eac3to - audio conversion tool

[SamuriHL]

Is Nero 7 even compatible with W7 64? I've read a lot of reports that it no workie. I've never tried. I don't even know if I still have the email with my key kicking around. Is there a reason you're wanting to get it to work so badly?

[rica]

Normally all 32 bit SW should work on 64 bit OS.


Is there a reason you're wanting to get it to work so badly?

No reason; i just wonder :)

[SamuriHL]

Yea, I don't have my license for 7. I only have a license for 8. I never bothered with anything after that cause it became ridiculously bloated. I have no intention of installing Nero again. As for eac3to and TrueHD, I've had no problem at all with the built in libav code. Works perfectly.

[rica]

OK, thanks Samuri.

So, i think nautilus7 should revize the wiki.

[ACrowley]

Nero 7; what i've been speaking about which eac3to supposed to be used to use. With its HD patch.

Remember that theres a bigger Problem with the NeroAudio Decoder in eac3to! The DRC isnt removed properly so the Decoder is not recommended

[hubblec4]

No, I was meaning I need to sync audio and video, and the audio is 400ms shorter.

streching is possible in MKVmergeGUI.

you need the time of the video and the audio( audio= 100sek ; video 100,400sek) then you divide the videotime by the audiotime and you get the stretch-value.

[deado]

Is Nero 7 even compatible with W7 64? I've read a lot of reports that it no workie. I've never tried. I don't even know if I still have the email with my key kicking around. Is there a reason you're wanting to get it to work so badly?

Using Nero 7 with Win 7 64 here no problems.

[rebkell]

No, I was meaning I need to sync audio and video, and the audio is 400ms shorter.

I assume the audio is out of sync? If it drifts over the entire clip, you could just add some audio loops spaced equally over the entire audio track.

-edit=0:00:00,0ms loops or removes audio data at the specified runtime
-silence/-loop forces usage of silence (or looping) for audio edits


If it's 48Khz AC3 you could add 12 or 13 32ms edits spaced equally and get with 16ms.

[SamuriHL]

Using Nero 7 with Win 7 64 here no problems.

Well, I guess that's good. :) I could only find my key for Nero 8. Not that I care in either case. I'm perfectly content with the libav support for TrueHD in eac3to. And I use ImgBurn for all my burning needs. That's all I ever really used Nero for anyway...stable burning. Then they started bloating it up with all kinds of other useless stuff. Meh.

[stax76]

There is a improved dialog for Blu-ray demuxing in StaxRip:

http://i.imagehost.org/t/0562/Unbenannt_3.jpg (http://i.imagehost.org/view/0562/Unbenannt_3)

http://forum.doom9.org/showthread.php?p=1419618#post1419618

[rica]

Using Nero 7 with Win 7 64 here no problems.

Do you mean eac3to can use nero's forced decoder?



_ _ _ _ __

[Snowknight26]

How can I reproduce this? I've just tried "eac3to some.wav 1.aac 2.aac" and both AAC files are created just fine.

Was able to find a small enough sample to be able to reproduce the issue:

http://stfcc.org/misc/eac3to.aac.issue.zip

Run the bat, wait for eac3to to stop outputting. Might be a while, but it will. Either put the two files in the folder eac3to is in or modify the bat.

[tvjunky]

Hi Madshi!

Maybe this is a bug in eac3to:

I always rip my BRs directly with eac3to like this:
eac3to d: 1) -demux

This was functioning for 100% of my BRs the last years. With the new version 3.22 i get the following error on ~50% of my discs (example):

D:\_hd>eac3to y: 1)
Skipping 66240 bytes to work around damaged source file...
Hint: You can use the option "-fix" to clean up the damage.
The format of the source file could not be detected.

The parameter "-fix" does not help. With eac3to version 3.21 all BRs are ok and can be copied to hd.

[jdawn1]

the same reason why 24bit is converted to 16bit. Human hearing has limitations.

Word. Personally, I stick with as high as possible source sample rate and bit depth, and usually use lossy encoding. Much more efficient use of space, IMO. Especially when you can use something like AAC. I'd rather have a 5.1 96kHz/24bit (sourced) 96kHz AAC @ 650kbit/s then a 5.1 48Khz/16bit FLAC @ 3000kbit/sec. All that extra superfluous audio data is much better spent on the video. By superfluous, I would personally guarantee that no one here could spot the difference between a "high bitrate" AAC and a FLAC from the same source, on your choice of audio equipment. I am talking an AAC file 1/5 to 1/10th the size of the FLAC.

