eac3to - audio conversion tool

[weust]

I have the Blu-ray set of all four Hunger Games movies, and the Monkingjay Part 2 has a TrueHD Atmos audio track.

For some reason eac3to can't show me those TrueHD tracks.
The English audio track is a TrueHD Atmos track, while the French is a DTS-HD MA 5.1 track.
I know this, because MediaInfo and MakeMKV both see the audio tracks.

What I see with eac3to 3.31:
eac3to d: 1)
M2TS, 1 video track, 3 audio tracks, 3 subtitle tracks, 0:00:29, 24p /1.001
1: Chapters, 16 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
4: AC3, French, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
5: AC3, English, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
6: Subtitle (PGS), Dutch
7: Subtitle (PGS), French
8: Subtitle (PGS), French

What MakeKMV shows me is in the screenshot.

Any idea what might cause this? And if there is anything I can do to help solve this by giving information, please let me know.

[Soulvomit]

Which switches or options do I use to properly convert a portion of a track? Say it's three hours long and I only want the second hour. I've read the manual but I think it's outdated because only one cut is being made and it's at the wrong start time with the options I use.

[tebasuna51]

12 last post moved to Splitting TrueHD stream into separate songs (http://forum.doom9.org/showthread.php?t=173484)

[Overdrive80]

Hi, using eac3to for transcoding to pipe to ffmpeg with this:"%rea%eac3to" %1 stdout.w64 -progressnumbers -speedup -normalize -down32 | "%rff%ffmpeg" -y -i - -c:a ac3 -b:a 640k "Japones.ac3"

I get this advice: [ac3 @ 0317e840] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead. What do it means? Thanks

[Motenai Yoda]

about the new loudnorm of ffmpeg, it can be integrated into eac3to?
http://k.ylo.ph/2016/04/04/loudnorm.html
IIRC the only normalization filter eac3to has is a 2 pass peak detection

[tebasuna51]

Seven off topic post's moved to Split ac3 in 5.1 and 2.0 (http://forum.doom9.org/showthread.php?t=173525)

[Music Fan]

I have a strange problem with eac3to ; it does not detect any audio (only video) in some of my HDTV recordings (h264 in TS). And it's not due to the format because those whose sound is detected come from the same TV channel than the others (it's always ac3).
It's maybe related to TSDoctor which I use to remove some garbage and unneeded audio tracks. But with some TS files that passed through TSDoctor, there is no problem, thus I don't know what's happening.
And it does not seem either related to the size of the files.
And I like to use eac3to to demultiplex audio and video of my HDTV recordings because it detects the correct delay (unlike MediaInfo and TSMuxer) and cuts the unneeded part of the sound at the beginning (because there is a negative delay on all my recordings).

[stax76]

here is a problem reported by a StaxRip user:

another issues, when I'm encoding audio from AC3 using default AAC VBR 112kbps profile, audio duration doubled up
Audio source http://pastebin.com/Bc1MS4ea
Encoded audio (duration doubled) http://pastebin.com/pCp7PG15
Log http://pastebin.com/TtfWBvg3
profile used (default) http://s33.postimg.org/4odvh2pxb/Capture.png

[tebasuna51]

here is a problem reported by a StaxRip user:
There are something wrong in this ac3 file, maybe a mix of 2.0 and 5.1 frames.

An ac3 2.0, 48 KHz, 56m 9s decoded to wav 64 bits have a size:

3369 s * 48000 * 64 * 2 / 8 = 2587392000 bytes = 2.4 GB

but eac3to say:

Caution: The WAV file is bigger than 4GB. <WARNING>

Use DelayCut to fix the ac3 and put here the log.

[stax76]

@tebasuna51

Thanks for examining it, I notified the user about your reply.

[IbrahimKh]

There are something wrong in this ac3 file, maybe a mix of 2.0 and 5.1 frames.

An ac3 2.0, 48 KHz, 56m 9s decoded to wav 64 bits have a size:

3369 s * 48000 * 64 * 2 / 8 = 2587392000 bytes = 2.4 GB

but eac3to say:

Caution: The WAV file is bigger than 4GB. <WARNING>

Use DelayCut to fix the ac3 and put here the log.

Yes it's mix of 2.0 and 5.1
thanks for reply

[stax76]

Is my list incomplete?

