Program2 is not executed.

Program1 | Program2
connect Program1 STDOUT with Program2 STDIN, both programs are executed.

[Rockaria]

A detailed definition of redirection (http://www.microsoft.com/resources/documentation/windows/xp/all/proddocs/en-us/redirection.mspx?mfr=true) by MS.

[Inc]

C:\ffmpeg>ffmpeg -i INPUT.wma -acodec pcm_s16le | xxxxxxx ....

Shure ffmpeg provides also a waveheader at beginning when serving the data to stdout?

[Ivan Dimkovic]

New Version (1.0.0.2) is out, get it from:

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

Version Number: 1.0.0.2

Release Date: 18/05/2006
New Features:

* neroAacTag.exe for Nero Digital Audio Tagging
* Nero AAC Encoder
- Now writes mpeg-4 tags at the beginning of the MP4/3GP files
- Now supports multiple input files (-if) and produces a multichapter
MP4 file when more than one input file is present
- Backwards compatible signalling of SBR and PS Audio Object Types
(also when LC is used for encoding) for improved 3GPP
Compatibility
- Added progress display
* Nero AAC Decoder
- changed to new faster AAC/HE-AAC/HE-AAC v2 decoder
- Added support for the backwards compatible signalling and
corresponding behaviour


Version Number: 1.0.0.0

Release Date: 01/05/2006

[dimzon]

New Version (1.0.0.2) is out, get it from
Any quality-related changes in encoder?

[Ivan Dimkovic]

Yes, there were some small quality fixes, too.

[bond]

writes mpeg-4 tags? i assume those are not the apple-style tags? what specs are defining those tags?

[Ivan Dimkovic]

We do write Nero Digital-style tags, defined by the Nero Digital Audio specification.

In addition, compatibility tags are written, too - so other tagging systems will see the tags as well.

[bond]

We do write Nero Digital-style tags, defined by the Nero Digital Audio specification.in what way do they differ from apple's tags?

[Ivan Dimkovic]

They differ in a way that they enabled ND-Audio specific features (CD index maps, replay gain, chapter-specific album art, etc...)

It is of course understood that you have the freedom to use whatever tagging specification you wish - but at least I will make sure that Nero tags are catering to most wishes of the audio communities.

[bond]

interesting. can they be seen as an extension to the apple tags (in the sense that basic tags, like "name" or so, are identical in nero and apple. storage in the udta atom...) or are they different?

[fanbanlo]

C:\ffmpeg>ffmpeg -i INPUT.wma -acodec pcm_s16le | xxxxxxx ....

Shure ffmpeg provides also a waveheader at beginning when serving the data to stdout?

I tried adding -f wav like this:

C:\ffmpeg>ffmpeg -i INPUT.wma -acodec pcm_s16le -f wav - | neroaacenc_sse2 -q 0.
28 -ignorelength -if - -of output.mp4

FFmpeg version CVS, Copyright (c) 2000-2004 Fabrice Bellard
configuration: --enable-theora --enable-mp3lame --enable-libogg --enable-vorb
is --enable-faad --enable-faac --enable-xvid --enable-x264 --enable-mingw32 --en
able-a52 --enable-dts --enable-pp --enable-gpl --enable-memalign-hack --enable-a
mr_nb --enable-amr_wb
libavutil version: 49.0.0
libavcodec version: 51.1.0
libavformat version: 50.0.0
built on Jan 29 2006 03:34:22, gcc: 4.0.2
Input #0, asf, from 'INPUT.wma':
*************************************************************
* *
* Nero Digital Audio Reference MPEG-4 & 3GPP Audio Encoder *
* Copyright (C) 2006 Nero AG *
Duration: 00:41:10.4, start: 0.608000, bitrate: 20 kb/s
Stream #0.0: Audio: wmav2, 22050 Hz, mono, 20 kb/s
Output #0, wav, to 'pipe:':
Stream #0.0: Audio: pcm_s16le, 22050 Hz, mono, 352 kb/s
Stream mapping:
Stream #0.0 -> #0.0
* All Rights Reserved Worldwide *
* *
* Package build date: May 1 2006 *
* *
* See -help for a complete list of available parameters. *
* *
*************************************************************
ERROR: could not parse WAV file
size= 106376kB time=2470.0 bitrate= 352.8kbits/s
video:0kB audio:106376kB global headers:0kB muxing overhead 0.000040%


Problem with piping? 'cause I tried:
> ffmpeg -i input.wma -f wav temp
> neroaacenc_sse2 -q 0.28 -ignorelength -if temp -of output.mp4

and it works... Nero's problem or XP's piping problem?

update: tested with 1.0.0.2, same result.

Thank you.

[Avish]

New Version (1.0.0.2) is out, get it from:

http://www.nero.com/nerodigital/eng/down-ndaudio.phpHow one could use the provided decoder? Is it usable to general users?

[fanbanlo]

How one could use the provided decoder? Is it usable to general users?
yes, read End-User License Agreement for details.

[Avish]

yes, read End-User License Agreement for details.Since when "How to use" details are being written in EULA?:p

I think I didn't put my question in proper way. What I meant was Is the decoder usable for general ppl like me who have no clue how to use these EXEs, simple usage like using it to decode AACs through some player? Or is it only for Devs who use it in their programs by calling through intarnal scripts? Programs like meGUI or BeHappy. ;)

Thanks.

