eac3to - audio conversion tool

[frumble]

Thank you for your help! So I made WAV files with eac3to and want to convert them to WavPack, but the Encoder says: "can't handle .WAV files larger than 4 GB (non-standard)!" - sadly booth streams are over 4 GB. Also Monkey's audio can't encode them: "Error: 1002".
Export to AC3 with ea3to works fine but I absolutely want a lossless codec (it feels better ;) ). In my desperation I even tried to merge the movie MKV with the WAVE files but MKVmerge GUI gives the error "87".
Both WAVE streams seams to be a little bit strange: Non of my audio players can play more than a few minutes (even they display only a few minutes of playtime) and also Audacity can only show me a few seconds!
The link from @Sparktank is very confusing for me. Maybe I try it later.
Downmixing the stream would be the last option but I would be very pleased if it's possible to preserve the original channels...

[Midzuki]

^ @ frumble:

1) forget Monkey Audio, it doesn't support multichannel ;

2) try "wavpack --help" in the command prompt ;)

[Midzuki]

Off this topic...

It's really great that there are freeware tools like eac3to but I can't understand why the authors of such media helpers don't make the source code available...
But since 2010 no progress with eac3to. In this time the program could have been matured and bugs could have been fixed from others but they can't do it because they don't have the source code.

Sadly you're not entirely correct. Open source-code is NO guarantee that there will ever be other capable programmers interested in fixing the bugs of the software. Take a look at MaestroSBT, K-Meleon, mpeg2enc, mkisofs, etc.

[frumble]

@Midzuki Sorry, the WaPack help page hasn't a option close to "allow input files bigger than 4 GB", or what do you mean?
You are right, the fact that a software is OSS doesn't mean that there are other developers capable with the task. But look at ffmpeg/libav, x264, VLC and mplayer in the media field. It's not soo bad. And consider the chance @madshi, eac3to's author can't work on freeware projects anymore or dies. Then the community would be forced to rewrite the tool from scratch. OSS is freedom and opportunity, not constraint.

[Midzuki]

@Midzuki Sorry, the WaPack help page hasn't a option close to "allow input files bigger than 4 GB", or what do you mean?
...

-i ignore length in wav header

[MuteyM]

Hi, I am decoding a DTS-ES 6.1 soundtrack to wav format. If I use the Sonic decoder then everything works correctly. However if I use libav, then I end up with a wav file with a corrupt channel mask. It looks like eac3to is not taking into account libav's lack of back channel decoding:

eac3to j:\backup 1) 4: c:\temp\orig.wav -libav
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 1:38:52, 24p /1.001
1: Chapters, 20 chapters
2: MPEG2, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
4: DTS-ES, English, 6.1 channels, 24 bits, 1509kbps, 48kHz, dialnorm: -4dB
5: Subtitle (PGS), English
6: Subtitle (PGS), French
7: Subtitle (PGS), Spanish
8: Subtitle (PGS), English
a04 The libav DTS decoder doesn't decode the back channels.
a04 Extracting audio track number 4...
a04 Removing DTS dialog normalization...
a04 Removing XCh extension...
a04 Decoding with libav/ffmpeg...
a04 Reducing depth from 64 to 24 bits...
a04 Writing WAV...
a04 Creating file "c:\temp\orig.wav"...

I end up with a wav file containing 5.1 channels of data, but with a channel mask of 0x70f (6.1) instead of the expected 0x60f (5.1). This confuses the heck out of any software that attempts to play it :)

I thought using -down6 might force eac3to to work properly, but this ended up with the same problem. So I think the solution is for madshi to update eac3to to write the correct channel mask when libav decodes 6.1 (and probably 7.1) DTS files.

[tebasuna51]

I end up with a wav file containing 5.1 channels of data, but with a channel mask of 0x70f (6.1) instead of the expected 0x60f (5.1).
Yes, is a know bug.