I am still amazed when I see movie encodes people make with 10GB file size: 5GB of video and 5GB of a DTS-HD-MA audio track. :rolleyes: Makes no sense whatsoever. Make it 9.2GB of video and 800MB of audio (AAC for PC playback or DTS/AC3 core if you need to "bitstream") and it's overall much better quality.

Digital: In the world of today, 192kHz audio should already be something quite trivial. :)

Analog: In the world of today, speakers and audio amplifiers that support up to 96kHz still are very rare, and (IMHO) somewhat pointless. :rolleyes:ss......


Yeah. If you mean by frequency response, 96KHz could represent 48Khz frequency audio. Given I can't hear much above 16KHz, yeah, speaker reproduction of 48kHz would be pretty stupid.

Currently, are there any speaker systems/receiver's that actually support 192Khz? Now THAT would be the 9th Wonder of the World.

Well, speaker support is irrelevant, they're analog. My cheapie receiver supported it just fine, even the original bitstream (DTS-HD-MA 5.1 192 / 24)
Funnily enough, it actually sounds better if I playback a version of it downsampled to 48KHz as my receiver supports Audyssey but can't process 5.1 streams over 48Khz. The audio benefit of running with Audyssey enabled far exceeded listening to the track at 96 or 192 KHz...
My digital copy of the movie is 96/24 audio encoded to AAC, which I have Windows downsample to 48khz (LCPM over HDMI) so the receiver can enable Audyssey.

OK... I'll be the first to ask... Why would you want to take such a beautiful audio track and throw away 2/3 of the data?

Because it is a pointless waste of space if it's to be stored on a PC. In my case, resampling from 192 to 96 and encoding to AAC (Nero, -q 0.5) resulted in "throwing away" something like 94% of the data, lol. 13.4 GB to less then 1 GB. Yet, you can't hear a damn bit of difference on a nice audio setup; sounds great. I'd dare you to even try.

It's still a beautiful audio track even if you remove a few samples or use a transparent lossy encode. How it SOUNDS is what's important, and what makes a great score. File size has, and should not have, in anyone's mind, anything to do with it.

Using eac3to, any way I can downconvert a ~25GB 192 Khz (!) LPCM track (Akira Blu-Ray) to something more human, like, say, 48 Khz?

Of course! Just look at the help. It's really easy. I own the blu-ray, and I make .MKV's for my media server. This is the only movie I had to re-sample. I re-sampled to 96KHz / 24bit and encoded to AAC with Nero, file went from 13+ GB (DTS-HD MA) to less then 1 GB and sounds awesome. :)

The movie went from taking most of a BD-50 to only about 13.4 GB using the original video and AAC audio. Nothing lost quality wise, lol.

I did keep a copy of that sucker on my server for testing purposes, though. My receiver seems to work just fine with it... a few years old and a cheapie, too. Onkyo 606. Kinda neat to see it say 5.1ch DTS-HD MA 192 khz (bitstreamed from a PC with ATI 5xxx HDMI card). My old PS3 plays it back without a hitch either. 5.1 ch 192 kHz PCM shows up on the receiver, not that I would be able to hear a difference even if it was not fully decoding or down/upsampling, lol.

[lintran]

Hello
I just tried extract DTS-ES Discrete 6.1 to wav(input.dts output.wavs) with Arcsoft Decoder (version 3.0.160) and it produced 7 individual WAV files, but there was something wrong with them. Too much high-frequency and noise.
Same occurs when I try to create an AC3 file.
So please tell me where do i wrong? Or how to do it proper?
Thank you so much.

[Atak_Snajpera]

Arcsoft decoder cannot properly decode 6.1 dts. I had the same problem.

[deathlord]

Arcsoft decoder cannot properly decode 6.1 dts. I had the same problem.

Sonic works well on these!

I re-sampled to 96KHz / 24bit and encoded to AAC with Nero

Why not 48KHz, 96 is a waste of diskspace, too!

[Atak_Snajpera]

Why not 48KHz, 96 is a waste of diskspace, too!
because people like in placebo effect ;)

[deathlord]

@ madshi,

Could you allow for longer -edits, in particular silence? At the moment the maximum possible (positive) edit seems to be 4999ms.

[longtom]

I want to extract from a BluRay the 2.0 PCM soundtrack to wav.
The output should be outputchapter1.wav outputchapter2.wav etc...
So that i have all songs in a extra wav files and not in one LARGE output.wav file!