AC3
AC3 EX
AC3 Surround
DTS
DTS Express
DTS Hi-Res
DTS Master Audio
DTS-ES
E-AC3
E-AC3 Surround
RAW/PCM
TrueHD/AC3
TrueHD/AC3 (Atmos)

1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
4: DTS, English, 2.0 channels, 768kbps, 48kHz

[tebasuna51]

Is my list incomplete?

Nope, if is a list of audio from BD's.

But eac3to can decode other audio formats MP1, MP2, MP3, FLAC, MLP, AAC (with Nero 7 installed)...

[MonoS]

Hi, some times, when i try to convert some audio tracks, lossless and lossy, to ac3 i get distorted and inaudible audio.

The problem appear randomly and re-executing the conversion fix the problem without changing anything and, if this doesn't seems to work, executing the conversion on the single audio tracks seems to fix for good.
The only thing i'm sure about the state of the pc is that it's always under heavy load when it happens [4 cores at 100%], other recurring thing are that this happens mostly demuxing blurays or converting massively multiple tracks.

This is the sample of the result, it happens randomly so i can't offer a source sample: https://mega.nz/#!OlIDVRQZ!hKwCU-3Qo0G0BFDrsp2sYqaxSIhfggH1nVOMrGf1GK8

Thanks a lot for the attention, if i can provide further information let me know

[stax76]

Nope, if is a list of audio from BD's.

But eac3to can decode other audio formats MP1, MP2, MP3, FLAC, MLP, AAC (with Nero 7 installed)...

I have two lists with formats defined in StaxRip's source code regarding eac3to, BD codecs as posted and supported input file extensions:

ac3
dts
dtshd
dtshr
dtsma
eac3
evo
flac
m2ts
mlp
mp2
mpa
pcm
raw
thd
thd+ac3
ts
vob
wav

It's somehow important that my BD codecs list is complete because for a unknown codec StaxRip will:

throw a unhandled exception

prompt the user to send the log file

terminate

It happened last week for 'E-AC3 Surround', that's why I ask.

[tebasuna51]

...supported input file extensions:
You can add: mkv, mpls, mp1, mp3, w64 and rf64

It happened last week for 'E-AC3 Surround', that's why I ask.
I never see a E-AC3 Surround and don't know if can be a problem for eac3to.

BTW I can talk you about 'AC3 Surround':

eac3to show the qualifier 'Surround' based in a flag in AC3 header than say the user: this stereo audio have surround channels encoded in DPL style.

When a player, with DPL decoder inside, see that flag can do the extraction of surround channels.

But eac3to do nothing with that info and decode the AC3 like 2.0 because don't have a DPL decoder inside.

I only can supose than eac3to try to decode 'E-AC3 Surround' only like 2.0 ignoring the 'Surround' qualifier like with 'AC3 Surround'

[tebasuna51]

Hi, some times, when i try to convert some audio tracks, lossless and lossy, to ac3 i get distorted and inaudible audio.

Yes, your output is usseless and I have the same problem sometimes.

And don't know for what that happen.

[stax76]

@tebasuna51

Thanks for the info. :thanks:

[junior_l3oss]

I use Eac3to with nero without installing nero.
Eac3to only needs a couple files and registry settings
Nero Files:
- AdvrCntr2.dll
- NeAudio2.ax
- NeEacDec.dll

Copy these files in a folder of your choice. I have copied them to my eac3to folder.
Use the following command lines to register the dll's.
regsvr32.exe C:\tools\eac3to\Nero\NeAudio2.ax
regsvr32.exe C:\tools\eac3to\Nero\AdvrCntr2.dll

The following registry keys have to be added to the registry to register the nero plugin.

[HKEY_LOCAL_MACHINE\SOFTWARE\Ahead\Installation\Families\Nero 7\Info]
etc.

[HKEY_LOCAL_MACHINE\SOFTWARE\Ahead\Installation\Families\Plugins\Info]
etc.