[frumble]

@Midzuki Thank you very much! The conversion worked with all streams and the end MKVs play nice with the WavPack codec! :)

[Midzuki]

^ Glad to see it worked :)

Now let's spread the word,
WavPack is the future! :sly: :D

[Bigmango]

Is it possible to convert TrueHD to DTS-HDMA (TrueHD > WAVS > DTS HD Encoder) and keep the same sound for the TrueHD tracks that apply DRC to improve the sound? (at least I think it's the DRC that's doing this ?).

This problem arises with the Transformers 3 TrueHD 7.1 track (and people tell me Iron Man is probably the only other movie doing the same). The converted DTS-HDMA track does not sound the same as the original TrueHD. Some sounds at different moments in the movie have more presence with the original TrueHD track. It makes the sound better, more dynamic.

I think this is caused by the DRC (?) that's applied differently on the different channels at different moments.

The TrueHD track sounds better with PowerDVD 12 compared to the converted DTS-HDMA, but with MPC I don't seem to hear a difference (this leads me to think that MPC doesn't apply the effects and so it doesn't handle the TrueHD metadata correctly (?).

PCM and DTS-HDMA can't modify the sound at playback time in the way TrueHD does. How can I save the wavs with the full TrueHD sound experience?

To summarize, I would like to convert TrueHD to DTS-HDMA and have it sound as good on hardware receivers and PowerDVD (now with eac3to the audio source quality is the same, but many sounds are lacking presence in different parts of the movie).


(This is the first movie I am having this problem with. It seems the Transformers 3 sound engineers wanted to TrueHD track to be reproduced with improvements applied to the Lossless track it contains - and the whole experience is indeed more enjoyable to my ears with these improvements).



PS: sorry mods, I have opened this thread (http://forum.doom9.org/showthread.php?p=1570771#post1570771) before, but I think here is the right place as it concerns eac3to

[tebasuna51]

Bimango, please continue this discussion in your original thread if you want.
The discussion about this topic in this thread was closed in first month.

eac3to always ignore DRC because is a effect to apply at play time (with 'Night mode' for instance), never when recoding, and don't improve the audio just make it less dynamic.

[Bigmango]

Bimango, please continue this discussion in your original thread if you want.
The discussion about this topic in this thread was closed in first month.

eac3to always ignore DRC because is a effect to apply at play time (with 'Night mode' for instance), never when recoding, and don't improve the audio just make it less dynamic.

Ok so if it's not DRC it's another effect that's applied to the sound. (continued in my other thread (http://forum.doom9.org/showthread.php?t=164724)).

Regarding eac3to: Can eac3to be used with another decoder that can save the "complete" TrueHD to wavs? It seems the open source decoders don't fully support TrueHD as they only extract the lossless track without the effects that enhance it by giving more presence to specific sounds at specific times (most movies only play the lossless track as is, so only a handful of movies seem to be concerned by this issue).

Thanks.

[tebasuna51]

eac3to save the complete TrueHD to wav's. Don't exist "effects that enhance it by giving more presence to specific sounds".

[Joniii]

I'm converting 5.1 AC3 from DVD (25.000 -> 23.976), Is it normal that eac3to remaps channels?

eac3to f:\al.ac3 g:\al.ac3 -slowdown
AC3, 5.1 channels, 1:26:43, 384kbps, 48kHz, dialnorm: -27dB
The Nero decoder doesn't seem to work, will use libav instead.
Removing AC3 dialog normalization...
Decoding with libav/ffmpeg...
Remapping channels...
Changing FPS from 25.000 to 23.976...
Encoding AC3 <640kbps> with libAften...
Creating file "g:\al.ac3"...
eac3to processing took 3 minutes, 42 seconds.
Done.

[tebasuna51]

Yes, internal AC3 channel order is different than standard channel order used to resample and send to encoder.
Don't worry.