[tebasuna51]

You can use the BeSplit command line, or with CUE_Splitter GUI, to split the full wav file.
For CUE_Splitter you need a .cue file, but maybe you can convert the Chapters file to a cue file easily with a text editor.

[longtom]

You can use the BeSplit command line, or with CUE_Splitter GUI, to split the full wav file.
For CUE_Splitter you need a .cue file, but maybe you can convert the Chapters file to a cue file easily with a text editor.

Great, Thanks!
I'll try with Cue_splitter!

[tebasuna51]

WARNING:
- When the language field of a track (in mkv) is empty this mean the language is the default: eng (from MkvMerge v4.0.0)
- eac3to don't manage audio tracks (MP2/MP3/DTS/AC3?) from mkv's generated with MkvMerge v4.1.0 defaults (header compression).
- eac3to crash (without messages) with video track H264 from mkv's generated with MkvMerge v4.2.0 defaults (header compression).

@madshi

Do you have any plan to support these new defaults?

[frank]

Test of mkvtoolnix 4.2.0 header removal compression
(deletes repeating header infos)

Remuxing of a ts Stream to mkv:
8.292.939 kB without header removal compression, like before
8.292.174 kB with Compression

= 0,00923 % difference!!

Wow, what a great saving! :devil: Forget that new default.
The mkvtoolnix developer bunkus should remove that annoying standard setup!!

First Aid:
Disable for every track the option Compression: to None.
Not empty.

[setarip_old]

@frank

Hi!Wow, what a great saving! Forget that new default.
The mkvtoolnix developer bunkus should remove that annoying standard setup!!If you read the MKVToolnix thread, you'll find that "bunkus" is adamant that since this is technically correct, it will remain as the far-from-obvious default setting...

[dansrfe]

Does eac3to do encodings to .mp3?

[Inspector.Gadget]

dansrfe - Not directly. But you can pipe output to the LAME executable. tebasuna51 has posted a command template several times around here, and I imagine you'll find it pretty easily searching for eac3to lame.exe or something similar,

[mr.duck]

Is it possible for eac3to demux a TS file and pipe it directly into mkvtoolnix?

If that's too hard I can just demux the files and then run mkvtoolnix when eac3to is done. Any recomendations?

[longtom]

Great, Thanks!
I'll try with Cue_splitter!

I've splitted it now with foobar!

[LeXXuz]

How do you convert AC3 2.0 Surround to another 2.0 Surround file and keep the surround information?

I'd like to make a speedup on an AC3 2.0 Surround f.e.:
eac3to input.ac3 output.ac3 -224 -speedup

However the output file is only AC3 2.0 (when reopening the output file, eac3to reports "AC3 2.0" and not "AC3 2.0 Surround" as for the input file) and my AVR does not show the Surround channel any more.

Is there any switch I need to use to keep the output file as 2.0 surround?

[tebasuna51]

How do you convert AC3 2.0 Surround to another 2.0 Surround file and keep the surround information?
You need use the external encoder Aften.exe.

Suggested command line:
eac3to input.ac3 stdout.wav -speedup | Aften -b 224 -m 0 -dsur 2 -pad 0 -readtoeof 1 -exps 32 -s 1 - output.ac3

[tebasuna51]

Is it possible for eac3to demux a TS file and pipe it directly into mkvtoolnix?
Not now, eac3to can only write the demuxed track files in hard disk.
If that's too hard I can just demux the files and then run mkvtoolnix when eac3to is done. Any recomendations?
You can try the GUI UsEac3to (http://forum.doom9.org/showthread.php?t=145574).

You can order extract the video to mkv, one or two audios to the desired format, and then push 'Run and MkvMux'.
After eac3to finish, the output files are sended to MkvMerge automatically.

[deathlord]

Is it possible for eac3to demux a TS file and pipe it directly into mkvtoolnix?

If that's too hard I can just demux the files and then run mkvtoolnix when eac3to is done. Any recomendations?

You can also let mkvmerge show you the command line. Then you can make a batch file with the eac3to command followed by the mkvmerge command.
This is very efficient for tv shows.

[LeXXuz]

You need use the external encoder Aften.exe.

Suggested command line:
eac3to input.ac3 stdout.wav -speedup | Aften -b 224 -m 0 -dsur 2 -pad 0 -readtoeof 1 -exps 32 -s 1 - output.ac3

Worked like a charm. Thanks a lot tebasuna51! :thanks:

[robpdotcom]

Hi,

I was searching this thread for problems with TrueHD in seamless branching discs, and read this:

It's not a good idea to extract THD tracks from seamless branching Blu-Rays. Audio and video will drift out of sync, because eac3to can't fix the audio overlaps with THD tracks, when demuxing them. You should convert to THD/WAVs directly from the original Blu-Ray structure. Then those "lossless check failed" warnings should go away and audio and video will stay in sync.

