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

[Rockaria]

Originally Posted by dimzon

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]

Originally Posted by dimzon

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]

Originally Posted by 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

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]

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 |

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]

Originally Posted by 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

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]

Originally Posted by Inc

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?...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?...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/...t=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.

[Pasqui]

Originally Posted by 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!

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!