In the readme file is says the usage is:

neroAacEnc.exe [options] -if <input-file> -of <output-file>

It then says:

Specify - to encode from stdin.

If I want to transcode from flac to mp4 using neroAacEnc.exe, what would I type in the command line?
Similarly if I want to transcode from wavpack to neroAacEnc.exe, what would I type in the command line?

Thanks for the info so far.

Ax

Nero AAC encoder only works with PCM WAV files (it can support 16, 24, 32 bit depths as well as floating point)

In order to process FLAC, MP3 or any other files to AAC with Nero command-line encoder, you need to feed the encoder with already decoded PCM frames.

So you can't set flac.exe to stdout its output and set neroenc to stdin flac output? Sorry for the dumbness of this question!

I can only say from neroAacEnc side - it should work when you pipe it with wave data as STDIN

Thanks

Im at work at the moment so I cant test it. I found the following which pipes from flac to mp3

FLAC --decode --stdout file.flac | LAME -V2 --vbr-new - file.mp3

So can anyone test if the following works from the commandline?

FLAC --decode --stdout <input-file> | neroAacEnc.exe [options] -if - -of <output-file>

Thanks

Ax

Tested. Bit-compares to file created from original wave.

NB:I've not seen so many people huddled in a thread since 0.9 was released.

Well done Garf and Ivan.

Working here on AMD now BTW.

Edit: Command lines used:

Thanks

NB: You should really let foobar 0.9 do this for you, so that tags are converted also.

If you do this on the command line all your FLAC tags will simply be lost.

See the Nero AAC Recommended Settings thread for info on setting up foobar.

IC!

How would you go about tagging the files without foobar? Can it be done in 1 step using the commandline? So, pipes from flac to mp4 and tagging. I'm guessing foobar is using one of the popular tagging commandline programs?

foobar just slaps on the tags itself after encode, just set the right CLI encoder options

Thanks

Ok, I couldn't wait, I downloaded it to test it out!

I have to questions for the developers:

1. Why is there no percentage report on the commandline when using -2pass
2. Why is it not possible to use stdin and -2pass at the same time

Thanks

Because that can't possibly work unless the encoder caches all audio data somewhere, which would be quite painful if you're encoding a DVD soundtrack? You can't seek backwards in standard input.

You could use AtomicParsley to do something like this:
- M.

But you need to extract the meta data from the FLAC file somehow first...

Tag.exe has the --fromfile switch, but it won't tag MP4.

Ooh, this is getting a little OT... maybe if I thank Ivan and Garf again I can get away with it...

Yes, it is a fantastic program! I was just being picky. Really Sorry!

Garf, I see your point.

Thanks

hi, i just got the new file NeroDigitalAudio.zip (md5: c9e36ca81f7eee438c729f4fe8b9abf3) and i'm getting this error with any wav file i try to encode using the non ss2 version (neroaacenc -q 0.5 -if 001.wav -of 001.mp4):
ERROR: could not open WAV file

the pc here at work is a p2 300mhz using win98 se.
could this be some unicode issue?

Yes - this package is in fact fully Unicode.

thx Ivan, can you add this Note on the download page?

Unicode & ecnoder/decoder... Unicode is REALLY needed for such kind of tasks?
But... I want to ask a question about decoder part:
I want decode MONO AAC, but output wav is a "stereo". It's a old... "feature" of FAAD, but can you fix it, please?

Unicode is needed for sure: Unicode filenames, paths, and tags. They're not uncommon.

In fact it is needed for handling tags and other metadata which is in Unicode - we do conform to Unicode for all our metadata and also for all handling of the file names etc..., and I don't think switching back to half-hacks is a good idea.



We will work on this - there is a solution.

i'm glad it's fully unicode, it's one less thing to worry about when trying to encode files with "strange" characters.
if it wasn't, sooner or later people would ask for it

Thx for making this available for the non sse2 processors.

1. The quality (-q) versus the abr (-br) setting:

Are the algorithms similar or does the abr use a rather conservative algorithm in the coice of bitrates at a certain point?

2. What are the min/max values for -br?

VBR(-q) only looks at what the psymodel says. ABR may need to compensate to meet the target average bitrate.