[ilomambo]

Hello, I am having the following problem:

eac3to v3.24
command line: "c:\Program Files\Utils\eac3to\eac3to.exe" "track01.dts" "track01.wav" -libav -simple
------------------------------------------------------------------------------
VOB, 1 audio track, 0:08:57
1: DTS-96/24, 5.1 channels, 24 bits, 1510kbps, 96kHz
Track 1 is used for destination file "track01.wav".
[a01] Extracting audio track number 1...
[a01] Decoding with libav/ffmpeg...
[a01] Reducing depth from 64 to 24 bits...
[a01] Writing WAV...
[a01] Creating file "track01.wav"...
[a01] Clipping detected, a 2nd pass will be necessary. <WARNING>
[a01] Starting 2nd pass...
[a01] Extracting audio track number 1...
[a01] Decoding with libav/ffmpeg...
[a01] Reducing depth from 64 to 24 bits...
[a01] Writing WAV...
[a01] Applying -0.73dB gain...
[a01] Creating file "track01.wav"...
eac3to processing took 1 minute, 27 seconds.
Done.


The wav file seems fine, I can see it plays all 5.1 channels
But look at the file sizes:

>>dir track01*.*
Volume in drive K is Music
Volume Serial Number is 884C-4DCD

Directory of K:\DTS

29/04/2012 14:09 105,261,056 track01.dts
05/05/2012 23:54 987 track01 - Log.txt
05/05/2012 23:54 463,804,460 track01.wav
3 File(s) 569,066,503 bytes
0 Dir(s) 88,401,051,648 bytes free


Is this correct, WAV has a factor of 4.5 times the DTS size?
I don't know, other WAV DTS files I have are much smaller (given the average MB/min of audio, this one is 8:56 min)
Is there any command line switch to keep the WAV size smaller?

[tebasuna51]

...
Is this correct, WAV has a factor of 4.5 times the DTS size?

eac3to/libav decode your DTS-96/24, 5.1 channels, 24 bits, 1510kbps, 96kHz to a WAV 5.1 channels, 24 bits, 48kHz with a bitrate of:
6 channels x 24 bits x 48KHz = 6912 kb/s

6912/1510 = 4,57

Then, yes, is correct.

Is there any command line switch to keep the WAV size smaller?
You can use -down16 to obtain a WAV 5.1 channels, 16 bits, 48kHz with a bitrate of:
6 channels x 16 bits x 48KHz = 4608 kb/s

4608/1510 = 3,05

[ilomambo]

You can use -down16 to obtain a WAV 5.1 channels, 16 bits, 48kHz with a bitrate of:
6 channels x 16 bits x 48KHz = 4608 kb/s

4608/1510 = 3,05

Thanks.

I am dumb regarding the technical details. I think the DTS file has the same information as the final WAV, that's why the increase in size seemed too much.
But, I assume from your explanation, that much of the DTS info is replicated to create the 6 CH WAV, and that's why the file is so much bigger, isn't it?

EDIT:
I just looked in another song I have in DTS 5.1 WAV format (in the AC3 filter properties while it was playing) and it showed this:


Decoder:
Stream format: DTS 3/2.1 (5.1) 44100Hz
Bitstream type: 14bit low endian
Frame size: free format
Samples: 1024
Bitrate: unknown
SPDIF stream type: 0xc
Frame interval: 4096
Actual bitrate: 1411kbps
DTS
speakers: 3/2.1 (5.1)
sample rate: 44100Hz
bitrate: 1411kbps
stream: 14bit LE
frame size: 3584 bytes
nsamples: 1024
amode: 9
No CRC

Tebasuna51, If I follow your math 6ch x 14bit x 44KHz = 3696 kbps != 1411 kbps reported by AC3
It is 13:40 min song and the file only takes 141MB
Something is not fitting, according to my understanding

On the other hand when I play the file created by eac3to, AC3 filter shows this:


Input format: PCM24 3/2.1 (5.1) 48000
User format: PCM16 - 0
Output format: PCM16 3/2.1 (5.1) 48000

That's why I think I am using the wrong tool. I just wanted to wrap the DTS file in WAV format, not to convert it. It seems the file I got is pure PCM.

