eac3to - audio conversion tool

[asarian]

Is there already a solution to the below error with TrueHD? This is from the Gantz Blu-Ray.


eac3to 00000.m2ts 3: c:\video\temp.pcm

M2TS, 1 video track, 1 audio track, 2 subtitle tracks, 2:10:40, 24p /1.001
1: Chapters, 17 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, Japanese, 5.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
4: Subtitle (PGS), Japanese
a03 Extracting audio track number 3...
a03 Extracting TrueHD stream...
a03 Decoding with libav/ffmpeg...
a03 Swapping endian...
a03 Remapping channels...
a03 libav Substream 0 parity check failed
a03 libav Substream 0 checksum failed
a03 libav Substream 0 length mismatch.
a03 The libav decoder reported error -1 while decoding.
Aborted at file position 1048576.

[Thunderbolt8]

try the -nero switch (nero is limited to 5.1 channels, but this is not a problem here)

[asarian]

try the -nero switch (nero is limited to 5.1 channels, but this is not a problem here)

Thanks. I don't have Nero 7 any more, though, since my migration to Windows 7. I'll contact their support desk.

[Rodeo]

No matter the source/decoder or encoder, when you downmix 6.1 to 5.1 with eac3to need use:

-0,1,2,3,5,6,4 -down6

First of all, thanks for figuring this out!

I have a few questions (mostly just to make sure I understand it correctly).

Assuming I have an MKV source with a DTS-HD Master Audio 6.1 track:

1: h264/AVC, 1920x1080 23.976p
2: DTS Master Audio, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz)

Let's say I want to convert it to FLAC 5.1. I can do it in a single step, like this:

eac3to.exe input.mkv 2: output.flac -0,1,2,3,5,6,4 -down6

Right? And this should give me a good downmix?

And if I also want a stereo downmix, I should do this:

eac3to.exe input.mkv 2: output_stereo.flac -0,1,2,3,5,6,4 -down2

or maybe this:

eac3to.exe input.mkv 2: output_stereo.flac -0,1,2,3,5,6,4 -down6 -down2

but not this:

eac3to.exe input.mkv 2: output_stereo.flac -down2

Right?

Finally, if I reduce the bit depth from 24- to 16-bit, I get the following in my eac3to log:

eac3to v3.24
command line: eac3to.exe input.mkv output.flac -0,1,2,3,5,6,4 -down6 -down16
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 0:06:15, 24p /1.001
1: h264/AVC, 1920x1080 23.976p
2: DTS Master Audio, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz)
Track 2 is used for destination file output.flac.
[a02] Extracting audio track number 2...
[a02] Remapping channels...
[a02] Decoding with ArcSoft DTS Decoder...
[a02] Remapping channels...
[a02] Mixing surround channels...
[a02] Reducing depth from 24 to 16 bits...
[a02] Encoding FLAC with libFlac...
[a02] Creating file output.flac...
[a02] The original audio track has a constant bit depth of 24 bits.
[a02] Processed audio track, L+R+C+SL+SR: constant bit depth of 16 bits.
[a02] Processed audio track, LFE: no audio data.
Video track 1 contains 8997 frames.
eac3to processing took 54 seconds.
Done.

I'm worried about this:

[a02] Processed audio track, LFE: no audio data.

It's not there if I keep the original bit depth:

eac3to v3.24
command line: eac3to.exe input.mkv output.flac -0,1,2,3,5,6,4 -down6
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 0:06:15, 24p /1.001
1: h264/AVC, 1920x1080 23.976p
2: DTS Master Audio, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz)
Track 2 is used for destination file output.flac.
[a02] Extracting audio track number 2...
[a02] Remapping channels...
[a02] Decoding with ArcSoft DTS Decoder...
[a02] Remapping channels...
[a02] Mixing surround channels...
[a02] Encoding FLAC with libFlac...
[a02] Creating file output.flac...
[a02] The original audio track has a constant bit depth of 24 bits.
[a02] The processed audio track has a constant bit depth of 24 bits.
Video track 1 contains 8997 frames.
eac3to processing took 1 minute, 8 seconds.
Done.