[fanbanlo]

Since when "How to use" details are being written in EULA?:p

I think I didn't put my question in proper way. What I meant was Is the decoder usable for general ppl like me who have no clue how to use these EXEs, simple usage like using it to decode AACs through some player? Or is it only for Devs who use it in their programs by calling through intarnal scripts? Programs like meGUI or BeHappy. ;)

Thanks.
it is very easy to use.
almost every CD ripper can use this.
it just takes in a .wav file and convert into a .mp4 with aac codec.

Read this thread!

[Avish]

it is very easy to use.
almost every CD ripper can use this.
it just takes in a .wav file and convert into a .mp4 with aac codec.

Read this thread!I'm asking about the decoder exe, not the encoder exe. :)

[fanbanlo]

I'm asking about the decoder exe, not the encoder exe. :)
that's for AAC (mp4) -> PCM (.wav), is this what u want?
> neroaacdec -help

*************************************************************
* *
* Nero Digital Audio Reference MPEG-4 & 3GPP Audio Decoder *
* Copyright (C) 2006 Nero AG *
* All Rights Reserved Worldwide *
* *
* Package build date: May 26 2006 *
* *
* See -help for a complete list of available parameters. *
* *
*************************************************************
Usage: neroaacdec -if <input file> -of <output file> [-chapter <number>]

So you can enter:

neroaacdec -if input.mp4 -of output.wav

in your Command Prompt.

[Avish]

that's for AAC (mp4) -> PCM (.wav), is this what u want?
> neroaacdec -help

*************************************************************
* *
* Nero Digital Audio Reference MPEG-4 & 3GPP Audio Decoder *
* Copyright (C) 2006 Nero AG *
* All Rights Reserved Worldwide *
* *
* Package build date: May 26 2006 *
* *
* See -help for a complete list of available parameters. *
* *
*************************************************************
Usage: neroaacdec -if <input file> -of <output file> [-chapter <number>]

So you can enter:

neroaacdec -if input.mp4 -of output.wav

in your Command Prompt.

Yes. That's the answer I was looking for. What is the exact use of that decoder exe. Its not what I thought. ;)

Thank You. :)

[obieobieobie]

I'm having problems with -2pass and -br


C:\>neroAacEnc_SSE2.exe -br 30 -2pass -if "Star Wars Theme.wav" -of test2
*************************************************************
* *
* Nero Digital Audio Reference MPEG-4 & 3GPP Audio Encoder *
* Copyright (C) 2006 Nero AG *
* All Rights Reserved Worldwide *
* *
* Package build date: May 26 2006 *
* *
* See -help for a complete list of available parameters. *
* *
*************************************************************

ERROR: no valid SBR configuration found

C:\>

I can't see what I'm doing wrong.

[fakey]

obieobieobie, please read the readme.txt.

You are trying to encode at 30bps.

[obieobieobie]

oh holy crap, how could I have missed that. Thank you, fakey. :)

edit: ooh that's very nice. At 31 kbps, AAC + SBR + PS sounds a whole lot better than vorbis aoTuV b4b at q=-2 (which is approx. the same bitrate)

stellar encoder, ivan and nero digital :)

[stucky]

Would it be terrible of me to want 2 pass encoding AND a quality based bitrate target?

[GodofaGap]

?

You can either do a quality based encode, and then you don't know the resulting bitrate, or you can do a 2-pass encode with a target bitrate.

A 2-pass quality based encode doesn't make sense, and I'm not sure what you mean with a "quality based bitrate target".

[stucky]

Please correct me if I'm wrong:

1 The basic idea behind Npass encoding is that you get a higher quality encoding for the same number of bits used.

2 Quality based encoding allows me to trust the encoder to use the number of bits required to achieve a certain quality of encoding, but not more bits than required.

As a human, I don't know how many bits a specific track is going to require, but I do know is that I want the smallest possible file that sounds the best it possibily can.

[Oline 61]

1-pass quality tries to reach a certain quality. It analyzes each sample and determines how many bits it needs to reach the specified quality.

2-pass bitrate tries to reach a certain bitrate. The first pass analyzes each sample and weights it comparatively to the other samples in the stream. Then, the bitrate curve produced in pass 1 is scaled to reach an average bitrate, and then encoded in the second pass.

Multipass encoding only gives you a higher quality per bit when you are trying to reach a bitrate target.

Bottom line: If you are trying to reach a specified bitrate, use 2-pass bitrate based encoding, if you have no bitrate target use 1-pass quality based encoding. Since quality doesn't have a set size quota, using 2-pass is redundant and unessecary.

[stucky]

That makes sense.

[ReinerSchweinlin]

Thanx a lot to Ivan and Nero for releasing this very fine peace of software.
After using foobar with this new encoder, I decided to convert some of my MP3s to play on my ipod and came across the limitations of the apple-device. As far as I understand right now, advanced Features like SBR oder "-he" are not supported by the plattform apple uses (portableplayer, I assume?).
Am i right that forcing LC-Mode with "-lc" would give compatible Files for playing on ipod? Can someone please confirm, as I can´t find any detailed tech-info on what is supported by Apple.

[bond]

he-aac is backwards compatible, meaning every lc-aac device can play he-aac files

and afaik apple doesnt support he-aac explicitely (which doesnt mean the files wont play)

[DeathTheSheep]

HE may be backwards compatible for sure, but the high spectrum will be cut off (only 22khz will be played if the device doesn't explicitly support HE-AAC).

