eac3to - audio conversion tool

[qyot27]

What?
Don't exist difference between Dolby 5.1 and DTS 5.1 or any 5.1 config, is always 3 front channels, 2 surround channels and LFE.
The Surround channels must be at +-120º from Central channel (more or less)

With 7.1 config the order in wav files is: FL FR FC LF BL BR SL SR
and mean:
BL BR = Back Channels at 150º (more or less)
SL SR = Side Channels at 90º (more or less)

Like you can see don't exist (in wav specs) a exact correspondence with Surround channels in 5.1 (120º)
Before XP SP2 the FL FR FC LF BL BR maskchannels was used for 5.1, after the FL FR FC LF SL SR was recommended but the previous one is also accepted.
I figured that the difference between Back and Side channels were their angle compared to the Center, and that the difference does matter in 7.1 far more than it does in 5.1.

I took that Dolby/DTS snippet from one of Microsoft's pages detailing Channel Masks (http://www.microsoft.com/windows/windowsmedia/howto/articles/audio9procmmt.aspx#understandingchannelmasks) (albeit in the case of WMA 9 Pro):

"The 5.1 and 7.1 Dolby configurations make use of SL and SR speakers, while the 5.1 DTS system, the default used by Windows Media, places the side speakers in the back (BL and BR). The 7.1 SDDS configuration, which is the default used in Microsoft Windows® XP, makes use of five front speakers. Although this configuration is used commonly in large commercial theaters, it may not provide the best experience in a home system."

[tebasuna51]

I took that Dolby/DTS snippet from one of Microsoft's pages detailing Channel Masks (http://www.microsoft.com/windows/windowsmedia/howto/articles/audio9procmmt.aspx#understandingchannelmasks)

I never see this:
5.1 (DTS) FL, FR, C, LFE, BL, BR (63)
5.1 (Dolby) FL, FR, C, LFE, SL, SR (1551)

Of course is a big absurdity, the users don't need move the surround speakers to 90º whe listen AC3 and to 140º when listen DTS.

[Momber]

TMT 3 has version 1.1.0.1 of "dtsdecoderdll.dll"
TMT 3.5 has version 1.1.0.10 of "dtsdecoderdll.dll"

have you tried this new versions?

the latest 1.1.0.10 doesn't work with eac3to though. anyway, it's interesting if they fixed the problem with DTS-HD MA tracks reported by eac3to as "7.1 (strange setup)". as you know such tracks are not bit-perfect decoded by old 1.1.0.0 versions of "dtsdecoderdll.dll"
Ugh! So what is the version of choice we should all use?
I just confirmed I'm still on 1.1.0.0 - will I need to update? Can I replace just this dll (I don't really want to update the whole TMT software suite, because they keep on getting worse as the build numbers progress).

[Chumbo]

@madshi,
When you have time and get back to eac3to, would you be kind enough to add a switch to be able to turn off the audio alerts please? I was watching one of my shows with eac3to running some operations in the background and it was really annoying to hear the alerts over the program. Thank you.

[Atak_Snajpera]

remove all .wavs

[TinTime]

If you delete the success and error wavs then this will silence eac3to.

EDIT - Atak_Snajpera beat me to it!

[hubblec4]

@madshi,
When you have time and get back to eac3to, would you be kind enough to add a switch to be able to turn off the audio alerts please? I was watching one of my shows with eac3to running some operations in the background and it was really annoying to hear the alerts over the program. Thank you.

delete the success.wav and the error.wav

[hoju3508]

eac3to is reporting 189 chapters when there's only 16 reported by BdInfo.

What kind of information should I send to madshi?



eac3to v3.16
command line: "C:\RipBot264v1.14.2\Tools\eac3to\eac3to.exe" "D:\" 1)
------------------------------------------------------------------------------
M2TS, 1 video track, 6 audio tracks, 17 subtitle tracks, 1:29:33, 24p /1.001
1: Chapters, 189 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
4: TrueHD/AC3, French, 5.1 channels, 48khz, dialnorm: -26dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -26dB)
5: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -26dB
6: TrueHD/AC3, Portuguese, 5.1 channels, 48khz, dialnorm: -26dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -26dB)
7: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
8: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
9: Subtitle (PGS), English
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), French
13: Subtitle (PGS), Spanish
14: Subtitle (PGS), Portuguese
15: Subtitle (PGS), Arabic
16: Subtitle (PGS), Dutch
17: Subtitle (PGS), French
18: Subtitle (PGS), Spanish
19: Subtitle (PGS), Portuguese
20: Subtitle (PGS), Dutch
21: Subtitle (PGS), English
22: Subtitle (PGS), French
23: Subtitle (PGS), Spanish
24: Subtitle (PGS), Portuguese
25: Subtitle (PGS), Dutch

[setarip_old]

Can "eac3to" be used to "timestretch" (or shorten) an .AC3 audiostream of a BluRay?

If so, how?

[buzzqw]

-slowdown or speeup

BHH

[setarip_old]

@buzzqw

Hi!

Thanks for the suggestion(s), but based on their definitions ("-slowdown convert 25.000 and 24.000 content to 23.976 fps"
"-speedup convert 23.976 and 24.000 content to 25.000 fps") they appear to be specifically for NTSC to PAL and PAL to NTSC conversion. (If my interpretation is incorrect, please advise)

What I'm seeking at the moment is to evenly, over the duration of the audiostream, shorten the audiostream by 700ms...

[rapscallion]

Ugh! So what is the version of choice we should all use?
I just confirmed I'm still on 1.1.0.0 - will I need to update? Can I replace just this dll (I don't really want to update the whole TMT software suite, because they keep on getting worse as the build numbers progress).
See the 2nd link, in the quote, below. If your's is dated 4/25/08 vs 4/21/08, then you're OK. If not, then you should update to the dll ver 1.1.0.1

You need use ArcSoft decoder.

This:
"Using eac3to with ArcSoft decoder there are a problem, the Side and Back channels are interchanged" (http://forum.doom9.org/showthread.php?p=1265484#post1265484)

Was solved with a more recent version of dtsdecoderdll.dll (http://forum.doom9.org/showthread.php?p=1266679#post1266679) <-----

[rapscallion]

Hey guys, I'm looking for a simple program that will display volume levels of an audio track (DTS, AC-3) graphically, I suppose.
There's a VU meter built into either VLC or MPC, but it's indicators hardly move so there's no easy way to compare tracks. I've done a search and can't find anything. Thanks

[Chumbo]

remove all .wavs

If you delete the success and error wavs then this will silence eac3to.

EDIT - Atak_Snajpera beat me to it!

delete the success.wav and the error.wav

Thank you all. :) Believe it or not, I actually thought of that after I posted the message. ;) It would still be a useful switch though. I find the audiible warnings useful except under certain circumstances. It would be easier to not have to delete or rename them just for those cases.

[Midzuki]

Hey guys, I'm looking for a simple program that will display volume levels of an audio track (DTS, AC-3) graphically, I suppose.
There's a VU meter built into either VLC or MPC, but it's indicators hardly move so there's no easy way to compare tracks. I've done a search and can't find anything. Thanks

AC3Filter and/or MatrixMixer should be able to do the trick.

But if you prefer the stupid nerdish methods :D ,
you could as well "AudioDub" a BlankClip and apply
Histogram("AudioLevels") onto it. :)

{ http://avisynth.org.ru/docs/english/corefilters/histogram.htm }

[rapscallion]

Thanks ! MatrixMixer works great.

[htpc@rischer.at]

1.0 Channels 64 kpbs
Hi!

I want to demux the german audio E-AC3 1.0 64 kbps.
I tried to convert to .wav and to .ac3. Everytime ffmeg allerts that it cannot do it.
Have anyone an idea?

command line: eac3to h: bruce.log
------------------------------------------------------------------------------
EVO, 1 video track, 6 audio tracks, 12 subtitle tracks, 1:42:33
"Feature Presentation"
1: Joined EVO file
2: Chapters, 29 chapters with names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: E-AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: E-AC3, French, 1.0 channels, 64kbps, 48khz, dialnorm: -27dB
6: E-AC3, German, 1.0 channels, 64kbps, 48khz, dialnorm: -27dB
7: E-AC3, Italian, 1.0 channels, 64kbps, 48khz, dialnorm: -27dB
"Italian"
8: E-AC3, Spanish, 1.0 channels, 64kbps, 48khz, dialnorm: -27dB
"Cast"
9: E-AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
"Producer Commentary"


This now decodes properly with libavcodec from FFmpeg's current SVN.

Thanks madshi for contacting me directly about this file. It does use spectral extension. I had implemented the feature in the "sandbox" repository from the original Google Summer of Code project, but had not merged to FFmpeg SVN because I only had 1 sample file to test... until now!! With this sample and another (also provided by madshi) working properly, I decided to merge the changes to FFmpeg SVN.

@jruggle:
Do you remember my problem?
Do you have some informations how do convert the file into another format like .wav?
I tried it with ffmpeg 0.5, but i failed.

thank you very much...

[Thunderbolt8]

how can I transform a .thd track to .dts again? when I enter it that way 'eac3to source.thd dest.dts' then I get 6 wav files first, but which are apparently not accepted by the surcode dts encoder and the process is cancelled.

[AnryV]

How can I disable "Patching bitdepth to 24 bits..." during extracting Core from DTS-HD?

[christinedawson]

Thanks for the tool.

It's good to know that it's finally possible to convert TrueHD.
As I've been playing around with Nero these past days, could you tell what sort of working graph permit the decoding of TrueHD ?
Is it Nero Splitter -> Nero Audio Decoder 2 ?
Do you need to specify anything in the Nero Audio Decoder 2 options ?

[tebasuna51]

How can I disable "Patching bitdepth to 24 bits..." during extracting Core from DTS-HD?
I can't reproduce your problem extracting dts core:
eac3to v3.16
command line: "D:\eac3to.exe" "D:\Queen.dtshd" "D:\Queen_core.dts"
------------------------------------------------------------------------------
DTS Hi-Res, 5.1 channels, 0:00:14, 24 bits, 3018kbps, 96khz
(core: DTS, 5.1 channels, 0:00:14, 24 bits, 1509kbps, 48khz)
Creating file "D:\Queen_core.dts"...
eac3to processing took 1 second.
Done.
Please :logfile:

[rapscallion]

I've been playing around with Eac3to More GUI and find it interesting that a AC-3 640/448 track can be converted to a DTS 1536/768 track. (decoding with Nero and encoding w/Surcode)

If a BD souce disc has only a AC-3 track, under what circumstances would one want to convert it to DTS and what kind of quality, by comparison, would result. (I'm assuming that the original AC-3/ 640 is lower quality than DTS/ 1536...but ,of course realize there are many opinions on that)

[TinTime]

If you convert AC3 to DTS then you will lose quality. It's not a question of DD vs. DTS.

The only circumstance you'd want to do this is for compatibility reasons. I doubt that there are any devices that can decode DTS but not AC3 though.

[rapscallion]

Ok, thanks, was really curious about this.

[rack04]

Is Nero the only AAC decoder that will work with eac3to?

[wswartzendruber]

How does eac3to find the codecs? Can I just drop the ArcSoft DTS-HD files in there and call it good, or do I have to install it? I know how to use regsvr32.

[tebasuna51]

How does eac3to find the codecs? Can I just drop the ArcSoft DTS-HD files in there and call it good, or do I have to install it? I know how to use regsvr32.
Read the first post in this thread.

[tebasuna51]

Is Nero the only AAC decoder that will work with eac3to?
Yes.
See this post. (http://forum.doom9.org/showthread.php?p=1262463#post1262463)

[wswartzendruber]

I'm wondering what the channel mapping will be of this output file:

eac3to input.thd output.raw
I can't seem to figure it out.

:confused:

[tebasuna51]

The raw audio data must be identical than wav or w64, then the channel map must be FL,FR,FC,LF,BL,BR,SL,SR (if 7.1)

[wswartzendruber]

The raw audio data must be identical than wav or w64, then the channel map must be FL,FR,FC,LF,BL,BR,SL,SR (if 7.1)
Thanks, I've actually spent those few hours chasing an MPlayer bug. :mad:

GStramer seems to handle multichannel FLACs correctly.

[jruggle]

@jruggle:
Do you remember my problem?
Do you have some informations how do convert the file into another format like .wav?
I tried it with ffmpeg 0.5, but i failed.

thank you very much...
The patch for FFmpeg to fix this is still pending final review.
This is the most recent patch if you want to try to apply it and compile your own binary:
http://article.gmane.org/gmane.comp.video.ffmpeg.devel/92029

[Thunderbolt8]

how can I transform a .thd track to .dts again? when I enter it that way 'eac3to source.thd dest.dts' then I get 6 wav files first, but which are apparently not accepted by the surcode dts encoder and the process is cancelled.anyone able to tell?

eac3to v3.16
command line: eac3to E:\input.thd E:\output.dts
------------------------------------------------------------------------------
TrueHD, 5.1 channels, 48khz
Decoding with libav/ffmpeg...
Writing WAVs...
Creating file "E:\output.R.wav"...
Creating file "E:\output.SR.wav"...
Creating file "E:\output.L.wav"...
Creating file "E:\output.C.wav"...
Creating file "E:\output.LFE.wav"...
Creating file "E:\output.SL.wav"...
[libav] End of stream indicated <WARNING>
The original audio track has a constant bit depth of 16 bits.
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.29.0.
Surcode says/asks: "At least one valid source file must be specified to encode."
.
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

[regisso]

anyone able to tell?

eac3to v3.16
command line: eac3to E:\input.thd E:\output.dts

I think it's because you don't have the DD core in your thd file, you should make a thd+ac3 file.

[TinTime]

anyone able to tell?

eac3to v3.16
command line: eac3to E:\input.thd E:\output.dts
------------------------------------------------------------------------------
TrueHD, 5.1 channels, 48khz
Decoding with libav/ffmpeg...
Writing WAVs...
Creating file "E:\output.R.wav"...
Creating file "E:\output.SR.wav"...
Creating file "E:\output.L.wav"...
Creating file "E:\output.C.wav"...
Creating file "E:\output.LFE.wav"...
Creating file "E:\output.SL.wav"...
[libav] End of stream indicated <WARNING>
The original audio track has a constant bit depth of 16 bits.
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.29.0.
Surcode says/asks: "At least one valid source file must be specified to encode."
.
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

If you convert to wavs using eac3to and encode manually with Surcode does it work?

[Thunderbolt8]

you mean manually encode with surcode by using that program? perhaps, but I don't want to have it like that. I remember I already had the same problem, I just dont know any more how I solved it back then :S

I think it's because you don't have the DD core in your thd file, you should make a thd+ac3 file.would the ac3 track as lossy file be needed for that? shouldnt the same information be included in the lossless .thd anyway?

[TinTime]

you mean manually encode with surcode by using that program? perhaps, but I don't want to have it like that. I remember I already had the same problem, I just dont know any more how I solved it back then :S

I didn't mean as a permanent solution :) It was just to test this particular file and try to narrow down the problem. Is it the wavs eac3to produces or the way eac3to calls Surcode or Surcode itself?

Does it work if you go from the original source rather than demuxing the thd track first?

[pascalwil]

Anyone knows how to use Surcode DVD DTS Encoder in command line mode instead of using the GUI? Thanks.
eac3to does it this way. I would love to have an example of the script to use. Many thanks.

[rapscallion]

I'm trying to extract the True HD (embedded core) audio stream from a *.m2ts file, using Eac3to and More GUI.

The "catch22" I'm running into is that the only way it will demux the correct stream (there's also a ac3/640 stream) is if I select "thd" as the save to file type. Then it extracts with a "thd" file extension, which Tsmuxer does not recognize. If I change the extension to "ac3", Tsmuxer says unsupported format. If I try to extract the stream as a ac3 file extension then the ac3/640 stream is extracted.

Yet, If I demux , using Tsmuxer, the THD stream is extracted correctly w/a ac3 file extension. Anyone's comments as to what the problem is with eac3to would be appreciated. Maybe I should be using a different/better GUI ?

[TinTime]

I'm trying to extract the True HD (embedded core) audio stream from a *.m2ts file, using Eac3to and More GUI.

I don't use any eac3to GUI or Tsmuxer but if you have a TrueHD with embedded DD track then specifying output.thd in eac3to will demux just the TrueHD, output.ac3 will demux just the DD and output.thd+ac3 will demux the complete stream. Perhaps you need to do this last option?

[regisso]

output.thd+ac3 will demux the complete stream
That's what I do when I convert hd-dvd to BD with tsMuxer.

[TinTime]

I guess that means that Tsmuxer wants an AC3 core for its TrueHD tracks.

HD-DVD TrueHD streams don't contain an AC3 core IIRC. In this case eac3to will generate the core if output.thd+ac3 is specified.

[rapscallion]

I don't use any eac3to GUI or Tsmuxer but if you have a TrueHD with embedded DD track then specifying output.thd in eac3to will demux just the TrueHD, output.ac3 will demux just the DD and output.thd+ac3 will demux the complete stream. Perhaps you need to do this last option?
The problem is that the last option is NOT an option in Eac3to and More. Any suggestions for the program to extract that stream ?

[Snowknight26]

You can do thd+ac3 in eac3to. Sounds like a GUI problem, not an eac3to problem, which have their own threads.

[Inspector.Gadget]

Anyone know if TMT3 decoders currently work in eac3to? Thanks.

[spida_singh]

Anyone know if TMT3 decoders currently work in eac3to? Thanks.

did not work for me, so went back to tmt2....i don't think there is a differencein the decoder, but, in tmt3 it does not allow third party apps to make use of it....thats the last i knew anyway

[SomeJoe]

Does this happen on every movie with TrueHD or is from movie to movie with TrueHD i ask because im having what seems to be the same problem, theres movies that are messed up and others that are not and im using a Pioneer receiver aswell. If the audio has problems do you hear it right away meaning through out the movie or is it parts of the audio during the movie?

I take it you have no problems with DTS-HD Master or PCM audio when demuxed and then remuxed?

My apologies for the very late reply on this. Been very busy at work. :)

No, I have no problem with TrueHD tracks if they are decoded to PCM, those work perfectly. Also have never had any problem with any flavor of DTS (DTS, DTS-HD, DTS-MA).

There are some TrueHD tracks that seem to work, but most give distorted audio. The distortion varies, and there are portions of the audio where you probably can't tell at first listen, and then there are sections where there is obviously something wrong.

I use the Iron Man BD to test this. The very first scene is majorly affected - the dialogue/discussion in the jeep at the very start of the film is barely audible under the eac3to.exe demux/TSMuxer remux scenario.

However, I have indeed finally solved this problem. It is indeed eac3to.exe's issue, and the root of the problem is the removal of dialogue normalization.

Eac3to removes dialogue normalization by default. When it does that, the sound is distorted when played through my Pioneer receiver.

Adding "-keepdialnorm" on the eac3to.exe command line has completely fixed the problem. I've tried my particularly problematic TrueHD track (Iron Man) which was very obviously distorted when played back after a demux with eac3to.exe and a remux with TSMuxer. The TrueHD track now plays perfectly and identically to the original when the -keepdialnorm option is used during the eac3to.exe demux.

This must be something particular to Pioneer receivers, evidently they pay very close attention to the dialogue normalization metadata and get confused if the values are incorrect, missing, etc.

[rack04]

Why does eac3to remove the fullrange flag from h264 streams?

Because there are almost always wrong, mostly due to a bug in the old Tandberg h.264 encoder that most European tv stations use.

Is there a way to override the removal of the fullrange flag from h264 streams?

EDIT: Nevermind I found "-keepFullRange"

[LeXXuz]

When I use eac3to to demux evo-Containers with the -demux switch, eac3to demuxes not all the streams (especially subtitles with ID: 28 and up).

Is there a way to fix this or can I demux multiple streams at once WITHOUT converting them?

[picman1]

After ready through some of the posts on talking about TMT 2 and 3 to use for converting DTS 7.1 to LPCM 7.1 it sounds like if I don't have TMT 2 I cannot use TMT 3 to do the convertion? If that is the case does anyone know where I can get a copy (buy) of TMT 2? Thank you

[frenchglen]

Hey madshi and folks,

Is there a way to split an MLP file with eac3to if you have a cue sheet or somehow just typing in where (in time/samples etc) to split the audio file?

Thanks very much...Haven't used eac3to in a while - I can only imagine all the excellent new features since the last time I updated it!

[StephenB]

Anyone have any ideas on what might be wrong here.

eac3to v3.16
command line: "C:\VID2EVA\tools\eac3to\eac3to.exe" "F:\BDMV\STREAM" 1) "Z:\VIDEOS\Vid2eva.mkv" -seekToIFrames -stripPulldown -lowPriority
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 5 subtitle tracks, 1:44:05, 24p /1.001
1: Chapters, 24 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 6.1 channels, 24 bits, 48khz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48khz)
4: AC3, Spanish, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, -13ms
5: AC3, French, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, -13ms
6: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB, -19ms
7: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB, -19ms
8: Subtitle (PGS), English
9: Subtitle (PGS), French
10: Subtitle (PGS), Spanish
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), English
Creating file "Z:\VIDEOS\Vid2eva - Chapters.txt"...
[a03] The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
[a03] The libav DTS decoder doesn't decode the back channels. <WARNING>
[a03] The libav DTS decoder doesn't decode the full DTS-HD information. <WARNING>
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a05] Extracting audio track number 5...
[a06] Extracting audio track number 6...
[a06] Removing AC3 dialog normalization...
[a07] Extracting audio track number 7...
[s09] Extracting subtitle track number 9...
[a03] Extracting audio track number 3...
[s10] Extracting subtitle track number 10...
[s11] Extracting subtitle track number 11...
[s08] Extracting subtitle track number 8...
[s12] Extracting subtitle track number 12...
[a05] Removing AC3 dialog normalization...
[a04] Removing AC3 dialog normalization...
[a04] A remaining delay of -13ms could not be fixed.
[a07] Removing AC3 dialog normalization...
[a03] Extracting DTS core...
[a05] A remaining delay of -13ms could not be fixed.
[a06] Applying (E-)AC3 delay...
[a07] Applying (E-)AC3 delay...
[a07] A remaining delay of +13ms could not be fixed.
[a06] A remaining delay of +13ms could not be fixed.
[a03] Decoding with libav/ffmpeg...
[a03] Remapping channels...
[a03] Reducing depth from 64 to 24 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "Z:\VIDEOS\Vid2eva - 3 - DTS Master Audio, English, 6.1 channels, 24 bits, 48khz.flac"...
[a04] Creating file "Z:\VIDEOS\Vid2eva - 4 - AC3, Spanish, 5.1 channels, 448kbps, 48khz.ac3"...
[a05] Creating file "Z:\VIDEOS\Vid2eva - 5 - AC3, French, 5.1 channels, 448kbps, 48khz.ac3"...
[a06] Creating file "Z:\VIDEOS\Vid2eva - 6 - AC3, English, 2.0 channels, 224kbps, 48khz.ac3"...
[a07] Creating file "Z:\VIDEOS\Vid2eva - 7 - AC3, English, 2.0 channels, 224kbps, 48khz.ac3"...
[s12] Creating file "Z:\VIDEOS\Vid2eva - 12 - Subtitle (PGS), English.sup"...
[s08] Creating file "Z:\VIDEOS\Vid2eva - 8 - Subtitle (PGS), English.sup"...
[s10] Creating file "Z:\VIDEOS\Vid2eva - 10 - Subtitle (PGS), Spanish.sup"...
[a03] Clipping detected, a 2nd pass will be necessary. <WARNING>
[a03] The original audio track has a constant bit depth of 64 bits.
[a03] The processed audio track has a constant bit depth of 24 bits.
[s09] Creating file "Z:\VIDEOS\Vid2eva - 9 - Subtitle (PGS), French.sup"...
[s11] Creating file "Z:\VIDEOS\Vid2eva - 11 - Subtitle (PGS), Spanish.sup"...
[a03] The libav DTS decoder doesn't decode the back channels. <WARNING>
[a03] The libav DTS decoder doesn't decode the full DTS-HD information. <WARNING>
[a03] Starting 2nd pass...
[a03] Extracting audio track number 3...
[a03] Extracting DTS core...
[a03] Decoding with libav/ffmpeg...
[a03] Remapping channels...
[a03] Reducing depth from 64 to 24 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "Z:\VIDEOS\Vid2eva - 3 - DTS Master Audio, English, 6.1 channels, 24 bits, 48khz.flac"...
[a03] Writing the destination file failed. <ERROR>
[a03] One of the FLAC encoder's callbacks returned a fatal error. <ERROR>
Aborted at file position 56623104. <ERROR>

[tebasuna51]

Maybe eac3to try to encode a 6.1 flac but the decoder output (libav) is only 5.1

Try to decode first:
"C:\VID2EVA\tools\eac3to\eac3to.exe" "F:\BDMV\STREAM" 1) 3: "Z:\VIDEOS\Vid2eva.wav"
and encode to flac after:
"C:\VID2EVA\tools\eac3to\eac3to.exe" "Z:\VIDEOS\Vid2eva.wav" "Z:\VIDEOS\Vid2eva.flac"

But if you don't have the ArcSoft decoder you can't decode the DTS-MA and use flac like output is not necesary, you can convert the wav output to ac3:
"C:\VID2EVA\tools\eac3to\eac3to.exe" "Z:\VIDEOS\Vid2eva.wav" "Z:\VIDEOS\Vid2eva.ac3"
or extract the core DTS-ES, 6.1
"C:\VID2EVA\tools\eac3to\eac3to.exe" "F:\BDMV\STREAM" 1) 3: "Z:\VIDEOS\Vid2eva.dts" -core

[StephenB]

Maybe eac3to try to encode a 6.1 flac but the decoder output (libav) is only 5.1

Try to decode first:
"C:\VID2EVA\tools\eac3to\eac3to.exe" "F:\BDMV\STREAM" 1) 3: "Z:\VIDEOS\Vid2eva.wav"
and encode to flac after:
"C:\VID2EVA\tools\eac3to\eac3to.exe" "Z:\VIDEOS\Vid2eva.wav" "Z:\VIDEOS\Vid2eva.flac"

But if you don't have the ArcSoft decoder you can't decode the DTS-MA and use flac like output is not necesary, you can convert the wav output to ac3:
"C:\VID2EVA\tools\eac3to\eac3to.exe" "Z:\VIDEOS\Vid2eva.wav" "Z:\VIDEOS\Vid2eva.ac3"
or extract the core DTS-ES, 6.1
"C:\VID2EVA\tools\eac3to\eac3to.exe" "F:\BDMV\STREAM" 1) 3: "Z:\VIDEOS\Vid2eva.dts" -core

This same call works on other 6.1 DTS-MA files though.

[tebasuna51]

This same call works on other 6.1 DTS-MA files though.
Maybe, but with "core: DTS-ES, 6.1 channels"?

[picman1]

Ok. Since no one seems to know where I can get a TMT version 2 to decode DTS 7.1 to LPCM. Does anyone know if there is any other way to do this conversion without TMT 2? Thanks

[<><]

Ok. Since no one seems to know where I can get a TMT version 2 to decode DTS 7.1 to LPCM. Does anyone know if there is any other way to do this conversion without TMT 2? Thanks

There's a 30 day trial version available form here: http://www.avsforum.com/avs-vb/showthread.php?t=1033822, read #1 Exceptions

[picman1]

Great! Thank you. I'll give a try this weekend.

[StephenB]

Maybe, but with "core: DTS-ES, 6.1 channels"?

Good point. I don't know if I've seen a DTS-ES 6.1 core with DTS-MA before.

[Atak_Snajpera]

Ok. Since no one seems to know where I can get a TMT version 2 to decode DTS 7.1 to LPCM. Does anyone know if there is any other way to do this conversion without TMT 2? Thanks
ArcSoft Decoder in TMT 3.0 works as well with eac3to.

[picman1]

Thanks Atak that's good to know.

I downloaded the trial version and after trying different gui's and finally from the command line using these examples from a previous post :eac3to app.dts app.pcm -0,2,1,3,4,5,6,7
pcm2tsmu app.pcm output.pcm -c 8

I got a good 7.1 with correct speaker layouts.

Then I used Tsmuxer to bring the video and audio together and the video and audio looks and sounds great.



If I upgrade this version to TMT 3 will I have to use the same commands or is there a better way to get the correct 7.1 stream with the new version?

What version of TMT 3 will I need to upgrade to, standard, gold or platinum?

Thanks for the help.

[Fiffy]

ArcSoft Decoder in TMT 3.0 works as well with eac3to.Could you tell us how you made it work? Eac3to doesn't recognize the Arcsoft filters with a vanilla TMT3 install.

[xkodi]

Could you tell us how you made it work? Eac3to doesn't recognize the Arcsoft filters with a vanilla TMT3 install.

1) TMT2 installs dtsdecoderdll.dll version 1.1.0.0
2) TMT3 installs dtsdecoderdll.dll version 1.1.0.1
3) TMT 3.0.1.133 Patch installs dtsdecoderdll.dll version 1.1.0.5

http://www.arcsoft.com/downloads/digitaltheatre/arcsoft/totalmediatheatre_beta_3.0.1.120_3.0.1.133_update_all.exe

you need to have installation of TMT2 on which eac3to recognize the dtsdecoderdll.dll, i.e. "eac3to -test" returns:

ArcSoft DTS Decoder (1.1.0.0) works fine

then just replace dtsdecoderdll.dll version 1.1.0.0 with version 1.1.0.1 or version 1.1.0.5 from TMT3 and then when you run "eac3to -test" you should get:

ArcSoft DTS Decoder (1.1.0.1) works fine

or

ArcSoft DTS Decoder (1.1.0.5) works fine

4) TMT 3.5 installs dtsdecoderdll.dll version 1.1.0.10, but it doesn't work with eac3to.

BTW, there is also new CinemasterAudio.DLL version 4.3.0.236 from the new version 2.6.3 of SONIC CineVision, which works good with eac3to. if you look here:

http://forum.doom9.org/showthread.php?p=1158014#post1158014

and here:

http://forum.doom9.org/showthread.php?p=1212881#post1212881

you will see that this is the first new version of CinemasterAudio.DLL from very long time that works good with eac3to

[Fiffy]

then just replace dtsdecoderdll.dll version 1.1.0.0 with version 1.1.0.1 or version 1.1.0.5 from TMT3 and then when you run "eac3to -test" you should get:I don't understand. Are you saying I should replace the dll in a TMT2 installation with one from TMT3? That's not what I want. I want to switch to TMT3, but retain a DTS filter that is usable from eac3to. Before the current TMT3 beta, it was possible to keep TMT2 and 3 installed in parallel, but the beta (and presumably the upcoming official update) doesn't allow that anymore, so it is currently not easily possible to use TMT3 beta while keeping a filter that is usable from eac3to.

[Joniii]

I got some errors while ripping one Blu-ray. Do I need to worry about this (audio/video out of sync or something) or did eac3to fix everything?

1: MPEG2, 1080p24 /1.001 (16:9)
2: DTS Master Audio, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
[a02] Extracting audio track number 2...
[v01] Extracting video track number 1...
[a02] Extracting DTS core...
[v01] Creating file "e:\koh.mpeg2"...
[a02] Creating file "e:\koh.dts"...
[a02] [1:30:49] The source file seems to be damaged (discontinuity). <WARNING>
[v01] [1:30:49] The source file seems to be damaged (discontinuity). <WARNING>
[a02] [1:30:49] The source file seems to be damaged (discontinuity). <WARNING>
[a02] This track is not clean. <WARNING>
[a02] [3:09:33] The source file seems to be damaged (discontinuity). <WARNING>
[v01] Video overlaps for 14 frames at playtime 1:30:48. <WARNING>
[a02] Audio overlaps for 61ms at playtime 1:30:49. <WARNING>
[a02] Starting 2nd pass...
[a02] Realizing DTS gaps...
[a02] Creating file "e:\koh.dts"...
Video track 1 contains 272688 frames.
eac3to processing took 16 minutes, 34 seconds.
Done.