Is it a problem or should I just ignore it?

Note: it's only a 6-minute sample - I don't see the message if I convert the 2-hour source to FLAC, even with -down16.
My guess/hope is that maybe the LFE is indeed empty in the sample, but for some reason eac3to doesn't mention it if I don't specify -down16.

[tebasuna51]

- Yes

- Use this one, the -normalize is recommended when use -down2 :

eac3to.exe input.mkv 2: output_stereo.flac -0,1,2,3,5,6,4 -down2 -normalize

- Don't worry about the LFE message, appears only when the audio is analyzed to change the bitdepth.

[Rodeo]

Awesome! Thanks.

[mzso]

That's write. The checkactivate.dll was installed with previous version of TMT. The checkactivate.dll is only needed for
eac3to, TMT 3 doesn't need it to function. Eac3to check's if this file is installed, if not then eac3to will not use TMT.
Get hold of an older version of TMT (I think version 2) and extract the checkactivate.dll

That sucks. It wasn't updated since TMT2?

By the way why does eac3to automatically encode the dts core to flac if it fails to use the arcsoft decoder? Isn't that like universally condemned?

[mzso]

1) Copy following files from ArcSoft Total Media Theater 2 to Windows\system32

ASAudioHD.ax
dtsdecoderdll.dll
MagCore.dll
MagPCMac.dll
MagUIEngine.dll
MagUIInter.dll

put checkactivate.dll in \system32 folder as well as eac3to folder

2) Register ASAudioHD.ax -

regsvr32.exe ASAudioHD.ax

Played around with it a bit. It seems that it works with TMT5 files too. And it turns out only the following are needed:
checkactivate.dll
ASAudioHD.ax
dtsdecoderdll.dll
MagPCMac.dll
MagUIEngine.dll
MagUIInter.dll
Unfortunately though the checkactivate.dll isn't in TMT5 anymore so you still have to get that separately.

[rapscallion]

I'm extracting to wavs, a Dolby True HD 7.1 track, using eac3to_more and libav/ffmeg TrueHD decoder.

Does libav/ffmpeg remove dialogue normalization by default ?

When I extrac 5.1 DTHD, I use the Nero decoder and notice that the message always says " removing DN" .
No such message when using libav/ffmpeg.

Edit: a quote from ACrowley, post #9353 : "Im wondering that i cant see the "removing Dialog Normalization" Message anymore when decoding A(E)C3,TrueHD ?
I think because the DialNom removal is a standard Operation and latest eac3to wont output that Message everytime? " There never was an answer, altough one responder stated he saw the msg when extracting to ac3.

[jruggle]

I'm extracting to wavs, a Dolby True HD 7.1 track, using eac3to_more and libav/ffmeg TrueHD decoder.

Does libav/ffmpeg remove dialogue normalization by default ?

short answer, yes. libav does not apply the dialogue normalization to the decoded output. i also suspect eac3to does not have its own mechanism to extract/apply it.

[kws53]

Demuxed a 96kbps, 24 bit THD stream using eac3to. When converting to DTS, Surcode [1.0.29.0] has indicated that it can't handle the sampling rate.

I used the following to try to convert the THD stream to 48kbps:

>eac3to "00000 - 4 - TrueHD+AC3, English, 5.1 Channels, 96kHz.thd+ac3" "dion.thd+ac3" -resampleTo48000

TrueHD/AC3, 5.1 channels, 96kHz
<embedded: AC3, 5.1 channels, 448kbps, 48kHz>
Creating file "dion.thd+ac3"...

The conversion took 7minutes and essentially copied over the file. No sampling rate conversion was done. The file is still at 96kbps.

Any ideas or help?

Thanks,
Kurt

[Thunderbolt8]

is the file just named truehd+ac3 or is it really truehd+ac3? afaik eac3to can only demux single thd or single ac3? or has that been changed?

[nevcairiel]

You can't resample while still keeping it TrueHD. You would need to make it output WAV or FLAC, then it should be able to resample.

