eac3to - audio conversion tool

[madshi]

Originally Posted by nautilus7

I found out some more features that you didn't tell.
A very good bug report. Yeah, i had a crash. It's the same file i mentioned a few post above. Can you tell anything about this?

Not sure. Can you please mail that bug report to me? A sample of that file would help, too (if the crash also occurs with the sample).

[madshi]

Originally Posted by shambles

by the way, i'm guessing noise-shaping is used with the TPDF dithering?

No noise shaping. It's "simple" TPDF dithering. Should be good enough, as long as you don't go under 16bit. Noise shaping only reduces the noise that you hear. I don't think TPDF noise is audible in 16bit.

Originally Posted by shambles

any idea how the quality is compared to ssrc/wavewizard?

No idea. The best idea would be: Try it out and report it here!

[madshi]

Originally Posted by nautilus7

C:\Tools>eac3to ice.greek.ac3 ice.greek.new.ac3 -slowdown -resampleto48000
AC3, 5.1 channels, 1:26:53, 384kbit/s, 48khz
Creating/writing file "ice.greek.new.ac3"...
Done.

Tried something different this time.
The process finished almost in no time and the output file is identical to the source.
Did i do something wrong?

Ah, that's a bug. Here's what eac3to thinks:

"Source file is ac3, destination file is ac3, and no different bitrate was specified. So I'm not supposed to reencode. I'm only supposed to remove dialnorm (if even necessary)."

You can work around that by doing it in 2 steps:

(1) eac3to ice.greek.ac3 ice.greek.wav -slowdown
(2) eac3to ice.greek.wav ice.greek.new.ac3

Please note that the "-resampleto48000" is probably superfluous in your case. AC3 files coming from DVDs are usually already 48000. And "-slowdown" does not change the samplerate.

[nautilus7]

I 'll send you the bug report, but how? I don't use an email client. Only webmail (gmail). And a sample too.

For the 2nd problem, i 'll do what you suggested.

[madshi]

Originally Posted by shambles

really interested to see what kind of results 24->20bit conversion will bring..

Ah yes, that's one thing I forgot to mention: Bitdepth can be reduced to anything between 14 bits and 23 bits now. And for FLAC encoding every bit helps getting the file size down. Roger Dressler (Dolby) recently stated on AVSForum that in his opinion 20 bits may be the sweet spot. So personally I might now reduce all my 24 bit tracks to 20 bit. Haven't fully decided yet...

[hristoff2]

You certainly should re-think the name 'eac3to'

We're so reliant on you and your work, thanks a million.

[Mtz]

Can somebody post a command line for an eac3 source and the desired target to be AC3 640kbps using the sonic filter?
This is not working anymore for me:
eac3to stream1.eac3 stream2.ac3 -48 -640 -sonic

enjoy,
Mtz

[madshi]

Originally Posted by nautilus7

I 'll send you the bug report, but how? I don't use an email client. Only webmail (gmail). And a sample too.

You can PM me the bug report or you can post it into gmail. Once the bug report window is visible on screen, the bug report is automatically stored into the clipboard. So just open PM or gmail and press Ctrl+V.

[madshi]

Originally Posted by hristoff2

You certainly should re-think the name 'eac3to'

Yeah, that's true. I was already thinking about alternative names. But the name "eac3to" is well known now, even if it doesn't really fit, anymore. So I'm not sure if I should change the name...

[madshi]

Originally Posted by Mtz

Can somebody post a command line for an eac3 source and the desired target to be AC3 640kbps using the sonic filter?
This is not working anymore for me:
eac3to stream1.eac3 stream2.ac3 -48 -640 -sonic

What is that "-48" supposed to mean? Just remove that. It should work then. The "-640" is probably not necessary, either, cause it's the default value used by eac3to for multichannel audio tracks. So just "eac3to stream1.eac3 stream2.ac3 -sonic" should do.

[hristoff2]

Originally Posted by madshi

Yeah, that's true. I was already thinking about alternative names. But the name "eac3to" is well known now, even if it doesn't really fit, anymore. So I'm not sure if I should change the name...

Indeed it's well known, you've got a point there. Guess that's the same reason why 'avimux' is still named avimux although it supports matroska etc etc.