[Atak_Snajpera]

Could you tell us how you made it work? Eac3to doesn't recognize the Arcsoft filters with a vanilla TMT3 install.
You just need following files from TMT 3.0 package

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

Once you have them you can remove TMT from your computer if you don't like it.
Then copy all .dlls to System32 and register ASAudioHD.ax with regsvr32.exe

[rica]

Hi madshi, where the hell are you? :search: :)
We've all missed you here, at least me :D

_ _ _ _ _ _ _

[Thunderbolt8]

You just need following files from TMT 3.0 package

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

Once you have them you can remove TMT from your computer if you don't like it.
Then copy all .dlls to System32 and register ASAudioHD.ax with regsvr32.exeis there any advantage of using the tmt3 files compared to the standard tmt1 files (when not using the tmt prog itself)?

[jj666]

I got some errors while ripping one Blu-ray. Do I need to worry about this (audio/video out of sync or something) or did eac3to fix everything?

1: MPEG2, 1080p24 /1.001 (16:9)
2: DTS Master Audio, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
[a02] Extracting audio track number 2...
[v01] Extracting video track number 1...
[a02] Extracting DTS core...
[v01] Creating file "e:\koh.mpeg2"...
[a02] Creating file "e:\koh.dts"...
[a02] [1:30:49] The source file seems to be damaged (discontinuity). <WARNING>
[v01] [1:30:49] The source file seems to be damaged (discontinuity). <WARNING>
[a02] [1:30:49] The source file seems to be damaged (discontinuity). <WARNING>
[a02] This track is not clean. <WARNING>
[a02] [3:09:33] The source file seems to be damaged (discontinuity). <WARNING>
[v01] Video overlaps for 14 frames at playtime 1:30:48. <WARNING>
[a02] Audio overlaps for 61ms at playtime 1:30:49. <WARNING>
[a02] Starting 2nd pass...
[a02] Realizing DTS gaps...
[a02] Creating file "e:\koh.dts"...
Video track 1 contains 272688 frames.
eac3to processing took 16 minutes, 34 seconds.
Done.

Rerip your disk, I'd be concerned with such a log.

-jj-

[Thunderbolt8]

Is there a way to override the removal of the fullrange flag from h264 streams?

EDIT: Nevermind I found "-keepFullRange"what is the fullrange flag actually for, whats its purpose?

[73ChargerFan]

It specifies that the video was encoded to use 0-255 instead of 16-235 values.

[Inspector.Gadget]

You just need following files from TMT 3.0 package

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

Once you have them you can remove TMT from your computer if you don't like it.
Then copy all .dlls to System32 and register ASAudioHD.ax with regsvr32.exe

I did this on Vista x64 because I hate having TMT3 installed - I put all dlls in both SysWOW64 and System32 and then registered the filter in System32 - and eac3to doesn't see Arcsoft DTS decoder as installed. Argh! Any suggestions?

[Kythe]

Guys, I'm getting an odd error. It's cropped up every now and then, and I'm wondering whether anyone else has seen it...

When demuxing an HD track (either DTS Master Audio or TrueHD) to flac from a movie file, the decoding sometimes quits with the error "The FLAC decoder thread seems to hang". This happened most recently during the second pass of a demux. Eac3to noted multiple overlaps in the demuxed audio track.

Does anyone have any idea what could be causing this, and how to fix it? I'm running this on older hardware (a single core, 1 GHz Pentium).

[drlove]

Hi, after using eac3to w/o -keepdiagnorm my TrueHD track has no diagnorm info at all and my A/V interpret that as -0dB and normalize to -31dB = the volume is very low. If I run the eac3to ones again on the track the log shows no diagnorm value at all, shouldn't eac3to sett the diagnorm to -31dB and not remove the value?
Can I add diagnorm info, of say -27dB, to a TrueHD track using eac3to? I don't have the source at my summer house so I can't redo the rip from the source BD at the moment. I only have the TrueHD track and the video file at my laptop/PCH.

Best regards L.

[DoomBot]

Hi, after using eac3to w/o -keepdiagnorm my TrueHD track has no diagnorm info at all and my A/V interpret that as -0dB and normalize to -31dB = the volume is very low. If I run the eac3to ones again on the track the log shows no diagnorm value at all, shouldn't eac3to sett the diagnorm to -31dB and not remove the value?
Can I add diagnorm info, of say -27dB, to a TrueHD track using eac3to? I don't have the source at my summer house so I can't redo the rip from the source BD at the moment. I only have the TrueHD track and the video file at my laptop/PCH.

Best regards L.

That would be nice if you could restore diagnorm info some how, if it could be done how do we know what the db would be to restore it after it was removed in the past. Is it always -27db or what?

[Atak_Snajpera]

I did this on Vista x64 because I hate having TMT3 installed - I put all dlls in both SysWOW64 and System32 and then registered the filter in System32 - and eac3to doesn't see Arcsoft DTS decoder as installed. Argh! Any suggestions?

1) Copy all dlls to SysWOW64
2) Register: x:\Windows\sysWOW64\regsvr32.exe x:\Windows\sysWOW64\ASAudioHD.ax

Still nothing?

[Inspector.Gadget]

That's exactly what I did. Weird. I'll try it again after reinstalling TMT3 and see if it's some sort of conflict. Thanks for the help.

[Atak_Snajpera]

I also tested on W7 x64. No problems.

[Inspector.Gadget]

Thanks for the help, Atak, but it's not working. When I have TMT3 installed, eac3to can't find the DTS decoder. When I uninstall it, reboot, and copy the relevant DLLs and decoder into SysWOW64 and use exactly the commands above, I get a success message but eac3to still can't find the decoder. I'm beginning to think that this is some sort of version incompatibility.

[yesgrey]

When backing up the movies, I would like to remove the final credits, so I can save some space. If I use the -edit option I know I can remove the last part of the audio data, but is there in eac3to (or any other tool) an option for removing the last part of the video data?

[honai]

Not yet. madshi once said that he'll consider it for his todo list.

[yesgrey]

Do you know if is there any other tool that could do it? cut the last part of a n h264 video file?

[TinTime]

Do you know if is there any other tool that could do it? cut the last part of a n h264 video file?

What are you backing up to? If it's in an mkv container you can split it with mkvmerge.

[yesgrey]

What are you backing up to? If it's in an mkv container you can split it with mkvmerge.
That's good enough for what I need. Since I have to use mkvmerge to join the video/audio/subs/chapters, it would be the best solution... I should have looked more carefully in mkvmerge before asking for it...:o

Thank you very much!:)

[Cela]

eac3to does not like most of my "Xvid.avi" recordings produced with Total Recorder.
As a consequence, multiAVCHD and tsMuxeR produce issues.

A short example for debugging is here (ttp://rapidshare.com/files/252569731/Birds.rar) (9606 KB).
A longer example (application of eac3to within multiAVCHD, uncrop) is here (http://rapidshare.com/files/252553272/Xvid_uncrop_audio_and_video_issues.rar) (19331 KB).

Now, the multiAVCHD workflow must be interrupted since eac3to refuses to demux these files. Demuxing must be performed with VirtualDub. VirtualDub's wav must be seperately imported into multiAVCHD.
RipBot's ffmpeg demuxing finishes, but may also produce issues (sometimes adds weird clicks as 'watermarks').

VirtualDub has no problem to edit, append and demux these avi files, though.

It would be a great advantage for the workflow, if eac3to did not refuse the demuxing of these "Xvid.avi" recordings.

Please upgrade eac3to/eac3to_more_gui accordingly.

Thanks.

[tebasuna51]

@Cela
You don't need samples to prove this, eac3to (or TsMuxer) don't support avi container.

[Cela]

Yes, I was afraid to get this response.

But, why?
I would like it did. I would need it.

I wanted to do my share to get things moving.

Edit:
Dean, the author of multiAVCHD, adviced me to update to newest eac3to and to correct an input typo.
Now multAVCHD's workflow, using newest eac3to.exe, works like a charm! :)

And, within multiAVCHD, tsMuxeR does not need to deal with the original avi any more.

[LeXXuz]

When I use eac3to to demux evo-Containers with the -demux switch, eac3to demuxes not all the streams (especially subtitles with ID: 28 and up).

Is there a way to fix this or can I demux multiple streams at once WITHOUT converting them?

Any ideas?

[Moondust]

I've encountered a very strange problem with EAC3TO, at least to me. I loaded my copy of Ice Age 2 in eac3to. It said the length of the movie was 1h 30m 38s (all playlists). I ripped the Dutch 768kbps audiotrack. But when I played that seperate track in MPC-HC it was 1h 32m 15s. Other players and MediaInfo revealed the same. EAC3TO says the seperate audiotrack is 1h 30m 38s. I am confused. Who is right? And why?

EAC3TO source info.
D:\EAC3TO>eac3to e:\downloads\newsleecher 1)
M2TS, 1 video track, 5 audio tracks, 5 subtitle tracks, 1:30:38, 24p /1.001
1: Chapters, 24 chapters
2: MPEG2, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: DTS, Dutch, 5.1 channels, 24 bits, 768kbps, 48khz
5: DTS, Dutch, 5.1 channels, 24 bits, 768kbps, 48khz
6: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
7: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
8: Subtitle (PGS), English
9: Subtitle (PGS), Dutch
10: Subtitle (PGS), Dutch
11: Subtitle (PGS), Dutch
12: Subtitle (PGS), English

EAC3TO seperate audio track info.
D:\EAC3TO>eac3to e:\IceAge2Dutch.dts
DTS, 5.1 channels, 1:30:38, 24 bits, 768kbps, 48khz

[TinTime]

When I use eac3to to demux evo-Containers with the -demux switch, eac3to demuxes not all the streams (especially subtitles with ID: 28 and up).

Is there a way to fix this or can I demux multiple streams at once WITHOUT converting them?


You can demux streams by specifying the target as the same type as the source. For example, if track 3 is ac3, 4 is dts...

eac3to.exe source 3: output3.ac3 4: output4.dts

...will demux those tracks.

[AnryV]

I can't reproduce your problem extracting dts core:
eac3to v3.16
command line: "D:\eac3to.exe" "D:\Queen.dtshd" "D:\Queen_core.dts"
------------------------------------------------------------------------------
DTS Hi-Res, 5.1 channels, 0:00:14, 24 bits, 3018kbps, 96khz
(core: DTS, 5.1 channels, 0:00:14, 24 bits, 1509kbps, 48khz)
Creating file "D:\Queen_core.dts"...
eac3to processing took 1 second.
Done.
Please :logfile:

I have 16 bits DTS Hi-Res.

[Snowknight26]

It doesn't matter what it says as DTS-HD HR doesn't have a bit depth. No lossy formats do.

[rik1138]

It doesn't matter what it says as DTS-HD HR doesn't have a bit depth. No lossy formats do.

Sure they do... DTS HD retains the bit depth of the source files used to create it (or, at the very least, it's recording the bit-depth in the header information...). I can encode a 24-bit DTS-HD HR file and a 16-bit DTS-HD HR file (using 24-bit or 16-bit source WAVs). eac3to decodes them differently (but reports the bit depth correctly). I tried it with a stereo file and a 5.1 file, the 5.1 one was the only one that eac3to said it was 'Patching bitdepth to 24 bits...'

Obviously, it didn't say this on the 24 bit 5.1 file, and it didn't do it on the 16 bit 2.0 file (not sure why).

Here's a 16-bit 5.1 DTS-HD HR file if you want to try it:

http://www.sendspace.com/file/usnwqd

Rik

[tebasuna51]

Sure they do... DTS HD retains the bit depth of the source files used to create it (or, at the very least, it's recording the bit-depth in the header information...)
Only the last part is true: "recording the bit-depth in the header"
Like Snowknight26 say you, only losless encoders (like DTS-MA) can preserve the bitdepth.

I say to madshi don't put this header info in Eac3to logs because can induce to mistakes.
But I agree with madshi when change the header mark with 24 no matter the original source was. Always is better decode to 24 bits instead 16. Don't worry.

[tebasuna51]

I loaded my copy of Ice Age 2 in eac3to. It said the length of the movie was 1h 30m 38s (all playlists). I ripped the Dutch 768kbps audiotrack.

But when I played that seperate track in MPC-HC it was 1h 32m 15s.

EAC3TO says the seperate audiotrack is 1h 30m 38s. I am confused. Who is right? And why?

Use DelayCut to fix the DTS (remove internal pad's) and you can obtain the same value with all methods.

[shambles]

Hi, after using eac3to w/o -keepdiagnorm my TrueHD track has no diagnorm info at all and my A/V interpret that as -0dB and normalize to -31dB = the volume is very low. If I run the eac3to ones again on the track the log shows no diagnorm value at all, shouldn't eac3to sett the diagnorm to -31dB and not remove the value?

eac3to has used -31db instead of 0db for a while now for that exact reason.. unless something is broken in the latest build, just running the file through eac3to again should set the flag to -31db eventhough eac3to doesn't show it (with -31db and 0db, no dialnorm value is shown)

[AnryV]

It doesn't matter what it says as DTS-HD HR doesn't have a bit depth. No lossy formats do.
I asked not about it, but how to forbid eac3to to change the original information.

[Thunderbolt8]

it doesn't change the original information, because as shadowknight said, lossy tracks have no bit depth. therefore it doesn't matter if theres written 16 or 24 bit.

[AnryV]

it doesn't change the original information, because as shadowknight said, lossy tracks have no bit depth. therefore it doesn't matter if theres written 16 or 24 bit.
It change the information about 16 or 24 bit of the source.

[Snowknight26]

Which is completely irrelevant and sometimes not even correct.

[tebasuna51]

@AnryV
The field info at header is useless, any DTS, also the core or with original bitdepth 16 bits, must be decoded to 24 bits to preserve the maximum precision.

The free dts decoders, at least, never read this field to output a bitdepth or other.

[AnryV]

@AnryV
The field info at header is useless, any DTS, also the core or with original bitdepth 16 bits, must be decoded to 24 bits to preserve the maximum precision.

The free dts decoders, at least, never read this field to output a bitdepth or other.
I know. :)
I spoke about core extraction - a procedure at which does not have decoding. I do not understand what for to change something in an original file of a core.

[tebasuna51]

You know, madshi always try to improve the output:)

[killa_kid]

Hey madshi I have a script which parses the output of eac3to to automatically add video/audio information to a database. This worked beautifully up to 3.15. For some reason PHP (the language the script is written in) will not capture any of the output from 3.16. Are you doing something different with the output now? I've tried redirecting output, which didn't work. According to PHP the program has no output, except my command line window says something different :P

Any help would be appreciated, I would like to upgrade if possible

Thanks!

[tebasuna51]

You can use the eac3to parameter:

-log="X:\folder\file.log"

to obtain the output in a logfile.

[buzzqw]

@killa_kid

yep.. i got the same error in my application (but i read stdout)
the last line is truncated :confused:

BHH

[Atak_Snajpera]

That's exactly what I did. Weird. I'll try it again after reinstalling TMT3 and see if it's some sort of conflict. Thanks for the help.
sorry but I forgot that this checkactivate.dll file is also required.

[SomeJoe]

@killa_kid

yep.. i got the same error in my application (but i read stdout)
the last line is truncated :confused:

BHH

Yes, I can confirm the same. I'm using the ShellExecute method from a .NET 2.0 front end, and the last line of eac3to.exe 3.16's output is not present when reading StdOut.

I have reverted to 3.14 for the time being.

[Inspector.Gadget]

sorry but I forgot that this checkactivate.dll file is also required.

Ah, no worries. I gave up and reinstalled but now I'll know for next time. Thank you.

[odin24]

Would demuxing a TrueHD stream, using 384kb/s as the core... as opposed to 640kb/s have a negative effect on the performance of the TrueHD stream?

I often include TrueHD audio on DVD9 recodes, using a 384k core would save some MBs.

Thanks.

[LeXXuz]

You can demux streams by specifying the target as the same type as the source. For example, if track 3 is ac3, 4 is dts...

eac3to.exe source 3: output3.ac3 4: output4.dts

...will demux those tracks.

Got to try that out. :thanks:

[MrVideo]

Here is the situation. I have a transport stream that has three MPEG-2 audio streams that comprise the 6 channels for AC3 encoding. They map as follows:

1) L/R
2) C/LFE
3) LS/RS

How do I tell eac3to to use those three dual channel streams as input to aften AC3 encoding?

Can it even be done?

Thanks.

[Snowknight26]

No, not with eac3to.

[MrVideo]

No, not with eac3to.

Ok, thanks.

Rats!

[rik1138]

Would demuxing a TrueHD stream, using 384kb/s as the core... as opposed to 640kb/s have a negative effect on the performance of the TrueHD stream?

I often include TrueHD audio on DVD9 recodes, using a 384k core would save some MBs.

Thanks.

The core stream shouldn't have any affect on the TrueHD stream at all. Obviously your core stream won't sound quite as good, but if you are playing the THD anyway, that shouldn't matter.

[odin24]

The core stream shouldn't have any affect on the TrueHD stream at all. Obviously your core stream won't sound quite as good, but if you are playing the THD anyway, that shouldn't matter.

Thanks for the clarification.

[DoomBot]

eac3to has used -31db instead of 0db for a while now for that exact reason.. unless something is broken in the latest build, just running the file through eac3to again should set the flag to -31db eventhough eac3to doesn't show it (with -31db and 0db, no dialnorm value is shown)

So can the dialnorm info be restored to the file if eac3to has removed it? How would you restore it if its possible?

[73ChargerFan]

re-rip the audio track

[DoomBot]

re-rip the audio track

re-rip from the file that has no dialnorm info?

[73ChargerFan]

No, from the original source disk, using the option " -keepDialnorm ".

Also, log file created when you ran eac3to will have the dialnorm setting. I don't know of any way to put it back into the stream.

[DoomBot]

No, from the original source disk, using the option " -keepDialnorm ".

Also, log file created when you ran eac3to will have the dialnorm setting. I don't know of any way to put it back into the stream.

Thats what i was asking was a way to put it back into the stream.

[73ChargerFan]

I've looked on the net, and the only thing I can see is convert it to wav, then to ac3 using aften which allows you to set a dialnorm. The discussions about how to determine the appropriate value for dialnorm didn't seem to have a conclusion.

[a4840639]

Mkvmerge will drop the core when input TrueHD, the coreless THD will not be recognized by tsMuxeR，can eac3to adds a function to fix this issue？（Like combine a normal ac3 and a coreless TrueHD into a normal TrueHD)

Additional question
I do think THD has its own bits depth because it is lossless, why don't eac3to show it?

[snpetit]

I'm running Windows 7 x64 RC1 and everything work fine, nero BD/HDDVD plugin, arcsoft, nero aac, surcode...

I have the same problem under Vindows7 x64:
"eac3to doesn't see Nero Audio Decoder (Nero 6 or older). But I have Nero 7.11.10.0 and the Blu-ray/HD-DVD plug-in registered."

Can anybody tell me, how this codecs works under win7 x64.
Thanks in advance! :)

[nurbs]

eac3to supports Nero 6 or older versions. You have Nero 7 which is obviously newer than Nero 6. Nero 7 doesn't work and there is AFAIK no way to make it work with eac3to.

[73ChargerFan]

Nero 7 doesn't work and there is AFAIK no way to make it work with eac3to.
wrong, see the first post

[nurbs]

I mixed up the Nero version numbers, because I always use ImgBurn to write my discs and only have Nero installed for eac3to. In fact I have Nero 7 myself and it works fine. Nero 8 doesn't.

[darthcloud]

I have the same problem under Vindows7 x64:
"eac3to doesn't see Nero Audio Decoder (Nero 6 or older). But I have Nero 7.11.10.0 and the Blu-ray/HD-DVD plug-in registered."

Can anybody tell me, how this codecs works under win7 x64.
Thanks in advance! :)

Should work just fine without special tuning. If this can help I installed Nero 7 micro english from http://updatepack.nl/downloads/nero-lite/nero-7-lite/ instead of the full version and them I registered the BD plugin.

do you use "eac3to -test" to see what eac3to detect?

[buzzqw]

while demuxing an mkv i got this message
Video has a gap of 42 frames at playtime 0:12:49.

what's mean ?

BHH

[Seraphic-]

For best AC3, E-AC3 and AAC decoding you need:
(1) Nero 7 (Nero 8 won't work!)
(2) Nero HD DVD / Blu-Ray plugin

For best DTS decoding you need:
(1) ArcSoft DTS Decoder - version 1.1.0.0 or newer


I have a few DDWAVs to decode and wanted more info about the required programs.

So do you need Ahead Nero 7 Ultra Edition? I mean, what program is needed from the bundle to do the decoding? Also, where do you find the Nero HD DVD / Blu-Ray plugin? Did not see it listed on Nero website as a plugin. Is it for Nero 7?

And is the ArcSoft DTS Decoder part of the Arcsoft Total Media Theater bundle?

[TinTime]

So do you need Ahead Nero 7 Ultra Edition? I mean, what program is needed from the bundle to do the decoding? Also, where do you find the Nero HD DVD / Blu-Ray plugin? Did not see it listed on Nero website as a plugin. Is it for Nero 7?

Yes, you need Nero 7 and the plugin. Not sure which edition of Nero is needed. Unfortunately the plugin is no longer for sale so it doesn't much matter any more though. However eac3to will decode AC3 without Nero using libav. See the first post for madshi's views on the different decoders.

And is the ArcSoft DTS Decoder part of the Arcsoft Total Media Theater bundle?

Yes, it is.

[Seraphic-]

Yes, you need Nero 7 and the plugin. Not sure which edition of Nero is needed. Unfortunately the plugin is no longer for sale so it doesn't much matter any more though. However eac3to will decode AC3 without Nero using libav. See the first post for madshi's views on the different decoders. Yes, it is.

Well, looking over madshi's post about the decoders, and considering I want a perfect bit-for-bit decode of the file, it looks like I would need Nero (E-)AC3/Nero TrueHD and ArcSoft DTS decoders. However, from what you said, seems I can no longer get the required plugin. Am I out of luck? And does anyone know which edition of Nero 7 is needed?

[nurbs]

eac3to uses libav for TrueHD decoding due to the limitations of the nero decoder. AC3 is also no problem and E-AC3 basically works fine with libav. Only the DTS variants are problematic if you don't only want to decode the core.

[Fiffy]

However, from what you said, seems I can no longer get the required plugin. Am I out of luck? And does anyone know which edition of Nero 7 is needed?Any version of Nero 7 should do. I'm using the "Premium" edition. The trick is getting the plugin. BTW, all required functionality is already included in Nero. The plugin is just a registration code that unlocks the functionality.

[Seraphic-]

Any version of Nero 7 should do. I'm using the "Premium" edition. The trick is getting the plugin. BTW, all required functionality is already included in Nero. The plugin is just a registration code that unlocks the functionality.

Is the correct/official name for the plugin "Blu-ray/HD DVD Video Plug-in"?

Also, do you have to install the full Nero 7 "Premium" package for decoding to work?
Or do you know what program(s) are needed for just the AC3, E-AC3 and AAC decoding with eac3to?

[TinTime]

Is the correct/official name for the plugin "Blu-ray/HD DVD Video Plug-in"?

Yes, that's right.

Also, do you have to install the full Nero 7 "Premium" package for decoding to work?
Or do you know what program(s) are needed for just the AC3, E-AC3 and AAC decoding with eac3to?

You just need to install Nero ShowTime, and activate the plug-in.

[codres]

Hi everybody. I was trying to encode a raw pcm file extracted from a bluray and I getting this error message:

eac3to v3.16
command line: F:\BLURAY\INDIAN~1\EAC3TO\EAC3TO.EXE 1_3_audio.raw Indiana -448
------------------------------------------------------------------------------
This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
The RAW/PCM file seems to be little endian.
The RAW/PCM file seems to have a bitdepth of 16 bits.
The RAW/PCM file seems to have 6 channels.
RAW/PCM, 5.1 channels, 2:02:32, 16 bits, 4608kbps, 48khz
This audio conversion is not supported. <ERROR>

It's possible that some RAW files aren't supported yet?

[jj666]

Wrong command line, add the file extension: F:\BLURAY\INDIAN~1\EAC3TO\EAC3TO.EXE 1_3_audio.raw Indiana.ac3 -448

Cheers,

-jj-

[codres]

Wow, what a lame I was. I forgot about extension. I'm really ashame...

[Thunderbolt8]

is there a way to use -slowdown for 24fps sources, to slow them down to 23.976fps? it seems that eac3to always automatically assumes the source has 25 fps and it therefore uses a 1,024fps slowdown instead of only 0,024fps. the resulting length in my case is ~5 mins longer, while it should only be ~7 seconds longer.

edit: found a workaround, when I slow it muxed together with the according video file, then its accepted as 24fps input. but what to do in cases I don't have a video file?

[TinTime]

is there a way to use -slowdown for 24fps sources, to slow them down to 23.976fps? it seems that eac3to always automatically assumes the source has 25 fps and it therefore uses a 1,024fps slowdown instead of only 0,024fps. the resulting length in my case is ~5 mins longer, while it should only be ~7 seconds longer.

You can use -changeTo23.976

If you've already demuxed the audio you need to specify the source speed too, like...

eac3to.exe input output -24.000 -changeTo23.976

[Thunderbolt8]

-24.000 after input output? or is it input -24.000 output -changeto23.976, as usual?

[TinTime]

Yes, after input output as above. Maybe it works the other way too - I've never tried it.

[Thunderbolt8]

nope, works only the way you described. thanks ;)

[odin24]

I rencently converted a 25fps DTS-HD MA to 24fps LPCM. The source was 16 bit, the output LPCM was 24 bit. I did not use any extra parameters, except the "-changeto" switch.

I'd just like to understand why it went from 16 bit DTS-HD MA to 24 bit LPCM. If anybody could explain that would be great.

Here's an image of the process... sorry couldn't wait for the log file, on my way to work... :)

EDIT: Oh yeah, I first tried converting to w64. It produced a 16 bit LPCM, but couldn't mux with tsMuxeR, got several sync errors, I used the same convert switch. So I decided to try converting to PCM, everything muxes fine... and here I am.

Thanks.

http://img20.imageshack.us/img20/2599/19527983.jpg

[tebasuna51]

I'd just like to understand why it went from 16 bit DTS-HD MA to 24 bit LPCM. If anybody could explain that would be great.
Easy, the timestretch you make is a lossy conversion then the output can't be exact and is better a 24 bit aproach.
I recommend you convert to ac3 640 Kb/s because you can't preserve the lossless quality.

EDIT: Oh yeah, I first tried converting to w64. It produced a 16 bit LPCM, but couldn't mux with tsMuxeR, got several sync errors, I used the same convert switch. So I decided to try converting to PCM, everything muxes fine... and here I am.
If you have sync problems with w64 you got the same problems with LPCM, the audio data is identical.

[odin24]

Easy, the timestretch you make is a lossy conversion then the output can't be exact and is better a 24 bit aproach.
I recommend you convert to ac3 640 Kb/s because you can't preserve the lossless quality.


If you have sync problems with w64 you got the same problems with LPCM, the audio data is identical.

I believe you when you say they are identical, however, the sync error only occured when muxing with w64, no such error occured while muxing pcm. I don't actually mean "lip sync", I mean the muxing process could not complete because of an error... something about "byte sync error, or something like that.

So, when stretching, or shortening the time it is considered lossy and the LPCM track will not equal the source DTS-HD MA... will it even be close? Wouldn't the LPCM still be better than AC3 @ 640k?

Thanks for your response.

[tebasuna51]

I believe you when you say they are identical, however, the sync error only occured when muxing with w64, no such error occured while muxing pcm. I don't actually mean "lip sync", I mean the muxing process could not complete because of an error... something about "byte sync error, or something like that.
Then seems TsMuxer still don't support w64.

So, when stretching, or shortening the time it is considered lossy and the LPCM track will not equal the source DTS-HD MA... will it even be close?
Yes, but you can't recover the original audio then must be considered lossy.

Wouldn't the LPCM still be better than AC3 @ 640k?
Of course because the uncompressed data in LPCM is the source to encode to ac3, then you add more loss when encode. But the LPCM bitrate is 6912 Kb/s (seems the video :rolleyes: )

[Fiffy]

So, when stretching, or shortening the time it is considered lossy and the LPCM track will not equal the source DTS-HD MA... will it even be close? Wouldn't the LPCM still be better than AC3 @ 640k?Stretching and shortening requires resampling, so yes, it is a lossy operation. But if done right (and I'm sure eac3to does it right) the loss should be minimal and is not comparable to the losses caused by perceptual codecs (whether or not you can actually hear either of them is a different question :)).

If you find the resulting PCM file too large, I would recommend to reduce the word length back to 16 bits (-down16 in eac3to) and then use a lossless compression such as FLAC. Lossy encoding e.g. to AC3 is a much more severe manipulation of the signal.

[Thunderbolt8]

I wonder, does eac3to recognize AC3 4.0 channels correctly as such? what does 4.0 mean, 2 front and 2 rear channels? because eac3to recognized it as 3.1 so there might be a difference? its from an old movie and unfortunately I don't know whether its really supposed to be 2 front 2 rear.

[tebasuna51]

I wonder, does eac3to recognize AC3 4.0 channels correctly as such? what does 4.0 mean, 2 front and 2 rear channels? because eac3to recognized it as 3.1 so there might be a difference?

Yep, eac3to can difference 4 channel ac3 with:
AC3, 2/2 channels (FL,FR,BL,BR)
AC3, 3/1 channels (FL,FR,FC,BC)
AC3, 2/1.1 channels (FL,FR,LF,BC)
AC3, 3.1 channels (FL,FR,FC,LF)

for more info see this post:
http://forum.doom9.org/showthread.php?p=1134710#post1134710

[Thunderbolt8]

k, I guess that 3/1 thing should be correct then ;)

thanks!

[pascalwil]

Hi

Anyone to help me use SurCode DVD DTS Encoder alone (ie outside of eac3to) in Command Line mode?
I would like to batch convert the 6 wav files I create with eac3to into one DTSWAV file using a script instead of the Surcode GUI.
An example of the script to use would be most welcome to automate the process of creating DTS audio CDs from my 5.1 DVDs.
Thanks for your help.

Cheers

[leeperry]

am I going nuts, or do DTSHD(MA and HR)>FLAC conversions always end up w/ the dialogs center channel being too low?

TrueHD or LPCM>FLAC is perfectly fine(same dialogs volume as core DTS), and I've tried both Sonic 4.3 and Arcsoft(they give identical CRC btw).

I've encountered this problem many times, DTSHD>FLAC is unusable for me...I use ffdshow to decode if that means anything(DRC is disabled).

I've also tried to enable -keepDialnorm in eac3to, but it doesn't chance anything(same CRC for the FLAC).

[Thunderbolt8]

does it change when you use madflac to decode?

[leeperry]

yep, center is still too low compared to the core DTS...even w/ madflac.

this time it was a DTS HR 24bit track that I'm downsampling to 16 :

eac3to track.dts N:\eng_16.flac -sonic -down16
DTS Hi-Res, 5.1 channels, 1:51:33, 24 bits, 3840kbps, 48khz
(core: DTS, 5.1 channels, 1:51:33, 24 bits, 1509kbps, 48khz)
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Reducing depth from 24 to 16 bits...
Encoding FLAC with libFlac...
Creating file "N:\eng_16.flac"...
eac3to processing took 15 minutes, 19 seconds.
Done.

I'll try to come up w/ some proper sample, was just wondering if anyone else encountered this problem :)

[tebasuna51]

am I going nuts, or do DTSHD(MA and HR)>FLAC conversions always end up w/ the dialogs center channel being too low?

TrueHD or LPCM>FLAC is perfectly fine(same dialogs volume as core DTS), and I've tried both Sonic 4.3 and Arcsoft(they give identical CRC btw)...