[kws53]

is the file just named truehd+ac3 or is it really truehd+ac3? afaik eac3to can only demux single thd or single ac3? or has that been changed?

This is the name that eac3to gave the file - it is a THD file with an embedded AC3 "core". I've run a separate conversion to AC3 and to FLAC without any problems, so the file format is supported by eac3to.

And according to the WiKi, I should be able to change sampling rate on any audio format that eac3to supports.

My objective is to get it into DTS format. I do not want to lose the fidelity by converting to AC3 first.

Kurt

[kws53]

You can't resample while still keeping it TrueHD. You would need to make it output WAV or FLAC, then it should be able to resample.

So I can convert to FLAC [already done], then resample down to 48kHz and then use the Surcode codec to convert to DTS?

Sounds like a plan if this is true.

Kurt

[kws53]

Played around with it a bit. It seems that it works with TMT5 files too. And it turns out only the following are needed:
checkactivate.dll
ASAudioHD.ax
dtsdecoderdll.dll
MagPCMac.dll
MagUIEngine.dll
MagUIInter.dll
Unfortunately though the checkactivate.dll isn't in TMT5 anymore so you still have to get that separately.

How do you "deregister" an older version of codec? I'm trying to upgrade my 1.0.29.0 version to 1.1.10.0 and the original keeps coming back...

I've replaced all of the appropriate files with new versions and run regsvr, but to no avail. I had NO problem installing to a virgin system. By the way, you do not need to put anything in the system32 folder - this is incorrect information.

Kurt

[kws53]

How do you "deregister" an older version of codec? I'm trying to upgrade my 1.0.29.0 version to 1.1.10.0 and the original keeps coming back...

I've replaced all of the appropriate files with new versions and run regsvr, but to no avail. I had NO problem installing to a virgin system. By the way, you do not need to put anything in the system32 folder - this is incorrect information.

Kurt

Sorry - I'm referring to the Surcode DTS encoder.

Kurt

[rapscallion]

short answer, yes. libav does not apply the dialogue normalization to the decoded output. i also suspect eac3to does not have its own mechanism to extract/apply it.
Thanks !

[nibus]

How do you "deregister" an older version of codec? I'm trying to upgrade my 1.0.29.0 version to 1.1.10.0 and the original keeps coming back...

I've replaced all of the appropriate files with new versions and run regsvr, but to no avail. I had NO problem installing to a virgin system. By the way, you do not need to put anything in the system32 folder - this is incorrect information.

Kurt

regsvr32 /u filename.dll


You don't have to put the files in System32 as long as you have them in the system path somewhere.

[tebasuna51]

Demuxed a 96kbps, 24 bit THD stream using eac3to. When converting to DTS, Surcode [1.0.29.0] has indicated that it can't handle the sampling rate.

I used the following to try to convert the THD stream to 48kbps:

>eac3to "00000 - 4 - TrueHD+AC3, English, 5.1 Channels, 96kHz.thd+ac3" "dion.thd+ac3" -resampleTo48000

Use:
eac3to "00000 - 4 - TrueHD+AC3, English, 5.1 Channels, 96kHz.thd+ac3" "dion.wavs" -resampleTo48000

And the wav's can be used like input for your DTS encoder.

[Anakunda]

Hello, how can I extrct video stream from m2ts? It's of format MPEG-4 AVC. It can't be to matroska, so extract to bare video track. But what extension to give it? I've tried .m4v and .avc but eac3to failed

This is the video track I need to extract:

ID : 4113 (0x1011)
Menu ID : 1 (0x1)
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L4.1
Format settings, CABAC : Yes
Format settings, ReFrames : 2 frames
Codec ID : 27
Duration : 1h 34mn
Bit rate mode : Variable
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16:9
Frame rate : 23.976 fps
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive

eac3to "I:\video\BDMV\PLAYLIST\" 1) 2: I:\video\BDMV\bluray.mkv

gives a freeze on 1% of MKV creation (perhaps encryption or some bug)
what else format I can use to extract this track and what output extensions are supported by eac3to? Thank U!

