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Old 27th June 2013, 15:23   #12361  |  Link
yesgrey
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Originally Posted by yesgrey View Post
I wanted to start using flaccl instead of flac inside eac3to, because it will save me a lot of time, but I want to be sure that it gives accurate results, hence why I wanted to use eac3to to convert a flaccl generated file to a flac eac3to flac, to compare with the original one created with eac3to, which I consider as the reference.

The tests I did using graphstudio and madflac to output pcm files were not very good to flaccl, because not only the files are different than the eac3to generated one, but they are also different between them if I compare two different compression levels. It seems flaccl is messing the data in someway...
Just to let you know that the latest flaccl version fixed the problems with the multiple channels files. Now eac3to can read the files and I have checked with eac3to to decode and then re-encode the flaccl created files and they were exactly the same as the ones created directly with eac3to.

Now I can remux my movies much faster...
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Old 1st July 2013, 21:43   #12362  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
Video frames have a duration of 40 ms (25 fps) then delays of +/- 20 ms are absolutely undetectable.
From an old post, however I couldn't find the answer to this:
What is the video frame duration @23.976 fps?

Next question. I extracted a Dolby THD track to wavs, where the track has a video delay of -32ms.

The log shows (note lines 2 and 7):

Code:
[a03] Extracting audio track number 3...
[a03] This track begins with a non-major frame. <WARNING>
[a03] Extracting TrueHD stream...
[a03] Removing TrueHD dialog normalization...
[a03] Decoding with DirectShow (Nero Audio Decoder 2)...
[a03] DirectShow reports 5.1 channels, 24 bits, 48kHz
[a03] Applying RAW/PCM delay...
[a03] Writing WAVs...
I then converted the wavs to DTS-MA.

So, does the "applying RAW/PCM delay..." mean that I don't have to apply any delay when muxing the DTS track back with the video?
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Old 1st July 2013, 21:56   #12363  |  Link
tebasuna51
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Quote:
Originally Posted by rapscallion View Post
What is the video frame duration @23.976 fps?
1000 ms / 23.976 = 41.7 ms

Quote:
So, does the "applying RAW/PCM delay..." mean that I don't have to apply any delay when muxing the DTS track back with the video?
Correct.
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Old 1st July 2013, 22:18   #12364  |  Link
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@tebasuna51

As always, thank you for the quick and knowledgeable response.

I'm always a bit confused by audio/video delays.
Also, this is the first time I've seen "This track begins with a non-major frame. <WARNING>"
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Old 2nd July 2013, 08:43   #12365  |  Link
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Also, this is the first time I've seen "This track begins with a non-major frame. <WARNING>"
I can't help you with this.
I don't know TrueHD specs.
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Old 2nd July 2013, 20:41   #12366  |  Link
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A TrueHD stream is composed of Major sync frames and non-major frames. Apparently tracks are supposed to start with a major frame.

This may have accounted for the -32ms delay and the warning.
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Old 3rd July 2013, 21:10   #12367  |  Link
DoctorM
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Love how useful this tool is. I have 2 feature requests:

1) A SoundTouch type Tempo adjustment. The ability to speed up and a slow down a track without any change to the pitch is sometimes preferable.

2) Dolby Pro Logic II/IIx/IIz Upmix. I often want to expand the number of tracks on a stereo source to 5.1 or 6.1. Manually doing this is labor intensive and requires multiple programs and in between conversions. Be nice to just: Eac3to StereoTrack.wav SurroundTrack.ac3 -upDpl6
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Old 3rd July 2013, 21:39   #12368  |  Link
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Quote:
Originally Posted by DoctorM View Post
Love how useful this tool is. I have 2 feature requests:

1) A SoundTouch type Tempo adjustment. The ability to speed up and a slow down a track without any change to the pitch is sometimes preferable.

2) Dolby Pro Logic II/IIx/IIz Upmix. I often want to expand the number of tracks on a stereo source to 5.1 or 6.1. Manually doing this is labor intensive and requires multiple programs and in between conversions. Be nice to just: Eac3to StereoTrack.wav SurroundTrack.ac3 -upDpl6
Those features are non-trivial to implement I.M.H.O., BUT if you donate some hundreds or thousands of $$$ to madshi,
then possibly he'll do that for you

Last edited by filler56789; 3rd July 2013 at 21:44. Reason: typo
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Old 3rd July 2013, 22:01   #12369  |  Link
Nico8583
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Hi
Any news about fix for left view / right view with MVC flag ?
This case is open in Mantis : http://bugs.madshi.net/view.php?id=86
Thanks !
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Old 4th July 2013, 06:08   #12370  |  Link
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Thanks, filler56789, float me a loan? ;-)
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Old 5th July 2013, 06:08   #12371  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
Like I never see this kind of files please confirm:
- If this stream can be muxed in any container and extracted by eac3to.
- If eac3to can decode this stream.