I can't reproduce your problem:
Core dtshd
--------------- ------------------ --------------------
RMS power ch0: 3.21% (-29.88 dB) 3.40% (-29.37 dB)
RMS power ch1: 2.82% (-30.99 dB) 2.99% (-30.49 dB)
RMS power ch2: 4.93% (-26.15 dB) 5.22% (-25.64 dB)
RMS power ch3: 1.59% (-35.96 dB) 1.86% (-34.61 dB)
RMS power ch4: 0.88% (-41.11 dB) 0.93% (-40.60 dB)
RMS power ch5: 0.69% (-43.22 dB) 0.73% (-42.72 dB)
Please upload a sample.

[leeperry]

ok, thanks for the test...I was hoping for some measurements...how did you do that? you're comparing the FLAC'ed DTS-HD against the core DTS?

I've already had the problems many times on DTS HD tracks :(

for instance on this stream "DTS Master Audio, 5.1 channels, 24 bits, 48khz"(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz), the dialogs were hardly audible, where they were perfectly fine on the core...and -keepDialnorm doesn't work(maybe by design?)

[Revgen]

^If you upload a sample, they may be able to help.

[tebasuna51]

I was hoping for some measurements...how did you do that? you're comparing the FLAC'ed DTS-HD against the core DTS?
I campare a dtshd decoded to wav with ArcSoft (DTS-MA) and decoded with libav (DTS core), the values are statistics from Wavosaur (free audio editor)
...and -keepDialnorm doesn't work(maybe on design?)
Don't use -keepDialnorm never with ac3 or dts (I don't know with TrueHD), let eac3to work.

[leeperry]

righty, I'll do my homework and come back w/ a proper sample...maybe ffdshow's mixer is playing tricks on me after all :confused:

I use my own stereo downmix Logic7 matrix, and this one works fine on all movies...except FLAC'ed DTS-HD :o

[Revgen]

1) TMT2 installs dtsdecoderdll.dll version 1.1.0.0
2) TMT3 installs dtsdecoderdll.dll version 1.1.0.1
3) TMT 3.0.1.133 Patch installs dtsdecoderdll.dll version 1.1.0.5

http://www.arcsoft.com/downloads/digitaltheatre/arcsoft/totalmediatheatre_beta_3.0.1.120_3.0.1.133_update_all.exe

you need to have installation of TMT2 on which eac3to recognize the dtsdecoderdll.dll, i.e. "eac3to -test" returns:



then just replace dtsdecoderdll.dll version 1.1.0.0 with version 1.1.0.1 or version 1.1.0.5 from TMT3 and then when you run "eac3to -test" you should get:



or



4) TMT 3.5 installs dtsdecoderdll.dll version 1.1.0.10, but it doesn't work with eac3to.

BTW, there is also new CinemasterAudio.DLL version 4.3.0.236 from the new version 2.6.3 of SONIC CineVision, which works good with eac3to. if you look here:

http://forum.doom9.org/showthread.php?p=1158014#post1158014

and here:

http://forum.doom9.org/showthread.php?p=1212881#post1212881

you will see that this is the first new version of CinemasterAudio.DLL from very long time that works good with eac3to

I'm having no luck getting arcsoft to work either. Am I screwed if I can't install TMT 2.0?

I bought TMT version 3.0 with the 1.1.0.5.dll and it will not work with EAC3 no matter what steps I take. Only the regular asdecoder.ax works in graphedit. Is there a way I can get a hand on a previous dtsdecoder.dll version?

[xkodi]

I'm having no luck getting arcsoft to work either. Am I screwed if I can't install TMT 2.0?

I bought TMT version 3.0 with the 1.1.0.5.dll and it will not work with EAC3 no matter what steps I take. Only the regular asdecoder.ax works in graphedit. Is there a way I can get a hand on a previous dtsdecoder.dll version?

look here:

http://forum.doom9.org/showthread.php?t=148324

[tebasuna51]

The spanish audio from Blu-Ray movie La Cosa is :
Bitrate 384Kbps
Sample Rate: 48KHz
Channels: 2
299MB, 1h 48min

Without problem when use ArcSoft decoder, but is buggy when use libav.

Audio sample (http://rapidshare.com/files/263798729/DTS_384_48Khz_2.0_La_Cosa_2_min.zip)

Using ffmpeg to decode (or Foobar2000 or MeGUI/BeHappy) is also OK.

[odin24]

When you encode with Surcode use the .cpt (compact) format instead the .dts (padded) format. After rename to dts.


I tried searching, but no luck, I even searched through google. I was curious as to what exactly is "DTS zero padding"?

Thanks.

[Octo-puss]

is Madshi still MIA?

[tebasuna51]

I tried searching, but no luck, I even searched through google. I was curious as to what exactly is "DTS zero padding"?.
These 4 dts files (20 sec long) was encoded with Surcode, the padded ones add 0's bytes after the core frame to reach the 2048 boundary (to easy write and search on hardware)

File ........: D:\Test\dts\Jap_padD.dts ... 3840000 bytes
----------------------------------------- First Frame Info
Primary Frame Byte Size .....: 2012 ( 2013 bytes/frame)
Transmission Bit Rate .......: 24 (1536 Kb/s)
...
--------------------------------------------- Revised Info
Total Frames ......: 1875
Duration ..........: 20 seconds. ( 0 h. 0 m. 20 s.)
Pad Bytes min./max.: 35 / 35 (bytes between frames)
------------------------------------------------- End Info

Real frame = Core frame + pad bytes = 2013 + 35 = 2048
Real bitrate = 3840000 x 8 / 20 = 1536 Kb/s
=================================================================

File ........: D:\Test\dts\dJap_cptD.dts ... 3774375 bytes
----------------------------------------- First Frame Info
Primary Frame Byte Size .....: 2012 ( 2013 bytes/frame)
Transmission Bit Rate .......: 24 (1536 Kb/s)
...
--------------------------------------------- Revised Info
Total Frames ......: 1875
Duration ..........: 20 seconds. ( 0 h. 0 m. 20 s.)
------------------------------------------------- End Info

Real Bitrate = 3774375 x 8 / 20 = 1509.75 Kb/s
=================================================================

File ........: D:\Test\dts\Jap_pad7.dts ... 3840000 bytes
----------------------------------------- First Frame Info
Primary Frame Byte Size .....: 1005 ( 1006 bytes/frame)
Transmission Bit Rate .......: 15 (768 Kb/s)
...
--------------------------------------------- Revised Info
Total Frames ......: 1875
Duration ..........: 20 seconds. ( 0 h. 0 m. 20 s.)
Pad Bytes min./max.: 1042 / 1042 (bytes between frames)
------------------------------------------------- End Info

Real frame = Core frame + pad bytes = 1006 + 1042 = 2048
Real bitrate = 3840000 x 8 / 20 = 1536 Kb/s
=================================================================

File ........: D:\Test\dts\Jap_cpt7.dts ... 1886250 bytes
----------------------------------------- First Frame Info
Primary Frame Byte Size .....: 1005 ( 1006 bytes/frame)
Transmission Bit Rate .......: 15 (768 Kb/s)
...
--------------------------------------------- Revised Info
Total Frames ......: 1875
Duration ..........: 20 seconds. ( 0 h. 0 m. 20 s.)
------------------------------------------------- End Info

Real bitrate = 1886250 x 8 / 20 = 754.5 Kb/s

[Pho3n1x]

I have one Question,is there a Decoder for TruHD wich decodes all 7.1 channels with eac3to to mono waves or is the only way 5.1 channel with the Nero or the libav decoder?

[tebasuna51]

I have one Question,is there a Decoder for TruHD wich decodes all 7.1 channels with eac3to to mono waves or is the only way 5.1 channel with the Nero or the libav decoder?
Eac3to decode TrueHD 7.1 to 8 wavs with libav without problems

[SomeJoe]

Don't use -keepDialnorm never with ac3 or dts (I don't know with TrueHD), let eac3to work.

For my AV receiver (Pioneer VSX-1018AH-K), TrueHD will not play properly unless it is extracted with eac3to using -keepDialnorm.

Have never had a problem letting eac3to remove it with ac3 or DTS, however.

[Snowknight26]

You sure you're using the latest version? Even when its removing dialog normalization, its setting the value to -31dB. In some earlier versions, it set it to 0dB, which might be causing your issue.

[Revgen]

is Madshi still MIA?

Madshi is currently fighting Krell on Altair IV. He'll be back next month.

[M-Blaster]

Hi, by downcoding Renaissance HD-DVD comes the following error.

eac3to l:\mainevo.evo e:\ren
EVO, 1 video track, 2 audio tracks, 1 subtitle track, 1:45:09
1: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
2: DTS Master Audio, German, 5.1 channels, 16 bits, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
3: DTS Master Audio, English, 5.1 channels, 16 bits, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
4: Subtitle (DVD), German
Track 2 is used for destination file "Ren".
This audio conversion is not supported.

even the option -core does not help.

DOes somebody have an idea how to strip this movie evo?

[yesgrey]

I don't know if you can simply use e:\ren to specify the directory for the demuxed files.
Try switching to e:\ren directory and then run from there eac3to l:\mainevo.evo -demux

[tebasuna51]

Try with:
eac3to l:\mainevo.evo e:\ren.*

[M-Blaster]

how stupid I was :(

Now it runs with
"eac3to l:\mainevo.evo e:\ren.mkv" :D

[SomeJoe]

You sure you're using the latest version? Even when its removing dialog normalization, its setting the value to -31dB. In some earlier versions, it set it to 0dB, which might be causing your issue.

I normally use v3.14 because latest version has some type of problem where the last line of output to stdout is not present, so my front end cannot parse certain data.

But I have also tried v3.16 and get the same issue.

Apparently, my AV receiver is highly sensitive to a TrueHD stream. Unless -keepDialnorm is used, TrueHD output is very distorted.

It is no big deal, I have dialogue normalization turned off on the AV receiver anyway.

[pascalwil]

Hi

Anyone knows if/where eac3to source code is available? Thanks.

Cheers.

[kypec]

AFAIK the source code to eac3to has never been released by madshi. Considering how long he's not been around this thread it seems rather disturbing. Development of such a great tool should continue because even revision 3.16 is still not "perfect" I think and many things can be improved on it.

[leeperry]

I don't see why he should go GPL, he's very busy IRL...be glad he made it public in the first place!

[jj666]

It would be easier to do the same route as Neuron2 and make it payware.

-jj-

[M-Blaster]

hello I want to downmix a song from the credits of a mkv movie file.
Soundstream is 1,5 mBit DTS 5.1 with 24 bits.
I splitted the credits with mkvtoolnix.

eac3to e:\moviecredits.mkv e:\credits.flac -down2 -mixlfe
eac3to e:\moviecredits.mkv e:\credits.flac -down2
eac3to e:\moviecredits.mkv e:\credits.flac

Problem is that the classical song of the credits is ok, but when the song from the artist begins the sound is totally centered on the left.
I tried with the 3 parameters.
I don not habe arcsoft dts so that "libav" was used.

[Guest]

It would be easier to do the same route as neuron2 and make it payware. The donation I require is modest but it does give me an ongoing incentive to continue development and support. It doesn't earn me a fortune but it pays for my web hosting and some hardware/software purchases.

[jmvh]

Back to getting the best decoders for use with eac3to under Windows 7 x64...I can confirm the earlier posts that eac3to can work with the the TMT 2.0 codecs on Windows 7 x64 RTM. My PC originally had eac3to working with Nero 7 and TMT under winXP. I copied the required files from my winXP installation into the eac3to directory and used the Regserv32 command under Windows 7 x64.

All is good for DTS-MA using TMT. However, for the life of me, I can not get Nero 7 decoders to install. Despite successfully installing in my winXP, and activating with by BD/HD-DVD plugin, I can not repeat that under Win7. I've installed Nero as admin and normally, I've activiated the plug-in. But Showtime doesn't work (errors saying can't build graph on any content - DVD blu HD - and doesn't seem to recognize that BD is active since it doesn't have the BD enabled skin).

Anyone have success with Nero 7 win 7 x64 and care to share how you accomplished that feat?

[Jeff Flowerday]

Back to getting the best decoders for use with eac3to under Windows 7 x64...I can confirm the earlier posts that eac3to can work with the the TMT 2.0 codecs on Windows 7 x64 RTM. My PC originally had eac3to working with Nero 7 and TMT under winXP. I copied the required files from my winXP installation into the eac3to directory and used the Regserv32 command under Windows 7 x64.

All is good for DTS-MA using TMT. However, for the life of me, I can not get Nero 7 decoders to install. Despite successfully installing in my winXP, and activating with by BD/HD-DVD plugin, I can not repeat that under Win7. I've installed Nero as admin and normally, I've activiated the plug-in. But Showtime doesn't work (errors saying can't build graph on any content - DVD blu HD - and doesn't seem to recognize that BD is active since it doesn't have the BD enabled skin).

Anyone have success with Nero 7 win 7 x64 and care to share how you accomplished that feat?

How important is nero just for DD+ decoding? libavcodec will also decode DD+ by default with eac3to.

I've already ripped my HD-DVDs when my machine was vista now that it's 7 and I'll probably never rip any quantity of HD-DVDs again I'll live with libavcodec.

[nurbs]

libavcodecs dd+ support is not complete, but nobody ever provided sample files for the features that aren't supported so it's probably safe to assume there are no HD-DVDs that use them.

[Revgen]

^DD+ 7.1 audio is virtually non-existent. HD-DVD's never used it and Blu-Ray never will have it since the discs can hold up to 50gb. 7.1 audio typically uses True-HD or DTS-HD.

[Fiffy]

The libav E-AC3 codec does support 7.1, so that is not the problem. The concern is whether the decoder (which is still quite new) works correctly. Apparently it produces different output than the Nero codec (which is Dolby certified). Also take a look at this thread:

http://forum.doom9.org/showthread.php?t=145636

[xxx666yyy777]

I normally use v3.14 because latest version has some type of problem where the last line of output to stdout is not present, so my front end cannot parse certain data.

But I have also tried v3.16 and get the same issue.

Apparently, my AV receiver is highly sensitive to a TrueHD stream. Unless -keepDialnorm is used, TrueHD output is very distorted.

It is no big deal, I have dialogue normalization turned off on the AV receiver anyway.

SomeJoe - how do you turn if off? which receiver?

Thx

[<><]

I have retail version of Arcsoft TMT 2.0 installed on Win7 (with latest patch/update), added the bin path the path Environment Variable, but it still says that it is not installed when running the -test command....any ideas?

[Revgen]

The libav E-AC3 codec does support 7.1, so that is not the problem. The concern is whether the decoder (which is still quite new) works correctly. Apparently it produces different output than the Nero codec (which is Dolby certified). Also take a look at this thread:

http://forum.doom9.org/showthread.php?t=145636

libav does not support EAC3 (Dolby Digital Plus) 7.1. Only 5.1. It supports 7.1 for Dolby True-HD aka MLP.

http://forum.doom9.org/showthread.php?p=1313704#post1313704

AFAIK, there are no 7.1 EAC3 Blu-Rays or HD-DVD's. Discs that use 7.1 are usually Dolby True-HD or DTS-HD.

[jruggle]

The libav E-AC3 codec does support 7.1, so that is not the problem. The concern is whether the decoder (which is still quite new) works correctly. Apparently it produces different output than the Nero codec (which is Dolby certified). Also take a look at this thread:

http://forum.doom9.org/showthread.php?t=145636
Both AC3 and E-AC3 use random noise as part of the decoding process. From the tests I have done so far, the difference between the output of libavcodec and Nero is approx. the same as the difference between using 2 different types of random number generators in the libavcodec decoder. The type of RNG used is not defined in the specification. It even says that the level of the noise, which should be 0.707, can be approximated to 0.5 and still be compliant.

[RonaldoSan]

I always extract the best audio track to wav (better source material, better encoding result), unfortunately some Scandinavian Blu-rays only offer the standard AC-3 track (The Departed for instance). I noticed I was getting some large wav files from those tracks (compared to TrueHD tracks).

I'm not very confident in the Scandinavian publisher, besides lacking quality audio tracks on Blu-rays, The Dark Knight has a separate AC-3 track that is exactly the same as the embedded AC-3 track.

The Dark Knight

3: AC3, English, 5.1 channels, 640kbps, 48khz
4: TrueHD/AC3, English, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 640kbps, 48khz)

Anyway, I extract the TrueHD and AC-3 tracks to wav and the output looks like this:

[a03] Extracting audio track number 3...
[a03] Decoding with DirectShow (Nero Audio Decoder 2)...
[a03] DirectShow reports 5.1 channels, 24 bits, 48khz
[a03] Writing WAV...
--------------------------------------------------------------
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Decoding with libav/ffmpeg...
[a04] Writing WAV...
--------------------------------------------------------------
AC3.wav - 6912 kbps 7,34 GB
TrueHD.wav - 4608 kbps 4,89 GB

TrueHD is lossless and AC-3 is compressed. When you extract to wav I assume you don't loose any information, so how come the AC-3 extraction is much bigger?

[tebasuna51]

...
AC3.wav - 6912 kbps 7,34 GB
TrueHD.wav - 4608 kbps 4,89 GB[/CODE]

TrueHD is lossless and AC-3 is compressed. When you extract to wav I assume you don't loose any information, so how come the AC-3 extraction is much bigger?

You decode to wav (not extract).
This TrueHD track seems have 16 bitdepth (audio samples with 16 bits), when decode from compressed audio you obtain 24 bitdepth then the size is 24/16 = 1,5 greater.

Of course the TrueHD.wav is the exact source audio.

[RonaldoSan]

You decode to wav (not extract).
This TrueHD track seems have 16 bitdepth (audio samples with 16 bits), when decode from compressed audio you obtain 24 bitdepth then the size is 24/16 = 1,5 greater.

Of course the TrueHD.wav is the exact source audio.
I see. Thanks for clarification.

[LeXXuz]

I have demuxed my Harry Potter 3 HD DVD with eac3to and got a TrueHD track with a delay of 2002ms.
I got a file named: "feature_1 - 4 - TrueHD, 5.1 channels, 48khz DELAY 2002ms.thd".
As a next step I wanted to transcode this file with eac3to into dts (using SurCode).
From what I understand eac3to first decodes this TrueHD track into several WAVE tracks and feeds them to the Surcode Encoder afterwards.
Is there a way to get rid of this delay during this process?

[nurbs]

Look at the first post. I assume adding +2002ms to the command line would do the trick.

[LeXXuz]

Look at the first post. I assume adding +2002ms to the command line would do the trick.

Ah thank you. I must have missed this as I looked into the command line options last night. :o

Why was the delay not removed during demuxing like for the E-AC3 tracks?

[TinTime]

Why was the delay not removed during demuxing like for the E-AC3 tracks?

I think eac3to can't apply delays to TrueHD tracks - that's why the delay is written to the file name.

However, as you're converting to DTS anyway why not go straight from the source EVOs to DTS, instead of demuxing the TrueHD track first? If you transcode then eac3to will apply the delay as needed.

[LeXXuz]

However, as you're converting to DTS anyway why not go straight from the source EVOs to DTS, instead of demuxing the TrueHD track first? If you transcode then eac3to will apply the delay as needed.

Because Nero AC3 Decoder does not allow to process more than one track at a time. That's why I first demux all the tracks I need (usually 2 E-AC3 tracks at least + TruHD) and process them afterwards.

[TinTime]

Because Nero AC3 Decoder does not allow to process more than one track at a time.

Fair enough. In that case demux the tracks you need except for TrueHD ones - convert them. Multiple libav decodes are ok.

[BlackJack1]

How to disable fixes using AV gaps during eac3to demuxing ts source?
After eac3to finishing job I've got out of sync mkv while ts source is in sync.
Eac3to found 11 gaps and discontinuous in stream.
Only one ts part has problems. Five others w/o :)
Thx in advance.

[Fiffy]

libav does not support EAC3 (Dolby Digital Plus) 7.1. Only 5.1. It supports 7.1 for Dolby True-HD aka MLP.You're right, I mixed that up.
AFAIK, there are no 7.1 EAC3 Blu-Rays or HD-DVD's. Discs that use 7.1 are usually Dolby True-HD or DTS-HD.I am not aware of any either.

[Fiffy]

Both AC3 and E-AC3 use random noise as part of the decoding process. From the tests I have done so far, the difference between the output of libavcodec and Nero is approx. the same as the difference between using 2 different types of random number generators in the libavcodec decoder. The type of RNG used is not defined in the specification. It even says that the level of the noise, which should be 0.707, can be approximated to 0.5 and still be compliant.Maybe I'm missing something (haven't looked into the spec in a long time), but as far as I know, an AC3 decoder only uses random noise if dithflag is set in the stream. Apart from that the decoder is deterministic (of course an implementation may still use dithering to reduce the word length for output). I'm sure Dolby takes this into account in the certification process.

[jruggle]

Maybe I'm missing something (haven't looked into the spec in a long time), but as far as I know, an AC3 decoder only uses random noise if dithflag is set in the stream. Apart from that the decoder is deterministic (of course an implementation may still use dithering to reduce the word length for output). I'm sure Dolby takes this into account in the certification process.
Yes, and the dithflag is typically set to true for nearly every block. That is the recommendation for encoders, except for blocks where the short block MDCT is used for transients or in other rare cases.

(E-)AC-3 decoding is not deterministic. Selected parts of it are, but certainly not the format as a whole. The specification hints that 24-bit fixed point would be a typical implementation for AC-3, but it is not explicitly required or even suggested. There are 16-bit fixed point tables, and even some hints at 6-bit fixed point for downmixing. For E-AC-3, some floating point tables and calculations are shown, but again it is not stated what the implementation must be. I'm sure Dolby does have some some certification tests and requirements, but those are not publicly available.

[BLKMGK]

Am stuck pulling AC3 out of a claimed TrueHD track, feature list for 3.16 claims it can be done however -> can extract AC3 stream from Blu-Ray TrueHD/AC3 tracks. When I attempt this I get an error stating that the audio conversion isn't supported. I am running NO 3rd party plug-ins although I do have Nero7-Ultra just not ever bought the plug-in. Output is as follows ->

D:\Video\eac3to>eac3to x: 1) 2: f:\bunny\bunny-vid.mkv 3: f:\bunny\bunny-ac3 1: f:\bunny\bunny-chap.txt
M2TS, 1 video track, 5 audio tracks, 10 subtitle tracks, 1:37:18, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 640kbps, 48khz)
<snip>
This audio conversion is not supported.

Where did I go wrong? I've used eac3to forever and this has me stumped!:confused:

Thanks!

P.S. Movie is The House Bunny on BluRay...

[Killroy™]

Am stuck pulling AC3 out of a claimed TrueHD track, feature list for 3.16 claims it can be done however -> . When I attempt this I get an error stating that the audio conversion isn't supported. I am running NO 3rd party plug-ins although I do have Nero7-Ultra just not ever bought the plug-in. Output is as follows ->

D:\Video\eac3to>eac3to x: 1) 2: f:\bunny\bunny-vid.mkv 3: f:\bunny\bunny-ac3 1: f:\bunny\bunny-chap.txt
M2TS, 1 video track, 5 audio tracks, 10 subtitle tracks, 1:37:18, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 640kbps, 48khz)
4: TrueHD/AC3, French, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 640kbps, 48khz)
5: AC3, Spanish, 5.1 channels, 640kbps, 48khz
6: TrueHD/AC3, Portuguese, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 640kbps, 48khz)
7: AC3, Thai, 5.1 channels, 640kbps, 48khz
8: Subtitle (PGS), English
9: Subtitle (PGS), English
10: Subtitle (PGS), French
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), Portuguese
13: Subtitle (PGS), Chinese
14: Subtitle (PGS), Chinese
15: Subtitle (PGS), Indonesian
16: Subtitle (PGS), Korean
17: Subtitle (PGS), Thai
This audio conversion is not supported.

Where did I go wrong? I've used eac3to forever and this has me stumped!:confused:

Thanks!

P.S. Movie is The House Bunny on BluRay...

3: f:\bunny\bunny-ac3 should be 3: f:\bunny\bunny.ac3 (period, not dash)

[BLKMGK]

3: f:\bunny\bunny-ac3 should be 3: f:\bunny\bunny.ac3 (period, not dash)

OMG! Geez, I sat staring at that for freaking ever and read multiple pages here looking to see if this was unique. D'oh!!:eek:

Thank you for pointing out the obvious. I'll edit the previous to shorten it up <sigh>

[deathlord]

Hi

I have a big problem:
eac3to does not write to my raid array.
When I write to a normal disk, everything is ok. As soon as I write to a raid 5 array (highpoint controller), eac3to becomes extremely slow (essentialy zero).
Can anyone help?

[hoju3508]

I'm trying to re-encode a Blu-ray that's 24fps instead of the usual 23.976fps. Video and audio are fine, but the sub is out of sync.


-slowdown convert 25.000 and 24.000 content to 23.976 fps


Will this option fix my subtitle sync issue? Or will this cause other problems?

:thanks: -Hoju

[setarip_old]

@hoju3508

Hi!I'm trying to re-encode a Blu-ray that's 24fps instead of the usual 23.976fps.That's rather unusual. May I ask the name of the BluRay?

[raquete]

first time using eac3to with flac-1.2.1-win

log file:
"eac3to v3.16
command line: eac3to RingoWave.wav RingoWave.flac
------------------------------------------------------------------------------
WAV, 5.1 channels, 0:03:55, 24 bits, 6912kbps, 48khz
Reading WAV...
Encoding FLAC with libFlac...
Creating file "RingoWave.flac"...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 1 minute, 8 seconds.
Done."

RingoWave.wav : 193Mb
RingoWave.flac : 145Mb

comment:
wave sounds brilliant, .flac result seems dry(opac)

2 questions:
how to get more compression?
as flac is lossless why sounds worse?

[Snowknight26]

You don't, eac3to compresses FLAC with the best compression. And yes, it's lossless, so it's just an illusion or your software is altering it on playback.

[raquete]

You don't, eac3to compresses FLAC with the best compression. And yes, it's lossless, so it's just an illusion or your software is altering it on playback.

is the best compression? ok....lol, i need more. :)

illusion? no, i really can hear (8 speakers, 2 sub-woofers and 6.1 amplifier/decoder here).
but is possible 3 differents softwares altering it on playback?
(VLC media player, adobe audition(in multitrack view), sound forge)

curious: winamp play lots of 5.1 formats(dts, ac3,waves...) but is crashing with this 5.1.flac.

[Midzuki]

but is possible 3 differents softwares altering it on playback? (VLC media player, adobe audition(in multitrack view), sound forge)

Yes, it is possible (although not probable). :D

curious: winamp play lots of 5.1 formats(dts, ac3,waves...) but is crashing with this 5.1.flac.

It seems the FLAC plugin was designed to deal with stereo (and mono) files only.

[Thunderbolt8]

anyhow possible to use madflac as winamp plugin? like its quality better

[raquete]

Yes, it is possible (although not probable). :D

It seems the FLAC plugin was designed to deal with stereo (and mono) files only.
yes,...lol
of course i don't blame eac3to for altering the audio result but flac plugin or any other mysterious issue.

anyhow possible to use madflac as winamp plugin? like its quality better
after search about madflac features i could not find answers if can be used as winamp plugin.

[Fiffy]

is the best compression? ok....lol, i need more. :)The compression ratio depends heavily on the content. Audio with a high degree of entropy is more difficult to compress. Some extreme cases (white noise) cannot be compressed at all. So try a few other examples.

[raquete]

The compression ratio depends heavily on the content. Audio with a high degree of entropy is more difficult to compress. Some extreme cases (white noise) cannot be compressed at all. So try a few other examples.
please, take a look in this post from this link in another thread: http://forum.doom9.org/showpost.php?p=1317155&postcount=12
Well, I've just created a 6-channel "test.wav", and by using the "normal"/"well-known" options, I haven't managed to make flac 1.2.0 generate a 24-bit && 88.2 kHz "test.flac". :(

OTOH, eac3to worked as it should and gave me no HeadAC3hes: :)

eac3to input.wav output.flac
was done following this recommendations and work very fine!

now look here in the bottom of this other post where i wrote "edit" :
http://forum.doom9.org/showpost.php?p=1317250&postcount=20

now help me to understand the reason of differents sizes in the result.

cheers!

[qyot27]

now look here in the bottom of this other post where i wrote "edit" :
http://forum.doom9.org/showpost.php?p=1317250&postcount=20

now help me to understand the reason of differents sizes in the result.
My guess is that FLAC's default compression level differs between using the standalone and using it via eac3to (FLAC uses a scale from 0 to 8, with 0 being fastest but largest filesize, and 8 being slower but producing smaller files). That's the only thing I can think of to produce a size difference like that.

If you try compressing other files in both, does roughly the same size difference occur as well? If you recompress the eac3to-generated FLAC in FLAC.exe, does it happen also?

[TinTime]

eac3to uses best compression I believe, but an older version of FLAC - v1.2.0. This might account for the size difference.

[Fiffy]

now help me to understand the reason of differents sizes in the result.I didn't read the long guide in your second link, so I'm not sure what you did there. The standard flac encoder uses compression level 5 by default, eac3to uses level 8 according to an earlier post by madshi (which is unfortunate IMO, since level 8 is quite a bit slower without significant gain in the compression when using the standard encoder). For some reason, eac3to seems to produce a slightly larger file (by roughly 10% in my tests) despite the higher compression level. I can reproduce that behavior here as well. Perhaps madshi will have an answer when he returns.

[hoju3508]

The Secret of Moonacre.

@hoju3508

Hi!That's rather unusual. May I ask the name of the BluRay?

[setarip_old]

@hoju3508

I, for one, will be unable to look into your problem any further because:

1) I don't own "The Secret of Moonacre" on BluRay, and

2) My research indicates that "The Secret of Moonacre" doesn't appear to have been released in the U.S. yet (You indicate that you're in Austin, TX)

[raquete]

@ qyot27, TinTime, Fiffy

was done with tebasuna51's guide and eac3to with the same flac version: "flac-1.2.1-win"
source wave: 193Mb (multichannel 5.1 24b/48k)

using the tebasuna51's guide, flac have 3 options to chose:
best, default and fast...was used "default", result: 127Mb

with eac3to was used this single command line(.bat file) :
eac3to RingoWave.wav RingoWave.flac, result: 145MB
then, i don't know the compression level used by eac3to...

was the default flac level used with eac3to?
(seems impossible as was the same flac version used in both and eac3to works like a charm too)

[deathlord]

madshi,


eac3to does not write to my raid array.
When I write to a normal disk, everything is ok. As soon as I write to a raid 5 array (highpoint controller), eac3to becomes extremely slow (essentialy zero).

As someone suggested a few pages back, this is clearly caused by highpoint's various sector size feature (which allows you to have partitions > 2TB e.g. in Win XP).
I could confirm this by creating a new array that is identical to the old one except I have left the sector size at the default value. Eac3to then has no trouble writing the the array.
For arrays with non-standard sector size: while eac3to can read from such raid arrays, it can't write to them.
Maybe you can add support for this?

[TinTime]

(seems impossible as was the same flac version used in both and eac3to works like a charm too)

Are you sure it's the same version of FLAC? According to foobar2000 any FLAC files I create with eac3to use v1.2.0.