[Doom9User123]

First, let me just commend the author of this wonderful program for all of the time and effort spent. This program has really simplified my life!

Does anyone know if it's possible to use this program to extract audio/video streams w/o producing a log file in the process?

[nibus]

First, let me just commend the author of this wonderful program for all of the time and effort spent. This program has really simplified my life!

Does anyone know if it's possible to use this program to extract audio/video streams w/o producing a log file in the process?

add -log=NUL to the command line

[looney]

How to keep decoded wav that eac3to temporarily produce in dts transcoding 1536k->768k (libav) and after manually encode it in dts? Original audio track is 5.0 only and mixed-up channels and Surcode doesnt support that input (at least i dont know how), and decoded wavs are automatically deleted by eac3to when Surcode fails to encode it. I would rather convert it manually in Surcode.

[tebasuna51]

eac3to input.dts output.wavs

[me7]

No matter the source/decoder or encoder, when you downmix 6.1 to 5.1 with eac3to need use:

-0,1,2,3,5,6,4 -down6

What about 7.1 sources? Do I need to remap the channels as well?

[tebasuna51]

What about 7.1 sources? Do I need to remap the channels as well?
Nope, downmix 7.1 -> 5.1 work fine.

[rapscallion]

That raises another questiom. I notice when I extract Dolby THD 7.1 track to wavs, that there is a msg displayed in the dialog box "remapping channels".

Why does it do that and will the resulting wavs be incorrect when I enter them into the DTS encoder ?

[tebasuna51]

The remap is between the internal channel order (different in each format: ac3, dts, ogg, aac, ...) and the WAV order.
Don't worry about the message.

[rapscallion]

Thanks tebasuna, I was interpreting as remapping their positions and it kind of threw me even though I've seen the msg before and never paid attention to it.

But today, all of a sudden, I had an epithany "huh" ?

[7ekno]

Hi all :)

Just after a quick clarification!

With this command:
eac3to -input.dtshd -output.ac3 -down6 -down16 -resampleTo48000 -640 -normalize -libav

Does the "-libav" on the end force decoding of the DTS-MA via "libav", or does it force the encoding of "output.ac3" to be done via "libav" ...

Just trying to get the best quality AC3 as possible and I believe the AC3 encoder in libav is much more actively and recently updated ...

Would it be better to start with an MKV, extract the audio and specify AC3 and "-libav" in the extraction (will this force encoding via libav since MKV can't be decoded via libav)? Like:

eac3to -input.mkv -output.ac3 -down6 -down16 -resampleTo48000 -640 -normalize -libav

Thanks for anybody that can help,
7ek

[the_weirdo]

With this command:
eac3to -input.dtshd -output.ac3 -down6 -down16 -resampleTo48000 -640 -normalize -libav

Does the "-libav" on the end force decoding of the DTS-MA via "libav", or does it force the encoding of "output.ac3" to be done via "libav" ...
Decoding. However, libav doesn't support decoding DTS-MA at all (at least for now), it just fallback to DTS core.

Just trying to get the best quality AC3 as possible and I believe the AC3 encoder in libav is much more actively and recently updated ...
eac3to currently doesn't support encoding AC-3 via libav directly. Even it does, libav came with eac3to is old. You can, however, pipe decoded data from eac3to to Libav/FFmpeg executable binaries, for example:
eac3to -input.dtshd stdout.wav -down6 -down16 -resampleTo48000 -normalize | ffmpeg -i pipe:0 -ab 640k output.ac3

[tebasuna51]

eac3to -input.dtshd -output.ac3 -down6 -down16 -resampleTo48000 -640 -normalize (-libav)

Just trying to get the best quality AC3 as possible ...
Then, for what -down16?