[junglemike]

Hi, I'm sorry if this was asked. Couldn't find it. I've experimented little bit with free nero encoder and Foobar2000. I have some questoins. Sorry if this is too obvious I read faq and neroaacenc.exe -help and couldn't find the answers.

Q1) I notices that for some low bitrates nero downmixes to mono automatically. What is bitrate treshold that it starts to happen?

Q2)Can i force it be stereo? even for very low bitrates?

Q3) How can I force it be mono? (no such option in -help file)

Q4)What settings would be good for compressing audio books?

Q5) I compare latest Ogg with q -2 on 22khz/mono files (audio books) - which gives me ~20kbps and nero encoder. In general, i see that for regular music (44khz) at very low bitrates <64 - nero is better than ogg, but for 22khz/mono files nero sound much worse , Why?

[DeathTheSheep]

Q1 and Q2) Check your decoder. In my experience, Nero defaults to stereo, even at the lowest of bitrates. However, some decoders cannot recognize the stereo elements of low-bitrate streams due to the fact that these decoders don't support PS (Parametric Stereo).

Q4) Depending on your decoder, preference, etc, I'd suggest as low as Quality 0.1 with standard settings. Sometimes lower.

Q5) Try forcing LC profile. To do this, check the -help.

[Kurtnoise]

Q3) How can I force it be mono? (no such option in -help file)
For the moment, we can't...

Q5) Try forcing LC profile.
huh ? Considering question 5 asked by Junglemike, that makes no sense...Could you develop a little bit more ?

[InfoCynic]

I'm having terrible trouble with this program randomly hanging my machine. I've tried using it directly (after using azid to create a 6ch wav) or with Belight (from source 6ch AC3) with replacement bsn. Sometimes I can get through an entire encode (45 min file, about 15-20 minutes at HE q = .3), sometimes not. Various files (from a season of a TV show), so not all the same file.

Other CPU-intesive tasks like XviD encoding don't hang the system.

I have an Athlon XP 2500+ @ 3200+ and 2 GB of RAM.

[Sharktooth]

try removing the overclock and see if it helps.

[elmimmo]

They differ in a way that they enabled ND-Audio specific features (CD index maps, replay gain, chapter-specific album art, etc...)

Ivan, could you do a step by step on how it should be done to go from a CD to MP4 with CD index maps and chapter-specific album art?

Please, excuse me if this was explained elsewhere. I found nothing.

[DeathTheSheep]

Q5) Try forcing LC profile.
huh ? Considering question 5 asked by Junglemike, that makes no sense...Could you develop a little bit more ?
Ah ya. Maybe for some reason Nero's doing your 22khz file in HE-AAC, which is bound to sound just terrible. Forcing LC profile will keep the encoder clean. ...eh, just a try-and-see suggestion, haven't actually tested this myself.

:o

[Kurtnoise]

Well...If an encoder uses automatically a lowpass or a resampling, it's for a good reason imo. Why not let the encoder to choose by himself the appropriate settings ? Today, most of the encoders are well tuned.

Also, keep in mind that this is a lossy compressor.

[rootkit]

What is this?

[~/Documenti/Film2iPOD]$ file audiodump.wav
audiodump.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz
[~/Documenti/Film2iPOD]$ neroaacenc -lc -q 0.15 -if audiodump.wav -of test.mp4
ERROR: could not parse WAV file
*************************************************************
* *
* Nero Digital Audio Reference MPEG-4 & 3GPP Audio Encoder *
* Copyright (C) 2006 Nero AG *
* All Rights Reserved Worldwide *
* *
* Package build date: May 26 2006 *
* *
* See -help for a complete list of available parameters. *
* *
*************************************************************

[Kurtnoise]

Why you didn't run trough wine ?

[otherdarom]

that was under wine...
wo. wine u get this:
"run-detectors: unable to find an interpreter for ./neroAacEnc.exe"

it seems that neroenc doesnt work under wine...

[Sharktooth]

sadly it doesnt.

[Kurtnoise]

it seems that neroenc doesnt work under wine...

sadly it doesnt.

Too bad for you...:p

http://img244.imageshack.us/img244/7523/winendjg5.png

Works fine here...Anyway, if you're looking for a HE-AAC encoder which works without wine, have a look at Helix Producer or 3GPP reference encoder.

[otherdarom]

Too bad for you...:p

http://img244.imageshack.us/img244/7523/winendjg5.png

Works fine here...Anyway, if you're looking for a HE-AAC encoder which works without wine, have a look at Helix Producer or 3GPP reference encoder.

Now i understand. The error was not to set the filenames in "quotation marks".

It works for me that way:
wine neroAacEnc.exe -q 0.15 -if "/usr/share/sounds/startup.wav" -of "/tmp/startup.mp4"

But not that:
wine neroAacEnc.exe -q 0.15 -if /usr/share/sounds/startup.wav -of /tmp/startup.mp4

Maybe NeroEnc tries to read the file as given /.../...wav and not as sum of path and filename.

I still wait for avisynth and neroenc as native versions for linux.
Any idea when that happen ?