[raquete]

Are you sure it's the same version of FLAC? According to foobar2000 any FLAC files I create with eac3to use v1.2.0.
gee..:eek: you're right, i see now what is different.
eac3to use libFLAC.dll(07/22/2007) and in tebasuna51 guide was used flac.exe(flac-1.2.1-win 09/16/2007).

i never had used eac3to, i download the file in 08/21/2009 09:33Am, not sure but i think that i got the file from the first post of this thread.(well, the first post is the best reference for me to got one trusty file and i don't read the complete thread searching new versions because it's a too big thread.)
or i download old eac3to version or may have a newer libFLAC.dll(or newer eac3to).

is the reason of differents final sizes?...(must be).
please, confirm and tell me if have and where have new libFLAC.dll version.
thanks! :)

[tebasuna51]

I don't see relevant differences using eac3to (libflaac) and BeHappy (flac.exe):

Tommy_orig.wav 793.553.084
Tommy_eac3.flac 458.619.467 last eac3to v3.16, libflac 2007-07-22
Tommy_Best.flac 458.627.594 BeHappy 'best', flac 1.2.1 2007-09-16
Tommy_Defa.flac 460.107.514 BeHappy 'default', flac 1.2.1 2007-09-16

[raquete]

tebasuna,
i did all again using the same process and adjusts as before and still got the same results from source 193Mb wave multi 5.1:
BeHappy (flac.exe) 127Mb
eac3to (libFLAC.dll) 147MB

looking yours source(793.553.084) and results sizes( 458MB~460MB) comparing with mine, 197MB(wave 5.1) to 127Mb(flac) seems the right compression(flac default level).

libFLAC.dll(07/22/2007) is the last version?

edit: log files...

eac3to v3.16
command line: eac3to RingoWave.wav RingoWave.flac
------------------------------------------------------------------------------
WAV, 5.1 channels, 0:03:55, 24 bits, 6912kbps, 48khz
Reading WAV...
Encoding FLAC with libFlac...
Creating file "RingoWave.flac"...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 1 minute, 51 seconds.
Done.
147MB

BeHappy (flac default)
Starting job RingoWave.wav->RingoWave.flac
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=48000Hz
encoder\flac.exe --force -o "D:\00 D files\Tutoriais 5.1\EAC3to\RingoWave.flac" --silent --force-raw-format --endian=little --channels=6 --bps=24 --sample-rate=48000 --sign=signed -
Writing PCM data to encoder's StdIn
Finalizing encoder
Complete
127MB

:confused:

[Snowknight26]

Run both files through eac3to and see if they're the same length, samplerate, bit-depth, etc.

[tebasuna51]

Yes, using 5.1 sample seems there are big differences:

Espinado_orig.wav 219.230.276
Espinado_eac3.flac 172.426.732
Espinado_def.flac 148.999.614
Espinado_best.flac 148.305.326

Of course the 3 flac recover the original wav bitexact

[TinTime]

Curious. I just tried encoding the soundtrack of the film "21" to flac - 16bit, 5.1, ~2 hours.

Both eac3to and the flac command line encoder (v1.2.1 using --best compression) produced the same size files (1.2GB), give or take a couple of KB.

I also tried a small sample of a 24 bit, 5.1 track and again the results were almost identical.

[hoju3508]

Correct, it's an import from the UK.

@hoju3508
2) My research indicates that "The Secret of Moonacre" doesn't appear to have been released in the U.S. yet (You indicate that you're in Austin, TX)

UPDATE: I'm using Clown_BD (which runs eac3to in the background) with the -slowdown option but now the audio and subtitle are both out of sync :(.

[raquete]

sorry to quote everybody but each detail is important.

Run both files through eac3to and see if they're the same length, samplerate, bit-depth, etc.
no differents files Snowknight26, is the same file from same path: "D:\00 D files\Tutoriais 5.1\EAC3to\RingoWave.wav".

Yes, using 5.1 sample seems there are big differences:

Espinado_orig.wav 219.230.276
Espinado_eac3.flac 172.426.732
Espinado_def.flac 148.999.614
Espinado_best.flac 148.305.326

Of course the 3 flac recover the original wav bitexact
yes, the same happens here tebasuna. i was thinking in upload my 5.1 source wave but now i see that no need as happened that same with yours.

Curious. I just tried encoding the soundtrack of the film "21" to flac - 16bit, 5.1, ~2 hours.

Both eac3to and the flac command line encoder (v1.2.1 using --best compression) produced the same size files (1.2GB), give or take a couple of KB.

I also tried a small sample of a 24 bit, 5.1 track and again the results were almost identical.
i used eac3to command line and flac.exe v1.2.1default compression in BeHappy and got less size.
after change the flac configuration to "best" in BeHappy, the result have the same file size as using default...127Mb.
lol, now i'm really curious of what is happening TinTime.

cheers for all.

[raquete]

from FLAC 1.2.2 (@@-@@@-2007) changelog page:
http://flac.cvs.sourceforge.net/*checkout*/flac/flac/doc/html/changelog.html
"libraries:
libFLAC encoder was defaulting to level 0 compression instead of 5" ( http://sourceforge.net/tracker/?func=detail&aid=1816825&group_id=13478&atid=113478 )

is the reason why eac3to is giving less flac compression?

[kypec]

raquette, please see my previous post (http://forum.doom9.org/showthread.php?p=1266310#post1266310) on this behaviour. Now I ran another round of tests just for curiosity and found the following facts quite interesting.
I used the first VOB of "Land of Women (PAL edition)" as the source.eac3to v3.16
command line: "C:\Program Files\AVTools\eac3to\eac3to.exe" VTS_03_1.VOB 2: c:test.wav
------------------------------------------------------------------------------
VOB, 1 video track, 3 audio tracks, 2 subtitle tracks, 0:19:26
1: MPEG2, 576i50 (16:9)
2: AC3, 5.1 channels, 448kbps, 48khz
3: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -28dB
4: AC3, 5.1 channels, 448kbps, 48khz
5: Subtitle (DVD)
6: Subtitle (DVD)
[a02] The Nero decoder doesn't seem to work, will use libav instead.
[a02] Extracting audio track number 2...
[a02] Decoding with libav/ffmpeg...
[a02] Remapping channels...
[a02] Reducing depth from 64 to 24 bits...
[a02] Writing WAV...
[a02] Creating file "c:test.wav"...
[a02] The last (E-)AC3 frame is incomplete and thus gets skipped. <WARNING>
Video track 1 contains 29162 frames.
eac3to processing took exactly 1 minute.
Done.

With that 5.1 multichannel WAV for testing I tried eac3toeac3to v3.16
command line: "C:\Program Files\AVTools\eac3to\eac3to.exe" test.wav test_eac.flac
------------------------------------------------------------------------------
WAV, 5.1 channels, 0:19:26, 24 bits, 6912kbps, 48khz
Reading WAV...
Encoding FLAC with libFlac...
Creating file "test_eac.flac"...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 2 minutes, 16 seconds.
Done.
test_eac.flac = 435 640 726 bytes

When encoded directly with flac.exe (version 1.2.1)flac.exe test.wav --best --no-seektable --no-padding --tag=VALID_BITS=24 --tag=HDCD=0 -o test_cli.flactest_cli.flac = 435 306 274 bytes

Finally I replaced libFLAC.dll in eac3to directory with the latest version 1.2.1, encoded as before and added one tag manually because it was missing in the file:metaflac.exe --set-tag=WAVEFORMATEXTENSIBLE_CHANNEL_MASK=0x060F test_eac.flac test_eac.flac = 435 306 274 bytes :)
Although they both have same filesize, they're not 100% identical! :confused: There are 2 differences in the data content1: 3 bytes changed at offset 169E8B19
90 E8 96 <> 50 E9 16
2: 2 bytes changed at offset 169EB552
1A 49 <> 86 46
BUT...when they are decoded back to WAV then those WAVs are perfectly identical and match with original test.wav so everything is alright. How come they are encoded differently yet still provide same output when decoded is beyond my knowledge... Maybe someone with deeper understanding of FLAC algorithms can shed some light onto this.

[kypec]

If one is encoded with version 1.2.1 and the other with version 1.2.0, and the version number is included in the file, then the files cannot match... but the audio data match.;)
You didn't read my post thoroughly, did you? :mad:
The data difference I point out occur in two files which were both encoded with version 1.2.1 of FLAC (one through libFLAC.dll, other through standalone flac.exe CLI).
Moreover, the difference happen to be in the DATA area of FLAC files, not in their headers where metadata are stored. Therefore I played with manually tagging file produced by eac3to in order to get them as much identical as possible. Which I almost succeeded by the way, it's just that difference exhibiting here cannot be fixed by any tagging, padding or any other harmless operations on files.

[raquete]

raquette, please see my previous post (http://forum.doom9.org/showthread.php?p=1266310#post1266310) on this behaviour. .
yes, i read now....i don't saw before!
Finally I replaced libFLAC.dll in eac3to directory with the latest version 1.2.1, encoded as before and...
and i did the same. download flac-1.2.1-devel-win, replaced libFLAC.dll and run eac3to.... :cool: :
file size now 127MB exactly size like using tebasuna51 guide!

great! :)

[tebasuna51]

Then seems there are good reasons to update libFLAC to 1.2.1 and this answer from madshi to kypec must be reconsidered:

Is there any practical reason for updating? IIRC someone recently tried the newest libFLAC.dll and the FLAC files produced with the new dll were bit by bit identical to the ones produced with the dll shipping with eac3to. Updating to a newer dll could break something. So I don't really want to do that, unless there is a real benefit.

@kypec
If you are using libFLAC 1.2.1, do you know any problem running eac3to with this version?

[kypec]

If you are using libFLAC 1.2.1, do you know any problem running eac3to with this version?
I must admit I do not use it very extensively (yet) but so far I had no issues whatsoever with latest libFLAC.dll.

[TinTime]

please see my previous post (http://forum.doom9.org/showthread.php?p=1266310#post1266310) on this behaviour.

I thought I had a sense of deja vu about this discussion!

Anyway, I've just been testing eac3to with libFLAC 1.2.1 and I'm not getting any significant filesize differences. I've tried three different movie soundtracks, all 5.1, one 16 bit and two 24 bit.

I wonder what's different about your test files, raquete and tebasuna51, that produces such dramatic differences?

I'm going to stick with 1.2.1 for the time being anyway and I'll report back if I hit any problems.

[tebasuna51]

I wonder what's different about your test files, raquete and tebasuna51, that produces such dramatic differences?
With my first sample stereo, 16 bits, 44100 KHz the sizes was equivalent, but with the second sample 5.1, 24 bits, 44100 KHz the difference in size is significant:
libFLAC 1.2.0 -> 172426732 bytes (ratio 78.6%)
libFLAC 1.2.1 -> 148297157 bytes (ratio 67.6%)

[raquete]

I wonder what's different about your test files, raquete and tebasuna51, that produces such dramatic differences?
i posted my old results 2(or3) times in this same thread and in the tebasuna guide thread ...
http://forum.doom9.org/showpost.php?p=1317155&postcount=12

now after change to libFLAC 1.2.1 the results are equals.

later i will post the result of wave source 5.1 24b/96k with 1:08:00:928 with the "old" and with the "new" libFLAC in eac3to...i still don't know the wave file size, are in seperateds channels(L,R,C...etc) and i need to join(encode) as multichannel.

[raquete]

crazy pc, crazy flac, crazy world and now the user is crazy too. :p

7.050.129.452 source wave 5.1 24b/96K (time 01:08:00)

eac3to:

3.752.647.704 old libflac (07/22/2007)
eac3to v3.16
command line: eac3to 01Wave.wav 01Wave.flac
-----------------------------------------------------
-------------------------
WAV, 5.1 channels, 1:08:00, 24 bits, 13824kbps, 96khz
Reading WAV...
Encoding FLAC with libFlac...
Creating file "01Wave.flac"...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 31 minutes, 33 seconds.
Done.

3.745.901.910 new libflac (09/16/2007)
eac3to v3.16
command line: eac3to 01Wave.wav 01Wave.flac
-----------------------------------------------------
-------------------------
WAV, 5.1 channels, 1:08:00, 24 bits, 13824kbps, 96khz
Reading WAV...
Encoding FLAC with libFlac...
Creating file "01Wave.flac"...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 31 minutes, 24 seconds.
Done.

BeHappy/Flac(09/16/2007)
but have errors.
opening as RaWavesource:
Starting job 01wave.wav->01wave.flac
Error: BeHappy.AviSynthException: Script error:
there is no function named "RaWavSource"
em BeHappy.AviSynthClip..ctor(String func, String arg,
AviSynthColorspace forceColorspace, AviSynthScriptEnvironment env)
em BeHappy.Encoder.encode()

1.464.637.781 BeHappy/Flac.exe(09/16/2007)
openinig as WavSource
error too, run in less than 5 minutes, file size is short but
was encoded only 26:34.422(time in minutes)
and i did twice, just the same.
Starting job 01wave.wav->01wave.flac
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=96000Hz
encoder\flac.exe --force -o "H:\eac3to\01wave.flac"
--silent --force-raw-format --endian=little --channels=6
--bps=24 --sample-rate=96000 --sign=signed -
Writing PCM data to encoder's StdIn
Finalizing encoder
Complete

now who can explain that?!? :o

[yesgrey]

You didn't read my post thoroughly, did you? :mad:
kypec,
sorry, I 've read it but somehow I've mixed it in my mind with some other previous posts.:o
I've already deleted that post because is completelly nonsense.;)

By the way, thanks for your findings. I will also try with new libflac, maybe I can gain some extra space on my backups.:)

[tebasuna51]

...
7.050.129.452 source wave 5.1 24b/96K (time 01:08:00)
...
there is no function named "RaWavSource"
...
openinig as WavSource
error too

now who can explain that?!? :o

1) WavSource can't manage wav files >4GB, you need open the file with RaWavSource

2) RaWavSource function is in NicAudio.dll v2.0. Downnload (see my signature) and put in AviSynth plugins folder

[Midzuki]

metaflac.exe --set-tag=WAVEFORMATEXTENSIBLE_CHANNEL_MASK=0x060F test_eac.flac

Another "undocumented" option of the flac applications? :(

Another very-good reason for me to prefer TTA and WavPack from now on! :cool:

[raquete]

1) WavSource can't manage wav files >4GB, you need open the file with RaWavSource

2) RaWavSource function is in NicAudio.dll v2.0. Downnload (see my signature) and put in AviSynth plugins folder
Great, i got the file. :)
in few hours new result.

Another very-good reason for me to prefer TTA and WavPack from now on! :cool:
i don't know TTA( ? ) format but Wavpack always give me problems when i encode waves with high volumes(levels), some peaks appear clippeds(saturations, straight lines/squares). i can see in any wave editor doing analysis.
i really don't like.
don't happens with your encodes?

[Midzuki]

i don't know TTA( ? ) format

http://www.true-audio.com/

but Wavpack always give me problems when i encode waves with high volumes(levels), some peaks appear clippeds(saturations, straight lines/squares). i can see in any wave editor doing analysis.
i really don't like.
don't happens with your encodes?

What ??? :confused:
Are you sure you weren't using the «lossy» compression mode ?
Also, if the answer is "yes", are you sure the sound editors you used were decompressing the lossless .WV files correctly???

[raquete]

http://www.true-audio.com/
What ??? :confused:
Are you sure you weren't using the «lossy» compression mode ?
Also, if the answer is "no", are you sure the sound editors you used were decompressing the lossless .WV files correctly???


first, thanks for the link.
i really don't know TTA and need to read about this format. :)

WV lossy? :confused:
i don't knew that one lossless compression can be "lossy". :p

adjusted "lossless(high)" to encode (no hibrid) 32 bit float
and adobe audition to open and check WV.

example from audition waveform statistics
(selected one channel where have one clip):
Mono
Min Sample Value: 21785
Max Sample Value: 32767
Peak Amplitude: -13.87 dB
Possibly Clipped: 1
DC Offset: 79.736
Minimum RMS Power: 996.99 dB
Maximum RMS Power: -inf dB
Average RMS Power: -3.01 dB
Total RMS Power: -18.65 dB
Actual Bit Depth: 16 Bits

Using RMS Window of 50 ms

[raquete]

tebasuna51,
after 8 minutes with ~60% of the file encoded i got this message:
Error: System.IO.IOException: O pipe foi finalizado.
Starting job 01wave.wav->01wave.flac
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=96000Hz
encoder\flac.exe --force -o "H:\eac3to\01wave.flac"
--silent --force-raw-format --endian=little
--channels=6 --bps=24 --sample-rate=96000 --sign=signed -
Writing PCM data to encoder's StdIn
Error: System.IO.IOException: O pipe foi finalizado.

[kypec]

Another "undocumented" option of the flac applications? :(
No, it's not undocumented. FLAC has very extensive support of tagging, look in the documentation, --help switch is your friend.
I wonder what's different about your test files, raquete and tebasuna51, that produces such dramatic differences?
In my case the size difference wasn't that dramatic at all.
435 306 274 : 435 640 726 = 99.923%, the real difference is less than 0.1%, no big deal.

[tebasuna51]

tebasuna51,
after 8 minutes with ~60% of the file encoded i got this message:
Error: System.IO.IOException: O pipe foi finalizado.

Yep, this is a know problem with flac.exe. Can't write flac files >2GB
Maybe in next v1.2.2...

See:
http://sourceforge.net/tracker/index.php?func=detail&aid=1834949&group_id=13478&atid=363478
http://sourceforge.net/tracker2/?func=detail&aid=1800350&group_id=13478&atid=113478

Until next flac.exe version you must use only eac3to if flac file output is >2GB.

Edit: With an audio track tested don't exist many difference:
8032850564 Pulp.wav
3918536382 Pulp_120.flac 48.8%
3893160387 Pulp_121.flac 48.5%

[Midzuki]

No, it's not undocumented. FLAC has very extensive support of tagging, look in the documentation, --help switch is your friend.

metaflac - Command-line FLAC metadata editor version 1.2.1
Copyright (C) 2001,2002,2003,2004,2005,2006,2007 Josh Coalson

Usage:
metaflac [options] [operations] FLACfile [FLACfile ...]

Use metaflac to list, add, remove, or edit metadata in one or more FLAC files.
You may perform one major operation, or many shorthand operations at a time.

Options:
--preserve-modtime Preserve the original modification time in spite of edits
--with-filename Prefix each output line with the FLAC file name
(the default if more than one FLAC file is specified)
--no-filename Do not prefix each output line with the FLAC file name
(the default if only one FLAC file is specified)
--no-utf8-convert Do not convert tags from UTF-8 to local charset,
or vice versa. This is useful for scripts, and setting
tags in situations where the locale is wrong.
--dont-use-padding By default metaflac tries to use padding where possible
to avoid rewriting the entire file if the metadata size
changes. Use this option to tell metaflac to not take
advantage of padding this way.

Shorthand operations:
--show-md5sum Show the MD5 signature from the STREAMINFO block.
--show-min-blocksize Show the minimum block size from the STREAMINFO block.
--show-max-blocksize Show the maximum block size from the STREAMINFO block.
--show-min-framesize Show the minimum frame size from the STREAMINFO block.
--show-max-framesize Show the maximum frame size from the STREAMINFO block.
--show-sample-rate Show the sample rate from the STREAMINFO block.
--show-channels Show the number of channels from the STREAMINFO block.
--show-bps Show the # of bits per sample from the STREAMINFO block.
--show-total-samples Show the total # of samples from the STREAMINFO block.

--show-vendor-tag Show the vendor string from the VORBIS_COMMENT block.
--show-tag=NAME Show all tags where the the field name matches 'NAME'.
--remove-tag=NAME Remove all tags whose field name is 'NAME'.
--remove-first-tag=NAME Remove first tag whose field name is 'NAME'.
--remove-all-tags Remove all tags, leaving only the vendor string.
--set-tag=FIELD Add a tag. The FIELD must comply with the Vorbis comment
spec, of the form "NAME=VALUE". If there is currently
no tag block, one will be created.
--set-tag-from-file=FIELD Like --set-tag, except the VALUE is a filename
whose contents will be read verbatim to set the tag value.
Unless --no-utf8-convert is specified, the contents will
be converted to UTF-8 from the local charset. This can
be used to store a cuesheet in a tag (e.g.
--set-tag-from-file="CUESHEET=image.cue"). Do not try
to store binary data in tag fields! Use APPLICATION
blocks for that.
--import-tags-from=FILE Import tags from a file. Use '-' for stdin. Each line
should be of the form NAME=VALUE. Multi-line comments
are currently not supported. Specify --remove-all-tags
and/or --no-utf8-convert before --import-tags-from if
necessary. If FILE is '-' (stdin), only one FLAC file
may be specified.
--export-tags-to=FILE Export tags to a file. Use '-' for stdout. Each line
will be of the form NAME=VALUE. Specify
--no-utf8-convert if necessary.
--import-cuesheet-from=FILE Import a cuesheet from a file. Use '-' for stdin.
Only one FLAC file may be specified. A seekpoint will be
added for each index point in the cuesheet to the
SEEKTABLE unless --no-cued-seekpoints is specified.
--export-cuesheet-to=FILE Export CUESHEET block to a cuesheet file, suitable
for use by CD authoring software. Use '-' for stdout.
Only one FLAC file may be specified on the command line.
--import-picture-from=FILENAME|SPECIFICATION Import a picture and store it in a
PICTURE block. Either a filename for the picture file or
a more complete specification form can be used. The
SPECIFICATION is a string whose parts are separated by |
characters. Some parts may be left empty to invoke
default values. FILENAME is just shorthand for
"||||FILENAME". The format of SPECIFICATION is:
[TYPE]|[MIME-TYPE]|[DESCRIPTION]|[WIDTHxHEIGHTxDEPTH[/COLORS]]|FILE
TYPE is optional; it is a number from one of:
0: Other
1: 32x32 pixels 'file icon' (PNG only)
2: Other file icon
3: Cover (front)
4: Cover (back)
5: Leaflet page
6: Media (e.g. label side of CD)
7: Lead artist/lead performer/soloist
8: Artist/performer
9: Conductor
10: Band/Orchestra
11: Composer
12: Lyricist/text writer
13: Recording Location
14: During recording
15: During performance
16: Movie/video screen capture
17: A bright coloured fish
18: Illustration
19: Band/artist logotype
20: Publisher/Studio logotype
The default is 3 (front cover). There may only be one picture each
of type 1 and 2 in a file.
MIME-TYPE is optional; if left blank, it will be detected from the
file. For best compatibility with players, use pictures with MIME
type image/jpeg or image/png. The MIME type can also be --> to
mean that FILE is actually a URL to an image, though this use is
discouraged.
DESCRIPTION is optional; the default is an empty string
The next part specfies the resolution and color information. If
the MIME-TYPE is image/jpeg, image/png, or image/gif, you can
usually leave this empty and they can be detected from the file.
Otherwise, you must specify the width in pixels, height in pixels,
and color depth in bits-per-pixel. If the image has indexed colors
you should also specify the number of colors used.
FILE is the path to the picture file to be imported, or the URL if
MIME type is -->
--export-picture-to=FILE Export PICTURE block to a file. Use '-' for stdout.
Only one FLAC file may be specified. The first PICTURE
block will be exported unless --export-picture-to is
preceded by a --block-number=# option to specify the exact
metadata block to extract. Note that the block number is
the one shown by --list.
--add-replay-gain Calculates the title and album gains/peaks of the given
FLAC files as if all the files were part of one album,
then stores them in the VORBIS_COMMENT block. The tags
are the same as those used by vorbisgain. Existing
ReplayGain tags will be replaced. If only one FLAC file
is given, the album and title gains will be the same.
Since this operation requires two passes, it is always
executed last, after all other operations have been
completed and written to disk. All FLAC files specified
must have the same resolution, sample rate, and number
of channels. The sample rate must be one of 8, 11.025,
12, 16, 22.05, 24, 32, 44.1, or 48 kHz.
--remove-replay-gain Removes the ReplayGain tags.
--add-seekpoint={#|X|#x|#s} Add seek points to a SEEKTABLE block
# : a specific sample number for a seek point
X : a placeholder point (always goes at the end of the SEEKTABLE)
#x : # evenly spaced seekpoints, the first being at sample 0
#s : a seekpoint every # seconds; # does not have to be a whole number
If no SEEKTABLE block exists, one will be created. If
one already exists, points will be added to the existing
table, and any duplicates will be turned into placeholder
points. You may use many --add-seekpoint options; the
resulting SEEKTABLE will be the unique-ified union of
all such values. Example: --add-seekpoint=100x
--add-seekpoint=3.5s will add 100 evenly spaced
seekpoints and a seekpoint every 3.5 seconds.
--add-padding=length Add a padding block of the given length (in bytes).
The overall length of the new block will be 4 + length;
the extra 4 bytes is for the metadata block header.

Major operations:
--version
Show the metaflac version number.
--list
List the contents of one or more metadata blocks to stdout. By default,
all metadata blocks are listed in text format. Use the following options
to change this behavior:

--block-number=#[,#[...]]
An optional comma-separated list of block numbers to display. The first
block, the STREAMINFO block, is block 0.

--block-type=type[,type[...]]
--except-block-type=type[,type[...]]
An optional comma-separated list of block types to be included or ignored
with this option. Use only one of --block-type or --except-block-type.
The valid block types are: STREAMINFO, PADDING, APPLICATION, SEEKTABLE,
VORBIS_COMMENT. You may narrow down the types of APPLICATION blocks
displayed as follows:
APPLICATION:abcd The APPLICATION block(s) whose textual repre-
sentation of the 4-byte ID is "abcd"
APPLICATION:0xXXXXXXXX The APPLICATION block(s) whose hexadecimal big-
endian representation of the 4-byte ID is
"0xXXXXXXXX". For the example "abcd" above the
hexadecimal equivalalent is 0x61626364

NOTE: if both --block-number and --[except-]block-type are specified,
the result is the logical AND of both arguments.

--application-data-format=hexdump|text
If the application block you are displaying contains binary data but your
--data-format=text, you can display a hex dump of the application data
contents instead using --application-data-format=hexdump

--remove
Remove one or more metadata blocks from the metadata. Unless
--dont-use-padding is specified, the blocks will be replaced with padding.
You may not remove the STREAMINFO block.

--block-number=#[,#[...]]
--block-type=type[,type[...]]
--except-block-type=type[,type[...]]
See --list above for usage.

NOTE: if both --block-number and --[except-]block-type are specified,
the result is the logical AND of both arguments.

--remove-all
Remove all metadata blocks (except the STREAMINFO block) from the
metadata. Unless --dont-use-padding is specified, the blocks will be
replaced with padding.

--merge-padding
Merge adjacent PADDING blocks into single blocks.

--sort-padding
Move all PADDING blocks to the end of the metadata and merge them into a
single block.

As you ¿can? see, there is no mentioning of "WaveFormatExtensible", nor "Channel_Mask".

[raquete]

Yep, this is a know problem with flac.exe. Can't write flac files >2GB
Maybe in next v1.2.2...

See:
http://sourceforge.net/tracker/index.php?func=detail&aid=1834949&group_id=13478&atid=363478
http://sourceforge.net/tracker2/?func=detail&aid=1800350&group_id=13478&atid=113478

Until next flac.exe version you must use only eac3to if flac file output is >2GB.

Edit: With an audio track tested don't exist many difference:
8032850564 Pulp.wav
3918536382 Pulp_120.flac 48.8%
3893160387 Pulp_121.flac 48.5%

no problems, eac3to is cool, work perfectly and easy to use.

thanks so much for the links, hints, suport, big patience and friendship. :)

[Midzuki]

first, thanks for the link.
i really don't know TTA and need to read about this format. :)

WV lossy? :confused:
i don't knew that one lossless compression can be "lossy". :p

WavPack is a very-versatile format, indeed. :D

adjusted "lossless(high)" to encode (no hibrid) 32 bit float
and adobe audition to open and check WV.

example from audition waveform statistics
(selected one channel where have one clip):
Mono
Min Sample Value: 21785
Max Sample Value: 32767
Peak Amplitude: -13.87 dB
Possibly Clipped: 1
DC Offset: 79.736
Minimum RMS Power: 996.99 dB
Maximum RMS Power: -inf dB
Average RMS Power: -3.01 dB
Total RMS Power: -18.65 dB
Actual Bit Depth: 16 Bits

Using RMS Window of 50 ms

Have you tried: 1) decompress the .WV file to a .WAV copy of the original source; 2) bit-compare the original .WAV and the copied .WAV; ???

I don't use Adobe Audition, so I can just suppose it doesn't deal very well with 32-bit .WV files...


BTW: Did you generate those 32-bit .WV archives from 32-bit sources --- or not ???

I mean: regardless of the possible bugs/imperfections of «WavPack.exe»,
I see no point in creating 32-bit .WV archives only because Adobe Audition is your favorite sound editor. :)

[raquete]

WavPack is a very-versatile format, indeed.
lol

audition always open as wave 32bit float but i don't remember if i generate those 32-bit WavPack archives from 32-bit sources ...i don't use WV no more after see that clips.
sound forge show exactly the same clips as audition but audition is my favorite, have amazing differents features.
i'm using now flacs or apes.
flac is my preference.

cheers!

[yesgrey]

I've tested with Baraka, the biggest movie audio file I have, 5.1 96kHz 24bit.
wav : 9,452,814,404
1.2.0: 5,279,451,099 (55.85%)
1.2.1: 5,247,885,754 (55.51%)

That's not a very big difference. I'm not sure it's worth the effort to reencode all my backups with 1.2.1, I think I will only use it with the new ones...

[kypec]

As you ¿can? see, there is no mentioning of "WaveFormatExtensible", nor "Channel_Mask".
Of course there isn't. They are just METADATA and therefore can have arbitrary names and values. It's up to the application processing such FLAC files to honor/ignore such tags. These tags are not the same as RIFF data structures which are mandatory in WAVE headers for example.
I just included one of that tags manually in order to obtain one FLAC file as closely similar to another one as possible for testing purposes.

[leeperry]

i'm using now flacs or apes.
flac is my preference.
APE compresse way better than FLAC for stereo, and WavPack is usually slightly better than FLAC for 5.1 audio.

there really isn't anything good about FLAC, except hardware compatiblity..

plus eac3to won't let you choose the FLAC encoding strength? it's stuck on "5" apparently.

[raquete]

APE compresse way better than FLAC for stereo, and WavPack is usually slightly better than FLAC for 5.1 audio.

there really isn't anything good about FLAC, except hardware compatiblity..

plus eac3to won't let you choose the FLAC encoding strength? it's stuck on "5" apparently.as i posted, i have some problems like clips with WV then was left alone and what i know and was posted here too is that eac3to use flac max compression.
as my target is compress 5.1 multichannel, ape can't be used and WV even giving more compression than flac give me problems compressing stereo, i never did tests with 5.1 multichannel.
Paul McCartney E.Arguments are in flacs too then is cool, agree? :)

http://flac.sourceforge.net/news.html

[Midzuki]

raquete wrote:

Wavpack always give me problems when i encode waves with high volumes(levels), some peaks appear clippeds(saturations, straight lines/squares). i can see in any wave editor doing analysis.

Are you sure that your troubles with WavPack did not come from those excessively-loud sources ???

P.S.: Did you read the PM I sent you @ VideoHelp ?

[raquete]

raquete wrote:
Are you sure that your troubles with WavPack did not come from those excessively-loud sources ???

P.S.: Did you read the PM I sent you @ VideoHelp ?
PM in videohelp?!? no, i will read after this post.

excessively-loud sources not means clips but something like loudness war(is what you mean? )
as i encode my own cds, is impossible that EACopy or any wave editor(audition/sound forge) used to rip will encrease the levels of the source.
i check everything with big patience(sorry, i'm really bored with audio) : the waveforms after extracteds and after encodeds...and lots more details.
i posted the result of audition waveform statistics showing "Possibly Clipped: 1" because was selected few seconds around the first clip that i found from WV but the wave source don't have this clip. the top of the waveform turn from sine to straight line. :eek:
do you want samples and pictures from this waveforms with clips? i can post.
sometimes i need one entire month to encode a single album in 5.1(DVD-A & DVD-Video with audio too), i don't like of "more or less" results.

now answer me: if you find clips in the results of any encoder, what you will do?

cheers! :)

edit:
idea...as what we are talking about now nothing have to do with eac3to, we must open a new thread about encoders, his features, problems and advantages to don't mess this thread as the program here works very very fine, agree? :)

[kypec]

there really isn't anything good about FLAC, except hardware compatiblity..Maybe, but it is THE FACTOR for many people around here I guess. Also it is open source, cross-platform compatible, HW decoding is quite simple...
plus eac3to won't let you choose the FLAC encoding strength? it's stuck on "5" apparently.
No, it's stuck on 8=best as been stated by madshi before and proven true by testers (like me and raquette) also.