[MrVideo]

I just wanted to wrap the DTS file in WAV format, not to convert it. It seems the file I got is pure PCM.

WAV is not a wrapper, it is a format, just like MPEG-2 and MPEG-4 are video formats, that get wrapped into various containers. Neither DTS or WAV are containers.

DTS is a compressed audio format, while WAV is not. Hence the reason that the WAV file is larger after you uncompressed the DTS file.

[Midzuki]

...

DTS is a compressed audio format, while WAV is not.

But yes, .WAV is a RIFF-based container, and may contain compressed audio. Regarding DTS-in-WAV especifically, there are two types, 1) normal, without SPDIF-padding, and with a .dca TwoCC, and 2) hacky, with SPDIF-padding, disguised as stereo PCM @ 32 / 44.1 / 48 kHz.

[MrVideo]

Interesting. I've never seen anything but WAV uncompressed PCM audio.

Then, that leads me to this question... why put DTS compressed data into a WAV file? I'd just leave it as a DTS file.

[tebasuna51]

But, I assume from your explanation, that much of the DTS info is replicated to create the 6 CH WAV, and that's why the file is so much bigger, isn't it?
The audio data in DTS is encoded (compressed like a zip) the audio data in WAV file is decoded (uncompressed PCM data)

I just looked in another song I have in DTS 5.1 WAV format
A dts_5.1_wav is not a correct WAV file because the header don't show the content. The fake header show a PCM 2.0 44.1KHz just to lies a burner and burn a CD-Audio.

I just wanted to wrap the DTS file in WAV format, not to convert it.
Then you want burn CD-Audios 5.1
Try spdifer (http://ac3filter.net/wiki/AC3Filter_tools), a AC3Filter tool.

Then, that leads me to this question... why put DTS compressed data into a WAV file?

Like you can read, only to burn a CD Audio.
Some CD players can send the DTS 5.1 and play surround.

[ilomambo]

Then you want burn CD-Audios 5.1
Try spdifer (http://ac3filter.net/wiki/AC3Filter_tools), a AC3Filter tool.


tebasuna51, you are the man! :goodpost:

:thanks:

spdifer worked like a charm! I got my WAV file 5.1ch, at the same size as the DTS original file !!

... and I learned something about DTS and WAV files.

[RazvanuZu]

I get an error when I try to use eac3to with SurCode DVD–DTS v1029. Is this a problem with eac3to or SurCode DVD–DTS v1029? I mean, is eac3to incompatible with SurCode DVD–DTS v1029?
When I use SurCode DVD–DTS v1021 I get no error.

The error sounds like: "Pressing the Surcode "Encode" button didn't seem to work", but when I directly use SurCode DVD–DTS v1029 I don't get any errors at all.

What is the latest compatible version of SurCode DVD–DTS?

Thanks.

[ramicio]

I wish there was a way to have eac3to in Linux without wine. I don't have a GUI installed, so I can't do wine. I run eac3to on my Windows machine to rip discs, but sometimes I forget something and need to fiddle with large files after I've written them, and my disc is across a network. Being able to have it on my Linux would speed things up quite a bit. I would only need basic libavcodec functionality.

[Furiousflea]

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

Is this still needed, may I ask. Changelog mentions a bug fixed in version 3.21...

fixed: 6.1 DTS decoding with ArcSoft resulted in wrong channel order

Many thanks.

[tebasuna51]

The decode is OK, only need a remap when downmix 6.1 to 5.1

[Furiousflea]

The decode is OK, only need a remap when downmix 6.1 to 5.1

Thanks for confirmation :)

[Joniii]

I have previously converted 25.000 fps audio to 23.976 with -slowdown switch, how do I convert 25.000 to 24.000?