[7ekno]

Thanks guys,

Was mislead from the Wiki entry (http://en.wikibooks.org/wiki/Eac3to/How_to_Use):

Convert a DTS track to a AC3 one, using libav encoder:
eac3to.exe input.dts output.ac3 -libav

Great suggestion with the pipe ;)

As for 16 bit, I was under the impression that was standard for AC3 encoder, love to be corrected if that is indeed wrong :)

7ek

[tebasuna51]

As for 16 bit, I was under the impression that was standard for AC3 encoder, love to be corrected if that is indeed wrong

Yes is wrong, don't limit the precission at all.

[Rodeo]

So, I was making sure that channel assignments and/or downmixing worked well, and tested eac3to on a bunch of channel checks I had available.

The FFmpeg TrueHD decoder was swapping the side and back surround channels. That was fixed recently: http://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=6daf513

Given that eac3to 3.24 is much older than that fix, it should have been affected by the bug - but to my surprise, it wasn't.

I used a TrueHD 7.1 channel check available on demo-world.eu

I tested it this way:

eac3to.exe input.mkv 2: output.wavs

and all the channels seemed to be in the correct file, in particular:

output.SL.wav -> left side surround
output.SR.wav -> right side surround
output.BL.wav -> left back surround
output.BR.wav -> right back surround

So I tested it on a DTS-HD MA 7.1 channel check from the Disney World of Wonder Blu-ray, and got incorrect results:

output.SL.wav -> left back surround
output.SR.wav -> right back surround
output.BL.wav -> left side surround
output.BR.wav -> right side surround

eac3to.exe -test

eac3to (v3.24) is up to date
Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed
http://www.nero.com/eng/store-blu-ray.html
CAUTION: You need Nero 7. Nero 8 won't work with eac3to.
ArcSoft DTS Decoder (1.1.0.0) works fine
Sonic Audio Decoder (3.24.0.0) doesn't seem to be installed
Haali Matroska Muxer (2011-03-03) is up to date
Nero AAC Encoder could not be located
http://www.nero.com/eng/nero-aac-codec.html
Copy NeroAacEnc.exe to the eac3to or to the Windows folder.
Surcode DTS Encoder doesn't seem to be installed
http://www.surcode.com
MkvToolnix (5.0.0.0, release version) is up to date


There are a couple possibilities:

1)
- the FFmpeg TrueHD decoder is patched in eac3to to fix the channel swap bug, and either:
- - a. the Arcsoft DTS decoder is swapping channels
- - b. eac3to is remapping channels incorrectly from DTS-HD MA 7.1 sources

2)
- the side and back surround channels are swapped internally in eac3to, affecting all 7.1 sources
- the FFmpeg and eac3to bugs cancel each other out

In the latter case, remapping should always be used when dealing with 7.1 sources, whether downmixing or not:

-0,1,2,3,6,7,4,5

…except when using the FFmpeg decoder and the source is TrueHD 7.1.

Here are the WAV files output by eac3to:

no remapping: http://www.mediafire.com/?3r35vr7gpg3v008

-0,1,2,3,6,7,4,5: http://www.mediafire.com/?68q021d69doy8va

[wlee]

I have a TrueHD track without AC3 core, it's not recognized by tsMuxer, is there a way to convert to TrueHD+AC3 track? so that I can mux into m2ts container.

Thanks.

[tebasuna51]

I have a TrueHD track without AC3 core, it's not recognized by tsMuxer, is there a way to convert to TrueHD+AC3 track? so that I can mux into m2ts container.

Thanks.

eac3to input.thd output.thd+ac3

[tebasuna51]

...
There are a couple possibilities:

...

In the latter case, remapping should always be used when dealing with 7.1 sources, whether downmixing or not


I don't think so. For me work fine, maybe other possibility:

3) Your test channel is wrong. Please put a exact link (or upload) to the test channel.

Please test this test-channel I created now with DTS HD Encoder Suite, here:
ss = Side
sr = Back

http://www.mediafire.com/?sk7mud5h28ol2m4

[Thunderbolt8]

eac3to input.thd output.thd+ac3this would create a ac3 track from the .thd track though. is it also possible to use the orginal interweaved and demuxed ac3 track as input file for this process, together with the thd track?