[leeperry]

oh ok?! but last time I transcoded it to "8" w/ dBPowerAmp and the file was smaller :confused:

well DVDA-Explorer works just fine w/ WavPack and crashes w/ FLAC for me, so that kinda closes the deal(files are not >2GB)...FLAC for HD audio BD tracks(in eac3to), WavPack for DVD-A and APE for CDDA :)

comparing TrueHD to 384kb/s AC3 is like comparing 64kbit MP3 to WAV...I'm such a lossless whore these days :D

[kypec]

oh ok?! but last time I transcoded it to "8" w/ dBPowerAmp and the file was smaller :confused:That's because eac3to uses older version 1.2.0 of FLAC library. Replace it with newer 1.2.1 as suggested by me & raquette and you'll get smaller FLAC files also! :p

[yesgrey]

If you don't know how, you need to get the libflac pack for developers here (http://sourceforge.net/projects/flac/files/flac-win/flac-1.2.1-win/flac-1.2.1-devel-win.zip/download).
Then, you grab the libflac.dll contained in the pack (inside Lib dir) and substitute the one in eac3to's dir.

[Atak_Snajpera]

Replace it with newer 1.2.1 as suggested by me & raquette and you'll get smaller FLAC files also!
http://img175.imageshack.us/img175/3171/new1.th.png (http://img175.imageshack.us/i/new1.png/)

[raquete]

Atak_Snajpera,
your wave 5.1 with 1h 35mn have round 3Gb size?
or better, what is the size of your source that result 798Mb flac??

[kypec]

Atak_Snajpera,
it's not always that you get smaller files with 1.2.1 vs 1.2.0
Try with some 24-bit resolution samples and perhaps you'll see some improvement.
It also happened to me that 1.2.0 produced same size as 1.2.1 when compressing down-sampled 16-bit sources.

[Atak_Snajpera]

your wave 5.1 with 1h 35mn have round 3Gb size?
or better, what is the size of your source that result 798Mb flac??
It was encoded directly from DTSMA via eac3to with ArcSoft DTS decoder installed.

[raquete]

Atak_Snajpera,
i asked because seems too short the flac final size 798MB "only" as your source is wave 1h 35 min with 6.1 channels can have round 3GB, i mean must be too big size source to result in 798MB.
i'm losing details somewhere, i can be wrong and i'm confused.

cheers. :)

[Atak_Snajpera]

MY SOURCE IS NOT WAVE 6.1! It was encoded directly from DTSMA without any temporary wave files! What part you don't understand now? 16 bit files compress alot better than 24bit.

[raquete]

MY SOURCE IS NOT WAVE 6.1! It was encoded directly from DTSMA without any temporary wave files! What part you don't understand now? 16 bit files compress alot better than 24bit.

ah, sure...no 6.1(was a typo, i was thinking to write 5.1)....and because i don't know what is DTSMA. :confused:
of course 16b encode alot better than 24b.

in the end, what size have your source? :p (have a way to measure i think)
after know the size source we can tell about flac compression, right?

[yesgrey]

i don't know what is DTSMA.
DTS-HD Master Audio. Is the lossless format from DTS. One of the formats used in Blu-ray. If you want to know more go to the dts website...

[Atak_Snajpera]

n the end, what size have your source? (have a way to measure i think)
after know the size source we can tell about flac compression, right?

http://img382.imageshack.us/img382/5386/capturegfw.th.png (http://img382.imageshack.us/i/capturegfw.png/)

[raquete]

as you know, i'm not familiar with DTSMA (thanks for dts website hint) but for me from 1.28GB(5.1 dtsma variable bit rate) to 798MB flac seems very good compression.

[yesgrey]

DTS-HD MA is not very efficient, because it consists of a lossy dts core at 1536kbps and another part that contain the difference between the original and the lossy core.

[Revgen]

^FLAC even beats True-HD when it comes to lossless compression. True-HD, to my knowledge, does not have a lossy core. The only thing it has that FLAC doesn't is metadata telling True HD decoders how to downmix. I can't imagine that the metadata would put so much size on a file.

[leeperry]

you know, i'm not familiar with DTSMA (thanks for dts website hint)
the worst is DTS-HD HR, it was just as big as DTS-MA on the few discs I encountered it...and yet, it's lossy.

a sneaky way to still provide customers w/ lossy audio in my book..

yes, FLAC beats TrueHD :cool:

[M-Blaster]

Hey, when demuxing a movie with DTS HD master audio
"eac3to l:\mainevo.evo e:\ren.mkv -core"
I am getting only big flac audio files. that can not integrated in the mkv movie.

How can I get dts files (1508kbit core only ) from an movie with DTS Master?

[tebasuna51]

1) flac is supported in mkv and is more efficient than DTS-MA, then is the default option.

2) To extract the core of the track 2 (if is the DTS-MA you want)

eac3to l:\mainevo.evo 2: e:\ren.dts -core

[raquete]

another question for you tebasuna.
how to eac3to command line to extract dts.wav (5.1 44.1/16b) to waves (seperateds channels in 48k/24b) ?
thanks in advance.

[tebasuna51]

another question for you tebasuna.
how to eac3to command line to extract dts.wav (5.1 44.1/16b) to waves (seperateds channels in 48k/24b) ?
thanks in advance.

¿dts.wav (5.1 44.1/16b)?

I don't know that. I know:
- dts.wav (2.0, 44.1, 16b) if the fake wav header is readed
- dts.wav (5.1, 44.1, some bitrate) the dts in the fake wav container without bitdepth but bitrate

To obtain mono wavs 48KHz 24 bits you can use:

eac3to dts.wav mono.wavs -resampleTo48000

(24 bits is the default)

[raquete]

yes, dts.wav is what i get ripping dts-cds with EAC....

to burn multichannel dts-cd is needed 5.1 44.1k 16b files(burn as audio cd)and ripping dts.wav from dts-cds give 5.1 44.1k 16b too!

as 24 bits is the default when extract with eac3to the command line posted is perfect.
Thanks so much.

edit:
and how to(command line) extract from the same source in 96k/24b?

[Snowknight26]

It's documented when you run the program and in the first post.

[raquete]

It's documented when you run the program and in the first post.

ah...thanks...great document, i read all.
is cool for who knows command lines. :(

following tebasuna example seems easy to use, i only have doubt what came first in the command line...the bitrate or the (up)sample?!? (i can't find this detail in the documment)

[Snowknight26]

Well, seeing as -changeTo48000 changed the sampling rate to 48KHz....

[raquete]

correct. :)
edit:
and how to(command line) extract from the same source in 96k/24b?
what came first in the command line...the bitrate or the (up)sample?!? (i can't find this detail in the documment)

i know that 24b is default but and if i want 32b or 8b?
the doubt remains Snowknight26.

[tebasuna51]

@raquete
You can use:
eac3to dts.wav mono.wavs -down16 -resampleTo48000
---------------------------------------------------------
DTSWAV, 5.1 channels, 0:02:07, 24 bits, 1235kbps, 44.1khz
Reading DTSWAV...
Decoding with ArcSoft DTS Decoder...
Resampling to 48khz...
Reducing depth from 64 to 16 bits...
Writing WAVs...

Always the process (here only decode and resample) is done with the high precission (here 64 bits float), at the end the depth is changed to the desired.

No matter the parameter order.

[raquete]

no matter the parameter order, very nice, very clever!
i have dozen works to start now.

thank you so much tebasuna51.

[M-Blaster]

1) flac is supported in mkv and is more efficient than DTS-MA, then is the default option.

2) To extract the core of the track 2 (if is the DTS-MA you want)

eac3to l:\mainevo.evo 2: e:\ren.dts -core

Error with abort : This audio conversion is not supported.

[tebasuna51]

Error with abort : This audio conversion is not supported.

Please put a MediaInfo report of your evo file

[NanoBot]

Hi,

Error with abort : This audio conversion is not supported.


tebasuna51 accidentally added a white space in his suggested command line. If you entered the command exactly like suggested by copy and paste, it cannot work.

The correct command line is

eac3to l:\mainevo.evo 2:e:\ren.dts -core

without the white space between 2: and e:\ren.dts -core

C.U. NanoBot

[Snowknight26]

No, the space there doesn't matter. Since it doesn't matter, then the only logical explanation is that track #2 doesn't have a DTS core.

[Chumbo]

@madshi,
When you have time to get back to this, would you consider adding support for FLV (Flash Video) please? It would be great if eac3to can demux and give info on this file type. Hopefully this will help. I use a tool called FLV Extract but would rather use eac3to. It's available here with source: http://www.moitah.net/. The direct file link is: http://www.moitah.net/download/latest/FLV_Extract.zip. Thanks for considering it.

[Bozster]

Can someone help me out. I have Face Off HD DVD which has 2 audio streams worth looking at.

One is E-AC3 (DD+) 1536kbps 5.1 24bit 48khz

the other is

DTS-ES 1509kbps 6.1 24bit 48khz

I'm guessing that DTS-ES would be better considering it has 6.1 channels and is close to the bitrate of E-AC3

However I'm not sure.. Which is better for me to extract for my home library?


Second problem I have is when I extract the DTS track I get only 5.1 reported in TSMuxer or 6 channels in RipBot 264 .. not sure why I'm not getting 6.1.

When I do run eac3to audio.dts again it shows DTS-ES 6.1.

Will RipBot encode this DTS track into MKV as 6.1 if it says X.X Stream (muxing in)?


I extracted the DTS track with

eac3to 1) 5: audio.dts ..

also tried:

eac3to 1) 5: audio.dts -1536

in hopes that SurCode DTS encoder would recognize and create a proper DTS-ES 6.1 stream.. no luck..

Is it just reporting issue with aforementioned TSMuxer and RipBot or eac3to is not extract DTS stream properly?

My filters installed are as follows:

- Arcsoft
- Surcode


Thanks

[tebasuna51]

E-AC3 (DD+) 1536kbps is much better quality than DTS-ES 1509kbps.
Other question is if you have a 6.1 audio system with a DTS-ES decoder

If eac3to shows DTS-ES 6.1 then is DTS-ES 6.1, don't worry.

[Bozster]

E-AC3 (DD+) 1536kbps is much better quality than DTS-ES 1509kbps.
Other question is if you have a 6.1 audio system with a DTS-ES decoder

If eac3to shows DTS-ES 6.1 then is DTS-ES 6.1, don't worry.

Yes I do... I have a 6.1 Klipsch DIY custom speakers setup and Denon 2809ci AVR.

That's why I have a dillema.. I think that 6.1 benefit over 5.1 would be better then somewhat better compression.

I archive all my movies as DTS 1.5mbps because I can't hear the difference above that even with DTS-MA or TrueHD. The DD (480,640) and DTS (768) are different story. I can hear those being softer when compared to HD audio but anything at 1.5mbps I can't hear it.

[tebasuna51]

...
I archive all my movies as DTS 1.5mbps because I can't hear the difference above that even with DTS-MA or TrueHD. The DD (480,640) and DTS (768) are different story. I can hear those being softer when compared to HD audio but anything at 1.5mbps I can't hear it.
Is your choice.
But try encode a DTS-MA or TrueHD to AC3 640 Kb/s and compare with DTS 1.5Mb/s

[raquete]

Is your choice.
But try encode a DTS-MA or TrueHD to AC3 640 Kb/s and compare with DTS 1.5Mb/s
i never could hear differences using wave sources in 24b/48k.
AC3 640K is really "the way to go" with short size and play in all standalones(i never saw standalones that can't play 640K but i can be wrong as i don't know all models(of course)).

observation: i only use AFTEN(WAVtoAC3Enc) for AC3.(is the same AC3 encoder used in EAC3to)
with others encoders used before i always found "issues" and was posted somewhere in this audio forum.

[yesgrey]

try encode a DTS-MA or TrueHD to AC3 640 Kb/s and compare with DTS 1.5Mb/s
Just for curiosity, do you think that AC3 640kbps sounds better than DTS 1.5Mbps?

[tebasuna51]

I just say: AC3 is more efficient than DTS.
I think I put this comparative 3 times in this thread:
http://tech.ebu.ch/docs/tech/tech3324.pdf

[Blue_MiSfit]

I usually keep the AC3 track for compatibility purposes - and S/PDIF transport. At 448+kbps it's honestly "good enough" for me! If I must transcode something, it's always to Nero AAC-LC. The default Q=.5 sounds very good to me, and it's almost always smaller than 640kbps AC3.

I find the tasty high resolution / bitrate audio tracks tempting, but I'm rarely impressed by the difference. Once in a blue moon I'll FLAC a high-res track - but AAC is usually awesomely awesome for my tastes!

BTW - excellent link, tebasuna51! It was an interesting read, and further substantiates what folks have been saying on doom9 for years :)

~MiSfit

[yesgrey]

I think I put this comparative 3 times in this thread
Thanks! and sorry to make you repeat yourself...;)

[Bozster]

I just say: AC3 is more efficient than DTS.
I think I put this comparative 3 times in this thread:
http://www.ebu.ch/CMSimages/en/tec_doc_t3324-2007_tcm6-53801.pdf

I read the comparison..however I can tell you that DTS 1.5mbps is virtually indistinguishable from DTS-MA or TrueHD even in controlled conditions and blind tests by experts.

DD/DD+ at 640kbps can indeed be better than DTS 768 but based on tests the only versions that provided maximum quality at lowest sizes without sacrificing quality were DTS 1.5mbps and DD+ 1.5bmps and above.

You can read about comparisons here:
http://www.hemagazine.com/node/Dolby_TrueHD_DTS-MA_versus_Uncompressed_PCM

They visited both DTS and Dolby labs.

They did say that even though DTS-MA, TrueHD and LPCM offered high bitrates and excellent quality, DD and DTS held very good even at 448/640/768 bitrates. Even though they could distinguish the audio difference with these lower bitrate codecs (they say it did sound a bit softer and flatter) they would be good enough to majority of people while DTS 1.5 and DD+ 1.5+ couldn't be spotted at all in blind tests against DTS-MA/TrueHD/LPCM.

This is why I chose to encode my movies with DTS 1.5mbps. I even have SurCode encoder where I encode TrueHD as DTS 1.5mbps.

Just some thoughts.

[Bozster]

Is your choice.
But try encode a DTS-MA or TrueHD to AC3 640 Kb/s and compare with DTS 1.5Mb/s

So you think that I wouldn't hear a difference between AFTEN'ed TrueHD track as AC3 640 vs DTS 1.5?

I mean, I'd definitely do that because it would mean better compatibility with PS3 and xbox 360 which i have everywhere around my house as extenders but based on above articles I'm feeling more confident encoding as DTS 1.5 as I'm trusting these blind tests myself so I want to be sure.

[yesgrey]

DD/DD+ at 640kbps can indeed be better than DTS 768 but based on tests the only versions that provided maximum quality at lowest sizes without sacrificing quality were DTS 1.5mbps and DD+ 1.5bmps and above.
That's not what I've read...
They weren't capable of hearing the difference between the lossless formats and DTS-HD High Resolution / DD+ 1.5mbps.
DTS-HD High Resolution is between 2.0 and 6.0mbps, and the core DTS are 768kbps (DVD) and 1536kbps(BR). They were capable of noticing a slightly difference between the core DTS's and the lossless, but between DD 640kbps and the lossless only one of them was able to tell the difference. So, that test also seems to indicate that DD 640kbps is equal or slightly better than DTS 1536kbps.
There was an important distinction in the two tests, though. In Dolby it was a blind test, but in DTS it wasn't. It's "easier" to spot a difference when you know that should be there...;)

Edit: Don't forget also that in Dolby it was only a 10s test, and by their description of the sound it did not seem to be a very hard to compress piece. A soundtrack is more than 1H30m, so maybe it's not such a good idea to relly your decision in a comparison made with only 10s of a soundtrack...;)

[tebasuna51]

...
You can read about comparisons here:
http://www.hemagazine.com/node/Dolby_TrueHD_DTS-MA_versus_Uncompressed_PCM
...
This is why I chose to encode my movies with DTS 1.5mbps. I even have SurCode encoder where I encode TrueHD as DTS 1.5mbps.

Maybe my english is so bad, but at the link you provide there aren't comparasion between DTS and AC3, only between uncompressed audio and AC3/DTS and the conclusion is:

"The lossless, Dolby Digital Plus, and DTS-HD High Resolution compressed tracks were just a little more open and airy. I hate to say it, but they just sounded more realistic and transparent. The 448 kbps Dolby Digital and standard DTS tracks were less so, a little more closed off. Between the 640 kbps Dolby Digital and the uncompressed, the difference was even less noticeable."

I think he compare 448 Kb/s DD with 1.5 Mb/s DTS, and say 640 Kb/s DD is better.

[raquete]

That's not what I've read...
@ Bozster
chose one music(send me some options, maybe i have here)
i will encode in 5.1 and send: AC3-640k, DTS-1.5Mb, the seperateds waves useds as sources (L,R,C,LFE,LS,RS) and multichannel wave too.
after hear AC3 and DTS, tell us "who is who". after that compare the sound with the waves.
deal? :-)

[RonaldoSan]

I have a 6 channel wav file (created with eac3to) that I need to cut up into 6 wav files containing each their own channel.

Is this possible with eac3to?

[G_M_C]

I have a 6 channel wav file (created with eac3to) [...]

Reprocess the source, and use the commandline to save "WAVs" in stead of "WAV".

[RonaldoSan]

Reprocess the source, and use the commandline to save "WAVs" in stead of "WAV".
I was looking to avoid that, but I guess that is what I have to do.

Thanks for you help.

[tebasuna51]

I was looking to avoid that, but I guess that is what I have to do.
You can use the multichannel wav like input because is a lossless process:

eac3to multichan.wav mono.wavs

[amgeex]

Just a quick question. When one does a eac3to -demux movie.mpls, the audio is not touched, right? The .dtsma audio I get is pure and not dependant on any decoders I have (or don't have) installed on my computer, right?

Thanks in advance!

[Bozster]

I think he compare 448 Kb/s DD with 1.5 Mb/s DTS, and say 640 Kb/s DD is better.

Nope I disagree.. he meant in comparison to 448k DD and 768 DTS (as that's the standard DTS). The 1.5mbps DTS is not really standard.

[yesgrey]

The 1.5mbps DTS is not really standard.
It's the standard for Blu-ray.;)

[raquete]

It's the standard for Blu-ray.;)
and DTS 1.536 Mb/s (we can name as 1.5Mb) for dvd.
if DTS 768 is the standard, DD 448K win for miles and the competition between formats is without sense.

[Snowknight26]

A nice feature for eac3to would extracting a FLAC stream from an mkv file. Currently, the FLAC stream is decoded then encoded back to FLAC.

[hoju3508]

That's correct.

Just a quick question. When one does a eac3to -demux movie.mpls, the audio is not touched, right? The .dtsma audio I get is pure and not dependant on any decoders I have (or don't have) installed on my computer, right?

Thanks in advance!

[Killroy™]

A nice feature for eac3to would extracting a FLAC stream from an mkv file. Currently, the FLAC stream is decoded then encoded back to FLAC.

Does it do this if you just do a -demux? I have been using that when I know the mkv has a flac already and it did not appear to be re-encoded.

[Snowknight26]

It does.

[tal.aloni]

is there actually a way not to have eac3to apply a delay at all at demuxing

I found one way:
using a delay opposite to the audio delay,

for example:
4: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB, 83ms

if you use -83ms at the command, no delay will be applied to the AC3.

[<><]

I seem to be having some trouble w/ DTS Master Audio, which seems to affect 6.1/7.1 chan audio. Using Arcsoft Decorders.

For example, X-Men: The Last Stand has a DTS Master Audio 6.1 ch, which I conver to FLAC, but is not playable w/ madflac thru WMP12 and also gives an error in GraphStudio saying it cannot open the file.

Again, same movie/same audio track still converting to FLAC, this time using the -core flag, same results as above.

Again, same movie/same audio track still converting to FLAC, this time using the -libav flag, this time audio is playable and also is able to be graphed using GraphStudio. (but audio is out of sync when merged w/ movie using mkvmerge)

Rush Hour 3, same issues as above, DTS Master Audio 7.1 ch, converted to FLAC, not playable or graphable.

Death Race, which contains DTS Master Audio 5.1 ch converted to FLAC, no issues, is playable and graphable using GraphStudio (used ArcSoft decoders)

Any ideas?

[Mark_A_W]

I seem to be having some trouble w/ DTS Master Audio, which seems to affect 6.1/7.1 chan audio. Using Arcsoft Decorders.

For example, X-Men: The Last Stand has a DTS Master Audio 6.1 ch, which I conver to FLAC, but is not playable w/ madflac thru WMP12 and also gives an error in GraphStudio saying it cannot open the file.

Again, same movie/same audio track still converting to FLAC, this time using the -core flag, same results as above.

Again, same movie/same audio track still converting to FLAC, this time using the -libav flag, this time audio is playable and also is able to be graphed using GraphStudio. (but audio is out of sync when merged w/ movie using mkvmerge)

Rush Hour 3, same issues as above, DTS Master Audio 7.1 ch, converted to FLAC, not playable or graphable.

Death Race, which contains DTS Master Audio 5.1 ch converted to FLAC, no issues, is playable and graphable using GraphStudio (used ArcSoft decoders)

Any ideas?

I can playback Xmen3, with 6.1 FLAC with madFLAC and Zoom Player.

I have to ask, why on earth are you trying to do this with WMP???

Please try a player where you have proper controls over the filters that are being loaded!

[Snowknight26]

You have enough control with WMP to know that madFLAC will be loaded for FLAC streams, so that's most likely not the issue, especially when he mentioned GraphStudio.

[Mark_A_W]

Yeah, but no control after that..

What audio renderer is being loaded? Can it handle 6.1? Can the soundcard driver?

If ffdshow audio processor is loaded after madflac, then you can experiment with the channels setting in the Mixer controls and see what's happening.


WMP is good for casual use, but not for this.

[Snowknight26]

What audio renderer is being loaded? Can it handle 6.1? Can the soundcard driver?

If ffdshow audio processor is loaded after madflac, then you can experiment with the channels setting in the Mixer controls and see what's happening.

And those have nothing to do with WMP. Even if it does, it should be in it's own thread because its off topic.

[setarip_old]

- can extract Blu-Ray and HD DVD chapter information and subtitlesWhat would be the commandline entry for extracting the subtitles from a BluRay/.M2TS disc/file?

[Snowknight26]

Same as extracting any stream, but you have to use the correct extension (pgs for PGS, srt for SRT, sup for SUP, etc).

[setarip_old]

This is the example provided in the first post that includes extracting subtitles:eac3to source.evo 1: chapters.txt 2: video.mkv 3: audio.flac 5: subtitle.supWhat is the significance of the "5:"?

What would be the specific commandline entry for extracting the PGS subtitles from a BluRay/.M2TS disc/file? Would it be:

1) eac3to source.m2ts 5: subtitle.pgs

2) eac3to source.m2ts 1: subtitle.pgs

3) eac3to source.m2ts subtitle.pgs

4) Or, if something else, what?

[nurbs]

4) eac3to source.m2ts 5:subtitle.sup or eac3to "C:\path\source.m2ts" 5:"C:\path\subtitle.sup" for Blu-Ray

The 5: is the tracknumber. eac3to numbers all tracks in an m2ts file starting with the chapters, followed by video tracks, audio tracks and subtitles. You first use eac3to to open a file (eac3to source.m2ts) or select a folder (eac3to "C:\Path"). If you do the former you see a tracklist, if you do the latter you see a title list. In that case you select a title by its number (e.g. eac3to "C:\Path" 2) selects the 2nd title on the disc) to get to the track list.

[setarip_old]

@nurbs

Thank you ;>}

[tebasuna51]

...What is the significance of the "5:"?

What would be the specific commandline entry for extracting the PGS subtitles from a BluRay/.M2TS disc/file?
The track number, first you need run (to know the tracks):

eac3to source.m2ts

Please read http://en.wikibooks.org/wiki/Eac3to/How_to_Use

[73ChargerFan]

Or if you're not in a hurry, and will leave the computer unattended anyway, use -demux to get everything, and then delete the unwanted files later.

[YukonTrooper]

Hey, guys. Just wondering if using Nero's decoders for extracting the core DTS and DD audio streams will give me better sound-quality opposed to using eac3to's. I'm not doing any conversions, just extracting those core streams and remuxing them into .m2ts files with the untouched video stream.

Also, I realize I'll need Nero's decoders for 1.0 streams (Dog Day Afternoon), but how I do make eac3to point towards the decoders once I have them? Does eac3to automatically search for those decoders first if they are in the eac3to folder? I'm using Nero 8 Ultra right now, but the eac3to Wiki says that won't work.

[TinTime]

If you're just demuxing then no decoders will be used. They're only used for conversions.

[wolfbane5]

I've been using eac3to for a while now on vista and it works great; however, I'll be switching to windows 7 in a couple of weeks and I was wondering if anyone has encountered any issues running eac3to on 7?

[Jeff Flowerday]

I've been using eac3to for a while now on vista and it works great; however, I'll be switching to windows 7 in a couple of weeks and I was wondering if anyone has encountered any issues running eac3to on 7?

No it works fine.

[buzzqw]

same here...

just installed haali, ffdshow tryouts, xvid.. and used clsid reg tools for windows 7

BHH

[magic144]

hi,
just been converting an HD-DVD I own (Elizabeth: The Golden Age) and I came across the following issue.

Here's the log of eac3to's operation:
eac3to v3.16
command line: eac3to.exe L: 1) 2: chapters.txt 3: video.mkv 4: audio.dts 8: subs08.sup
------------------------------------------------------------------------------
EVO, 1 video track, 4 audio tracks, 3 subtitle tracks, 1:54:50
1: Joined EVO file
2: Chapters, 20 chapters with names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: TrueHD, English, 5.1 channels, 48khz, dialnorm: -27dB
5: E-AC3, English, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB, -8ms
6: E-AC3, French, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB, -8ms
7: E-AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -8ms
"Director Commentary"
8: Subtitle (DVD), English, "SDH"
9: Subtitle (DVD), French
10: Subtitle (DVD), French, "Forced"
Creating file "chapters.txt"...
[v03] Extracting video track number 3...
[a04] Extracting audio track number 4...
[v03] Writing new framerate "24fps /1.001" to bitstream.
[a04] Removing TrueHD dialog normalization...
[v03] Removing VC-1 pulldown...
[a04] Decoding with libav/ffmpeg...
[a04] Writing WAVs...
[v03] Muxing video to Matroska...
[a04] Creating file "audio.C.wav"...
[a04] Creating file "audio.L.wav"...
[a04] Creating file "audio.R.wav"...
[a04] Creating file "audio.SL.wav"...
[a04] Creating file "audio.LFE.wav"...
[a04] Creating file "audio.SR.wav"...
[s08] Extracting subtitle track number 8...
[s08] Creating file "subs08.sup"...
[a04] Original audio track, L+R+C+LFE: constant bit depth of 16 bits.
[a04] Original audio track, SL+SR: constant bit depth of 24 bits.
[a04] Audio has a gap of 10ms at playtime 0:00:00. <WARNING>
[a04] Starting 2nd pass...
[a04] Extracting audio track number 4...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Writing WAVs...
[a04] Realizing RAW/PCM gaps...
[a04] Creating file "audio.L.wav"...
[a04] Creating file "audio.SL.wav"...
[a04] Creating file "audio.C.wav"...
[a04] Creating file "audio.R.wav"...
[a04] Creating file "audio.SR.wav"...
[a04] Creating file "audio.LFE.wav"...
[a04] Processed audio track, L+R+C+LFE: constant bit depth of 16 bits.
[a04] Processed audio track, SL+SR: constant bit depth of 24 bits.
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.29.0.
Surcode encoding successfully started. Please wait...
Closing Surcode...
Added fps value to MKV header.
Video track 3 contains 165207 frames.
Subtitle track 8 contains 1093 captions.
eac3to processing took 43 minutes, 37 seconds.
Surcode encoding took 14 minutes, 35 seconds.
Done.

now everything appears fine, but when I later compared the .dts file here with the equivalent .ac3 file I got when separately extracting track 5 - the English E-AC3 audio is perfectly in sync with the video, whereas this .dts audio is 89ms too early (it needs a +89ms delay to be in-sync with the E-AC3 and the video) - I measured this delay by comparing the .wav versions
eac3to audio.dts audio.wavs
eac3to audio2.ac3 audio2.wavs
of the centre-channels of each soundtrack, using a multichannel .wav file editor.

I watched the original disc on my HD-DVD HD-A30 player and there was no noticeable sync issue with the TrueHD track.

The 89ms sync issue was noticeable when watching the video muxed with this derived .dts audio on my PC. (Again, the same video muxed with .ac3 derived from the E-AC3 track was also in-sync on my PC)

Can it be that eac3to hasn't read/observed some kind of delay (separate meta-data?) that should apply to the TrueHD track in this case? And would this delay/offset only turn up when extracting TrueHD from HD-DVD sources (vs BluRay TrueHD tracks)?

(has the HD-DVD side of the code been abandoned anyway?!)

Thanks in advance of any help or feedback.

ps - probably an easier question - what does (DVD) signify in the subtitle track listings? Obviously, these extract as HD subs in a .sup file (albeit presumably an HD-DVD specific .sup variant)

[magic144]

just tried another HD-DVD disc
"Pride & Prejudice" - this also has both TrueHD and EAC-3 English audio alternatives.
Extracted both audio tracks directly to .wavs using eac3to 3.16 (latest). Here is the log.

eac3to v3.16
command line: eac3to L: 1) 4: audio_thd.wavs 5: audio_eac3.wavs
------------------------------------------------------------------------------
EVO, 1 video track, 4 audio tracks, 3 subtitle tracks, 2:08:18
1: Joined EVO file
2: Chapters, 16 chapters with names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: TrueHD, English, 5.1 channels, 48khz, dialnorm: -27dB
5: E-AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, -8ms
6: E-AC3, French, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, -8ms
7: E-AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -8ms
"Director Commentary"
8: Subtitle (DVD), English, "SDH"
9: Subtitle (DVD), French
10: Subtitle (DVD), French, "Forced"
[a05] The Nero decoder doesn't seem to work, will use libav instead.
[a04] Extracting audio track number 4...
[a05] Extracting audio track number 5...
[a04] Removing TrueHD dialog normalization...
[a05] Removing E-AC3 dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a05] Decoding with libav/ffmpeg...
[a04] Writing WAVs...
[a05] Remapping channels...
[a05] Applying RAW/PCM delay...
[a05] Reducing depth from 64 to 24 bits...
[a05] Writing WAVs...
[a04] Creating file "audio_thd.L.wav"...
[a04] Creating file "audio_thd.R.wav"...
[a04] Creating file "audio_thd.SR.wav"...
[a04] Creating file "audio_thd.C.wav"...
[a04] Creating file "audio_thd.LFE.wav"...
[a04] Creating file "audio_thd.SL.wav"...
[a05] Creating file "audio_eac3.L.wav"...
[a05] Creating file "audio_eac3.LFE.wav"...
[a05] Creating file "audio_eac3.SR.wav"...
[a05] Creating file "audio_eac3.C.wav"...
[a05] Creating file "audio_eac3.SL.wav"...
[a05] Creating file "audio_eac3.R.wav"...
[a04] Original audio track, L+R+C+LFE: constant bit depth of 16 bits.
[a04] Original audio track, SL+SR: constant bit depth of 24 bits.
Video track 3 contains 184586 frames.
eac3to processing took 22 minutes, 25 seconds.
Done.
Whilst the centre-channel audio was distinctly closer in time in this case, there was still a measurable 10ms gap between the THD and EAC3 variants, again with the THD lagging (i.e. it would need a +10ms audio delay to match the EAC3).