I have import Blu-ray with strange 24p (not 24.000/1.001) h.264 and DTS. I'm trying to add AC3 audio into it from my older PAL DVD.

[ramicio]

Are you even sure the audio is going to line up? What is the fixation with the audio from the DVD?

Anyway, you would first add the "-25.000" switch, to tell it to assume the material is 25 FPS, and then "-changeto24.000".

[Joniii]

Are you even sure the audio is going to line up? What is the fixation with the audio from the DVD?

Anyway, you would first add the "-25.000" switch, to tell it to assume the material is 25 FPS, and then "-changeto24.000".

I've always adjusted the delay after conversion. I'm adding dubbed audio from DVD to import blu-ray for my 3 year old.

I've thought you need to convert audio with -slowdown not -changeto switch?

[ramicio]

I still don't get it. Slowdown is deprecated, and only for 25 to 23.976 conversion.

[Revgen]

Hi can someone please confirm for me what exactly the bug is with different versions of Arcsoft DTS Decoder, ?

Is it 1.1.0.5 or 1.1.0.0 for 6.1/6.0 audio and 1.1.0.8 for anything else ?
Some people claim that 1.1.0.0 is the best for 6.1/6.0 but others claim 1.1.0.5 is best because 1.1.0.0 lacks many bug fixes ?
I found the following pieces of information around doom9 forums but i'm still confused to which version to use to avoid any bug ?

- 1.1.0.0 can decode DTS(-HD) 6.1/6.0 but can't decode non-standard 7.1
- All versions above v1.1.0.0 do not accurately decode 6.1 DTS tracks with eac3to.
- 1.1.0.8 can't decode DTS(-HD) 6.1/6.0 but can decode non-standard 7.1
- Both decode DTS(-HD) 1.0 correctly, unlike 1.1.0.7.
lossy DTS
1) 1.1.0.0 and 1.1.0.1 always decode lossy DTS as 24 bit, and their decoding results differ from each other.
2) 1.1.0.5 and up decode in proper bitdepth.
3) Versions 1.1.0.5 and up decode lossy DTS identically.
4) 1.1.0.1 and 1.1.0.5 decode 24 bit lossy DTS identically.
5) Decoding of 16 bit lossy DTS was changed from 1.1.0.0 to 1.1.0.1 and from 1.1.0.1 to 1.1.0.5.
6) 1.1.0.7 and 1.1.0.8 decode 6.0 without back center channel.

The conclusion : for lossy DTS is 1.1.0.5. It decodes all configurations, in proper bitdepth, and decoding algorithm didn't change since this version (except 6.0 bug in .7 and .8).

Just to add to your list. I have 1.1.0.0, 1.1.0.5, 1.1.07, and 1.1.0.8. Only 1.1.0.8 seems to work with DTS MA 1.0 files. I don't know about 1.1.0.1 since I don't have it.

[xkodi]

- 1.1.0.0 can decode DTS(-HD) 6.1/6.0 but can't decode non-standard 7.1
- 1.1.0.8 can't decode DTS(-HD) 6.1/6.0 but can decode non-standard 7.1


i don't know why the same wrong information is continue spreading, when there are enough facts proving that's completely wrong statement about 1.1.0.0 vs. 1.1.0.8 regarding non-standard 7.1 (i.e. "strange setup") tracks:

http://forum.doom9.org/showthread.php?p=1471763#post1471763

[Boulder]

Only 1.1.0.8 seems to work with DTS MA 1.0 files.Is it the only one that can decode such files or what is the issue with other versions?

[Revgen]

Is it the only one that can decode such files or what is the issue with other versions?

It's the only version that works that I know of.