That all you need to use the nero plugin with eac3to

i tried but i cant detect nero audio decoder.
i installed nero 7 too but not working. i tried to edit regedit...
and again i install nero 7 but there isnt neaudio2.ax
i found torrent files include it but not worked again.
my system is win 7 64bit

http://image.prntscr.com/image/7d5d1b7f33a3413d83a6b36ef7568a24.png

[filler56789]

i tried but i cant detect nero audio decoder.
i installed nero 7 too but not working. i tried to edit regedit...
and again i install nero 7 but there isnt neaudio2.ax

<snip>

Does this work for you?

http://forum.doom9.org/showthread.php?p=1398214#post1398214

[tebasuna51]

I have Nero-7.11.10.0_europe_lite installed in W7 64 bits, and the registry keys:

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Ahead\Installation\Families\Nero 7\Info]
etc.

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Ahead\Installation\Families\Plugins\Info]
etc.

BTW the Nero7 decoder is only needed now by eac3to to decode AAC standalone files, and you have other tools to do that job: qaac, ffmpeg, faad...

If the AAC is in mkv container you can use eac3to to extract the .aac

[Music Fan]

If the AAC is in mkv container you can use eac3to to extract the .aac
Is the result different than when done with MKVExtract ?

[tebasuna51]

Is the result different than when done with MKVExtract ?

Is the same, in my tests.

[Music Fan]

Ok thanks.

[Ripman]

I downloaded 352.8khz wav files from HDtracks. I wanted to convert the wavs to 176400 so I can listen on my laptop also.

HDtracks uses a custom field in their wav files to store artwork. This field can sometimes contain many MBs of data.

I used the following command line. It all seems to work nicely. But each converted track has some pretty evil static at the very end.

Eac3to 352.wav 176.wav -resampleTo176400 -0.1dB

What should I change on the command line to get my down sampling to work properly?

[manolito]

Sorry if this has been asked before, I use eac3to only occasionally...

I need to normalize a clip, but not to 0dB, but to a lower value like -2dB or 97%. Is this possible with eac3to? The Wiki only mentions the -normalize parameter which normalizes to 0dB.


Cheers
manolito

[Overdrive80]

You could use -normalize for seeing ganancy that eac3to applies and after you can use +-0.0db parameters, are two passes.

[manolito]

Sorry this will not do it for me...
I use this audio conversion in StaxRip, everything should be automatic without any user intervention.

Mostly I will convert AAC audio to AC3 audio. When converting to a lossy compressed format it is never a good idea to normalize to 0dB, I mostly leave a headroom of 2dB.

The problem is that StaxRip cannot use BeSweet for AC3 audio as the target format (this would enable normalizing to arbitrary values), it only can use eac3to.


Cheers
manolito

[tebasuna51]

I downloaded 352.8khz wav files from HDtracks...
Nobody can help you because the forum rule 6 :

6) No warez, cracks, serials or illegally obtained copyrighted content! Links to content of a questionable nature (e.g. anything you don't own and/or have downloaded), asking for, offering, or asking for help/helping to process such content in any way or form is not tolerated.

Please read the forum rules.

[tebasuna51]

Sorry this will not do it for me...
I use this audio conversion in StaxRip, everything should be automatic without any user intervention...

Then we can't help you with that.

Ask in StaxRip thread or use MeGUI (or BeHappy) than allow this operation.

[Music Fan]

Sorry this will not do it for me...
I use this audio conversion in StaxRip, everything should be automatic without any user intervention.

Mostly I will convert AAC audio to AC3 audio. When converting to a lossy compressed format it is never a good idea to normalize to 0dB, I mostly leave a headroom of 2dB.

The problem is that StaxRip cannot use BeSweet for AC3 audio as the target format (this would enable normalizing to arbitrary values), it only can use eac3to.
You can do it with a free audio editor as Adobe Audition (that was earlier Cool Edit Pro).

[Groucho2004]

Nobody can help you because the forum rule 6
HDTracks is a legitimate site for music downloads.

[Ripman]

Nobody can help you because the forum rule 6 :

6) No warez, cracks, serials or illegally obtained copyrighted content! Links to content of a questionable nature (e.g. anything you don't own and/or have downloaded), asking for, offering, or asking for help/helping to process such content in any way or form is not tolerated.

Please read the forum rules.

HDtracks.com is a legitimate commercial site for hi-Rez downloads.

Anyway, they have about 80 or so new releases that are 352.8khz, and I'm trying to figure out if I can down sample these. I'll post a mediainfo report when I get in a little later.