As a complementary test, last night I compared the centre-channel of the separate THD and AC3 tracks from my BluRay version of 2010 (the THD track actually has AC3 embedded, but I used eac3to to only extract the THD for comparison with a separate AC3 track). In this case, there was no measurable time lag between the two.

Unfortunately (or fortunately?) I don't have any more HDDVD discs with THD audio, however I will dig out a few more BluRay discs for comparison over the coming days/weeks and report any significant discrepancies if I find any. Perhaps others could repeat my experiment with their own HDDVD THD/EAC3 discs and report their findings.

[magic144]

FYI, just compared the distinct TrueHD and AC3 Japanese audio tracks from my Letters from Iwo Jima BluRay - another small measureable difference of -9ms (THD actually is AHEAD of the AC3 in this case), this would be very hard to detect perceptually - nothing like the 89ms from eac3to's audio extractions from the Elizabeth HDDVD.

Maybe the "Audio has a gap of 10ms at playtime 0:00:00. <WARNING>" message in the Elizabeth HDDVD THD track extraction log is a clue to the discrepency.

[HunterXXX]

Help me extract to E-ac-3 to Wav???

Format : E-AC-3
Format/Info : Audio Coding 3
File size : 1.16 GiB
Duration : 1h 47mn
Overall bit rate : 1 536 Kbps

Audio
Format : E-AC-3
Format/Info : Audio Coding 3
Duration : 1h 47mn
Bit rate mode : Constant
Bit rate : 1 536 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Surround: L R, LFE
Sampling rate : 48.0 KHz
Stream size : 1.16 GiB (100%)

[Beastie Boy]

See the first post of this thread for eac3to usage examples.

Cheers, Beastie.

[Nnoitra]

I notice that in some movies, like Kill Bill 1&2, forced subtitles are not detected. There is one big English subtitle track with all subs, and another much smaller track with just the forced subs, but eac3to does not detect it as forced (nor does any other tool like BDSupToSub). Yet standalone Blu-Ray players somehow figure this out... how?

[TruckChase!]

Hey guys; I've searched the thread and addressed all the concerns regarding this issue that have been posted already with no luck.

OS: Win7 x64 (I had this working fine with Vixta x64 with the same installs)

E:\Encoding\video\eac3to>eac3to.exe -test
eac3to (v3.16) 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.

Nero 7.8.5.0 installed, including showtime, and the BluRay/HDDVD plugin is successfully "unlocked". The codecs are showing up correctly, but eac3to doesn't seem to like 'em.

Like I said, I have the same config installed in Vista x64 and it works fine. Has anyone else had probs with Win7?

[JoyMonkey]

I've got the same problem with Win7 x86.
I'm using Nero Ultra 7.10.1.0 and the Blu-Ray/HD-DVD plugin license, which was working fine with Windows XP.

[JoyMonkey]

I found a solution to the Windows 7 problem!

For some reason the file...
C:\Program Files\Common Files\Ahead\DSFilter\NeAudio2.ax
didn't get installed with Nero on my Windows 7 machine. This is Nero's DirectShow audio filter module. I copied that file over from my XP machine, then registered that module as a service with Windows by opening a command prompt in that folder and using the command...
regsvr32 neaudio2.ax

Now when I run eac3to -test it says that Nero Audio Decoder is working fine.
:)

[Jeff Flowerday]

eac3to will decode DD+ without Nero 7. Nero 7 was only really needed for DD+ because it a certified DD+ decoder.

Madshi has indicated many times he can't hear the different between Nero doing the DD+ decoding and the default eac3to library.

So the question becomes how important is it to have DD+ certified decoding of a lossy stream or can you survive with the default eac3to decoding which gives you pretty much the same results? How many HD-DVDs do you have to rip, because DD+ only existing on HD-DVDs and maybe a couple Blu-Rays?

[TruckChase!]

I found a solution to the Windows 7 problem!

For some reason the file...
C:\Program Files\Common Files\Ahead\DSFilter\NeAudio2.ax
didn't get installed with Nero on my Windows 7 machine. This is Nero's DirectShow audio filter module. I copied that file over from my XP machine, then registered that module as a service with Windows by opening a command prompt in that folder and using the command...
regsvr32 neaudio2.ax

Now when I run eac3to -test it says that Nero Audio Decoder is working fine.
:)

Thank you, that worked great! (copied from my Vista install)

eac3to will decode DD+ without Nero 7. Nero 7 was only really needed for DD+ because it a certified DD+ decoder.

Madshi has indicated many times he can't hear the different between Nero doing the DD+ decoding and the default eac3to library.

So the question becomes how important is it to have DD+ certified decoding of a lossy stream or can you survive with the default eac3to decoding which gives you pretty much the same results? How many HD-DVDs do you have to rip, because DD+ only existing on HD-DVDs and maybe a couple Blu-Rays?

1. I'm anal.
2. I installed this bloatware (Nero + Arcsoft) on my new Win7 install, I want to use it damnit. :D

[ACrowley]

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?

[magic144]

@ACrowley
I saw this message for both HDDVD and BluRay audio streams just this week using eac3to 3.16
(HD-DVD was E-AC3 extracting to .ac3, BluRay was TrueHD/AC3 extracting to .ac3, both sources reported as having dialnorm: -27dB)
What is your source/command-line/output?

[ACrowley]

@ACrowley
I saw this message for both HDDVD and BluRay audio streams just this week using eac3to 3.16
(HD-DVD was E-AC3 extracting to .ac3, BluRay was TrueHD/AC3 extracting to .ac3, both sources reported as having dialnorm: -27dB)
What is your source/command-line/output?

Standard cmd using Gui

something like:
eac3to input.truhd output.wavs

Decoder is libavcodec for Dolby. I dont get this DialNorm removal Message anymore for TrueHD and AC3

[magic144]

I assume if your input file (.thd or .ac3) was previously extracted from a disc source, the DialNorm removal will have been done then (and only then) - unless perhaps you initially used keepDialnorm (I've never tried).

I still see it when I do something like this (track 2 being an AC3 stream in the source):-
eac3to L: 1) 2: audio.ac3

[EpheMeroN]

Can do eac3to do timestretching to audio?

I used to use BeHappy to use timestretch for audio that I wanted to change the pitch of, but maintain the same tempo when going from 25>23.976.

[tebasuna51]

Can do eac3to do timestretching to audio?

I used to use BeHappy to use timestretch for audio that I wanted to change the pitch of, but maintain the same tempo when going from 25>23.976.

The -slowdown eac3to parameter is equal to use in AviSynth (for a 48 KHz audio):
SSRC(50050).AssumeSampleRate(48000)

Maybe the SSRC routines used in Eac3to are more recent than SSRC routines in AviSynth.

And is equivalent to TimeStretch(rate=98.0/1.001), the duration and the pitch change.

Eac3to can't do the others TimeStretch modes:
TimeStretch(tempo=98.0/1.001), change duration and preserve the pitch
TimeStretch(pitch=98.0/1.001), change pitch and preserve duration

[odyssey]

It didn't work for me either, in Nero licence management, it clearly shows the BluRay/HDDVD plugin being registered but eac3to just can't detect it. It detects Arcsoft, Sonic and Surcode just fine but something about the Windows 7 architecture is preventing eac3to from accurately checking Nero 7 status. To get around this, I am running eac3to in a WinXP virtual machine.

Are there any update on this issue? I can't get it to work at all on Windows 7.

[ACrowley]

Are there any update on this issue? I can't get it to work at all on Windows 7.

same Problem here on W7 x64 RTM

Anyway, i use libavcodec for AC3/THd. THD output should be Bitperfect from libavcodec and i cant hear any difference for AC3 decoding

[Tarnatos]

Hello,

why does eac3to don't maintain the tone pitch when in use the slowdown cmd?

The voices always sounds muffled!

I use Windows XP x64 - current eac3to version and libAv.

[Boulder]

eac3to is unable to process the 5.0 AC3 track from the Häxan Criterion Collection.

Here's the log:
eac3to v3.16
command line: eac3to.exe "f:haxan.ac3" f:\temp\captures\haxan_1922_audio2.m4a -quality=0.40 -normalize -down2
------------------------------------------------------------------------------
AC3, 5.0 channels, 0:00:50, 384kbps, 48khz, dialnorm: -27dB
Removing AC3 dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 5.0 channels, 24 bits, 48khz
Writing WAV...
Creating file "f:\temp\captures\haxan_1922_audio2.m4a.pass1.wav"...
Starting 2nd pass...
Reading WAV...
Reducing depth from 64 to 32 bits...
Encoding AAC <0.40> with NeroAacEnc...
The Nero AAC encoder aborted processing. <ERROR>
[NeroAacEnc] ERROR: could not open AAC encoder
Aborted at file position 262144. <ERROR>

Encoding the other two tracks (2.0) works just fine.

Here's a sample: http://www.mediafire.com/?sharekey=9b19b691e7eb0ac9d2db6fb9a8902bda

[tebasuna51]

eac3to is unable to process the 5.0 AC3 track from the Häxan Criterion Collection.
...
Here's a sample: http://www.mediafire.com/?sharekey=9b19b691e7eb0ac9d2db6fb9a8902bda

You don't need samples, is NeroAacEnc the culprit, can't encode 5.0 audio, must be 5.1

[Boulder]

But why doesn't the down2 parameter work?

[tebasuna51]

why does eac3to don't maintain the tone pitch when in use the slowdown cmd?

The method used don't maintain the pitch.

And madshi say is the correct method to convert the audio previously speedup to put in PAL DVD: recover the original pitch.

But if your audio is recorded with a PAL camera and want slowdown preserving the pitch you need use other method like TimeStretch(tempo=100.1/0.96) in AviSynth

[tebasuna51]

But why doesn't the down2 parameter work?
Sorry, I don't see your -down2 parameter.
Yes this is a eac3to problem, only downmix 5.1 audio.

If you have AviSynth you can try:
NicAc3Source("f:haxan.ac3", Channels=2)

the downmix is Dolby ProLogic, or other downmix options,
with SoundOut, Wavi, BeHappy or MeGUI methods

[dchard]

I am using the latest TotalMedia Theater (v3) platinum on win7 x64, and eac3to continously sais that the Arcsoft decoder is not installed.

Can anyone help me to solve this problem?

Dchard

[Nnoitra]

I notice that in some movies, like Kill Bill 1&2, forced subtitles are not detected. There is one big English subtitle track with all subs, and another much smaller track with just the forced subs, but eac3to does not detect it as forced (nor does any other tool like BDSupToSub). Yet standalone Blu-Ray players somehow figure this out... how?

Nobody knows the answer to this? There must be some flag in the BD structure somewhere that marks these files as forced that eac3to isn't checking.

[saint-francis]

I have the Criterion Kagemusha and it has 4 channel DTS MA. eac3to is unable to cope with it. It spits out various errors. I'm using Arcsoft. Any ideas? I can upload a sample if someone would be so kind as to describe in detail how to cut it (I'm a movie watching enthusiast, bit a programmer so I'm not familiar with the use of hex editors).

[tebasuna51]

With a common tool you can extract the first MB:

VirtualDub -> Tools -> Hex Editor -> File (open the DTS file) -> Edit -> Extract segment -> 0 - 1000000 -> OK

[saint-francis]

With a common tool you can extract the first MB:

VirtualDub -> Tools -> Hex Editor -> File (open the DTS file) -> Edit -> Extract segment -> 0 - 1000000 -> OK

Thanks tebasuna.

Here's a sample: http://www.mediafire.com/?sharekey=cd17d0be1d2e2563d6baebe61b361f7c2d988c7d23ec3ed6935cbde7375ca78c

I hope someone can make heads or tails of it.

It's a sample of the entire m2ts BTW. If I need to upload just the DTS MA file I can do that too.

[TinTime]

Just tried your sample and it seems ok here. I decoded the DTS-MA to wav and all seems correct - 4 channels, correct channel mask.

What errors are you getting?

[saint-francis]

Just tried your sample and it seems ok here. I decoded the DTS-MA to wav and all seems correct - 4 channels, correct channel mask.

What errors are you getting?

I cam decode to WAV also. Or at least I got eac3to to produce a WAV. I can't encode to AC3. Is AC3 only 2 and 6 channels? What other format can I use then?

[tebasuna51]

You need use the external encoder Aften with the sintax, if the audio track is the second:
eac3to input.m2ts 2: stdout.wav | Aften -readtoeof 1 - output.ac3

[saint-francis]

You need use the external encoder Aften with the sintax, if the audio track is the second:
eac3to input.m2ts 2: stdout.wav | Aften -readtoeof 1 - output.ac3

Thanks tebasuna. But I can't seem to get it to work.

G:\the hive\software\video tools\eac3to\eac3to.exe F:\KAGEMUSHA 2: stdout.wav "C:\Program Files (x86)\megui\tools\aften\Aften.exe" -readtoeof 1 - F:\KAGEMUSHA\test.ac3

Nothing happens. What am I doing wrong? :(

[TinTime]

You haven't copied tebasuna51's command exactly (missing pipe), and you need quotes around paths / filenames containing spaces. Try...

"G:\the hive\software\video tools\eac3to\eac3to.exe" F:\KAGEMUSHA 2: stdout.wav | "C:\Program Files (x86)\megui\tools\aften\Aften.exe" -readtoeof 1 - F:\KAGEMUSHA\test.ac3

[saint-francis]

OK. Making some progress here. Now I get this:

"G:\the hive\software\video tools\eac3to\eac3to.exe" F:\KAGEMUSH
A 2: stdout.wav | "C:\Program Files (x86)\megui\tools\aften\Aften.exe" -readtoeo
f 1 - F:\KAGEMUSHA\test.ac3

Aften: A/52 audio encoder
Version SVN
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

input format: Raw PCM Signed 16-bit little-endian 48000 Hz mono
output format: 48000 Hz mono (1/0)

SIMD usage: MMX SSE SSE2 SSE3
Threads: 4

progress: 100% | q: 532.0 | bw: 38.0 | bitrate: 96.0 kbps

And nothing else. That all happened right away. It immediately said progress 100%. So I'm assuming there's an error, or I did something wrong. The resulting ac3 is 384 bytes. Nothing is happening for over 20 minutes now. Are there some additional Aften parameters or something?

Thanks for all of the help tebasuna and TinTime.

[tebasuna51]

OK. Making some progress here. Now I get this:...

And nothing else....
But your input file is a folder: F:\KAGEMUSHA
and you say me is a m2ts.
If you have a complete BD structure please put the log of this command line:

"G:\the hive\software\video tools\eac3to\eac3to.exe" F:\KAGEMUSHA

[saint-francis]

Right, right. Stupid error.

"G:\the hive\software\video tools\eac3to\eac3to.exe" F:\KAGEMUSH
A
1) 00001.mpls, 00257.m2ts, 3:00:37
- Chapters, 31 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48khz
- DTS Master Audio, Japanese, multi-channel, 48khz

2) 00752.mpls, 00260.m2ts, 0:43:45
- Chapters, 20 chapters
- h264/AVC, 1080i60 /1.001 (16:9)
- AC3, Japanese, stereo, 48khz

3) 00751.mpls, 00259.m2ts, 0:41:01
- Chapters, 7 chapters
- h264/AVC, 1080i60 /1.001 (16:9)
- AC3, Japanese, stereo, 48khz

4) 00795.mpls, 00267.m2ts, 0:19:21
- h264/AVC, 1080i60 /1.001 (16:9)
- AC3, English, mono, 48khz

It's 00001.mpls

EDIT:
I think I got it now.

[tebasuna51]

Your DTS-MA is the track 2?
Please run this before:

"G:\the hive\software\video tools\eac3to\eac3to.exe" F:\KAGEMUSHA 1)

[saint-francis]

Your DTS-MA is the track 2?
Please run this before:

"G:\the hive\software\video tools\eac3to\eac3to.exe" F:\KAGEMUSHA 1)

No. It was #4. I got it to work. I think. I guess I have gotten lazy with the eac3to GUI in MeGUI and I spaced the track number. Now I need to encode the video and see if everything matches up upon muxing.

Thanks for all of the help.

BTW, Why couldn't eac3to handle this on it's own?

[ron spencer]

is development still ongoing for eac3to? just curious

[madshi]

I've encountered a very strange problem with EAC3TO, at least to me. I loaded my copy of Ice Age 2 in eac3to. It said the length of the movie was 1h 30m 38s (all playlists). I ripped the Dutch 768kbps audiotrack. But when I played that seperate track in MPC-HC it was 1h 32m 15s. Other players and MediaInfo revealed the same. EAC3TO says the seperate audiotrack is 1h 30m 38s. I am confused. Who is right? And why?
eac3to is right. Blu-Ray DTS tracks are very slightly different to DVD DTS tracks and that throws off the runtime calculation of many applications and filters.

yep.. i got the same error in my application (but i read stdout)
the last line is truncated :confused:
Don't know why. I've added a "flush" now after every log line. Hopefully that helps.

Mkvmerge will drop the core when input TrueHD, the coreless THD will not be recognized by tsMuxeR，can eac3to adds a function to fix this issue？（Like combine a normal ac3 and a coreless TrueHD into a normal TrueHD)
That feature already exists. Just ask eac3to to output a "something.thd+ac3" file.

I do think THD has its own bits depth because it is lossless, why don't eac3to show it?
All TrueHD tracks are stored in 24bit. Some of those may be zero. eac3to shows the true bitdepth after having decoded a TrueHD track.

How to disable fixes using AV gaps during eac3to demuxing ts source?
-no2ndpass

As someone suggested a few pages back, this is clearly caused by highpoint's various sector size feature (which allows you to have partitions > 2TB e.g. in Win XP).
I could confirm this by creating a new array that is identical to the old one except I have left the sector size at the default value. Eac3to then has no trouble writing the the array.
For arrays with non-standard sector size: while eac3to can read from such raid arrays, it can't write to them.
Maybe you can add support for this?
eac3to uses simple "WriteFile" calls. That's the very core win32 API for writing files. I don't see what I could change there. Maybe I'm writing too big chunks at a time? That's not really my fault, though. If your RAID array can't cope with that, I'd consider that a big bug of your RAID array - or of its driver or something. Maybe you change the drive's cache settings somehow? Don't know...

I'm trying to re-encode a Blu-ray that's 24fps instead of the usual 23.976fps. Video and audio are fine, but the sub is out of sync.

Will this option fix my subtitle sync issue? Or will this cause other problems?
If your only problem is the subtitle sync issue I'd suggest trying to modify the subtitle FPS. E.g. SubAdjust can do that. That's much less violent than changing video and audio FPS!!

When you have time to get back to this, would you consider adding support for FLV (Flash Video) please? It would be great if eac3to can demux and give info on this file type.
You mean with MKV containers? That would not be easy because I'd have to add a full FLV bitstream parser. Possible, but too time consuming right now. Maybe somewhen in the future...

A nice feature for eac3to would extracting a FLAC stream from an mkv file. Currently, the FLAC stream is decoded then encoded back to FLAC.
eac3to always decodes FLAC if you feed FLAC to it. The reason for that is that eac3to does not have a FLAC parser. Instead it relies on libFLAC. The only disadvantage I can see is that it takes more time. I don't consider that much of a problem, so no change planned here.

just been converting an HD-DVD I own (Elizabeth: The Golden Age) and I came across the following issue.

[...]

now everything appears fine, but when I later compared the .dts file here with the equivalent .ac3 file I got when separately extracting track 5 - the English E-AC3 audio is perfectly in sync with the video, whereas this .dts audio is 89ms too early (it needs a +89ms delay to be in-sync with the E-AC3 and the video)

[...]

Can it be that eac3to hasn't read/observed some kind of delay (separate meta-data?) that should apply to the TrueHD track in this case? And would this delay/offset only turn up when extracting TrueHD from HD-DVD sources (vs BluRay TrueHD tracks)?

(has the HD-DVD side of the code been abandoned anyway?!)
The HD DVD side has not been abandoned. But finding out exact delays is really difficult. I've found problems sometimes, too. To be honest, I'm not sure if eac3to is doing something wrong or if some discs are simply not encoded properly. In your case you've also reencoded to DTS. Maybe Surcode adds its own bit of delay. Anyway, I don't have much hope of improving the situation.

ps - probably an easier question - what does (DVD) signify in the subtitle track listings? Obviously, these extract as HD subs in a .sup file (albeit presumably an HD-DVD specific .sup variant)
It means that the type of the subtitles is similar to DVD subtitles. Blu-Ray subtitles are totally different. HD DVD subtitles are similar to DVD.

I notice that in some movies, like Kill Bill 1&2, forced subtitles are not detected. There is one big English subtitle track with all subs, and another much smaller track with just the forced subs, but eac3to does not detect it as forced (nor does any other tool like BDSupToSub). Yet standalone Blu-Ray players somehow figure this out... how?
Probably the Blu-Ray player fetches some additional information from somewhere else. Maybe the menus have it hard coded or something. eac3to reads out the default fields. If the information is not available there, there's nothing more eac3to can do.

why does eac3to don't maintain the tone pitch when in use the slowdown cmd?
There are 2 different kinds of algorithms. One kind changes the pitch, another one doesn't. Both are useful. Neither one is "better" than the other. Depending on the circumstances you should choose one or the other. If you choose the wrong one, audio quality will suffer. eac3to's algorithm changes the pitch. If you need an algorithm which doesn't change the pitch, try e.g. TimeFactory (expensive).

Is it possible to add Radscorpion's Monogram AAC Decoder support with the option of "-mmaac"?
Using this, i got the best results even with problematic aac files.
If you need any problematic aac _supposed to be- which eac3to can not figure out- i mean nero audio decoder2 or libav inside eac3to-, i can pm a sample to you.
Is the problem the decoding or the bitstream parsing? If the bitstream parsing doesn't work properly, then it's an eac3to problem and no external decoder can help.

Eac3to has a big problem with seemless branching movies with THD soundtracks eg. Spiderman 2, Cars (reg.B), Ratatouille (reg.b).
That sounds weird to me. I've done countless seamless branching discs and never had a problem. Can anybody reproduce this? Or is only CapnBob affected (for whatever reason)?

Hi, i'm using Hdconverttox that use eac3to to muxing/demuxing/decoding/encoding...
Recently i'm having this problem constantly when I start from a mkv with aac audio tracks.
Can I have a sample, please?

The transcoding to FLAC from PCM 3.0 of the English audio stream on the "Vicky Cristina Barcelona" BluRay failed miserably with respect to channel mapping.
The dialogue is in the left surround speaker of the finished FLAC stream.
That's problematic because the problem could also be caused by the playback system. 3.0 audio tracks are simply weird... :(

Does eac3to have any way to convert from OGG Vorbis audio to AAC?
Currently no.

strange thing with the "A bug's life" BD, according to the log its 39.395p. but is this only a display bug, because anything else tells me its 23.976fps, when I play the remuxed file (length is 1:34:53; playlist info tells 24/1.001 as well)?
There are different sources of framerate information. E.g. you can see in the video track listing that eac3to correctly identifies the video track as being 24/1.001. However, the container timestamps seem to be weird. They are not very important, though, eac3to mostly ignores them. So don't worry. If you want me to look at this, I'd need a sample which is big enough so that the problem can be reproduced with the sample.

I get the same problem as Thunderbolt8 with Cars BD. Not sure if the fps in the stream is set correctly, haven't remuxed yet, but it would be helpful if in the log the true fps were indicated
Ok, have added that.

Does the -down6 switch mix the entire rear channels with the sides, or does it work in a different way?
Rear channels are mixed with the sides.

Has anyone been able to rip the 'Waiting...' Blu-Ray? When I type in
c:\eac3to.exe e: 2)
with 2) being the main movie, it just hangs there for 20 seconds or so, and then brings me back to the prompt. Every other disc I have tried has worked just fine. Thoughts?
Weird. That's the first time I hear about such a problem. Maybe the rip is broken in a weird way. Can you please try reripping the disc, just to be sure?

If a Blu-ray has multiple 7.1 tracks, lets say all three lossless/uncompressed varieties, and I want to downmix to 5.1, does it matter which one I use? I'm guessing the PCM track might better since the layout is more straight-forward.
I'd suggest using the PCM track cause there can be no decoding issues whatsoever. Next best option would be TrueHD because there are no known bugs. DTS-MA should work fine in most cases, too, but there are some (rare) tracks where ArcSoft stumbles over the speaker configuration.

madshi, can you implement extracting the subtitles from HDTV recordings? The subtitles can be viewed if playing the files with VLC.

6: Subtitle (DVB), Dutch
I wouldn't even know in which format to save DVB subtitles. Probably there is some format which can hold this kind of subtitles, but I don't know it. And it would probably cost a lot of time to implement. So right now I don't have much hope to implement that anytime soon.

do you plan to add VobSub format support of subtitles to eac3to? It was already disscussed here that it is not supported. But is it planned?
Sooner or later, probably.

I have a bug report for eac3to when using seamless branching blurays and .w64 audio extension. eac3to creates a w64 file then it detects the gaps and re writes the w64 file. The first file is 4.5 gigs but after it re-writes the w64 file it is only 465 megs. It seems to only write the gap information up to the first gap then it stops. Using the same file with the dts or pcm file extension creates a file the correct size.
Are you sure that your harddisk had enough free space? Can anybody else reproduce this problem?

Hi, I am seem to be encountering the same troubles as some previous users with the 300 HD-DVD.
That's a weird behaviour. Can you fully reproduce it? I think I've done 300 once and it worked for me, IIRC.

How can I disable "Patching bitdepth to 24 bits..." during extracting Core from DTS-HD?
Not at all. I see no good reason for disabling that.

When I use eac3to to demux evo-Containers with the -demux switch, eac3to demuxes not all the streams (especially subtitles with ID: 28 and up).
Can I have a sample, please? Are you sure there's actually data in those subtitles tracks?

Is there a way to split an MLP file with eac3to if you have a cue sheet or somehow just typing in where (in time/samples etc) to split the audio file?
No, sorry.

[madshi]

Would be .mpls file reading be a reasonable feature to add? i.e. just load a mpls playlist file in eac3to and get the order the m2ts files will be played in, regardless if you have the rest of the disc structure or not. Would probably also be useful for GUIs versions.
That feature was already there, however, it required the m2ts files to exist. I've now changed that so that it works even if the m2ts files are no longer available.

Anyone have any ideas on what might be wrong here.

eac3to v3.16
command line: "C:\VID2EVA\tools\eac3to\eac3to.exe" "F:\BDMV\STREAM" 1) "Z:\VIDEOS\Vid2eva.mkv" -seekToIFrames -stripPulldown -lowPriority
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 5 subtitle tracks, 1:44:05, 24p /1.001
1: Chapters, 24 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 6.1 channels, 24 bits, 48khz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48khz)
4: AC3, Spanish, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, -13ms
5: AC3, French, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, -13ms
6: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB, -19ms
7: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB, -19ms
8: Subtitle (PGS), English
9: Subtitle (PGS), French
10: Subtitle (PGS), Spanish
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), English
Creating file "Z:\VIDEOS\Vid2eva - Chapters.txt"...
[a03] The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
[a03] The libav DTS decoder doesn't decode the back channels. <WARNING>
[a03] The libav DTS decoder doesn't decode the full DTS-HD information. <WARNING>
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a05] Extracting audio track number 5...
[a06] Extracting audio track number 6...
[a06] Removing AC3 dialog normalization...
[a07] Extracting audio track number 7...
[s09] Extracting subtitle track number 9...
[a03] Extracting audio track number 3...
[s10] Extracting subtitle track number 10...
[s11] Extracting subtitle track number 11...
[s08] Extracting subtitle track number 8...
[s12] Extracting subtitle track number 12...
[a05] Removing AC3 dialog normalization...
[a04] Removing AC3 dialog normalization...
[a04] A remaining delay of -13ms could not be fixed.
[a07] Removing AC3 dialog normalization...
[a03] Extracting DTS core...
[a05] A remaining delay of -13ms could not be fixed.
[a06] Applying (E-)AC3 delay...
[a07] Applying (E-)AC3 delay...
[a07] A remaining delay of +13ms could not be fixed.
[a06] A remaining delay of +13ms could not be fixed.
[a03] Decoding with libav/ffmpeg...
[a03] Remapping channels...
[a03] Reducing depth from 64 to 24 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "Z:\VIDEOS\Vid2eva - 3 - DTS Master Audio, English, 6.1 channels, 24 bits, 48khz.flac"...
[a04] Creating file "Z:\VIDEOS\Vid2eva - 4 - AC3, Spanish, 5.1 channels, 448kbps, 48khz.ac3"...
[a05] Creating file "Z:\VIDEOS\Vid2eva - 5 - AC3, French, 5.1 channels, 448kbps, 48khz.ac3"...
[a06] Creating file "Z:\VIDEOS\Vid2eva - 6 - AC3, English, 2.0 channels, 224kbps, 48khz.ac3"...
[a07] Creating file "Z:\VIDEOS\Vid2eva - 7 - AC3, English, 2.0 channels, 224kbps, 48khz.ac3"...
[s12] Creating file "Z:\VIDEOS\Vid2eva - 12 - Subtitle (PGS), English.sup"...
[s08] Creating file "Z:\VIDEOS\Vid2eva - 8 - Subtitle (PGS), English.sup"...
[s10] Creating file "Z:\VIDEOS\Vid2eva - 10 - Subtitle (PGS), Spanish.sup"...
[a03] Clipping detected, a 2nd pass will be necessary. <WARNING>
[a03] The original audio track has a constant bit depth of 64 bits.
[a03] The processed audio track has a constant bit depth of 24 bits.
[s09] Creating file "Z:\VIDEOS\Vid2eva - 9 - Subtitle (PGS), French.sup"...
[s11] Creating file "Z:\VIDEOS\Vid2eva - 11 - Subtitle (PGS), Spanish.sup"...
[a03] The libav DTS decoder doesn't decode the back channels. <WARNING>
[a03] The libav DTS decoder doesn't decode the full DTS-HD information. <WARNING>
[a03] Starting 2nd pass...
[a03] Extracting audio track number 3...
[a03] Extracting DTS core...
[a03] Decoding with libav/ffmpeg...
[a03] Remapping channels...
[a03] Reducing depth from 64 to 24 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "Z:\VIDEOS\Vid2eva - 3 - DTS Master Audio, English, 6.1 channels, 24 bits, 48khz.flac"...
[a03] Writing the destination file failed. <ERROR>
[a03] One of the FLAC encoder's callbacks returned a fatal error. <ERROR>
Aborted at file position 56623104. <ERROR>
Can I have a sample, please?

I have a question about audio gaps, I have the following:


[a02] Audio has a gap of 39ms at playtime 0:03:39. -> +32(-7)
[a02] Audio has a gap of 41ms at playtime 0:03:55. -> +32(-16)
[a02] Audio has a gap of 41ms at playtime 0:16:51. -> +32(-25)
[a02] Audio has a gap of 44ms at playtime 0:25:56. -> +32(-37)
[a02] Audio has a gap of 22ms at playtime 0:37:43. -> +32(-25)
[a02] Audio has a gap of 54ms at playtime 0:47:29. -> +32(-47)
[a02] Audio has a gap of 29ms at playtime 0:55:48. -> +32(-44)
[a02] Audio has a gap of 28ms at playtime 1:03:05. -> +32(-40)
[a02] Audio has a gap of 21ms at playtime 1:11:36. -> +32(-29)
[a02] Audio has a gap of 58ms at playtime 1:27:22. -> +32(-55)


I put in the red, just assuming you would pad 32ms, the number in parenthesis would be the running total left over in ms at each spot, just wondering how you would actually fill the gaps. The video is 23.976FPS and the audio is 48KHz AC3.
Your list assumes that eac3to strictly handles each gap separately. That's not how eac3to works. eac3to always keeps track on how far audio has drifted from where it should be. So e.g. in your "+32(-25)" line eac3to would add in 2 audio frames.