Does this mean that you can convert to THD? In other words, can eac3to fix overlaps if you extract as THD as opposed to THD+AC3? Or, must you convert to Flac or WAV?

[TinTime]

No, you can't convert to THD. I think it's just a typo. Possibly madshi meant to say "You should convert to FLAC/WAVs directly from the..."

[robpdotcom]

OK, thanks for clarifying.

What exactly is the difference between THD and THD+AC3, and why would you want to use one over the other?

[mr.duck]

You can also let mkvmerge show you the command line. Then you can make a batch file with the eac3to command followed by the mkvmerge command.
This is very efficient for tv shows.

If I understand you correctly this is how I have it set up at the moment. For TV shows too. Saves 10s of gigabytes with no transcoding! Performance and reliability are much better too, so pretty excellent IMO.


One recording I ran through eac3to and mkvmerge had chunk of audio or video missing because of some big spike of interference corrupting the data. So 1/2 way into the show, the audio and video went way out of sync by around a whole second.

So I notice eac3to says something like "... it may work, or not" as part of a warning message. For these files I want the batch file to abort and leave the original file as is rather than try and remux it to mkv with audio sync problems. So how can this be done? Maybe...
IF ERRORLEVEL = 1 (after eac3to is done) or...
eac3to.exe ..... | FIND "it may work, or not" <-- will that work doing it that way?

[TinTime]

Yes, try...

eac3to.exe commands -log="log.txt"
if %errorlevel% neq 0...
find /c /i "warning" "log.txt"
if %errorlevel%==0...

[Shamus76]

Hi guys,

I am having big problems encoding DTS. Example:

I:\KICKASS>eac3to bdmv 2) 3: Kick.Ass2.dts
M2TS, 1 video track, 1 audio track, 1:57:45, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: AC3, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
a03 The Nero decoder doesn't seem to work, will use libav instead.
a03 Extracting audio track number 3...
a03 Removing AC3 dialog normalization...
a03 Decoding with libav/ffmpeg...
a03 Reducing depth from 64 to 32 bits...
a03 Writing WAVs...
a03 Creating file "Kick.Ass2.L.wav"...
a03 Creating file "Kick.Ass2.R.wav"...
a03 Clipping detected, a 2nd pass will be necessary.
a03 Starting 2nd pass...
a03 Extracting audio track number 3...
a03 Removing AC3 dialog normalization...
a03 Decoding with libav/ffmpeg...
a03 Reducing depth from 64 to 32 bits...
a03 Writing WAVs...
a03 Applying -1,34dB gain...
a03 Creating file "Kick.Ass2.R.wav"...
a03 Creating file "Kick.Ass2.L.wav"...
Encoding DTS <768kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.29.0.
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...


I have tried encoding DTS like above on THD and AC3 tracks, same problem. Also tried to convert a THD-track to DTS, same problem.

I am running eac3to v3.22 and i have the exact same setup on another computer and there it works fine.

I have re-installed Surcode several times, no difference.

I read that surcode sometimes have path-length issues but i have run it on the root folder with same result.

[setarip_old]

@Shamus76

Hi! I am running eac3to v3.22 and i have the exact same setup on another computer and there it works fine. I have re-installed Surcode several times, no difference.

Does your purchase license of "Surcode" allow for it to run on more than one machine? If not, that could be your problem...

[Shamus76]

@Shamus76

Hi!

Does your purchase license of "Surcode" allow for it to run on more than one machine? If not, that could be your problem...

Yes, it should be allowed, any other ideas?

[Shamus76]

@Shamus76

Hi!

Does your purchase license of "Surcode" allow for it to run on more than one machine? If not, that could be your problem...

OMG, almost don't wanna say this but the problem is solved. I had forgotten to activate surcode :rolleyes:

:thanks:

[Lyle_JP]

How do you convert AC3 2.0 Surround to another 2.0 Surround file and keep the surround information

2 channel is 2 channel. The "surround information" is always there, out of phase between the left and right. If you mean the cute little surround light that pops on in your receiver's display, that's because Dolby Digital allows a surround "flag" to be added to a 2.0 stream. If you transcode to another format (AAC, MP3, etc.) you lose the flag, and those formats don't support such a surround flag AFAIK.

As long as you instruct your receiver to do surround processing on any stereo source, you won't notice a difference.