[Ripman]

I downloaded 352.8khz wav files from HDtracks. I wanted to convert the wavs to 176400 so I can listen on my laptop also.

HDtracks uses a custom field in their wav files to store artwork. This field can sometimes contain many MBs of data.

I used the following command line. It all seems to work nicely. But each converted track has some pretty evil static at the very end.

Eac3to 352.wav 176.wav -resampleTo176400 -0.1dB

What should I change on the command line to get my down sampling to work properly?

As my op describes, I used the following command line with a 352khz wav file as input:
Eac3to 352.wav 176.wav -resampleTo176400 -0.1dB

The mediainfo report seemed normal for the down sampled wav, but vlc reported it as a 32bit file.

So I used the following command line:
Eac3to 352.wav 176.wav -little -24 -resampleTo176400 -0.1dB

This also resulted in a down sampled file with static at the very end. Tbh, it seems to occur only with files that have a gain applied via the "#dB" command line option.

I appreciate any ideas on this one. Thanks.

[Groucho2004]

As my op describes, I used the following command line with a 352khz wav file as input:
Eac3to 352.wav 176.wav -resampleTo176400 -0.1dB
Use SoX for re-sampling.

[tebasuna51]

HDtracks.com is a legitimate commercial site for hi-Rez downloads.

Sorry.

Eac3to 352.wav 176.wav -little -24 -resampleTo176400 -0.1dB

Please post the log file to see how eac3to recognize this wav.

But each converted track has some pretty evil static at the very end.

Maybe the wav's have extrachunks at end of files not recognized by eac3to, maybe some metadata not compliant with wav spec.

EDIT: without problems in my test:
command line: eac3to.exe 352.wav 176.wav -resampleTo176400 -0.1dB
------------------------------------------------------------------------------
WAV, 2.0 channels, 0:00:20, 16 bits, 11290kbps, 352.8kHz
Reading WAV...
Resampling to 176.4kHz...
Reducing depth from 64 to 24 bits...
Writing WAV...
Applying -0.1dB gain...
Creating file "176.wav"...
The original audio track has a constant bit depth of 16 bits.
The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 2 seconds.
Done.

The resampling process is done at 64 bits, even the source is 16 bits, but by default the output is 24 bits.
Without static at end of file.

[stax76]

Sorry this will not do it for me...
I use this audio conversion in StaxRip, everything should be automatic without any user intervention.

Mostly I will convert AAC audio to AC3 audio. When converting to a lossy compressed format it is never a good idea to normalize to 0dB, I mostly leave a headroom of 2dB.

The problem is that StaxRip cannot use BeSweet for AC3 audio as the target format (this would enable normalizing to arbitrary values), it only can use eac3to.

What can be done is creating an audio profile based on the batch audio encoder instead of the GUI audio encoder.

it's created like so:

audio profiles menu > edit profiles > add > command line

and could look like so:

https://s31.postimg.org/q17st63az/aaa.png

[Ripman]

Sorry.



Please post the log file to see how eac3to recognize this wav.



Maybe the wav's have extrachunks at end of files not recognized by eac3to, maybe some metadata not compliant with wav spec.

EDIT: without problems in my test:


The resampling process is done at 64 bits, even the source is 16 bits, but by default the output is 24 bits.
Without static at end of file.

Thanks for your response buddy. Here is a log that I generated. I definitely looks like there is 24bit of audio data. Are you suggesting that I should forgo 24bit processing in favor of 16bit? (As I mentioned above, it seems like I get the "static" issue when I apply a gain.) I'm going to try SoX also.


eac3to v3.29
command line: eac3to "04-Violin Concerto, _The Red Violin__ III. Andante flautando.wav" "..\24-176.4-2\04-Violin Concerto, _The Red Violin__ III. Andante flautando.wav" -little -24 -resampleTo176400 -3.45dB
------------------------------------------------------------------------------
WAV, 2.0 channels, 0:06:29, 24 bits, 16934kbps, 352.8kHz
Reading WAV...
Resampling to 176.4kHz...
Reducing depth from 64 to 24 bits...
Writing WAV...
Applying -3.45dB gain...
Creating file "..\24-176.4-2\04-Violin Concerto, _The Red Violin__ III. Andante flautando.wav"...
The original audio track has a constant bit depth of 24 bits.
The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 1 minute, 11 seconds.
Done.