I have 2 Blu-Ray discs with DTS MA 1.0 tracks. The Big Heat (1953) and The Big Trail (1930). 1.1.0.0, 1.1.0.5, and 1.1.07 give errors everytime I try to decode with eac3to. I also can't play the DTS MA 1.0 audio in TMT either unless 1.1.0.8 is in the codecs folder. I don't have version 1.1.0.1, so I can't comment on whether that version can decode DTS MA 1.0 or not.

[pasadena]

Hi All,

I'm trying to convert this into a blu-ray structure ISO for playback on my media player.

I have no problems extracting all the files using either evodemux or HD-DVD steam extractor but cannot remux this particluar TrueHD track.

I've spent the past week trying to work out issues that exist with remuxing TrueHD tracks.

I tried to convert the TrueHD track to DTS but it comes up saying that it does not support the sampling rate.

Here's an ouput log:

eac3to v3.24
command line: "D:\Converter Programs\HdBrStreamExtractor_0.8\eac3to.exe" "D:\Black and White Night" 1) 4:"D:\bwnight\1_4_audio.dts" -progressnumbers
------------------------------------------------------------------------------
EVO, 1 video track, 3 audio tracks, 1 subtitle track, 1:04:34
1: Chapters, 18 chapters
2: h264/AVC, 1080i60 /1.001 (16:9)
3: E-AC3, English, 5.1 channels, 3024kbps, 48kHz, -66ms
4: TrueHD, English, 5.1 channels, 96kHz, -66ms
5: E-AC3, English, 2.0 channels, 1023kbps, 48kHz, -66ms
6: Subtitle (DVD), Japanese
[a04] Extracting audio track number 4...
[a04] Decoding with libav/ffmpeg...
[a04] Applying RAW/PCM delay...
[a04] Writing WAVs...
[a04] Creating file "D:\bwnight\1_4_audio.L.wav"...
[a04] Creating file "D:\bwnight\1_4_audio.R.wav"...
[a04] Creating file "D:\bwnight\1_4_audio.LFE.wav"...
[a04] Creating file "D:\bwnight\1_4_audio.C.wav"...
[a04] Creating file "D:\bwnight\1_4_audio.SR.wav"...
[a04] Creating file "D:\bwnight\1_4_audio.SL.wav"...
[a04] The original audio track has a constant bit depth of 24 bits.
Encoding DTS <1536kbps> with Surcode...
Surcode DTS Encoder doesn't support this samplingrate. <ERROR>

I use evodemux to get the THD file but I can't do anything else with it. Tsmuxer bombs out saying it can't detect stream type so it can't read and import it

Eac3to cannot convert it to thd+ac3.

eac3to v3.24
command line: eac3to demuxed.thd output.thd+ac3
------------------------------------------------------------------------------
TrueHD, 5.1 channels, 96kHz
Decoding with libav/ffmpeg...
Encoding AC3 <640kbps> with libAften...
Initialization of the AC3 encoder failed. <ERROR>
Aborted at file position 262144. <ERROR>

I've also used a program called pcm2tsmu which converted the TrueHD track to an LPCM stream sampled at 96KHz which tsmuxer was able to use and process. When I created a BD structure through Tsmuxer and played it through my media player, the audio cames out as white noise.

Is converting a TrueHD track to an LPCM stream going to be inferior in any way?

If anyone has this disc, can you please let me know if you had any luck successfully converting this to blu-ray with the TrueHD track intact.

Any help would be much appreciated.

Cheers

[tebasuna51]

@pasadena
- With THD track: I don't know if this can work because I can't test with a THD 96 KHz, but try this:

"D:\Converter Programs\HdBrStreamExtractor_0.8\eac3to.exe" "D:\Black and White Night" 1) 4:"D:\bwnight\1_4_audio.thd+ac3" -resampleTo48000 -progressnumbers

- With DTS-MA: you can't use Surcode because don't support 96 KHz and DTS-MA, only standard DTS is not recommended.
If you have DTS-HD Master Audio you can extract the monowavs and after encode to DTS-MA