[mr.duck]

Yes, try...

eac3to.exe commands -log="log.txt"
if %errorlevel% neq 0...
find /c /i "warning" "log.txt"
if %errorlevel%==0...

Unfortunately I have lots of errors and warnings. Typically I see something like this 10-20 times per recording...
[v01] [0:01:25] The source file seems to be damaged (transport error indicated). <WARNING>
But the video still seems to be able to remux to mkv ok and should not be aborted.



I found out what was causing audio sync issues (I said it was missing data due to interference). For TV recordings, it turns out...

eac3to.exe ...... -demux <-- Catastrophically bad audio sync. Was mkvmerge's fault for throwing away video (some complaint about GOPs).
eac3to.exe ...... 1: blah_blah.mkv ...... <-- Perfect audio sync with same file. mkvmerge has no complaints. eac3to is great :D

Obviously I'm now going to switch to using mkv output in eac3to instead of -demux. But just to improve my error handling for problem files, does anyone know about a warning message in eac3to that reads something like "... it may work, or not"? I don't have any files that produce this warning and I'm not sure as I may be making all this up having remembered it wrong. I had a look on the wiki but didn't find anything. Anyone know what I'm on about?

[Shamus76]

Hi again guys,

Got another problem. I am trying to extract an LCPM track from a Bluray and then merge it with MKVmerge.

I have tried several times extracting the LCPM track to a PCM file using no manual settings and once using "-16 -48000" setting. Result is this:

I:\KILLBILL1_GBR>eac3to bdmv 1) 3: Kill.Bill.1.pcm
M2TS, 1 video track, 8 audio tracks, 9 subtitle tracks, 1:50:44, 24p /1.001
1: Chapters, 19 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 5.1 channels, 16 bits, 48kHz
4: AC3, English, 5.1 channels, 640kbps, 48kHz
5: DTS, Italian, 5.1 channels, 16 bits, 1509kbps, 48kHz
6: AC3, Italian, 5.1 channels, 640kbps, 48kHz
7: DTS, Spanish, 5.1 channels, 16 bits, 1509kbps, 48kHz
8: AC3, Spanish, 5.1 channels, 640kbps, 48kHz
9: DTS, German, 5.1 channels, 24 bits, 1509kbps, 48kHz
10: AC3, German, 5.1 channels, 640kbps, 48kHz
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), Italian
14: Subtitle (PGS), Spanish
15: Subtitle (PGS), German
16: Subtitle (PGS), English
17: Subtitle (PGS), Italian
18: Subtitle (PGS), Spanish
19: Subtitle (PGS), German
a03 Extracting audio track number 3...
a03 Reading RAW/PCM...
a03 Swapping endian...
a03 Remapping channels...
a03 Swapping endian...
a03 Remapping channels...
a03 Creating file "Kill.Bill.1.pcm"...
a03 The original audio track has a constant bit depth of 16 bits.
Video track 2 contains 159287 frames.
eac3to processing took 9 minutes, 56 seconds.
Done.

Then when i drag'n'drop the file to MKVmerge 4.20, it takes a while and then i get a File Identification-error saying:

"C:\Program Files\MKVtoolnix\mkvmerge.exe" --output-charset UTF-8 --identify-for-mmg "I:\KILLBILL1_GBR\Kill.Bill.1.pcm"

Output:

Error end-of-file

Error: File I:\KILLBILL1_GBR\Kill.Bill.1.pcm has unknown type.

I cannot see any errors in the extraction, anyone got any ideas?

:thanks:

[b66pak]

@Shamus76 why don't you extract to .flac, .w64 or .wav?
_

[Shamus76]

@Shamus76 why don't you extract to .flac, .w64 or .wav?
_

Well i thought the uncompressed pcm track loose information when converting to another format, perhaps that isn't so?

Btw, .w64, what is that format?

[b66pak]

you don't lose anything...flac is lossless (you will save space!) and .w64 or .wav is as .pcm but with headers...

.w64 is short for .wav with 64bits...valid .wav files can't be bigger than 4gb so .w64 is the solution...
_

[Shamus76]

you don't lose anything...flac is lossless (you will save space!) and .w64 or .wav is as .pcm but with headers...

.w64 is short for .wav with 64bits...valid .wav files can't be bigger than 4gb so .w64 is the solution...
_

Oh i see, didn't know about .w64. Thnx for the tip.

Unfortunately .w64 isn't recognized by MKVmerge either.

I'll try standard .wav or .flac.

Thnx for your help.