Thanks for hints go to
Kurtnoise13

[Kurtnoise]

I still wait for avisynth and neroenc as native versions for linux.
Any idea when that happen ?
Avisynth 3.0 (i.e Linux port) is available but not usable yet. For NeroEnc, nobody knows...except Nero devs. :)

[Sharktooth]

Too bad for you...:p

http://img244.imageshack.us/img244/7523/winendjg5.png

Works fine here...Anyway, if you're looking for a HE-AAC encoder which works without wine, have a look at Helix Producer or 3GPP reference encoder.
You're right. And you helped me discoverying (by making me investigate on the problem) the "wine" static link was pointing to a quite old and badly configured version of wine.
Now everything works as it should. Thanks again ;)

[miztadux]

Hello,
I posted a question about AAC encoder delay, with some results of encoding/decoding tests.
The goal is to know if, in order to keep perfect sync in the playback, we got to compensate for this delay when encoding a movie soundtrack with AAC.
Surprisingly the Nero CLI encoder/decoder combo didn't introduce any sizable delay, that's one of the reason I link this question here...

I'd appreciate if people used to encode soudtracks in AAC would comment:

http://forum.doom9.org/showthread.php?p=902856#post902856

Kudos to the Nero team for providing a free high quality aac encoder.
Thanks.

[Boardlord]

New version (1.0.7.0) released today!

Changes:

neroAacEnc.exe:
- Quality tuning for some bitrates
- Added support for 16 and 22.05 kHz samplerates for HE AAC
- Fixed HE AAC configuration problems for 5.1 files (at q=0.0)
- Temporary file handling fixed on Windows Vista
- Various speed optimisations

neroAacDec.exe:
- Small speedups
- Reduced memory usage

neroAacTag.exe
- No changes

[Vytenis]

And wouldn't be interesting to see and compare vs. new aacplus_enc.exe
Recently Coding Technologies claims significant advance in new version:
The new encoder enhancements also lead to significantly improved speech quality and multi-channel audio fidelity at even lower bit rates. For example, delivering 5.1 Surround Sound now only requires 128 kbps, previous versions required 160 kbps. aacPlus is capable of delivering CD quality stereo at 48 kbps. In comparison, CD quality for MP3 is typically rated at 128 kbps.

[shon3i]

will be very interesting especilay 5.1 test, coz ct in earlier versions haved far better multichannel than other aac encoders IMO.

Both CT and Nero encoders are updated, new test will be wellcome.

[[)370|\|470!2]

No multiprocessor support, am i right?

[Kurtnoise]

You're right...

[Kurtnoise]

Not yet official but a new release is on the way soon...

changelog 2007-07-20 - Version 1.1.34.0

- neroAacEnc:
- Linux version
- New q value mappings
- Retuning of almost all bitrates
- Fix in HEv2 encoder/decoder delay
- Fixed incorrect streamlength written in MP4 files
- Significant speedup
- Removed -hinttrack option

neroAacDec:
- Linux version
- Fixed a memory leak

neroAacTag:
- Added -hinttrack option


No more wine tool...yeaaah. :)

http://www.listening-tests.info/mf-64-1/NeroDigitalAudio.zip

[menno]

Unless some serious bug comes up, this is the package we will release on the Nero website.

[Romario]

menno, what about proimised LC-AAC improvments, Ivan talked something about it?

[menno]

menno, what about proimised LC-AAC improvments, Ivan talked something about it?

They are in there too :)

[Gusar]

Is the linux build not optimized by any chance? Because there's a huge difference in speed:$ time a52dec -o wav6 "VTS_01_1 T01 3_2ch 384Kbps DELAY -248ms.ac3" 2> /dev/null | ./neroAacEnc -ignorelength -q 0.17 -if - -of linux.mp4
*************************************************************
* *
* Nero Digital Audio Reference MPEG-4 & 3GPP Audio Encoder *
* Copyright 2007 Nero AG *
* All Rights Reserved Worldwide *
* *
* Package build date: Jul 20 2007 *
* *
* *
* See -help for a complete list of available parameters. *
* *
*************************************************************

Processed 304 seconds...

real 1m28.181s
user 1m25.989s
sys 0m0.767s


$ time a52dec -o wav6 "VTS_01_1 T01 3_2ch 384Kbps DELAY -248ms.ac3" 2> /dev/null | wine neroAacEnc.exe -ignorelength -q 0.17 -if - -of win.mp4
*************************************************************
* *
* Nero Digital Audio Reference MPEG-4 & 3GPP Audio Encoder *
* Copyright 2007 Nero AG *
* All Rights Reserved Worldwide *
* *
* Package build date: Jul 20 2007 *
* Package version: 1.1.34.0 *
* *
* See -help for a complete list of available parameters. *
* *
*************************************************************

Processed 304 seconds...

real 0m52.108s
user 0m50.207s
sys 0m0.752

A 30 seconds difference on a 5 minute clip. Now, imagine if I would encode the entire movie, not just one chapter.

[menno]

Yeah, we didn't have time yet to do all optimizations for the linux version too. I guess if you have/use wine then using that will be better at the moment.
First thing was to actually get it working on linux, next step will be to incorporate all the optimizations.

[Gusar]

Yeah, we didn't have time yet to do all optimizations for the linux version too. I guess if you have/use wine then using that will be better at the moment.
First thing was to actually get it working on linux, next step will be to incorporate all the optimizations.Makes sense. Maybe this could be mentioned in the changelog. That said, the fact that you provide linux builds at all deserves respect. Thank you for providing linux users a high quality aac encoder.