12kbps to 465kbps for stereo

When i try to encode using the -br method, it just says SBR not found?

Oh man, I don't even know what to say. I've been rebooting into windows and going through the whole EAC + Itunes encode + Itunes fiasco to encode my new CDs. This way I'll be able to do it all from KAudioCreator!

You probably didn't read the help closely enough. (bits != kilobits)

Why HE profile can be used with higer rates than 160kbs, i try to encode to 256kbs with sbr



i get error message



And can you define what is max bitrates for all modes including 5.1, Thanks

What use should SBR have at bitrates higher than 96 kbps anyways?

I am doing some tests, a wanna see something. btw in 5.1 encoding i think is better to use for example 192kbs HE than LC

SBR works up to 192 kbps for 5.1 content - so basically there are two things that could be done:

a) erroring out (like it is done now)
b) adding presets for 256 kbps, but spitting out warning that LC-AAC might actually be a better choice

We will look into possibility b) whether it is feasible.

Thanks Ivan, but i have only to see how codec works on 88.2khz.

I think implementing SBR High Bitrate like CT then no posibilites to get erroring, like add switch -highsbr or something like this, and what you think about various stereo modes like CT.

I had seen it work @88.2 kHz some time ago ;-)



Although our encoder does algorithmically support "downsampled SBR" or "single-rate SBR" or "high-bitrate SBR" we do not export this option to the frontend.

The reason for this choice is because this particular codec mode puts tremendous burden to the audio decoder (cuts the decoding speed in half), while bringing no audible advantage over LC mode at the same bit-rate, and this has been tested in various listening tests by now.

I would like to point out - that the power of Nero Digital Audio will also be devices that play it in the future - with all the great features ND Audio brings.

If we allowed the single-rate SBR, I am afraid that many mobile device vendors would not be able to support this, as it would require very strong DSPs or embedded CPUs to support this - with no quality gains.



My opinion was always to leave encoder to decide what to use (LR, M/S, IS, etc...) - and thresholds for decision were set during the tuning process.

There is no need to allow modification I think, as there is much higher chance for non-experienced users to make settings that actually damage sound quality - and generate bad image for the complete technology when they listen to songs they made with "tweaked" codec (e.g. like with old MP3 encoders)

Good old encoder ;-)

You think Nero automaticly downsample 88->44khz. Why, i newer have bad things with CT encodings and devices

Thas is good reason, but i not say that for example 192kbsSBR is better than 192NONSBR. Did someone do tests with SBR @ higer rates comparing to LC

About IS, can you tell me when is activated. This is one of many reason to include this tweak. Me and many user want to know what is used in encoding. In my some test with AAC encoding M/S encoding not have sense at very high rates 220> like LR in very low rates <128. That aslo valid for MP3

This is codec for Pro users Right, then if somebody not know how to use codec, so then exist default settings like now. I just wanna make this tweak like LC,HE,HEv2 switches.

And i almost forgot i see that in mp3 section, ReplyGain. Until now somebody told that ReplyGain and AAC can't be. Can you confirm that.

Is there a way to specify MONO output from a dual-channel monaural file? For example, if an old album was recorded in mono, and I have the CD, ripping the CD will still produce a stereo *.wav.

Pesky, I know. Who listens to mono? (I do.)

- M.

Great work! I have tested out this encoder and it works like a charm! Could use a few little things like a better help section and a progress bar during the encoding process but this is just simply awesome either way. I've been waiting for something like this from Nero! There is now a viable and easy way to convert your music collection to Nero AAC. I can now use it easily with any ripper that supports command line encoders and the best part is it is free.

BTW, I'm not sure if I missed this in the above posts but how do you implement tagging with this encoder? Any switches to do that?