"D:\Converter Programs\HdBrStreamExtractor_0.8\eac3to.exe" "D:\Black and White Night" 1) 4:"D:\bwnight\1_4_audio.wavs" -progressnumbers

- With LPCM, same quality than THD but biger size.
You have two options:

1) Decode to w64 (tsMuxeR must accept PCM in w64 container):

"D:\Converter Programs\HdBrStreamExtractor_0.8\eac3to.exe" "D:\Black and White Night" 1) 4:"D:\bwnight\1_4_audio.w64" -progressnumbers

2) Decode to special PCM format for tsMuxeR with pcm2tsmu:

"D:\Converter Programs\HdBrStreamExtractor_0.8\eac3to.exe" "D:\Black and White Night" 1) 4: stdout.pcm -progressnumbers | "YOUR-PATH-TO\Pcm2Tsmu.exe" - "D:\bwnight\1_4_audio.pcm" -s 96000

[pasadena]

@tebasuna51

Thanks for getting back to me.

I'll give these a try in the morning and let you know the outcome.

Much appreciated.

Cheers.

[Vasch the stampede]

There will be any chance that eac3to become multithreaded?

[ramicio]

Another possible approach: drop FLAC :p and "move house" to WavPack :)

Except problems arise when you have files bigger than 4 GiB. Some filters handle it, some don't. For Windows you can use madFLAC for correct channel assignments, or WavPack's splitter filter and decoder handles files > 4 GiB. Outside of Windows, you're screwed on both fronts. Both codecs still need work, and it's pathetic that they've been basically "completed" years ago despite some problems. All FLAC would have to do is switch to using WAVEFORMATEXTENSIBLE channel masks.

[Joniii]

Stupid question, I have a m2ts file with DTS (delay -9ms). eac3to applied dts delay when demuxing but said a remaining delay of +1ms could not be fixed. I know 1ms is nothing but being a perfectionist, do I have to add -1ms or +1ms with mkvtoolnix when remuxing?

Little confused because the file has -9ms delay but after fixing it, it ends up having +1ms delay.

[hello_hello]

Maybe the extra 1ms delay is caused by some sort of padding used at the beginning of the audio file and that's as close as eac3to could get it? Just a guess....
To answer you question though I'd assume it's a positive delay of 1ms which needs to be added manually.

[tebasuna51]

Correct the delay without recode is only possible adding/cutting complete DTS frames (512 samples at 48KHz = 10.667 ms each frame).
Here 1 frame is cutted then 10.67ms - 9ms = 1.667ms.
Like audio have 1.667 ms less duration the delay in container must be +1, or +2 if you want, but is unnecesary.

Video frames have a duration of 40 ms (25 fps) then delays of +/- 20 ms are absolutely undetectable.

[Joniii]

Thx for the explanation, I have always wondered why it always leaves +1ms.

[ramicio]

I really wish logging could be disabled. There should be support for some sort of .ini file in the eac3to directory for a few settings, like which method for progress display, and if it's just the hyphen bar, then you should be able to specify how many characters wide your terminal is. Who uses the default width of 80? I don't get why progress numbers need to start a new line. It's difficult for me to judge progress because my terminals are 170 wide, and will likely grow as resolutions keep going up.

[sl1pkn07]

little question. why not use libmatroska/libebml instead haalisplitter/mkvtoolnix?

[mastrandrea]

Is this the right tool to convert a 3.0 AC3 track (Front: L R, Side: C) to a 2.0 AC3 track? If not, which software do you suggest me to use?

[tebasuna51]

eac3to only accept 5.1 to convert to 2.0.

You can try:
- BeHappy, with NicAc3Source(Down2) like Source and a DSP Normalize.
- BeLight, adjusting azid output like Stereo and Normalize.

[ramicio]

FFDShow, GraphEdit, and the WavDest filter. Set FFDShow to mix to 2.0 and normalize the levels.