Here is a mediainfo -f report for these files. (I removed the custom artwork tag.)

General
Count : 325
Count of stream of this kind : 1
Kind of stream : General
Kind of stream : General
Stream identifier : 0
Count of audio streams : 1
Audio_Format_List : PCM
Audio_Format_WithHint_List : PCM
Audio codecs : PCM
Complete name : C:\52\24-352.8-2\01-Phantasmagoria - Suite from The Ghosts of Versailles.wav
Folder name : C:\52\24-352.8-2
File name : 01-Phantasmagoria - Suite from The Ghosts of Versailles
File extension : wav
Format : Wave
Format : Wave
Format/Extensions usually used : wav
Commercial name : Wave
Internet media type : audio/vnd.wave
Codec : Wave
Codec : Wave
Codec/Extensions usually used : wav
File size : 2814506312
File size : 2.62 GiB
File size : 3 GiB
File size : 2.6 GiB
File size : 2.62 GiB
File size : 2.621 GiB
Duration : 1329205
Duration : 22mn 9s
Duration : 22mn 9s 205ms
Duration : 22mn 9s
Duration : 00:22:09.205
Duration : 00:22:09.205
Overall bit rate mode : CBR
Overall bit rate mode : Constant
Overall bit rate : 16939486
Overall bit rate : 16.9 Mbps
Stream size : 844934
Stream size : 825 KiB (0%)
Stream size : 825 KiB
Stream size : 825 KiB
Stream size : 825 KiB
Stream size : 825.1 KiB
Stream size : 825 KiB (0%)
Proportion of this stream : 0.00030
Title : Phantasmagoria - Suite from The Ghosts of Versailles
Album : Corigliano: Violin Concerto, "The Red Violin" - Phantasmagoria
Album/Performer : JoAnn Falletta
Track name : Phantasmagoria - Suite from The Ghosts of Versailles
Track name/Position : 01
Performer : Buffalo Philharmonic Orchestra
Composer : John Corigliano, Jr.
Genre : Classical Music, Orchestral
Recorded date : 2015
File creation date : UTC 2016-05-31 07:39:40.980
File creation date (local) : 2016-05-31 03:39:40.980
File last modification date : UTC 2016-05-31 07:42:47.380
File last modification date (local) : 2016-05-31 03:42:47.380
Cover : Yes
Cover description : Picture
Cover type : Cover (front)
Cover MIME : image/jpeg
Album Artist : JoAnn Falletta
Tool Name : HDtracks Downloader
Tool Version : 20.0.32

Audio
Count : 272
Count of stream of this kind : 1
Kind of stream : Audio
Kind of stream : Audio
Stream identifier : 0
Format : PCM
Commercial name : PCM
Format settings : Little / Signed
Format settings, Endianness : Little
Format settings, Sign : Signed
Codec ID : 1
Codec ID/Url : http://www.microsoft.com/windows/
Codec : PCM
Codec : PCM
Codec/Family : PCM
Codec/Info : Microsoft PCM
Codec/Url : http://www.microsoft.com/windows/
Codec/CC : 1
Codec settings : Little / Signed
Codec settings, Endianness : Little
Codec settings, Sign : Signed
Duration : 1329205
Duration : 22mn 9s
Duration : 22mn 9s 205ms
Duration : 22mn 9s
Duration : 00:22:09.205
Duration : 00:22:09.205
Bit rate mode : CBR
Bit rate mode : Constant
Bit rate : 16934400
Bit rate : 16.9 Mbps
Channel(s) : 2
Channel(s) : 2 channels
Sampling rate : 352800
Sampling rate : 352.8 KHz
Samples count : 468943520
Resolution : 24
Resolution : 24 bits
Bit depth : 24
Bit depth : 24 bits
Stream size : 2813661378
Stream size : 2.62 GiB (100%)
Stream size : 3 GiB
Stream size : 2.6 GiB
Stream size : 2.62 GiB
Stream size : 2.620 GiB
Stream size : 2.62 GiB (100%)
Proportion of this stream : 0.99970


Here is an audacity screen shot of the "noise" at the end of an eac3to 352-176 down sampled wav -- it's about 1/2 second. Again, the issue occurs when applying a gain. The source 352 wav is clean in audacity.
https://www.dropbox.com/s/ob7781uqvdy61pi/eac3to_screenshot.jpg?dl=0

[manolito]

What can be done is creating an audio profile based on the batch audio encoder instead of the GUI audio encoder.