[Bigmango]

Holly Molly !

Nero, thanks so much for Linux support :thanks:

[menno]

I tried adding -f wav like this:

C:\ffmpeg>ffmpeg -i INPUT.wma -acodec pcm_s16le -f wav - | neroaacenc_sse2 -q 0.
28 -ignorelength -if - -of output.mp4

FFmpeg version CVS, Copyright (c) 2000-2004 Fabrice Bellard
configuration: --enable-theora --enable-mp3lame --enable-libogg --enable-vorb
is --enable-faad --enable-faac --enable-xvid --enable-x264 --enable-mingw32 --en
able-a52 --enable-dts --enable-pp --enable-gpl --enable-memalign-hack --enable-a
mr_nb --enable-amr_wb
libavutil version: 49.0.0
libavcodec version: 51.1.0
libavformat version: 50.0.0
built on Jan 29 2006 03:34:22, gcc: 4.0.2
Input #0, asf, from 'INPUT.wma':
*************************************************************
* *
* Nero Digital Audio Reference MPEG-4 & 3GPP Audio Encoder *
* Copyright (C) 2006 Nero AG *
Duration: 00:41:10.4, start: 0.608000, bitrate: 20 kb/s
Stream #0.0: Audio: wmav2, 22050 Hz, mono, 20 kb/s
Output #0, wav, to 'pipe:':
Stream #0.0: Audio: pcm_s16le, 22050 Hz, mono, 352 kb/s
Stream mapping:
Stream #0.0 -> #0.0
* All Rights Reserved Worldwide *
* *
* Package build date: May 1 2006 *
* *
* See -help for a complete list of available parameters. *
* *
*************************************************************
ERROR: could not parse WAV file
size= 106376kB time=2470.0 bitrate= 352.8kbits/s
video:0kB audio:106376kB global headers:0kB muxing overhead 0.000040%


Problem with piping? 'cause I tried:
> ffmpeg -i input.wma -f wav temp
> neroaacenc_sse2 -q 0.28 -ignorelength -if temp -of output.mp4

and it works... Nero's problem or XP's piping problem?

update: tested with 1.0.0.2, same result.

Thank you.

Although it's old, bond asked me to look at this.
ffmpeg writes a length of 0 to the wave header when outputting to stdout. Even when the -ignorelength option is used this value 0 is not ignored in the aac encoder. The encoder assumes this length or less samples in the file.
Most software writes the maximum possible length to the wave header before it knows the actual output size. I consider it something that should be fixed in ffmpeg as pretty much every software I tried can not read this wave file.

[tebasuna51]

Although it's old, bond asked me to look at this.
ffmpeg writes a length of 0 to the wave header when outputting to stdout. Even when the -ignorelength option is used this value 0 is not ignored in the aac encoder. The encoder assumes this length or less samples in the file.
Most software writes the maximum possible length to the wave header before it knows the actual output size. I consider it something that should be fixed in ffmpeg as pretty much every software I tried can not read this wave file.

You are right about ffmpeg must fix this issue because also:

ffmpeg -i INPUT.wma -acodec pcm_s16le -f wav - > output.wav

make a unusable output wav and, of course, is not a NeroAacEnc problem.

But NeroAacEnc -ignorelength, can accept wav files with DataLength = 0, really the NeroAacEnc test to accept the wav (with -ignorelength) is:

RIFFLength = DataLength + DataChunkOffset

Then a standard PCM wav file (DataChunkOffset = 36) can be accepted with:
RIFFLength = 36
DataLength = 0

I think this test is really strange because if these fields are invalid (overflow for size > 4 bytes) you can't expect any relation between them.

For instance faad fill these two fields (when overflow) with 0xFFFFFF00 then are not accepted with NeroAacEnc. Also old NeroAacDec version output are not accepted.

Maybe the test is to ensure the 'data' chunk is the last ( if RIFFLength = DataLength + DataChunkOffset not exist other chunks after 'data' and all the bytes until eof must be audio samples) but I don't know any app. than write extrachunks at end of illegal wav files (> 4 GB).

If you want maintain the test maybe you can include a new parameter -ignore2length to really ignore the two invalid fields.

Thanks.

[bond]

:thanks: menno

[menno]

You are right about ffmpeg must fix this issue because also:

ffmpeg -i INPUT.wma -acodec pcm_s16le -f wav - > output.wav

make a unusable output wav and, of course, is not a NeroAacEnc problem.

But NeroAacEnc -ignorelength, can accept wav files with DataLength = 0, really the NeroAacEnc test to accept the wav (with -ignorelength) is:

RIFFLength = DataLength + DataChunkOffset

Then a standard PCM wav file (DataChunkOffset = 36) can be accepted with:
RIFFLength = 36
DataLength = 0

I think this test is really strange because if these fields are invalid (overflow for size > 4 bytes) you can't expect any relation between them.

For instance faad fill these two fields (when overflow) with 0xFFFFFF00 then are not accepted with NeroAacEnc. Also old NeroAacDec version output are not accepted.

Maybe the test is to ensure the 'data' chunk is the last ( if RIFFLength = DataLength + DataChunkOffset not exist other chunks after 'data' and all the bytes until eof must be audio samples) but I don't know any app. than write extrachunks at end of illegal wav files (> 4 GB).

If you want maintain the test maybe you can include a new parameter -ignore2length to really ignore the two invalid fields.