Demuxing M2TS files produced by an HD camcorder: Canon HF100. The camcorder produces almost a BD structure on the flash (or is it a complete BD structure?).

Anyway, it seems like files shorter than 10 seconds or so can be demuxed just fine:

http://dl.getdropbox.com/u/29966/demux/00001.MTS

eac3to v3.16
command line: eac3to 00001.MTS 1: 00001.mkv 2: 00001.ac3 -seekToIFrames
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 0:00:11, 60i /1.001
1: h264/AVC, 1080i60 /1.001 (16:9)
2: AC3, 2.0 channels, 256kbps, 48khz
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[v01] Muxing video to Matroska...
[a02] Creating file "00001.ac3"...
Added fps value to MKV header.
Video track 1 contains 658 fields.
eac3to processing took 1 second.
Done.

However, files longer than 20 sec or so cannot be demuxed - eac3to fails at some point.

http://dl.getdropbox.com/u/29966/demux/00002.MTS

eac3to v3.16
command line: eac3to 00002.MTS 1: 00002.mkv 2: 00002.ac3 -seekToIFrames
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 0:00:21, 60i /1.001
1: h264/AVC, 1080i60 /1.001 (16:9)
2: AC3, 2.0 channels, 256kbps, 48khz
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[v01] Muxing video to Matroska...
Unfortunately the Haali Muxer cannot handle this source file. <ERROR>
It doesn't contain enough seek/recovery points. <ERROR>
The "-seekToIFrames" option may work around that.
However, using this option may result in artifacts when seeking.
Aborted at file position 35651584. <ERROR>

The position where it's aborted is not always exactly the same, but the values are pretty similar - always 30 ... 40 million.

What's going on? Does it fail after a GOP or something?

The files are valid, I can open them in various programs and they work very well.
Have you bothered to read the warnings eac3to reported to you? :) They explain fairly well what happened and what options you have. There's nothing more I have to add to that. If you have specific questions, let me know.

In the log file, if the command was for something like stream 1) please list the .m2ts files that would be reported when listing 1), e.g. 00000.m2ts+00183.m2ts+00184.m2ts
Makes sense, I'll add that to my to do list.

Does r8brain (pro) produce better quality than SSRC for pal/ntsc conversions?
Check out this one:

http://src.infinitewave.ca/

According to those measurements eac3to's SSRC implementation might be ever so slightly superior to r8brain pro's linear phase implementation. I don't really know if these measurements are good enough to fully judge, though. In the end, only your ear can decide.

What FPS values are supported? Is it planned to make it possible to use any custom FPS?
SSRC is limited in what framerates it supports. eac3to only allows default framerates (23.976, 24.000, 24.975, 25.000, 29.970, 30.000, 47.962, 48.000, 50.000, 59.940, 60.000).
What arguments I can use with ssrc_hp.exe to get same settings?
eac3to uses special coefficients which are not available in ssrc_hp. So there's no way to reproduce eac3to's resampling with ssrc_hp.exe.

Did you compare SSRC with SoX Resampler (GPL)?
The SoX resampler's "VHQ Linear Phase" configuration looks virtually identical to eac3to's SSRC implementation in the comparison site linked to above. I don't have done direct comparison tests with real audio files, though. So I can't really judge. Why don't you compare for yourself?

eac3to cannot use SoX' resamplers because eac3to is not open source and thus cannot use GPL code.

Thank you for your help but as mentioned earlier when I muxed with tsMuxeR only (avoiding EAC3to), the disc created played with no problem at all. So that leaves the only different viarable as EAC3to.
eac3to sets AC3 and E-AC3 files to -31, but simply "clears" the dialnorm field of TrueHD tracks. The simple reason for that is that I don't have a TrueHD format specification, so I don't know *exactly* which fields in the TrueHD header are responsible for dialnorm. I'm simply clearing a number of fields and that makes Nero's decoder not apply dialnorm, anymore. If I had exact documentation available I could use -31, which might be better than clearing. Maybe I should not remove dialnorm from TrueHD tracks by default - except for decoding. The main problem I see with that is that it would make the parameters more complicated again. Currently there's only "-keepDialnorm". I would have to readd "-stripDialnorm".

Opinions?

[madshi]

EAC3TO has problems with mpg clips from all HP digicams.
eac3to supports VOB and EVO containers. And MPG containers which behave similar to VOB. This specific MPG file you uploaded is different: It doesn't have the video data packages in PES packets. Instead the video data is directly part of the MPG structure itself. eac3to doesn't support this type of file. On the first direct video data packet eac3to aborts processing.

[madshi]

eac3to D:\ -progressnumbers
eac3to D:\ 1) -progressnumbers
doesn't work as I imagined, eac3to goes straight into processing.
I fixed the first. Not sure what you mean with the 2nd? It seems to work fine for me.

also when doing this on a HD-DVD:

eac3to D:\ 1)

multiple times, the results are different each time - like a counter is not being reset so the (results are 1), the 2), then 3) etc)
Can you post 2 logs for the same HD DVD with different order? Thanks...

[madshi]

eac3to v3.17 released

http://madshi.net/eac3to.zip

* updated libFLAC to version 1.2.1
* added a flush after every log line to help GUIs
* "eac3to some.mpls" now also works if the stream files aren't there, anymore
* fixed: number of subtitles was not appended to demuxed subtitles' file name
* fixed: dialnorm removal (for Nero decoder) failed with some 2.0 TrueHD files

[Inspector.Gadget]

The return of the king :thanks:

I look forward to testing out this new version.

[Thunderbolt8]

thanks!

EDIT: is there something wrong? when I just enter 'eac3to' then it directly says 'HD DVD/Blu-ray disc structure not found' and does not display what eac3to can do as usual.

[madshi]

EDIT: is there something wrong?
Yes, stupid me forgot to remove some debugging code. Please download again (same link). You may have to empty your browser cache to make sure you get the fixed one.

[honai]

Great to see you back, madshi!

If time permits, would you ponder adding FLAC->Apple Lossless conversion? AFAIK libavcodec supports encoding to Apple Lossless.

[TinTime]

Thanks very much for your continuing work on this (and madVR). It's much appreciated.

:thanks:

[ron spencer]

thanks!!!!

[dat720]

Madshi, excellent work as usual...

I'd like to propose a couple of features, assuming you have the time and consider the ideas to be useful.

Extend the -demux option to allow stream selection rather than all streams, it would be nice to be able to demux streams to their native format without having to specify file name's and extensions.
Example: eac3to e: 1) -demux 2:,5:

Linux friendly, I remember earlier on in this thread you quoted that there would be no Linux build as it works perfectly under wine, happy to use it this way it work's fine, however since you introduced console text colouring wine now throws a heap of garbage on the screen (see below), OS Detection would be nice, ie if its not win* then don't colour the console... even a simple -nocol switch would be useful.

Also is there a way to use dll's directly rather than having to have things installed, ie Haali Media Spliter, ArcSoft etc, would it be difficult to set it up so there is a directory called addon's for instance and all support files are dropped in there, rather than having to set everything up manually, this would also allow Linux (hopefully) to use these filters and codecs and also allow other Windows systems on the network to use them without having to go through the whole deal of installing everything, adding arcsoft to the path etc.

That's all for now :) Keep up the good work, it is highly appreciated.

PS have you setup a donation system yet? I'd like to donate some money to say thanks for all the hard work.

$ wine eac3to.exe -test
fixme:reg:GetNativeSystemInfo (0x32fce0) using GetSystemInfo()
Hooking GetModuleFileNameA failed (Success).
Hooking WriteConsoleA failed (Success).
eac3to (v3.17) is up to date
err:ole:CoGetClassObject class {83e69a3d-56a5-4850-89c9-f6b6f003eb9b} not registered
err:ole:CoGetClassObject no class object {83e69a3d-56a5-4850-89c9-f6b6f003eb9b} could be created for context 0x1
err:ole:CoGetClassObject class {83e69a3d-56a5-4850-89c9-f6b6f003eb9b} not registered
err:ole:CoGetClassObject no class object {83e69a3d-56a5-4850-89c9-f6b6f003eb9b} could be created for context 0x1
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.
err:ole:CoGetClassObject class {1a89085b-b654-4ed8-8831-5ac629a05ff5} not registered
err:ole:CoGetClassObject no class object {1a89085b-b654-4ed8-8831-5ac629a05ff5} could be created for context 0x1
err:ole:CoGetClassObject class {1a89085b-b654-4ed8-8831-5ac629a05ff5} not registered
err:ole:CoGetClassObject no class object {1a89085b-b654-4ed8-8831-5ac629a05ff5} could be created for context 0x1
ArcSoft DTS Decoder doesn't seem to be installed
http://www.arcsoft.com/products/totalmediatheatre
err:ole:CoGetClassObject class {9f062738-cd84-4f54-a3c4-bd5eb44f416b} not registered
err:ole:CoGetClassObject no class object {9f062738-cd84-4f54-a3c4-bd5eb44f416b} could be created for context 0x1
err:ole:CoGetClassObject class {9f062738-cd84-4f54-a3c4-bd5eb44f416b} not registered
err:ole:CoGetClassObject no class object {9f062738-cd84-4f54-a3c4-bd5eb44f416b} could be created for context 0x1
Sonic Audio Decoder (3.17.0.0) doesn't seem to be installed
err:ole:CoGetClassObject class {564fd788-86c9-4444-971e-cc4a243da150} not registered
err:ole:CoGetClassObject no class object {564fd788-86c9-4444-971e-cc4a243da150} could be created for context 0x1
Haali Matroska Muxer doesn't seem to be installed
http://haali.net/mkv
Nero AAC Encoder (1.3.3.0) is up to date
Surcode DTS Encoder doesn't seem to be installed
http://www.surcode.com
MkvToolnix doesn't seem to be installed
http://www.bunkus.org/videotools/mkvtoolnix

[Xorp]

madshi any chance you could integrate an open source AAC decoder like libfaad2? I know Nero has better quality but is the difference really that dramatic? Myself and others have also had some issues getting eac3to to detect the Nero filters sometimes. A built-in decoder would streamline setup for those that just need a quick conversion.

A aesthetic bug I noticed is that eac3to reports the dialnorm setting different between Dolby and DTS. For a TrueHD track it might say -27dB (the DN setting), but for a DTS-MA track it says -4dB (the DN offset). It's unlikely the DTS-MA track was actually set 27dB lower than standard.

[Snowknight26]

eac3to spits this out when analyzing an mkv (or flac):

eac3to v3.17
command line: eac3to.exe "C:\Users\Admin\Videos\Unsorted\F8.2001.480p.DVD.MPEG-2.mkv"
------------------------------------------------------------------------------
This doesn't seem to be a valid EVO/VOB file ($0). <ERROR>
MKV, 1 video track, 1 audio track, 0:12:27, 60i /1.001
1: MPEG2, 480i60 /1.001 (16:9)
2: AC3, 5.1 channels, 448kbps, 48khz
"AC3, 5.1 channels, 448kbps"


And this when analyzing my HD DVD:

C:\unzipped\eac3to>eac3to.exe "E:"
1) FEATURE_1.EVO+FEATURE_2.EVO, 2:17:03
"Main Movie"
- VC-1, 1080p (16:9)
- E-AC3, English, 5.1, 48khz
- E-AC3, Spanish, 2.0, 48khz
- E-AC3, French, 2.0, 48khz

C:\unzipped\eac3to>eac3to.exe "E:" 1)
$3f800
$40000
$40000
$40000
$40000
$40000
$40000
$40000
$40000
$40000
$40000
$40000
$40000
$40000
---------^C
C:\unzipped\eac3to>

[nyonya]

My problem with the "Waiting..." Blu-Ray was fixed with an update to AnyDVD HD. Awesome work madshi, love using eac3to and madflac, and look forward to trying madVR once I upgrade to a more powerful CPU!

[9700PRO]

Helo Madshi. Thank you very much for you EAC3TO.
But I have problem with her.
I have Windows 7 64 bit and i'm installed Nero 7. Nero work good and correct, but EAC3TO don't see it. Please help.

[madshi]

If time permits, would you ponder adding FLAC->Apple Lossless conversion? AFAIK libavcodec supports encoding to Apple Lossless.
If I do that, the next person will come and say: "You write Apple Lossless files, you should also be able to read them". It's a never ending circle, I guess...

Currently my focus is on madVR, so eac3to will have to make do with bugfixes and minor improvements for a while.

Extend the -demux option to allow stream selection rather than all streams
I don't consider that important, because you can simply demux everything and then delete whatever you're not interested in. The only benefit of the feature you've suggested would be a small gain in speed. I usually don't consider small speed gains as worth spending many hours of programming time on.

since you introduced console text colouring wine now throws a heap of garbage on the screen (see below), OS Detection would be nice, ie if its not win* then don't colour the console... even a simple -nocol switch would be useful.
How can I detect I'm running under wine?

And why is everybody expecting *ME* to fix wine problems? I'm not doing any hacks to do colored console text. I'm using highly official win32 APIs. If wine doesn't like that, then that's a very clear bug in wine. Have you tried reporting this to the wine programmers? ;)

Also is there a way to use dll's directly rather than having to have things installed, ie Haali Media Spliter, ArcSoft etc, would it be difficult to set it up so there is a directory called addon's for instance and all support files are dropped in there, rather than having to set everything up manually, this would also allow Linux (hopefully) to use these filters and codecs and also allow other Windows systems on the network to use them without having to go through the whole deal of installing everything, adding arcsoft to the path etc.
That's all nice and fine but it would bypass the license checks of the commercial products. And I don't want to do that.

madshi any chance you could integrate an open source AAC decoder like libfaad2? I know Nero has better quality but is the difference really that dramatic? Myself and others have also had some issues getting eac3to to detect the Nero filters sometimes. A built-in decoder would streamline setup for those that just need a quick conversion.
I'm not sure what to do with AAC decoding. I'm still waiting for the freeware "does it all in perfect quality" decoder. I was hoping that the command line decoder shipping with Nero's AAC encoder package would do the trick, and I've suggested a few changes to the Nero folks. But last time I checked they haven't implemented that yet. Haven't checked for a while, though. I can't use libfaad2 because eac3to is not GPL and so I can't use any GPL libs. I can only use LGPL.

A aesthetic bug I noticed is that eac3to reports the dialnorm setting different between Dolby and DTS. For a TrueHD track it might say -27dB (the DN setting), but for a DTS-MA track it says -4dB (the DN offset). It's unlikely the DTS-MA track was actually set 27dB lower than standard.
Hmmmm... I think I'm reporting dialnorm in the same way for both AC3 and DTS. Dialnorm is rare for DTS, anyway. Do you have a DTS track where you know for sure that eac3to's dialnorm information is incorrect?

eac3to spits this out when analyzing an mkv (or flac): [...]

And this when analyzing my HD DVD: [...]
Argh, another few debug messages I forgot to remove!! I've now re-uploaded eac3to v3.17 another time, with these debug messages removed, too.

My problem with the "Waiting..." Blu-Ray was fixed with an update to AnyDVD HD.
Good to hear!

I have Windows 7 64 bit and i'm installed Nero 7. Nero work good and correct, but EAC3TO don't see it. Please help.
Please check out the other posts about this. IIRC some other users have been able to make it work somehow.

[allak]

Would you consider adding a flag like -simpleoutput to disable console text colouring and console character position handling ? That is, just output the plain text log messages without any formatting.

This way the Linux users and the Cygwin users (like me) could use your tool without having to fire up the Windows console. Maybe it could make the live easier for the people that wrote the GUI wrappers.

Just a thought, it's obviously not a priority. Thanks anyway for the great tool.

[shon3i]

madshi any chance you could integrate an open source AAC decoder like libfaad2? I know Nero has better quality but is the difference really that dramatic?What you talking about. Both decoders follow aac standard, both have exatcly same bit indentical output. FAAD is used in foobar2000 and i don't see point someone need Nero Decoder while is there full free implementation.

I can't use libfaad2 because eac3to is not GPL and so I can't use any GPL libs. I can only use LGPL.
Maybe you can use CT AAC decoder from winamp??

[madshi]

Would you consider adding a flag like -simpleoutput to disable console text colouring and console character position handling ?
Again: I could do this. But have you actually contacted the Wine programmers about this? It's *their* bug and not mine. Why should I implement workarounds for other software's bugs?

Maybe you can use CT AAC decoder from winamp??
Is it a winamp plugin? I wouldn't know how to communicate with it in that case...

[honai]

If I do that, the next person will come and say: "You write Apple Lossless files, you should also be able to read them". It's a never ending circle, I guess...

Currently my focus is on madVR, so eac3to will have to make do with bugfixes and minor improvements for a while.

Ok, I understand.

A while back you mentioned that you were working or had implemented a plugin system for eac3to with support for plugging custom filters. Would that suit my needs (I'm a programmer myself)? Any documentation for it?

[dat720]

Fair enough, I just wanted to put forward some suggestions, your choice whether you run with them or not.

[shambles]

hooray! madshi!

i have the Freedom bluray set and i would like to remux the english versions but eac3to can't do it correctly, as the playlists use only parts of certain m2ts files. it's the only bluray where i've seen such playlists.

the main m2ts for each episode has the title sequence in japanese followed by the main body, so the japanese version uses the whole m2ts file, but the other language versions have their own title/credit sequence m2ts files, and only use the main body from the main m2ts.

i uploaded the japanese and english version playlists for the first disc here: http://www.mediafire.com/?sharekey=d0332f673c7857a5af924764f9977b1de04e75f6e8ebb871

[Kurtnoise]

@madshi : any chance to update the lavc lib ? there are EAC3 Spectral & AAC LATM patches floating around in the FFmpeg-dev ML...

[Xorp]

Hmmmm... I think I'm reporting dialnorm in the same way for both AC3 and DTS. Dialnorm is rare for DTS, anyway. Do you have a DTS track where you know for sure that eac3to's dialnorm information is incorrect? I don't have the disc anymore, but eac3to reported the US Warner version of Watchmen to have -4dB DN on the DTS-MA track.

[saint-francis]

BTW, Why couldn't eac3to handle this on it's own?

Sorry. I don't mean to be a pest, but, can anyone shed any light on this? I have little grasp on how I got that process to work, no idea why it didn't work out of the box and I'm concerned that I'll need to do it again and will again be dependent on the good souls on this forum.

[Guest]

I don't have the disc anymore We can't help you with this now, per forum rule 6.

[Snowknight26]

My Aeon Flux Blu-ray shows something similar:

G:\Encoding Tools\eac3to>eac3to.exe "H:\" 1)
M2TS, 1 video track, 6 audio tracks, 4 subtitle tracks, 1:32:52, 24p /1.001
1: Chapters, 14 chapters
2: MPEG2, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
4: DTS, English, 5.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB
5: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
7: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
8: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
9: Subtitle (PGS), English
10: Subtitle (PGS), English
11: Subtitle (PGS), French
12: Subtitle (PGS), Spanish

[zn]

Did you compare SSRC with SoX Resampler (http://sox.sourceforge.net/SoX/Resampling) (GPL)? Code (http://sox.cvs.sourceforge.net/viewvc/sox/sox/src/rate.c?view=markup) is under active developement (http://sox.cvs.sourceforge.net/viewvc/sox/sox/src/rate.c?view=log) right now.

The SoX resampler's "VHQ Linear Phase" configuration looks virtually identical to eac3to's SSRC implementation in the comparison site linked to above. I don't have done direct comparison tests with real audio files, though. So I can't really judge. Why don't you compare for yourself?

eac3to cannot use SoX' resamplers because eac3to is not open source and thus cannot use GPL code.

I was little bit inaccurate, libsox is library under LGPL, so it can be used

[Chumbo]

You mean with MKV containers? That would not be easy because I'd have to add a full FLV bitstream parser. Possible, but too time consuming right now. Maybe somewhen in the future...
It's good to see you back. I hope you're doing well.

No not MKV but FLV files, i.e., file.flv that contains the video and audio stream. Using eac3to to get info and demux would be ideal, but you're right, you'd probably need a full parser for this. If you have time and decide to implement this in the future that would be great. Thanks for considering it.

[73ChargerFan]

> Originally Posted by Nnoitra
> I notice that in some movies, like Kill Bill 1&2,
> forced subtitles are not detected.

Probably the Blu-Ray player fetches some additional information from somewhere else. Maybe the menus have it hard coded or something. eac3to reads out the default fields. If the information is not available there, there's nothing more eac3to can do.
SupRip can determine which subtitles are forced, with no other information than what is in the SUP file.

[jmonier]

SupRip can determine which subtitles are forced, with no other information than what is in the SUP file.

So can eac3to, but both it and SupRip use the forced flag which is not always there. Sometimes there is a separate subtitle track that is selected on the disc by means external to the track with no indication in the track (i.e. no forced flag) of it's function.

[9700PRO]

sorry but I forgot that this checkactivate.dll file is also required.


Atak_Snajpera please help!
checkactivate.dll file i didn't find. Where is he?
And please, give me advice - TMT 3.0.1.133 Patch doesn't exit anymore. Where is could be dtsdecoderdll.dll version 1.1.0.5.
Thanks you, very much.

[Paddy97]

Is there any way to demux DVB subtitles with eac3to?
I've tried several ways but get that the conversion is not supported. So if anyone can give me either a tip on a commandline or inform me if I am trying something that is impossible I would be happy.

eac3to shows the following when applying it to the DVB ts file so it can see the subtitles at least.

TS, 1 video track, 1 audio track, 4 subtitle tracks, 0:14:34, 50i
1: h264/AVC, 1920x1088 50i (30:17)
2: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
3: Subtitle (DVB), Swedish
4: Subtitle (DVB), Danish
5: Subtitle (DVB), Norwegian
6: Subtitle (DVB), Finnish

[nwg]

I have a problem with Pixar Shorts Blu Ray. It can see all the mpls but when trying to do a movie only, it cannot find anything about video, audio, subs etc.

It has 13 titles that are seamless branched which eac3to finds but that is far as I go. I have been using it with clown_bd but that is not the problem, as HdBrStream Extractor does the same.

eac3to v3.17
-----------------------------------------------------------------------
1) 00037.mpls, 0:55:27
[35+37+38+36+7+11+8+6+12+14+10+13+9].m2ts
- Chapters, 15 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, stereo, 48khz
- AC3, English, stereo, 48khz
- RAW/PCM, Spanish, stereo, 48khz
- AC3, Spanish, stereo, 48khz
- AC3, Catalan, stereo, 48khz
- AC3, English, stereo, 48khz

2) 00038.mpls, 0:55:27
[35+37+38+36+7+11+15+6+41+14+40+13+39].m2ts
- Chapters, 15 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, stereo, 48khz
- AC3, English, stereo, 48khz
- RAW/PCM, Spanish, stereo, 48khz
- AC3, Spanish, stereo, 48khz
- AC3, Catalan, stereo, 48khz
- AC3, English, stereo, 48khz

3) 00039.mpls, 0:37:47
[35+37+38+36+7+11+42+6+43+14+45+13+44].m2ts
- Chapters, 14 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, stereo, 48khz
- AC3, English, stereo, 48khz
- RAW/PCM, Spanish, stereo, 48khz
- AC3, Spanish, stereo, 48khz
- AC3, Catalan, stereo, 48khz
- AC3, English, stereo, 48khz

4) 00014.mpls, 00029.m2ts, 0:23:33
- Chapters, 12 chapters
- MPEG2, 1080i60 /1.001 (16:9)
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz

Choosing 1, 2 or 3 prodcues this error.

eac3to v3.17
-----------------------------------------------------------------------
This playlist contains m2ts parts with different track properties (2). <WARNING>

[TinTime]

Is the first m2ts in the playlist a copyright message or something? Anyway, if this is the case you might have to do this the hard way and specify the individual files yourself, excluding the one that eac3to doesn't like.

eac3to.exe 00037.m2ts+00038.m2ts+00036.m2ts+...etc.

Thinking about it, you said this is Pixar Shorts. Are the playlists that don't work the "play all" ones? If so you could look for the playlists for each individual short instead and do them separately.

[nwg]

Yes it is the play all that do not work. Each m2ts is a short.

[TinTime]

Unfortunately in that case I think you're going to have to process each short separately. eac3to doesn't support track changes during processing.

[magic144]

@Paddy97 - have you tried extracting with the wildcard suffix, e.g. something like

eac3to 1) 4: subs04.*

that'd be my first guess anyway, I've never encountered this subtitle type before...

[Snowknight26]

That won't work. The usual extenion for subtitles is either .sup, .sup, .ass. Count try .dvb as eac3to allows you to extract unrecognized formats with an extension of the format name.. though in this case, eac3to does recognize it, so who knows.

[Paddy97]

@magic144 and @Snowknight26

Thanks for the tips. Still no luck though.

"This subtitle conversion is not supported. "

Is the reply I get. Might be a format issue and not a commandline issue after all.

[Snowknight26]

Seems like madshi hasn't implemented DVB subtitle demuxing, even though eac3to has a DVB subtitle reader.

Found the string "DVB subtitles are not supported yet." when digging through the exe.

[techstuff]

hi

i'am a newbie so sorry,
have been looking for days in the this thread for a way to make multi wav from 6 or 8 mono wav. (1 wav with 6 or 8 channel mono)
can you do this or is it only from other multi audio format like AC3 DTS etc.

and go from 23.976 and 25.000 fps audio to 24.000 (-slowdown, -speedup)

hope you can help.

thanks

[tebasuna51]

have been looking for days in the this thread for a way to make multi wav from 6 or 8 mono wav. (1 wav with 6 or 8 channel mono)
eac3to don't support multiple mono inputs.
Use Sox, WaveWizard, AviSynth methods, ...

and go from 23.976 and 25.000 fps audio to 24.000 (-slowdown, -speedup)
When input is audio we can't know the original video fps (because audio don't know the video fps), if you need change the audio duration you need include both original video fps and destination video fps like this:

eac3to input.ac3 output.ac3 -23.976 -changeTo24.000

With:
-speedup equivalent to: -23.976 -changeTo25.000
-slowdown equivalent to: -25.000 -changeTo23.976

[techstuff]

hi

thank you tebasuna51 for your help and info.

[asc28]

Extracting from the root directory of Almost Famous R1 Blu-Ray. Weird EVO error even though it's a m2ts file

eac3to v3.17
command line: eac3to .\ 1) 2:D:\af.h264 3:D:\af.flac
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 20 subtitle tracks, 2:41:30, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
4: AC3, Polish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: Subtitle (PGS), English
6: Subtitle (PGS), Polish
7: Subtitle (PGS), Portuguese
8: Subtitle (PGS), Spanish
9: Subtitle (PGS), Chinese
10: Subtitle (PGS), Korean
11: Subtitle (PGS), Thai
12: Subtitle (PGS), Arabic
13: Subtitle (PGS), Bulgarian
14: Subtitle (PGS), Croatian
15: Subtitle (PGS), Czech
16: Subtitle (PGS), English
17: Subtitle (PGS), Hebrew
18: Subtitle (PGS), Hindi
19: Subtitle (PGS), Hungarian
20: Subtitle (PGS), Icelandic
21: Subtitle (PGS), Romanian
22: Subtitle (PGS), Slovak
23: Subtitle (PGS), Slovenian
24: Subtitle (PGS), Turkish
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Extracting TrueHD stream...
[a03] Removing TrueHD dialog normalization...
[a03] Decoding with libav/ffmpeg...
[a03] Encoding FLAC with libFlac...
[v02] Creating file "D:\af.h264"...
[a03] Creating file "D:\af.flac"...
[a03] The original audio track has a constant bit depth of 16 bits.
[a03] Superfluous zero bytes detected, will be stripped in 2nd pass.
This doesn't seem to be a valid EVO/VOB file ($0). <ERROR>
[a03] Starting 2nd pass...
[a03] Decoding FLAC...
[a03] Reducing depth from 24 to 16 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "D:\af.flac"...
[a03] The processed audio track has a constant bit depth of 16 bits.
Video track 2 contains 232321 frames.
eac3to processing took 52 minutes, 16 seconds.
Done.

[Snowknight26]

Weird EVO error even though it's a m2ts file

I don't know why people don't bother reading the last couple of pages when they post things like this. Redownload eac3to. madshi fixed that issue but didnt change the version number.

[Paddy97]

Found the string "DVB subtitles are not supported yet." when digging through the exe.

That makes it easier for me, I will stop trying for the time being :-)

[asc28]

I don't know why people don't bother reading the last couple of pages when they post things like this. Redownload eac3to. madshi fixed that issue but didnt change the version number.my bad. thanks that fixed it

[ACrowley]

Any News about the libav Lossless check failed on TrueHD Tracks with libavcodec ?

I would use Nero Decoder ,but teh Deocder doesnt work on Windows7 anymore

[PatlaborForce]

I would use Nero Decoder ,but teh Deocder doesnt work on Windows7 anymore

It works just fine on Windows 7 for me.

[ACrowley]

It works just fine on Windows 7 for me.

on W7 x64 ? How did you install it ?

Ah, now it works. I use standard NEro7 installer now (only Showtime). The BluRAy/HDDVD plugin is properly registered

Its a TrueHD+AC3 Track, now i get this :

Extracting TrueHD stream...
Decoding with DirectShow (Nero Audio Decoder 2)...
The DirectShow audio decoder didn't accept the input stream. <ERROR>
Aborted at file position 262144. <ERROR>

[PatlaborForce]

on W7 x64 ? How did you install it ?

The normal way you install Nero.

[ACrowley]

The normal way you install Nero.

yes, it works...eac3to finds teh Nero7 Decoder.
Can you decode TrueHD with it ?
The DirectShow audio decoder didn't accept the input stream. <ERROR>

EDIT2:
It works! i had to register the.ax manually

[DrNein]

Glad to see you back madshi. What is the status of the PGS subtitle issue regarding Decoder Time Stamp? Thanks.

[oddwunn]

Hi...I just got started using eac3to (thanks for this great tool, madshi!) and I have a real noob question.

Is this the correct Nero Blu-ray plugin that I want to use:

http://www.nero.com/enu/store-blu-ray-disc-authoring.html

If not, could some kind soul please direct me to where I can purchase the correct one?

[DrNein]

That's not it (and only for Nero 9 as well). It seems "BD/HD DVD Playback Plug-in" license is now only available in Asian-Pacific region and costs about $30 USD.

http://www.nero8.com/cht/store-blu-ray.html

[oddwunn]

Thank you, DrNein. If the Google translation is correct, I will be buying a serial number only. Is the plugin already installed in Nero 7, or do I have to find the file somewhere else?

[rica]

Welcome back to your home madshi, we'd missed you :)

[saint-francis]

Thank you, DrNein. If the Google translation is correct, I will be buying a serial number only. Is the plugin already installed in Nero 7, or do I have to find the file somewhere else?

It's just a serial #. You pop it in and you then can use the plugin.

[PatlaborForce]

yes, it works...eac3to finds teh Nero7 Decoder.
Can you decode TrueHD with it ?
The DirectShow audio decoder didn't accept the input stream. <ERROR>

EDIT2:
It works! i had to register the.ax manually

Yeah it works without having to do anything additionally.

[oddwunn]

Thanks, saint-francis...much appreciated!

[oddwunn]

Another newbie question - When I demux the the streams of a .m2ts file, I end up with subtitles that have the .SUP extension and chapters with .TXT extension, neither of which can be muxed into a .MKV container using MKVToolnix. What am I doing wrong?

Please note, if there is a guide somewhere that will explain all of this to me, I will gladly read it. The only guide that I found was the wiki, and it was not clear at all, or at least not to someone as new at this as I am.

[Inspector.Gadget]

MKVMerge doesn't handle BD-sup yet. Use Suprip to get a SRT or BDSup2Sub to get a VOBSub file.