Thanks Stax,
I already did exactly this...

Obviously there is no way in StaxRip to use BeSweet for AC3 audio (except using the commandline profile). This is too bad because I think that for AC3 target format BeSweet is superior to eac3to. If you use the latest bsn.dll and aften.exe by KurtNoise BeSweet is much more versatile than eac3to.

Thanks and cheers
manolito

[stax76]

Obviously there is no way in StaxRip to use BeSweet for AC3 audio (except using the commandline profile). This is too bad because I think that for AC3 target format BeSweet is superior to eac3to. If you use the latest bsn.dll and aften.exe by KurtNoise BeSweet is much more versatile than eac3to.

do you have a link to the latest bsn.dll and aften.exe?

[manolito]

The original Kurtnoise free.fr site seems to be down. I uploaded the two files here:
http://www13.zippyshare.com/v/SkBAb6Av/file.html

For Aften I believe that the latest Wisodev builds also work, but the safest bet is to use the latest Kurtnoise builds.


Cheers
manolito

[Ripman]

Use SoX for re-sampling.

Thanks. I tried SoX 14.4.2 and it works just fine. Here is the sox command line I used to replicate the one I used with eac3to.

sox -V4 352.wav --rate 176400 176.wav gain -3.5 2>sox_log.txt

[tebasuna51]

Obviously there is no way in StaxRip to use BeSweet for AC3 audio (except using the commandline profile). This is too bad because I think that for AC3 target format BeSweet is superior to eac3to. If you use the latest bsn.dll and aften.exe by KurtNoise BeSweet is much more versatile than eac3to.

No way to use BeSweet for that job.

1) BeSweet can't decode AAC audio

2) BeSweet can't manage wav's (the decoded AAC) bigger than 2GB.
A track 5.1 from a movie is always bigger than 2 GB.

3) The encoded AC3 is the same, not superior, to eac3to output because both use Aften.exe like encoder (using the last bsn.dll in BeSweet).

The solution proposed by Stax76 must work.

I know than eac3to can be improved with this behaviour because if you try directly:

eac3to input.aac output.ac3 -normalize -2dB

first do the -2dB gain an after normalize, then the first operation is useless.
If first normalize and after apply -2dB, or better if take the value -2dB to limit to normalize, the process can be solved with only one pass.

[tebasuna51]

Are you suggesting that I should forgo 24bit processing in favor of 16bit?
Nope, was just a sample than show how eac3to manage the resampling.

Here:
...
WAV, 2.0 channels, 0:06:29, 24 bits, 16934kbps, 352.8kHz
Reading WAV...
Resampling to 176.4kHz...
Reducing depth from 64 to 24 bits...
Writing WAV...
Applying -3.45dB gain...

Here is a mediainfo -f report for these files.
...
Cover : Yes
Cover description : Picture
Cover type : Cover (front)
Cover MIME : image/jpeg
Album Artist : JoAnn Falletta
Tool Name : HDtracks Downloader
Tool Version : 20.0.32

Audio
Duration : 1329205 ms
Channel(s) : 2 channels
Sampling rate : 352800
Bit depth : 24 bits
Stream size : 2813661378


Here is an audacity screen shot of the "noise" at the end -- it's about 1/2 second.

The included Artwork tag seems the problem, the WAV structure is not designed to support that metadata. Without a sample I can't know if is a violation of WAV specs or a eac3to bug.

At least eac3to don't support that (incorrect duration calculated) and artwork data considered like audio data (last noise).

Audacity show noise from original file or from the eac3to converted?
Check if Audacity support the Artwork metadata or show also noise from original wav.

I tried SoX 14.4.2 and it works just fine.

Then problem solved.

[LigH]

I am curious about such a WAV file too, I wrote a RIFF header analyzing tool long ago (originally MS-DOS based, rebuilt for Win32 with Lazarus) and wonder which RIFF chunks it would report (the "data" chunk is not always "the whole rest of the file after the header")... Downloading Gigabytes to obtain a sample is insane, though. Possibly.