Thanks.

I'm not sure I understand exactly what you mean. Right now our wave library checks for a size of 4 (minimum) while 36 should be the minimum for even having a readable header.
Some sanity check has to remain, I think.

[tebasuna51]

I'm not sure I understand exactly what you mean. Right now our wave library checks for a size of 4 (minimum) while 36 should be the minimum for even having a readable header.
Some sanity check has to remain, I think.

Really I was wrong in one thing, The test is not:

RIFFLength = DataLength + DataChunkOffset

but

RIFFLength => DataLength + DataChunkOffset

Examples with standard PCM WAV files (DataChunkOffset=36):
RIFFLength=35, DataLength=0 ->Reject
RIFFLength=36, DataLength=0 ->Accepted
RIFFLength=37, DataLength=0 ->Accepted
RIFFLength=135, DataLength=100 ->Reject
RIFFLength=136, DataLength=100 ->Accepted
...

Examples with WAVE_FORMAT_EXTENSIBLE files (DataChunkOffset=60):
RIFFLength=59, DataLength=0 ->Reject
RIFFLength=60, DataLength=0 ->Accepted
RIFFLength=61, DataLength=0 ->Accepted
RIFFLength=159, DataLength=100 ->Reject
RIFFLength=160, DataLength=100 ->Accepted
...

Then the test is not to ensure the 'data' chunk is the last and become more strange for me. Ensure there are a readable header checking a field that can be with overflow (> 4 GB) or unknow when using 'pipe'?.

This field can be wrong, to ensure if there are a readable header try to read enough bytes from the stream, check if necessary parameters (Channels, Samplerate, BitsPerSample,...) are coherent and then continue reading and encoding until end of file.

This is, for me of course, the meaning of IgnoreLength parameter.

BTW, thanks for your free encoder, I dont want criticize but improve.

[MetalPhreak]

Just thought I'd let everyone know the new version has been "officially" released. I'm busy playing with it now.

[menno]

A new version of the Nero Digital Audio+ package has been released on http://www.nerodigital.com/

The package can be downloaded here: http://www.nero.com/nerodigital/eng/down-ndaudio.php

Changes since previous version:

* neroAacEnc
- Linux version
- New q value mappings
- Retuning of almost all bitrates
- Fix in HEv2 encoder/decoder delay
- Fixed incorrect streamlength written in MP4 files
- Significant speedup
- Removed -hinttrack option

* neroAacDec
- Linux version
- Fixed a memory leak

* neroAacTag
- Added -hinttrack option

[menno]

BTW, we found a bug when using -2pass option on 5.1 files, better not use that!
Old versions also had this bug, will be fixed in next version.

[deets]

any change of a multi threaded version :D

but thanks anyway, is great!

[MrCommunistGen]

Using MeGUI the SSE version isn't working for me anymore on my Athlon X2 3800+ its a S939 Manchester but the specific core shouldn't matter since all the X2s support SSE, SSE2, and SSE3. 1.0.7 and earlier worked just fine. The non-SSE version of 1.1.34.2 still works but I think its slower.
The MeGUI log file tells me:
"System.ApplicationException: Abnormal encoder termination 1
at MeGUI.AviSynthAudioEncoder.encode()
Fatal Error : This program was not built to run on the processor in your system."

-mcg

[Pookie]

You Nero guys are very cool for making this available to everyone. Thanks!

[menno]

It seems using the Intel compiler was not such a great idea, sorry to AMD users who want to use this.
But I think the current non-SSE build is not slower than the old SSE build, at least not much.

[menno]

any change of a multi threaded version :D


If you're encoding multiple files you could use foobar2000 (maybe some other GUI's support this too) to run multiple encoding threads at once.

[plane]

BTW, we found a bug when using -2pass option on 5.1 files, better not use that!
Old versions also had this bug, will be fixed in next version.

what kind of that bug?

http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=55827&view=findpost&p=501353
and is this another one bug?

[menno]

what kind of that bug?


Well, some tables get screwed up and the result is absolutely terrible sounding. Note: only for 5.1 and only with -2pass it is really bad.


http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=55827&view=findpost&p=501353
and is this another one bug?

I don't see any bug report in that post? You mean the fact that -br generates the same bitrate with and without -2pass? That is normal, it works to the same average bitrate, the distribution of the bits over the frames will just be different.

[plane]

I don't see any bug report in that post? You mean the fact that -br generates the same bitrate with and without -2pass? That is normal, it works to the same average bitrate, the distribution of the bits over the frames will just be different.

thanks for clear :)

[deets]

If you're encoding multiple files you could use foobar2000 (maybe some other GUI's support this too) to run multiple encoding threads at once.


yeah megui can now, but i have 4 cores and i wanna use them :D

go on, ill be your friend ;)

[MrCommunistGen]

@menno
I'm glad that the problem has a simple explanation. About the speed, its probably just my perception, especially since I just updated MeGui and it has a new way of showing encode speed. And of course my misguided hope that something using SSE/SSE2 etc will be faster... Anyhow thanks for joining up to answer questions and hear comments. Its always nice being able to communicate with the people "on the inside"

@deets
Is there a way to run more than one thread on audio? I know that some of the video codecs support multiple cores, but I didn't think that it worked for audio codecs because that would require support from the codec itself. I *think* there is an alpha version of LAME available that supports multithreading.

