eac3to - audio conversion tool

[Boulder]

Regarding audio delay: if DGIndex tells me that the audio track has +144ms delay, do I need to use +144ms or -144ms in eac3to's command line? It's a silly question but there is no obvious answer

[Marin85]

Originally Posted by Boulder

Regarding audio delay: if DGIndex tells me that the audio track has +144ms delay, do I need to use +144ms or -144ms in eac3to's command line? It's a silly question but there is no obvious answer

I feel like I am being mocked here What tebasuna51 (btw, thank you for the clear answer!) explained is - as I suspected - sort of the opposite of what eac3to is actually suggesting by the above message IMHO. It is confusing because eac3to message refers to the delay it tries to apply to the audio track, and not to the (resulting) delay the audio track already has.

[Boulder]

Originally Posted by Marin85

I feel like I am being mocked here

Not at all It's just very confusing at times. I think I've always used the opposite but since the delays are usually rather small, it's not easy to determine which way is the correct one.

[tebasuna51]

Originally Posted by Boulder

... if DGIndex tells me that the audio track has +144ms delay, do I need to use +144ms...

Yes, you must always respect the sign.
The info (also MediaInfo) always show the delay than you need apply.

[tormento]

There is a stream that eac3to really doesn't like:

9: DTS Express, English, 1.0 channels, 24 bits, 96kbps, 48kHz

I can demux it but nothing more. Tried with TotalMedia libraries, LibAV ones but whenever I am going to export to wavs, it tells me some kind of errors. Any idea about how to convert it?

[Brazil2]

Originally Posted by tormento

There is a stream that eac3to really doesn't like:

9: DTS Express, English, 1.0 channels, 24 bits, 96kbps, 48kHz

I can demux it but nothing more. Tried with TotalMedia libraries, LibAV ones but whenever I am going to export to wavs, it tells me some kind of errors. Any idea about how to convert it?

A sample would be nice for testing purposes

[tormento]

Originally Posted by Brazil2

A sample would be nice for testing purposes

If only could be possible to demux it somehow..

[Brazil2]

Originally Posted by tormento

If only could be possible to demux it somehow..

Well, you said you did:

Originally Posted by tormento

I can demux it but nothing more.

Anyway, post a muxed sample so we can check it ?

[Bryce2]

Hi to all!
Trying to convert a DTS track to AC3 using "eac3to" leads me to an unusual problem.
DTS (ES):2.77GiB, 4h 23mn = AC3: 1.34GiB, 5h 0mn.
I want to mention that this is only the 2nd time I'm facing this problem after more than 200 conversions. The first time, I found a solution in this forum with adding a parameter to eac3to, but now I'm searching hours & hours with no luck.

[Sparktank]

Originally Posted by Bryce2

Hi to all!
Trying to convert a DTS track to AC3 using "eac3to" leads me to an unusual problem.
DTS (ES):2.77GiB, 4h 23mn = AC3: 1.34GiB, 5h 0mn.
I want to mention that this is only the 2nd time I'm facing this problem after more than 200 conversions. The first time, I found a solution in this forum with adding a parameter to eac3to, but now I'm searching hours & hours with no luck.

Can you post the full log?

[Bryce2]

Originally Posted by Bryce2

Hi to all!
Trying to convert a DTS track to AC3 using "eac3to" leads me to an unusual problem.
DTS (ES):2.77GiB, 4h 23mn = AC3: 1.34GiB, 5h 0mn.
I want to mention that this is only the 2nd time I'm facing this problem after more than 200 conversions. The first time, I found a solution in this forum with adding a parameter to eac3to, but now I'm searching hours & hours with no luck.

Solved:
Using as usual the default command "eac3to.exe input.dts output.ac3" was the reason of this weird effect (for this & only dts stream).
After trial & error and changing the command to "eac3to.exe input.dts output.ac3 -libav" ..did the thing.
Anyway .. "Sparktank" for your interest to give a help.

[DJ-1]

Can I convert maintain DTS-MA track using eac32 ?
Sent from my GT-I9100 using Tapatalk 2

[DJ-1]

Oops, double post

[sneaker_ger]

eac3to can do that with the help of external libraries like Arcsoft. Read the start post.

[DJ-1]

Originally Posted by sneaker_ger

eac3to can do that with the help of external libraries like Arcsoft. Read the start post.

OK, will read it, I was trying out Multi avchd Wichita uses eac32.... tried converting dts-ma to LPCM, (as my media box does play dts, but not .dts-ma..... ) so I was trying g to passtgrough the pre-decoded LPCM stream.. got lots of hissing...nothing more.
I have an Onkyo Tx nr609, (does support dts-ma )
Sent from my GT-I9100 using Tapatalk 2

[frumble]

Hello,
I have a problem with eac3to that drives me crazy. I want to decode the DTS-HD MA streams of my release of x and encode them to FLAC so that my Linux audio players can play the full sound, not just the DTS core.
I have downloaded eac3to and the ArcSoft DTS Decoder 1.1.0.0 plus the HdBrStreamExtractor GUI.
The output WAVE files (or FLACs) have all 7 channels and the sound is nice but they appear to have the wrong channel mapping: Voices are very low and often come from only one side (I have only stereo speakers) and once in a while come with strange echo. But when I import them into Audacity and play them with the editor the output is correct. This however has no effect on the Audacity export: 6.1 channel export has the same mapping problems as the original file. My players are not the problem, they can play every DTS, AC3 and TrueHD stream correctly.
I read something about "strange setup" but I can't understand if this might be the problem. I have absolutely no knowledge about audio mastering. The issue affects both English and German streams. The Audacity export window offers the option to remap the channels but I have no clue what could be the right choice. So I really hope you can help me: I uploaded a piece of 6.1 export from Audacity in Ogg Vorbis and it would be really nice if someone could tell me the right mapping. Thank you very much in advance!
http://www.2shared.com/audio/boFrWT5...-wrong-ch.html - I hope not to violate against forum rules with this.

Off this topic I want to say something: It may sound ridiculous but I miss a "tutorial". I lost hours in trying to understand what these freeware tools do and how to set them up with decoder and codecs. The learning curve is very steep but not because I am a idiot but 'cause most of your tools lack a proper documentation. I am a Linux user and I am used to find answers in reading docs.

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. The last version of eac3to is from 2010? I read something about a GPU video encoder the author is writing since then and the promise to get back to work on eac3to when this encoder is final. 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. It is his right to hold the source for himself but I truly can't understand the reason. It doesn't seams to be his plan to make commercial profit out of it. But no hard feelings, this are just my thoughts.

[NanoBot]

Hi frumble,

this workaround might help: http://forum.doom9.org/showthread.ph...01#post1524301

[sneaker_ger]

Unfortunately 6.1 is not defined in FLAC. So players might decode it not as you expect, unless you do a downmix.

In addition to NanoBot's link:
http://forum.doom9.org/showthread.ph...30#post1538930

[Sparktank]

Originally Posted by sneaker_ger

Unfortunately 6.1 is not defined in FLAC.

This has actually been a bit of a constant problem for FLAC.
One user here goes on about it...
Dear (people), please stop using 6.1 FLAC

I remember reading about 6.1 FLAC in other places.

Personally, I'd stick to the CORE or even a conversion to AC3@640, as 640 is perceived to have transparent quality to the master file.

With some BD rips I do, I'm quite happy using only the core for maximum compatibility with my BD player that can playback MKV (with original h264/VC1/MPEG).

It's not a significant compromise between lossless and core, I'd stick to CORE audio. The human ear wouldn't really be able to pick pu the difference too much.

[Midzuki]

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