[Ripman]

Thanks for he responses. I don't think the custom artwork field is the problem - the source 352 wav files play fine through my gear, and there is no static at the end that can be heard or seen with audacity. (I put a Dropbox link to a screen shot from audacity in my prior post.) I have previously processed wavs with embedded artwork from hdt in the 44khz-192khz range without issue using eac3to and a gain command line argument.

The problem only occurs when a gain is applied via the #dB command line option. I wonder if it isn't caused by expanding to 32bits to apply gain.

I have a 352khz file that's about 650mb. I'll upload the whole thing when I get in later so people can experiment.

[manolito]

eac3to input.aac output.ac3 -normalize -2dB

first do the -2dB gain an after normalize, then the first operation is useless.
If first normalize and after apply -2dB, or better if take the value -2dB to limit to normalize, the process can be solved with only one pass.

Sorry I am not sure if I understand you correctly...

I did try your command line with "-normalize -2dB", and the result is that eac3to first applies a -2db gain decrease and afterwards normalizes to 0dB again. Here is the log:
eac3to v3.31
command line: "E:\Programme\StaxRip\Applications\eac3to\eac3to.exe" "I:\test temp files\test ID1 - iv-Undetermined 18ms.wav" "I:\test temp files\test ID1 - iv-Undetermined 18ms_Output.ac3" -448 -normalize -2dB -down16 -progressnumbers

------------------------------------------------------------------------------
WAV, 5.1 channels, 0:05:00, 16 bits, 4608kbps, 48kHz
Reading WAV...
Reducing depth from 64 to 16 bits...
Writing WAV...
Applying -2dB gain...
Creating file "I:\test temp files\test ID1 - iv-Undetermined 18ms_Output.ac3.pass1.wav"...
The original audio track has a constant bit depth of 16 bits.
The processed audio track has a constant bit depth of 16 bits.
Starting 2nd pass...
Reading WAV...
Reducing depth from 64 to 16 bits...
Remapping channels...
Encoding AC3 <448kbps> with libAften...
Applying 3.8dB gain...
Creating file "I:\test temp files\test ID1 - iv-Undetermined 18ms_Output.ac3"...
The processed audio track has a constant bit depth of 16 bits.
eac3to processing took 2 minutes, 2 seconds.
Done.

I want to limit the normalize gain to a max peak value of -2dB, how can I achieve this with eac3to?

Another question about the command line generated by StaxRip:
Is the "down16" parameter meaningful? Does libaften cause problems with an input which has a higher bit depth?


Cheers
manolito

[tebasuna51]

I am curious about such a WAV file too, I wrote a RIFF header analyzing tool long ago (originally MS-DOS based, rebuilt for Win32 with Lazarus) and wonder which RIFF chunks it would report (the "data" chunk is not always "the whole rest of the file after the header")...

I test, with eac3to, wav files with extra chunks after the data chunk without problems. For instance cue points created with GoldWave editor.

But, maybe, the eac3to behaviour can change with wav files greater than 2 GB, like here. Is know than there are soft than don't support this limit.

I can't understand how eac3to show a duration of 0:06:29 when seems (by size, channels, bitdepth and samplerate) the correct duration is 00:22:09.205 like show MediaInfo.

[tebasuna51]

Sorry I am not sure if I understand you correctly...

I want to limit the normalize gain to a max peak value of -2dB, how can I achieve this with eac3to?

Yes, you need two pass, don't work with only one pass.

For that stax76 solution:

eac3to "%input%" "%output%.flac" -normalize -progressnumbers
eac3to "%output%.flac" "%output%" -2dB -progressnumbers

(the bitrate is not needed, 640 Kb/s for 5.1, 448 Kb/s for 2.0)

Another question about the command line generated by StaxRip:
Is the "down16" parameter meaningful? Does libaften cause problems with an input which has a higher bit depth?


I don't see when "down16" is used. Is not needed for libaften, like in previous solution than use the default bitdepth 24.

[manolito]

eac3to "%input%" "%output%.flac" -normalize -progressnumbers
eac3to "%output%.flac" "%output%" -2dB -progressnumbers


Thanks, but would this intermediate flac file not have clipping? Isn't it necessary to use a 32-bit float wav file as the intermediate file?


Cheers
manolito