-mcg

[deets]

yeah i think i had only heard about the lame one as well. Its not usually a big deal for small clips, but if your converting a 2h20min DVD to aac then it could be handy, plus i have the cores doing nothing :P

but i knew that anyway, just wondered if it was even in the future as intel is pushing multi core support in software these days :)

[burfadel]

The Lame MT version doesn't process different sections of audio on each core, it simply does some of the acoustic modelling etc. on one thread and the actual encoding is still done on one thread only.

[menno]

@menno
I'm glad that the problem has a simple explanation. About the speed, its probably just my perception, especially since I just updated MeGui and it has a new way of showing encode speed. And of course my misguided hope that something using SSE/SSE2 etc will be faster... Anyhow thanks for joining up to answer questions and hear comments. Its always nice being able to communicate with the people "on the inside"

-mcg

If it's any comfort, the non-SSE build also uses a lot of SSE code :D
The difference is that for the SSE build version the compiler generates additional SSE code.

[Sharktooth]

blah... intel compiler crap...

[bond]

Although it's old, bond asked me to look at this.
ffmpeg writes a length of 0 to the wave header when outputting to stdout. Even when the -ignorelength option is used this value 0 is not ignored in the aac encoder. The encoder assumes this length or less samples in the file.
Most software writes the maximum possible length to the wave header before it knows the actual output size. I consider it something that should be fixed in ffmpeg as pretty much every software I tried can not read this wave file.they dont want to fix it in ffmpeg :(

[burfadel]

blah... intel compiler crap...

Yeah I wonder how the court case worked out? apparently the Intel compiler was designed so things compiled with it would work poorly on AMD systems. I guess if AMD won, the very latest compiler would be fine with both systems, I guess Intel won!

[Sharktooth]

still heard no words about that

[mizaru]

The SSE version of the encoder can easily be patched to run on AMD systems. However, I did not notice any speedup compared to the non-SSE version (tested on an X2 4200).

[menno]

The SSE version of the encoder can easily be patched to run on AMD systems. However, I did not notice any speedup compared to the non-SSE version (tested on an X2 4200).

Yes, the so-called SSE version is more there because it always was there. In the non-SSE version pretty much all places where SSE is possible, SSE is used, just only when your CPU supports it. In the next version we are considering removing the SSE build, as it seems to bring very little except increasing the download size :)

[scott1256ca]

I'm new to this, so please bear with me.

I running 1.1.34.2

Trying to convert a 6ch wav file to aac. All I get is 2 channel output.

neroAacEnc.exe -q0.3 -lc -if dead_aud2.wav -of dead_aud.mp4
also tried
neroAacEnc.exe -if dead_aud2.wav -of dead_aud.mp4

Is there some limitation? Is there a different download or version I should use?

I am opening the mp4 file in zoomplayer or mplayer and checking on the format in ac3filter. Decoder info says input format is PCM16 2/0 ...
When I open the .wav file, it says input format is pcm16 3/2.1

What am I doing wrong?

Thanks in advance

[menno]

I'm new to this, so please bear with me.

I running 1.1.34.2

Trying to convert a 6ch wav file to aac. All I get is 2 channel output.

neroAacEnc.exe -q0.3 -lc -if dead_aud2.wav -of dead_aud.mp4
also tried
neroAacEnc.exe -if dead_aud2.wav -of dead_aud.mp4

Is there some limitation? Is there a different download or version I should use?

I am opening the mp4 file in zoomplayer or mplayer and checking on the format in ac3filter. Decoder info says input format is PCM16 2/0 ...
When I open the .wav file, it says input format is pcm16 3/2.1

What am I doing wrong?

Thanks in advance

The MP4 file always says 2 channels, those players should get the info from the AudioSpecificConfig not from the mp4 track header itself. Try decoding it and you'll see that 6 channels come out.

[scott1256ca]

thanks. I'll try tomorrow.

[lolent]

I think there is a problem the nero encoder is not available anymore on nerodigital.com site ?

Someone has informations about this ?

[Schrade]

Huh. Wonder why it's off their page.

Ah well.. latest version is available here:

NeroDigitalAudio+ v1.1.34.0 (http://www.sendspace.com/file/57fc3f)

[Sharktooth]

1.1.34.2 (http://www.sendspace.com/file/4vrrly)

[menno]

It seems to have been lost in the webpage update for Nero 8. I'll see if I can get that solved :)

[arfster]

Is the audio ripper in Nero 8 of the same quality as this? (on highest quality settings, 2nd pass etc). Commandl ine is OK for a few files, but if I want to rip all my CDs the windows version looks a bit more practical :-)

[menno]

The current version in the web demo is the same version number as this command line encoder. Next webdemo will have an updated version IIRC.
There are also some batch tools around that can use this command line encoder, foobar2000 for example. Search this forum and you will find more. You can also have a look at http://www.hydrogenaudio.org

[Bigmango]

The current version in the web demo is the same version number as this command line encoder. Next webdemo will have an updated version IIRC.

According to a post from JohnV a couple months ago, Nero Recode does not do 2 pass audio encoding. Even if 2 pass is selected, it will do 2 pass for the video but not for the audio.

Has this changed ?

Also, the Nero wave aac export ui is very limited and you can't select 2 pass encoding either (even if saving the file to a wave). Is this going to improve ?

[menno]

According to a post from JohnV a couple months ago, Nero Recode does not do 2 pass audio encoding. Even if 2 pass is selected, it will do 2 pass for the video but not for the audio.