[nurbs]

As far as I know it would only create an AC3 track from THD if there is no interleaved AC3 in the track to begin with. If there is it should just demux.

[Thunderbolt8]

but if I demux, dont get thd and ac3 seperated automatically? that is what I would want to avoid.

[Rodeo]

I don't think so. For me work fine, maybe other possibility:

3) Your test channel is wrong. Please put a exact link (or upload) to the test channel.

Please test this test-channel I created now with DTS HD Encoder Suite, here:
ss = Side
sr = Back

http://www.mediafire.com/?sk7mud5h28ol2m4

Looks like we're getting different results:

eac3to.exe test-chan.dtshd channel.wavs


http://www.mediafire.com/?q9x6ovpz726waju

channel.SL.wav -> "beep… back left"
channel.SR.wav -> "beep… back right"
channel.BL.wav -> "beep… side left"
channel.BR.wav -> "beep… side right"

[b66pak]

@tebasuna51 i can confirm that SL/SR are swapped with BL/BR...

eac3to.exe test-chan.dtshd channel.wavs

eac3to 3.24
ArcSoft DTS Decoder (1.1.0.0) works fine
_

[tebasuna51]

I can't understand, here with:

eac3to (v3.24) is up to date
ArcSoft DTS Decoder (1.1.0.0) works fine

eac3to.exe test-chan.dtshd channel.wavs

25/07/2008 15:04 462.925 ASAudioHD.ax v1.3.2.60
15/04/2008 19:40 106.496 checkactivate.dll v1.0.0.2
17/01/2008 19:37 925.696 DtsDec.dll v1.0.2.2
25/04/2008 11:50 917.504 dtsdecoderdll.dll v1.1.0.0
02/11/2006 16:35 35.584 MagCore.dll v1.0.0.129
02/11/2006 16:28 60.160 MagPCMac.dll v1.0.0.129
02/11/2006 16:29 158.464 MagUIEngine.dll v1.0.0.129
02/11/2006 16:30 88.832 MagUIInter.dll v1.0.0.129

the channels aren't swapped.

Edit: I remember now what may be the problem (suffered by myself), check the dtsdecoderdll.dll date:
http://forum.doom9.org/showthread.php?p=1266679#post1266679

[Rodeo]

Hmm - I got my Arcsoft decoder from a very recent TMT5 trial, and wondered how I could have ended up with version 1.1.0.0 (which seemed old) of the DTS decoder.

It looks like eac3to.exe isn't picking up TMT5's decoder, but an old version of dtsdecoder.dll (from 21/04/2008) which found its way to my C:\Windows\SysWOW64 folder somehow…

:confused:

So I guess it's 1) a. then:

1)
- the FFmpeg TrueHD decoder is patched in eac3to to fix the channel swap bug, and either:
- - a. the Arcsoft DTS decoder (1.1.0.0 from 21/04/2008) is swapping channels

[wlee]

this would create a ac3 track from the .thd track though. is it also possible to use the orginal interweaved and demuxed ac3 track as input file for this process, together with the thd track?

If not possible to convert thd to thd+ac3, is it possible to convert thd to dts-hd ma track without any quality loss? tsMuxer should have no problem with dts-hd ma track.

[b66pak]

i can confirm that dtsdecoderdll.dll v1.1.0.0 (25 April 2008) is OK...

md5:
644aa3ade7742079533dcde2abf153e2 *dtsdecoderdll.dll
_

[rc71]

So, I was making sure that channel assignments and/or downmixing worked well, and tested eac3to on a bunch of channel checks I had available.

...................................

-0,1,2,3,6,7,4,5: http://www.mediafire.com/?68q021d69doy8va


I've had this issue myself. I was using Ripbot, which uses eac3to, when the problem arose. No matter what I did I couldn't get the mappings correct so I used MeGUI and set it to NeroAAC to convert the DTS to AAC HE and no problems.

[ramicio]

Is there any thought into being able to enter a dB gain in a precision other than an integer? The program can easily do it. If it detects clipping it applies a gain with a decimal, and it takes to below peak, so it's not just to get it to 0 dB.