[Mtz]

Still not working. Error log sent via PM. The problem appear after: Encoding AC3...
Sonic decoder version 4.2.

enjoy,
Mtz

[Thunderbolt8]

I have a problem with ratatouille remuxing (AVC, LPCM).
the movie has seamless branching so I joined each .m2ts part via "copy /b..". after that I used xport to extract video and audio from the joined .m2ts file and used eac3to to convert the LPCM stream (24-bit) to FLAC (everything fine).
but after I remuxed both together with mkvmerge I get an audio problem at playback (ffdshow, coreflac): audio suddenly disappeares at ~53 seconds movietime and also doesnt come back later on, shortly after that mpc crashes.
im not quite sure whether this could be a problem of the new eac3to version or because of seamless branching joining; however the error occurs before the first switch from the 1st to 2nd .m2ts file took place (length of 1st .m2ts file ~1:30 mins).

im currently using v1.23 to convert to flac again and will mux it all again together to see whether it might still crash then.

[madshi]

Originally Posted by Thunderbolt8

I have a problem with ratatouille remuxing (AVC, LPCM).
the movie has seamless branching so I joined each .m2ts part via "copy /b..". after that I used xport to extract video and audio from the joined .m2ts file and used eac3to to convert the LPCM stream (24-bit) to FLAC (everything fine).
but after I remuxed both together with mkvmerge I get an audio problem at playback (ffdshow, coreflac): audio suddenly disappeares at ~53 seconds movietime and also doesnt come back later on, shortly after that mpc crashes.
im not quite sure whether this could be a problem of the new eac3to version or because of seamless branching joining; however the error occurs before the first switch from the 1st to 2nd .m2ts file took place (length of 1st .m2ts file ~1:30 mins).

im currently using v1.23 to convert to flac again and will mux it all again together to see whether it might still crash then.

What happens if you play the external FLAC file? Do you get audio after those 53 seconds? Does madFlac also fail to work? Why are you using ffdshow/coreflac? madFlac should be better...

Usually after having joined TS files you need to run TsRemux over the result. However, I'm not sure if that does anything to audio. Maybe as a result of the seamless branching the joined audio data is corrupt. Would be strange but not impossible. E.g. if you add one byte in the middle of a valid PCM file you'll get totally trash sound after that byte.

[nautilus7]

Originally Posted by madshi

You can PM me the bug report or you can post it into gmail. Once the bug report window is visible on screen, the bug report is automatically stored into the clipboard. So just open PM or gmail and press Ctrl+V.

OK, i got it.

I also think a rename of eac3to is needed.

[madshi]

Originally Posted by Mtz

Still not working. Error log sent via PM. The problem appear after: Encoding AC3...
Sonic decoder version 4.2.

It seems that libAften is crashing for you. It's probably a multithreaded related bug. I'll switch libAften.dll to single threading in the next build, just to be safe.

[madshi]

New eac3to name suggestions:

- madAvConvert(er) - (AV for AudioVideo)
- madHcConvert(er) - (Hc for HomeCinema)
- madConvert(er)

The AudioVideo just in case. If I name it "AudioConverter" now, I'll have to change again in case I should ever decide to add some kind of video processing.

Your opinions? Any other suggestions?

[Thunderbolt8]

Originally Posted by madshi

What happens if you play the external FLAC file? Do you get audio after those 53 seconds? Does madFlac also fail to work? Why are you using ffdshow/coreflac? madFlac should be better...

Usually after having joined TS files you need to run TsRemux over the result. However, I'm not sure if that does anything to audio. Maybe as a result of the seamless branching the joined audio data is corrupt. Would be strange but not impossible. E.g. if you add one byte in the middle of a valid PCM file you'll get totally trash sound after that byte.

afaik madflac was only for external flac files, but my file was inside the .mkv
I tried to mux the flac file into .mka and wanted to play it outside, but I got an error directly before at the beginning of the parsing stage (and strangely enough this error was not reported or didnt stop the muxing of the previous .mkv with video and audio together).

[madshi]

Originally Posted by Thunderbolt8

afaik madflac was only for external flac files, but my file was inside the .mkv
I tried to mux the flac file into .mka and wanted to play it outside, but I got an error directly before at the beginning of the parsing stage (and strangely enough this error was not reported or didnt stop the muxing of the previous .mkv with video and audio together).

The newer versions of madFlac support both external files and also files inside of MKV or other containers.

[Thunderbolt8]

Originally Posted by madshi

The newer versions of madFlac support both external files and also files inside of MKV or other containers.

tried to use madflac instead of coreflac for the remuxed audio/video file, but it gave this grey filter screen directly at the beginning and no sound filter at all was loaded then.

the mkvmerge remux with the v1.23 flac file just finished parsing successfully, will soon find out whether it will work together with the video.