Has this changed ?


The new Recode (next release I think) will do proper 2pass for audio. It will also have better bitrate control for AAC.


Also, the Nero wave aac export ui is very limited and you can't select 2 pass encoding either (even if saving the file to a wave). Is this going to improve ?

Yeah, we consider adding some more options, but probably not 2pass, this is difficult considering the structure of the plugin manager (in some Nero apps the files will get written directly to a CD for example, this will not work well with 2pass, unless you use a CD-RW :p ).

[Bigmango]

The new Recode (next release I think) will do proper 2pass for audio. It will also have better bitrate control for AAC.

Awesome :thanks:


Also, (sorry this is off topic) if you communicate with them, please don't forget to bug the devel team for 720p AVCHD support (it now only supports 1080i & 1080p) :)

[SpAwN_gUy]

1.1.34.2 (http://www.sendspace.com/file/4vrrly)

i guess, something is fixed...
same version HERE: http://www.nero.com/eng/down-ndaudio.php

[Sharktooth]

yes. menno said they fixed the web pages in another thread

[Sharktooth]

leave alone the SSE version... there's no speedup... at least on your CPU...

[menno]

FYI here is a list of fixes in the current Nero 8 release (8.3.2.1)

Ndaudio version 1.1.34.13

* Greatly improved 5.1 encoding quality
* CBR now produces actual requested bitrate
* Improved 2nd pass encoding bitrate control
* Fixed MP4 file issue with 2nd generation iPod Shuffle


This encoder is used in:

* Nero Burning ROM (Encode Files, Save Tracks)
* Nero WaveEditor (save)
* Nero SoundTrax (save)
* Nero Recode (only with movie)
* Nero StartSmart (Audio Ripping)
* Nero MediaHome (if your receiving device supports AAC)

[Sharktooth]

is this encoder version also available with a CLI?

[menno]

Not yet, we will probably release a CLI version again when we are done with the next generation encoder.

[SpAwN_gUy]

wow.. next generation... cool...

*the only thing is.. how to properly(and open sorcerly, and.. easy) decode (or transcode) 5.1ch AAC streams... eh..

[SeeMoreDigital]

wow.. next generation... cool...

*the only thing is.. how to properly(and open sorcerly, and.. easy) decode (or transcode) 5.1ch AAC streams... eh..I'm currently product testing the Ziova CS615 media streamer, which is capable of playing/decoding 6Ch AC3 streams placed within the .MP4 container...

[dimzon]

I'm currently product testing the Ziova CS615 media streamer, which is capable of playing/decoding 6Ch AC3 streams placed within the .MP4 container...
AC3 in MP4 is hack == evil

[menno]

And I don't see what it has to do with playing 5.1 AAC

[Adub]

Maybe he meant AAC?

[LiFe]

Hi Menno,

Which Nero GUI lets you encode using the vbr quality option?

I use the cmd line to do all my 5.1 encoding via -q 0.4, and I don't seem to be able to replicate this using the nero provided GUIs?

Any chance we can get lucky and get an interim update of the cmd line before the next generation?

Thanks!!

[menno]

Hi Menno,

Which Nero GUI lets you encode using the vbr quality option?


Most actually use this, the Encode Files dialog for example. It just doesn't show exactly the q value (I guess you have to slide it 40% from the left...). This dialog will be improved greatly soon, also allowing ABR and CBR options.


Any chance we can get lucky and get an interim update of the cmd line before the next generation?


Probably not, sorry.

[plugh]

BTW, we found a bug when using -2pass option on 5.1 files, better not use that!
Old versions also had this bug, will be fixed in next version.
Can you clarify the nature of the bug? Does it apply across the board, for certain ranges of average bitrates, certain encoder modes (lc, he, hev2), ... ?

Or does this mean the only vbr-based 5.1 mode available is '-q'?

and BTW, what is the significance of q=0.37618960441 to the 'he' mode? (q=0.37618960442 fails)

[menno]

That 5.1 bug affected all modes. -q is the best to use if you want VBR (I personally prefer ABR, but that is no matter to you I guess :) ).

that -q number is probably exactly the limit after which HE AAC is not used anymore? Or do higher numbers work again?

[plugh]

That 5.1 bug affected all modes. -q is the best to use if you want VBR (I personally prefer ABR, but that is no matter to you I guess :) ).Actually I also wanted ABR. Was trying to understand where the issue was in order to avoid it. And I thought I saw a statement someplace that it was related to hev2, but I guess not...
that -q number is probably exactly the limit after which HE AAC is not used anymore? Or do higher numbers work again?Uh, yeah, it is. My question, out of curiosity, was the signifcance of that _particular_ value as the cut-off...
Or did someone just start grabbing random digits, tacked them after ".3" and said "yeah verily thou shalt not do 'he' beyond this quality" :rolleyes:

[menno]

Uh, yeah, it is. My question, out of curiosity, was the signifcance of that _particular_ value as the cut-off...
Or did someone just start grabbing random digits, tacked them after ".3" and said "yeah verily thou shalt not do 'he' beyond this quality" :rolleyes:

Hehe, I see, no, the limit for HE AAC is 80kbps. That q value is what the encoder relates to 80kbps. So it's not some human that spend a lot of time figuring it out, it's just the result from our bitrate to q mapping function :)