I have little ac3 6ch 448 Kbps. I make these convertions:
file wav 6 chanels 6ch.wav
file wav 2 chanels 2ch.wav (default matrix 6ch->2ch conversion for BS)
six files wav FR.wav, FL.wav, C.wav, LFE.wav, SR.wav and SL.wav

1) Convertion with neroAacEnc.exe
I use this command line for all wav file
neroAacEnc.exe -q 0.3 -lc -if input.wav -of output.mp4

and I find these size for each mp4 files
2ch done 2497 Ko and 114 Kbps
6h done 8764 Ko
FL done 1574 Ko
FR done 1569 Ko
C done 1567 Ko
LFE done 725 Ko
SL done 1481 Ko
SR done 1533 Ko

well it's really curious:
- same quality level for individual chanel done less bitrate than 6ch conversion (8.32 Mo vs 8.55 Mo)
- for same quality level bitrate for 6ch and 2ch sould be simple scalling something like between x2.0-x2.5 ... here it's 3.5


2) Convertion with FAAC.exe
I use this command line for all wav file
faac.exe input.wav -o output.mp4 --tns -q 100

and I find these size for each mp4 files
2ch done 2552 Ko and 116 Kbps
6h done 6559 Ko
FL done 1212 Ko
FR done 1211 Ko
C done 844 Ko
LFE done 2497 Ko
SL done 1233 Ko
SR done 1240 Ko

well here it's normal:
- same quality level for individual chanel done more bitrate than 6ch conversion (7.00 Mo vs 6.40 Mo)
- for same quality level bitrate for 6ch and 2ch sould be simple scaling something like between x2.0-x2.5 ... here it's 2.57

If quality level is the same for each encoding (1.0, 2.0, 5.1 ... etc) then bitrate for q 0.3 6ch file should be in 5000-6500 Ko interval. Chanel coupling seem work for 2.0 but not for 5.1. Bug for multichanel or different quality level ... ???


Well I try with the old nero aac version (aac.dll and aacenc32.dll) and I find that:

profil LC with quality "internet"

for quality "high" size scalling is x2.62
2ch done 2860 Ko
6ch done 7518 Ko

for quality "fast" size scalling is x3.32
2ch done 2713 Ko
6ch done 9022 Ko

for quality "high + pns" size scalling is x2.28
2ch done 2121 Ko
6ch done 4405 Ko

for quality "fast + pns" size scalling is x2.80
2ch done 1727 Ko
6ch done 4842 Ko

Perhaps an old bug for the "fast" preset ... I don't know. 5.1 Chanel coupling seem not work only in "fast" mode but perhaps simply that these quality mode are not equivalent ... ???

Excellent work, thank you very much.

The time matches perfectly as I just bought my Samsung D608. I can put a lot of MP4 music into it. Thanks.

@ M: If you use foobar, you can use the "Downmix Channels to Mono" DSP when converting. I have no idea if this is possible in EAC. But since this encoder uses joint-stereo (at least I'm quite sure it does), the gain in filesize will be rather small. Nevertheless, it's a mono-track, so I think it should be encoded mono.

But there is no point to it (encoding at 88.2kHz) and it seems to bring lower quality.




Yes, of course. SBR starts being pointless at about 85-90kbps.



I am sorry but this is complete bullshit. AAC's stereo system is not comparable to MP3, it makes no point to say that AAC is using M/S or L/R because it will consistently use both modes at the same time! And there's definitely an advantage in it even at extremely high bitrates.



Yeah, and now people are trying 2 pass VBR and 192kbps stereo HE-AAC. Uhm, I think even "pro users" can do amazingly silly things.



Whoever told you that had absolutely no idea what they were talking about. Nero Digital supports ReplayGain, and for example foobar2000 also does.





I'd recommend downmixing beforehand. But encoding it as stereo is fine too, because the encoder can and will figure out that it is really mono and encode it quite efficiently.




You will need to tag from an external application (for now). Switches will be added at a later time.

REACT v1.0 (just released) has support for this encoder.

It includes the encoder and a tagger app.
Three configs are available: MP4 (mp4 tracks), WV_MP4 (Wavpack image or tracks + mp4 tracks).

The WV_MP4-image.txt config is particular advanced, which creates a tagged Wavpack image, and then tagged mp4 tracks from the ripped image.

http://www.hydrogenaudio.org/forums/index....showtopic=36261

Enjoy.

OK but if me show some tests to prove me.

What about IS (Intesivity Stereo), i think that for example 96kbs sounds beter with LC and IS, from some earlier Ivan's tests.