OK, if we can call DTS Express like DTS High Definition.
BTW, DTS Express was already listed.



** MPEG1 = .mpeg1
I don't know, never tested.

** MS/VFW/FOURCC/XVID =
eac3to can't open avi container and can't extract this stream if present in mkv container.

** AC3 Surround = .ac3
Is a particular case of standard AC3, if you list this item separately seems you suggest than eac3to can apply a DPL decoder and is not true.

** DTS-HD (no core) = .dts
I don't know, never tested.

** RAW/PCM = .pcm/.raw
Both formats useless, must be extracted like .wav/.w64/.rf64

** DTSWAV = .dts
eac3to can't output .dtswav, only can convert DTSWAV to .dts

** AC3WAV = .ac3
eac3to can't output .ac3wav, only can convert AC3WAV to .ac3

** MP1 = .mp1
I don't know, never tested.

** (VobSub) =
eac3to can't extract this kind of sub.
OK thanks tebasuna51
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Old 5th July 2013, 06:17   #12372  |  Link
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Quote:
Originally Posted by Nico8583 View Post
Hi
Any news about fix for left view / right view with MVC flag ?
This case is open in Mantis : http://bugs.madshi.net/view.php?id=86
Thanks !
Temporary While waiting for a update eac3to
Small tool for detect AVC/MVC <=> MVC/AVC order
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Old 10th July 2013, 21:19   #12373  |  Link
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EDIT - Careful searching found that for any 6.1 format the remap must be done to accomplish -double7 processing correctly.

Hey everyone,

I have a 6.1 channel .w64 file with L C R LFE BC SL SR channel mapping, and I need to upmix to a 7.1 .w64 file for proper muxing with TSMuxer (TSM does not like 6.1, and converts it to 7.0, putting my LFE channel in one of my rear surrounds). Can I simply use the -double7 feature, or do I need to remap channels as I have found when searching for how to use the '-double7' feature?

I was wanting to just use:

eac3to input.w64 output.w64 -double7

Will this work?

Thanks in advance,

JSS

Last edited by maxmercy; 10th July 2013 at 21:47.
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Old 11th July 2013, 08:16   #12374  |  Link
tebasuna51
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After my test seems you need:

eac3to input.w64 output.w64 -0,1,2,3,5,6,4 -double7
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Old 12th July 2013, 12:33   #12375  |  Link
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Thanks! It worked great!

JSS
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Old 12th July 2013, 15:52   #12376  |  Link
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Would be possible implement option for output wav on 32 bit float??
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Old 14th July 2013, 17:00   #12377  |  Link
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Could we please get a replacement of the FLAC encoder with the one in ffmpeg 2.0 with "-compression_level 12" ?

It now properly supports 24bit. I have tested up to 24bit 7.1 channel FLAC and it plays fine on all the players I have tried (XBMC, MPC-HC, etc...).

"-compression_level 12" allows to shave off another 100 to 200Mb on my MKV backups.

It's a little cumbersome to compress to FLAC with eac3to and then to recompress it again to FLAC with compression level 12 with ffmpeg. It would be nice if we could do it in one go.

Thanks.
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Old 14th July 2013, 17:11   #12378  |  Link
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Can't you utilise piping to ffmpeg?
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Old 14th July 2013, 17:53   #12379  |  Link
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Quote:
Originally Posted by Boulder View Post
Can't you utilise piping to ffmpeg?
I am using the HD-DVD/Blu-Ray Stream Extractor (the MeGUI version as it allows to queue tasks - this way I can set up 20 or more MKV to FLAC taks in the queue and have it run in one go), so I don't think it's possible to setup piping with the gui.

Thanks for the feedback.
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Old 14th July 2013, 19:31   #12380  |  Link
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I tested simple piping to ffmpeg and it doesn't seem to work properly :/ The resulting file is only a part of the source, maybe it's due to some filesize restrictions. I couldn't find a way to force ffmpeg to ignore any audio length. The documentation of that application is just not very clear (I've ran into that many times in the past), but some googling told me that ffmpeg should support large files without a problem. There is something funny there anyway, I also tried decoding to wav and then using ffmpeg on the result but it didn't help either.
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