You sorry me but for me is bigest important thing. I know what is answer can be and you devs always find some stupid reason. Leave MP3 and his old technology alone. I don't have problems with M/S AAC at all but M/S can disturb quality especialy in high bitrates and very silent songs. Me and many other ppl know that using M/S on very high rates only disturb quality, like L/R on very low rates. AAC tecnology is more smarter and knows what doing where to put M/S where L/R, but M/S is M/S, not always good.

I think about ReplyGain like MP3

That is you mistake, you should put some info in readme, And i am i ppl area who tested VBR and 2-pass because this have sense.

>>What about IS (Intesivity Stereo), i think that for example 96kbs sounds beter with LC and IS, from
>>some earlier Ivan's tests.
It seems that you simply do not understand what are you talking about.

>>Me and many other ppl know that using M/S on very high rates only disturb quality
Prove it. You are welcome to post ABX test results for any aac encoder you wish.

>>That is you mistake, you should put some info in readme, And i am i ppl area who tested VBR and 2-pass
>>because this have sense.
This have NO sense at all.
NOTE: The special keys are for professionals.
If you do not understand, what is vbr and what is two pass, simply do not use them.
EDIT: to help you not to make such a mistake in the future: what is vbr, cbr and abr

1. Where will you keep us informed on encoder updates? Here on validated news page? AAC forum?
2. I'm not into aac: Whats supposed to be the correct file extension? mp4 or m4a?
3. Will you support tagging via command line? (useful for eac)

I tried the latest neroAacEnc.exe with SE test files and got:
- Nero ABR (-br 128000) => 130.7 kbit/s
- Nero 2-Pass (-br 128000 -2pass) => 138.8 kbit/s
- Nero Quality VBR (-q 0.462) => 128.0 kbit/s

Confused a little bit with 2-pass bitrate. Could someone clear up?

In Nero AAC Recommended Settings topic Garf said that 2 pass is no sense with VBR.
Ivan said that 2 pass VBR can be usefull for long samples with speech, music etc. ( i.e. soundtracks)
?

VBR mode here is q mode. Garf want certainely say that -q X -2pass is no sense simply because -br X -2pass is already a VBR mode.
Ivan say that -br X -2pass is very good for very long source like movie simply because you use VBR with very variable complexity source with good target bitrate.

Very good encoder, wonderful gift, thank you very much, Nero devs.

I tried it on my 'standard problem samples' harp40_1, trumpet, and herding_calls.

-q 0.425 is enough to bring me to my not-at-all-annoying level.
Going lower qualitywise I tried -q 0.4, -q 0.37, and -q 0.31. Quality scales very well with these bad samples, and even -q 0.31 is quite acceptable.

If I will ever encode for my mobile phone again I guess I will use this encoder and a quality level of -q 0.37 or similar. As for this application I also tried -br 96000 -2pass and was a bit disappointed. Quality was pretty poor.
So it's very advisable to use the -q mode. It was aware of the samples being problematic and chose an average bitrate quite high compared to what is usually expected with these quality levels.

Excellent work, thanks again.

If I were allowed a wish and if it's not much work I'd welcome very much the possibility of outputting a raw aac file which I need for my mobile phone (Nokia 6230).
As for now I have to use mp4box, and I didn't manage to integrate it into foobar, and the foobar plugin I used with 0.8.3 for this purpose isn't available any more (at least I didn't find it).

Tagging *seems* to be handled through libmp4v2 - part of mpeg4ip. But then supplemental non-compliant information is appended. You can't call them atoms/boxes because they non-conforming to the ISO specification. It seems like the supplemental info was added to accommodate foobar2000 0.9.x similarly non-conforming (and redundant) tagging scheme.

From what I remember, these files won't stream, and the iPod Shuffle won't be able to play them either.

If you use EAC, REACT v1.0 will automatically tag the output files using AtomicParsley. There are configurations for either just creating MP4 track files, or creating additional Wavpack image or tracks in one go (WV_MP4 config).

The files play nicely with Winamp, and the tags shows up fine.

Note that AtomicParsley v0.8.0 win32 executable is included with REACT, as the latest v0.8.4 is buggy.