Chapters files have to be selected in the "Global" tab, not the one where you drag and drop other streams.

[radius]

Hi guys,

I've been using clown_bd for ripping stuff and it looks like there an issue with the command line it uses to demux audio, -keepDialnorm is not being used on each track,
it results that some tracks keep the original dialog normalization and some get it removed...
Would there be any way, without encoding, to set the DN back in the tracks where DN was removed ?
Any possibility to get such function in eac3to ?

Thanks

[oddwunn]

Thanks, Inspector.Gadget! I tried both programs - Suprip had tons of unidentified characters, so it really was not useful, but BDSup2sub worked like a charm...much better...:)

And now I have chapters installed in my .MKV files...very cool!

[K1ngp1ng]

i noticed that eac3to isn't capable of removing dialnorm from dts master tracks. is this a known issue?

[Thunderbolt8]

guess you are refering to here: http://www.avsforum.com/avs-vb/showpost.php?p=17407462&postcount=251 (Transformers 2 IMAX BD)

was already reported here: http://forum.doom9.org/showpost.php?p=1333726&postcount=9401

[deathlord]

Hi madshi

Great to read from you!

eac3to uses simple "WriteFile" calls. That's the very core win32 API for writing files. I don't see what I could change there. Maybe I'm writing too big chunks at a time? That's not really my fault, though. If your RAID array can't cope with that, I'd consider that a big bug of your RAID array - or of its driver or something. Maybe you change the drive's cache settings somehow? Don't know...

I have tried everything in this respect. Eac3to can't write if the variable sector size feature is used. I have not had any troule with any other application, ever.
Currently, I do not require eac3to to write to such an array. So personally, I'm ok at this time.
However, since this might become annoying to myself or others in the future, it would be cool if it could be fixed.
Maybe it is a bug in the highpoint driver. But since highpoint do not respond to emails, there is nothing I can do about it.
If you want to track down the error, you could (temporarily) add an option to change the size of the chunks eac3to writes, so I could check if it has an influence. I don't know if this is difficult for you to do, though.

Another thing:
My "CSI: The First Season" Blu-ray contains 1080i60 VC-1. The resulting mkv is stuttering heavily in MPC, as opposed to any other VC-1 (or any Blu-ray i have tried, for that matter). You have mentioned problems with this format, so I guess it is that. Just let me know if you need a sample.

[73ChargerFan]

deathlord, I've just ripped about a dozen 1080i60 "extras" and found that -keepdialnorm to a mkv in eac3to, and then in mkvmerge setting the frame rate for that file to 60/1001 works like a charm. No more problems.

eac3to 1) 1: extra1.mkv 2: extra1.ac3 -keepdialnorm
then mux in mkvmerge

[Snowknight26]

keepDialNorm has nothing to do with successfully remuxing 1080i60 content.

[deathlord]

deathlord, I've just ripped about a dozen 1080i60 "extras" and found that -keepdialnorm to a mkv in eac3to, and then in mkvmerge setting the frame rate for that file to 60/1001 works like a charm. No more problems.

eac3to 1) 1: extra1.mkv 2: extra1.ac3 -keepdialnorm
then mux in mkvmerge

Are you talking about *VC-1* 1080i60 material?
As 73ChargerFan sais, dialnorm has nothing to do with it. But I haven't tried setting the framerate manually in mkvmerge. Does eac3to not write this value to the demuxed mkv file?

[Snowknight26]

It does write it. The only thing you could try is using the -seekToIFrame command, but that'll probably be fruitless. Infact, the issue is probably either a splitter or a decoder issue, not an eac3to issue... but who knows.

[MatMaul]

@madshi : do yo think you can patch the ffmpeg eac3 decoder with this patch to support spectral extension ?
http://lists.mplayerhq.hu/pipermail/ffmpeg-devel/attachments/20090805/381dbc22/attachment.diff

I would like to use eac3to with some of my DVB records but the eac3 tracks have spectral extension.

thanks !

[deathlord]

It does write it. The only thing you could try is using the -seekToIFrame command, but that'll probably be fruitless. Infact, the issue is probably either a splitter or a decoder issue, not an eac3to issue... but who knows.

I will try to use the ffdshow decoder in place of the one from windows. Maybe that helps.
I remember madshi writing something about 1080i60 VC-1 handling being imperfect atm. But I can't find it right now.

[Jeff Flowerday]

I will try to use the ffdshow decoder in place of the one from windows. Maybe that helps.
I remember madshi writing something about 1080i60 VC-1 handling being imperfect atm. But I can't find it right now.

There are definately issues with 1080i VC-1. It wasn't high on madshi's priorities because of it's rarity.

[kypec]

I couldn't figure out how to make bitrate based (not quality) AAC audio through eac3to and NeroAAC encoder with GUIs available here so I made myself a command script/batch for this very purpose. Should work on all Win2K/XP/Vista/7.
Feel free to modify it to your needs.:cool:
Single file usage example: aac.cmd VTS_01_1.VOB
Multiple files usage example: aac.cmd VTS_01_1.VOB+VTS_01_2.VOB please note there MUST NOT be space around plus sign when joining files
@echo off
set decoder="%PROGRAMFILES%\AVTools\eac3to\eac3to.exe"
set encoder="%PROGRAMFILES%\AVTools\eac3to\neroAacEnc.exe"
cls
if '%1'=='' goto error1
echo Analyzing input file(s). Please wait...
%decoder% %1
echo.
:choosetrack
set /p track=Type the number of track you want to convert:
echo Extracting selected track, please wait...
%decoder% %1 %track%:output_%track%.wav -normalize -libav
if %ERRORLEVEL% EQU 0 (goto choosecompression) else (goto error2)
:choosecompression
echo.
echo Option 1 = 96 kbps (2.0 stereo, medium quality)
echo Option 2 = 128 kbps (2.0 stereo, high quality)
echo Option 3 = 256 kbps (5.1 multichannel, medium quality)
echo Option 4 = 320 kbps (5.1 multichannel, high quality)
set /p quality=Choose your desired bitrate option (1, 2, 3, 4):
if not '%quality%'=='' set quality=%quality:~0,1%
if '%quality%'=='1' goto option_a
if '%quality%'=='2' goto option_b
if '%quality%'=='3' goto option_c
if '%quality%'=='4' goto option_d
echo Invalid option specified, try again
goto choosecompression

:error1
echo You must specify input VOB file(s) on the command line as a parameter!
echo Example (single file input): aac VTS_01_1.vob
echo Example (multiple files input): aac VTS_01_1.vob+VTS_01_2.vob
goto end
:error2
echo Error occurred while extracting track number %track%!
echo Please try again.
goto choosetrack
:error3
echo Error occurred while encoding file output_%track%.wav!
echo Please try again.
goto choosecompression

:option_a
set aac_options=-2pass -br 96000
goto encode
:option_b
set aac_options=-2pass -br 128000
goto encode
:option_c
set aac_options=-2pass -br 256000
goto encode
:option_d
set aac_options=-2pass -br 320000
goto encode

:encode
echo Encoding selected track, please wait...
%encoder% %aac_options% -if output_%track%.wav -of output_%track%.mp4
if %ERRORLEVEL% EQU 0 (exit) else (goto error3)
:end

[RainyDog]

Due to Miramax giving lossless honours to the English dub only on their Hero blu-ray, I've remuxed the original Mandarin DTS-ES track from the R3 DVD with an x264 encode of the blu-ray video. However, I needed to add 7900ms of delay for the A/V to sync up and when I put the DTS track through eac3to for this the resulting track has 7900ms of 'noise' at the beginning instead of silence. Is there anything I can do to avoid this? I simply imported the DTS track into eac3to (using the GUI), added the delay, chose to save as Hero_delay, selected .dts as the output format, and ran the command line.

Thanks in advance for any help.

[73ChargerFan]

Perhaps you can apply no audio delay in eac3to, and instead use apply the delay for that track in mkvmerge?
Or, if this will be the only track, clip off the first 8 seconds of the video (I don't know what tool though.)

[RainyDog]

Perhaps you can apply no audio delay in eac3to, and instead use apply the delay for that track in mkvmerge?
Or, if this will be the only track, clip off the first 8 seconds of the video (I don't know what tool though.)

Thanks for the response, 73ChargerFan. I've already chopped off the first 20 odd seconds of the video (the Miramax logo) using mkvmerge and can't lose anymore without cutting into the film itself. I've also tried adding the audio delay using mkvmerge, which works of course, and could settle on that solution. But I'm just been a perfectionist and looking to have both streams exactly the same length :) I've also tried delaycut which does do the same job as eac3to and without the noise... But unfortunately, that doesn't keep the DTS-ES Matrix header.

[73ChargerFan]

Perfection: Only you know the hacks you did to make it work.
Everyone else "wow, perfect dude!"

[djloewen]

I'm having some trouble figuring out exactly when eac3to does, and does not, correct sync/delay issues.

1.) If I extract a blu-ray audio track as WAV (or mp3 or aac), it applies any necessary delay directly to the WAV file, so if I convert/mux it later I don't need to add any delay at that point - is this correct?
2.) If I extract that same track without doing any conversion (say, as AC3), does it do the same thing? Or might there be a delay issue in this case?

[tebasuna51]

1) Yes
2) Yes. No

[TinTime]

2.) If I extract that same track without doing any conversion (say, as AC3), does it do the same thing? Or might there be a delay issue in this case?

TrueHD won't have the delay applied if you demux it. However the delay will be added to the filename I believe. If you decode it (instead of demux) then the delay will be applied. All other audio formats will be corrected automatically.

[ron spencer]

I am getting the $40000 errors with I am Legend HD DVD...3.16 is fine though 3.17 has the issue

[raquete]

eac3to v3.17
command line: eac3to sampleflac.flac samples.waves
-----------------------------------------------------
FLAC, 5.1 channels, 0:03:11, 24 bits, 4446kbps, 48khz
This audio conversion is not supported. <ERROR>

what i'm doing wrong and getting error?!

[Midzuki]

raquete wrote:

command line: eac3to sampleflac.flac samples.waves

Try wavs instead of waves.

[raquete]

raquete wrote:



Try wavs instead of waves.

:rolleyes: lol, i knew, was my fault....and is very clever in the help file that i read lots of times. :p
thanks so much Midzuki! :)

[Snowknight26]

I am getting the $40000 errors with I am Legend HD DVD...3.16 is fine though 3.17 has the issue

Redownload eac3to, was fixed a couple pages back...

[honai]

Can someone confirm that the eac3to + Arcsoft 1.1.0.5 (registered manually, legit version) toolchain works fine under Windows 7 Professional?

[ron spencer]

Thanks...did not realize the bugfix.

[Atak_Snajpera]

Can someone confirm that the eac3to + Arcsoft 1.1.0.5 (registered manually, legit version) toolchain works fine under Windows 7 Professional?
http://img266.imageshack.us/img266/6038/new1q.jpg

[Blue_MiSfit]

I can verify this also. No problems whatsoever with any DTS / HD / MA track I've touched in the last ~ 4 months - since RC1 :)

~MiSfit

[honai]

Thanks!

[SomeJoe]

I have tried everything in this respect. Eac3to can't write if the variable sector size feature is used. I have not had any troule with any other application, ever.
Currently, I do not require eac3to to write to such an array. So personally, I'm ok at this time.
However, since this might become annoying to myself or others in the future, it would be cool if it could be fixed.
Maybe it is a bug in the highpoint driver. But since highpoint do not respond to emails, there is nothing I can do about it.
If you want to track down the error, you could (temporarily) add an option to change the size of the chunks eac3to writes, so I could check if it has an influence. I don't know if this is difficult for you to do, though.


I have played with various arrays that support this feature, where the base block size of the device is larger than the standard 512 bytes.

I highly recommend against using this feature in any array. The NTFS file system routines in Windows were never designed to work with block devices with sector sizes other than 512 bytes. I have seen very odd behavior on systems that have one of these arrays installed in this mode.

The proper, known-to-work methods to use an array >2 TB are:

1) Define multiple 2TB LUNs using 512-byte sectors and MBR layout. Map the first LUN into the file system on a drive letter. Map other ones into the file system on folders under that drive letter. This allows all the storage to be seen, but each folder's space can't be used by other folders.

2) Define multiple 2TB LUNs using 512-byte sectors and MBR layout. Use Windows dynamic disks to pool all the 2TB LUNs together into one dynamic disk.

3) Define a >2TB LUN using 512-byte sectors, using GPT disk layout (requires Windows XP64, Server 2003 SP1 x86 or x64, Server 2008 x86 or x64, Server 2008 R2 x64 only, Vista x86 or x64, or Windows 7 x86 or x64). Format the large LUN with NTFS. This option requires that your drivers and device support 48-bit LBA for IDE/SATA devices or 64-bit LBA / 16-byte CDB for SCSI API-compatible controllers. Not all devices do, so check with the device manufacturer to be sure.

[Thunderbolt8]

got a problem with a H.264 .ts 50i(?) capture. plays allright in its normal .ts container (if you dont use ffmpeg-mt), but sync & playback speed is off when remuxing it .mkv (using the gap file doesnt really seem to make a difference). audio speed seems to be normal, but video playback looks like its running at 2x speed. displayed length of the video in mpc is ~54mins, even though mediainfo tells 1h 48mins and the original .ts video also is 1h 48mins. cannot tell how much the audio is off, as the spoken language is not english.

50mb sample: http://www.sendspace.com/file/qltp8n

and the log: http://www.sendspace.com/file/tnywyl

[73ChargerFan]

I had the exact same experience a few weeks ago with some 60i sources.

-keeppulldown fixed it.

[Thunderbolt8]

what does pulldown mean here?

[73ChargerFan]

-keepPulldown disable removal of pulldown for MPEG2, h264 and VC-1 tracks

Now I'm confused. I had tried a few things, and iirc keeppulldown allowed the file to play correctly. It was with with the extras on Region 1 Superman/Batman Public Enemies BD.

Also when I found video gaps I re-ran the eac3to xxx.m2ts xxx.mkv command which used the gaps file.

EDIT - Sendspace really s**ks. I can't get the TS file.

[deathlord]

I have played with various arrays that support this feature, where the base block size of the device is larger than the standard 512 bytes.

I highly recommend against using this feature in any array. The NTFS file system routines in Windows were never designed to work with block devices with sector sizes other than 512 bytes. I have seen very odd behavior on systems that have one of these arrays installed in this mode.

The proper, known-to-work methods to use an array >2 TB are:

1) Define multiple 2TB LUNs using 512-byte sectors and MBR layout. Map the first LUN into the file system on a drive letter. Map other ones into the file system on folders under that drive letter. This allows all the storage to be seen, but each folder's space can't be used by other folders.

2) Define multiple 2TB LUNs using 512-byte sectors and MBR layout. Use Windows dynamic disks to pool all the 2TB LUNs together into one dynamic disk.

3) Define a >2TB LUN using 512-byte sectors, using GPT disk layout (requires Windows XP64, Server 2003 SP1 x86 or x64, Server 2008 x86 or x64, Server 2008 R2 x64 only, Vista x86 or x64, or Windows 7 x86 or x64). Format the large LUN with NTFS. This option requires that your drivers and device support 48-bit LBA for IDE/SATA devices or 64-bit LBA / 16-byte CDB for SCSI API-compatible controllers. Not all devices do, so check with the device manufacturer to be sure.

Thanks for the info!

[73ChargerFan]

got a problem with a H.264 .ts 50i(?) capture. plays allright in its normal .ts container (if you dont use ffmpeg-mt), but sync & playback speed is off when remuxing it .mkv (using the gap file doesnt really seem to make a difference). audio speed seems to be normal, but video playback looks like its running at 2x speed. displayed length of the video in mpc is ~54mins, even though mediainfo tells 1h 48mins and the original .ts video also is 1h 48mins. cannot tell how much the audio is off, as the spoken language is not english.

50mb sample: http://www.sendspace.com/file/qltp8n

and the log: http://www.sendspace.com/file/tnywyl
I've got it figured out:

eac3to test.ts -demux
(version 3.17, ignore erroneous gap warnings)

mkvmerge GUI v2.9.8:
- add test.h264, select stream, go to Format specific options tab, FPS "50000/1001", display width/height set to 1920x1080
- add test.ac3 stream
- merge to test.mkv

mpc-hc v1299 plays test.ts & test.mkv correctly.

BUG - 1 minute 50i stream "eac3to test.ts 1: test.mkv" detects gaps incorrectly. When run a second time, the gaps file generates a test.mkv video file that is over 4 minutes in length.

[heerschop]

Hello madshi,

I use Eac3to to quickly convert my mp3 files to ac3. Recently I have noticed that some ac3 files were distorted.
So I converted the mp3 to wav and they were also distorted.
After a bit experimenting I discovered the problem.
It turns out that libav/ffmpeg reports a wrong bit depth to eac3to.
The bit depth of the mp3 is 16 bits as reported by nero but libav/ffmpeg think it is 32 bits. This is causing the distorting. The problem is salved when I use nero to decode.

Hope you can fix the problem.

Greetings,
Ben

Used Eac3to 3.17

Let nero decode:

C:\tools\eac3to\eac3to.exe J:\term.mp3 J:\termb.ac3 –nero –down16

MP3, 2.0 channels, 1:57:18, 128kbps, 48khz
Disabling DRC for Nero (E-)AC3 decoding...
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 2.0 channels, 16 bits, 48khz
Encoding AC3 <448kbps> with libAften...
Creating file "J:\termb.ac3"...

Let libav/ffmpeg decode:

C:\tools\eac3to\eac3to.exe J:\term.mp3 J:\termb.ac3 -down16

MP3, 2.0 channels, 1:57:18, 128kbps, 48khz
Decoding with libav/ffmpeg...
Reducing depth from 32 to 16 bits...
Encoding AC3 <448kbps> with libAften...
Creating file "J:\termb.ac3"...

[Snowknight26]

MP3 files have no bit-depth so ffmpeg uses 32-bit floating point by default. -full which uses 64-bit floating point and -down16 which downsamples the audio to 16 bits still causes the issue, so it's not a bit-depth issue. All that eac3to probably needs is an updated libavcodec.

[Thunderbolt8]

I've got it figured out:

eac3to test.ts -demux
(version 3.17, ignore erroneous gap warnings)

mkvmerge GUI v2.9.8:
- add test.h264, select stream, go to Format specific options tab, FPS "50000/1001", display width/height set to 1920x1080
- add test.ac3 stream
- merge to test.mkv

mpc-hc v1299 plays test.ts & test.mkv correctly.

BUG - 1 minute 50i stream "eac3to test.ts 1: test.mkv" detects gaps incorrectly. When run a second time, the gaps file generates a test.mkv video file that is over 4 minutes in length.
thanks for testing!
so does the error occur, because eac3to mistakes the framerate as 25i instead of 50i?
as the log said: [v01] The video bitstream framerate field doesn't seem to match the timestamps. <WARNING>

[73ChargerFan]

I don't know where the bug is, but I think the gap warnings are incorrect.
Using the gaps file doubles the length of the video, more or less.
Ignoring the gaps yielded the correct result.

[Chumbo]

I'm looking for some help on a strange one that I've been unable to mux. Basically, when I load the M2V file into MPC-HC and let the player load the accompanying AC3 audio track, the movie plays fine and stays in sync all the way through. However, when I mux it, the audio/video go out of sync. I've tried everything I can think of, i.e., mux to MKV, TS and M2TS containers plus tried different frame rates and so on but it's always just off at the end. Probably around 3 seconds off when using 59.940 fps.

I also no longer have the original TS file which irritates me that I deleted it. Below is the eac3to log from when it was demuxed and you can obviously see the video overlaps are many. For brevity, I've removed the myriad of lines, but essentially they appear all the way through every 3 and 4 seconds apart.eac3to v3.17
command line: eac3to "l:\hd movies\my.movie_720p_dd51.ts" -demux
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 1:50:49, 59.960p
1: MPEG2, 720p60 /1.001 (16:9)
2: AC3, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[v01] Creating file "my.movie_720p_dd51 - 1 - MPEG2, 720p60.m2v"...
[a02] Creating file "my.movie_720p_dd51 - 2 - AC3, 5.1 channels, 384kbps, 48khz.ac3"...
[v01] Video overlaps for 1 frames at playtime 0:01:42. <WARNING>
[v01] Video overlaps for 1 frames at playtime 0:01:46. <WARNING>
[v01] Video overlaps for 1 frames at playtime 0:01:49. <WARNING>
[v01] Video overlaps for 1 frames at playtime 0:01:53. <WARNING>
[v01] Video overlaps for 1 frames at playtime 0:01:56. <WARNING>
...
[v01] Video overlaps for 1 frames at playtime 1:50:33. <WARNING>
[v01] Video overlaps for 1 frames at playtime 1:50:37. <WARNING>
[v01] Video overlaps for 1 frames at playtime 1:50:40. <WARNING>
[v01] Video overlaps for 1 frames at playtime 1:50:44. <WARNING>
[v01] Video overlaps for 1 frames at playtime 1:50:47. <WARNING>
Video track 1 contains 398661 frames.
eac3to processing took 9 minutes, 53 seconds.
Done.
Any ideas on remuxing this to an MKV container would be greatly appreciated. And yes I did try the listed frame rate of 59.960 and that didn't help either. :(

[EDIT] Wouldn't you know it, after I post this I realized something. The one thing I didn't try is tsmuxer with a custom frame rate setting in a meta file and running the command-line. The UI does NOT allow you to enter a custom FPS setting nor does it let you edit the Meta file section so I was limited to 59.9401. Any how, setting the fps to 59.960 did work. I'm not sure why I don't get the same result for an MKV, but I'm happy with having a playable M2TS file. Whew...

[EDIT2] I got the MKV mux working correctly now. The timecodes file alone with 59.960 wasn't enough. I also manually added 60020/1001 to the FPS input on the "Format specific options" tab for the M2V stream and that did the trick.

[Snowknight26]

I accidentally encoded a file with the wrong framerate, so when I tried to change it (and demux the audio at the same time), eac3to spat out a lot of video overlaps that shouldn't be there:

G:\Encoding Tools\eac3to>eac3to.exe ..\temp\ipcressfile.done.mkv 1: ..\temp\ipcr
essfile.h264 -changeto24.000 2: ..\temp\ipcressfile.ac3
MKV, 1 video track, 1 audio track, 1:47:40, 24p
1: h264/AVC, English, 1920x816 24p /1.001
2: AC3, English, 5.1 channels, 640kbps, 48khz
"AC3, 5.1 channels, 640kbps"
v01 The video bitstream framerate field doesn't match the container framerate.
v01 Extracting video track number 1...
a02 Extracting audio track number 2...
v01 Writing new framerate "24fps" to bitstream.
v01 Creating file "..\temp\ipcressfile.h264"...
a02 Creating file "..\temp\ipcressfile.ac3"...
v01 Video overlaps for 4 frames at playtime 0:05:27.
v01 Video overlaps for 2 frames at playtime 0:05:46.
v01 Video overlaps for 2 frames at playtime 0:05:51.
v01 Video overlaps for 3 frames at playtime 0:05:57.
v01 Video overlaps for 11 frames at playtime 0:06:05.
v01 Video overlaps for 1 frames at playtime 0:06:38.
v01 Video overlaps for 15 frames at playtime 0:06:40.
v01 Video overlaps for 4 frames at playtime 0:06:42.
v01 Video overlaps for 13 frames at playtime 0:07:38.
v01 Video overlaps for 1 frames at playtime 0:08:16.
v01 Video overlaps for 1 frames at playtime 0:08:19.
v01 Video overlaps for 2 frames at playtime 0:08:21.
v01 Video overlaps for 15 frames at playtime 0:08:29.
v01 Video overlaps for 2 frames at playtime 0:09:12.
v01 Video overlaps for 1 frames at playtime 0:09:18.
[...]

Also, if you Ctrl+C at the wrong time, the colors stay in the command prompt, but I'm not sure if theres a fix for that.
http://i38.tinypic.com/fvcsh0.jpg

[ACrowley]

I had Overlaps on some3 DTS HD Tracks too.

But the Output from eac3to was perfectly fine after 2nd Pass.

[Terrachild]

I have two questions for you:

1) I've recently gotten into a debate with some people online that use the Surcode plug-in. I need some facts. If I use eac3to to encode an ac3 file, is the quality substantially different then Surcode? Is FFmpeg or Aften better?

2) I created a 6 track wave file in audacity, when I try to convert it with eac3to to an ac3 file, I get an error that says : "

eac3to v3.17
command line: "C:\Users\Admin\Desktop\eac3to\eac3to.exe" "C:\Users\Admin\Desktop\Surround Test2.wav" "C:\Users\Admin\Desktop\Surround Test2.ac3"-448
------------------------------------------------------------------------------
WAV, 5.1 channels, 0:00:31, 16 bits, 4608kbps, 48khz
This audio conversion is not supported. <ERROR>"


I've tried different bitrates out of Audacity, but nothing works. What am I doing wrong.

[Beastie Boy]

eac3to v3.17
command line: "C:\Users\Admin\Desktop\eac3to\eac3to.exe" "C:\Users\Admin\Desktop\Surround Test2.wav" "C:\Users\Admin\Desktop\Surround Test2.ac3"-448
------------------------------------------------------------------------------
WAV, 5.1 channels, 0:00:31, 16 bits, 4608kbps, 48khz
This audio conversion is not supported. <ERROR>"


I've tried different bitrates out of Audacity, but nothing works. What am I doing wrong.

Try inserting a space before -448. It could be a syntax error.

Cheers, Beastie.

[Terrachild]

Thanks Beastie, that was it.
I'm using the GUI by The_Keymaker, version 2.00.
In the command line Preview it shows a space where you said to put it, but when the command window opens there is no space there. What's going on? Is that a bug in the GUI? Is there a better GUI to use?

[Beastie Boy]

Personally I use the command line, but this GUI (http://forum.doom9.org/showthread.php?t=141829) by Greif is a good option.

[tebasuna51]

1) I've recently gotten into a debate with some people online that use the Surcode plug-in. I need some facts. If I use eac3to to encode an ac3 file, is the quality substantially different then Surcode?

Read '>' like 'better quality than':
AC3 640 > DTS 1536 > AC3 448 > DTS 768

Is FFmpeg or Aften better?.
The same guy (Justin Ruggles) wrote Aften and ac3 code in ffmpeg, but Aften have more options to tune the encode.

[Terrachild]

Read '>' like 'better quality than':
AC3 640 > DTS 1536 > AC3 448 > DTS 768


The same guy (Justin Ruggles) wrote Aften and ac3 code in ffmpeg, but Aften have more options to tune the encode.

So are they equally up to date? In other words, if I'm doing 5.1 ac3 without settings anything other than the bitrate, is there any reason to chose one over the other?

Audacity uses FFmpeg for encoding ac3. Should I stick with that, or is there an advantage to eac3to with Aften?

Do they come close to Surcode for quality?

[leeperry]

AC3 640 > DTS 1536
got any source about this please? the more I upgrade my sound system, the less I stand 1.5mbit DTS :mad:

[Terrachild]

To answer another debate I've entered into, could someone tell me if it's legal to use an open-source ac3 encoder for non-commercial use. Just a guy sitting at home testing out surround sound. Does the act of simply making an ac3 break any laws in the U.S.?

[tebasuna51]

So are they equally up to date? In other words, if I'm doing 5.1 ac3 without settings anything other than the bitrate, is there any reason to chose one over the other?

Audacity uses FFmpeg for encoding ac3. Should I stick with that, or is there an advantage to eac3to with Aften?
Well, ffmpeg have many forks, branches, compiles, ... then I can't know what is your ffmpeg and can't say nothing about.

I can't test all ffmpeg versions, I always use Aften.

Do they come close to Surcode for quality?

I don't know blind test between Aften and Surcode.
For me, with my audio equipment and my ears, sound the same.

Does the act of simply making an ac3 break any laws in the U.S.?
I don't know in U.S. but we are talking about audio encoded (A/52) AC3 compatible, not Dolby Digital.

[MokrySedeS]

if I'm doing 5.1 ac3 without settings anything other than the bitrate, is there any reason to chose one over the other?

I've made a test but still... I'm not sure what is better.


Here it is:

Original wav............................................ Slowed down wav.................................... eac3to encode......................................... Aften encode
http://thumb.phyrefile.com/m/md/mdgedegedegedebaba/2009/11/05/wavoryg.png (http://img.phyrefile.com/mdgedegedegedebaba/2009/11/05/wavoryg.png) http://thumb.phyrefile.com/m/md/mdgedegedegedebaba/2009/11/05/wavslowed.png (http://img.phyrefile.com/mdgedegedegedebaba/2009/11/05/wavslowed.png) http://thumb.phyrefile.com/m/md/mdgedegedegedebaba/2009/11/05/eac3to.png (http://img.phyrefile.com/mdgedegedegedebaba/2009/11/05/eac3to.png) http://thumb.phyrefile.com/m/md/mdgedegedegedebaba/2009/11/05/aften.png (http://img.phyrefile.com/mdgedegedegedebaba/2009/11/05/aften.png)

Original wav is encoded using eac3to from 448kbps 25fps AC3 track.
Slowed down wav is made the same way original wav was, just slowed to 23.976fps.
eac3to encode is made from slowed down wav and the only used setting was bitrate at 448kbps.
Aften encode is made from slowed down wav, bitrate at 448kbps and tweaked -w option (bandwidth).

When I did an aften encode leaving -w set at default, the result was exactly the same as eac3to's.

So what do you guys think... should I use aften and tweak bandwidth or is it a bad idea?

[jruggle]

Read '>' like 'better quality than':
AC3 640 > DTS 1536 > AC3 448 > DTS 768


The same guy (Justin Ruggles) wrote Aften and ac3 code in ffmpeg, but Aften have more options to tune the encode.

That is not true. Though I am listed as the current maintainer of it, I did not write the AC3 encoder in FFmpeg. It was written by Fabrice, who started FFmpeg. I used his code as the starting point for Aften. The original FFmpeg AC3 encoder has not changed much since it was first written. I made a few tweaks and some speed improvements, but nothing significant.

So are they equally up to date? In other words, if I'm doing 5.1 ac3 without settings anything other than the bitrate, is there any reason to chose one over the other?
FFmpeg is a simpler encoder than Aften. It only allows 16-bit input, and it uses 16-bit fixed-point for the MDCT and other calculations. It does not have any options other than bit rate and frequency cutoff. Aften allows higher resolution input such as 24-bit and floating-point, it uses floating-point calculations internally, and it has many options. Aften is also multi-threaded.

That said, the quality is not significantly different. There is only so much you can do with the format. The core encoding features are basically the same, the 2 encoders just use different algorithms in several places.

[jruggle]

So what do you guys think... should I use aften and tweak bandwidth or is it a bad idea?
The only downside of increasing the bandwidth is that the encoder has to spread the available bits across more frequency coefficients, so it can decrease overall quality.

[tebasuna51]

That is not true. Though I am listed as the current maintainer of it, I did not write the AC3 encoder in FFmpeg... I made a few tweaks and some speed improvements, but nothing significant...

Sorry, my fault. I was thinking your tweaks like current maintainer was more important.

[Abradoks]

I have recently made an objective comparison between proprietary Dolby encoders and aften. Aften has much worse SNR than DD. Even 224 kbps AC3 encoded with DD beats aften 448 kbps in SNR. When using PEAQ metric aften is closer to DD, but still worse. :(
Also, if you want to have frequency cutoff similar to DD encoders, you should use w=48 for 448 kbps and w=8 for 224 kbps. This tweak gives a little worse SNR but better PEAQ on 448 kbps (w=8 gives worse PEAQ than default settings).
If somebody is interested, I can repost here that comparison. But, it seems that development of AC3 encoders is dead.

[tebasuna51]

...
If somebody is interested, I can repost here that comparison. But, it seems that development of AC3 encoders is dead.
Two post before you have jruggle the Aften developper.

Maybe he need some support to improve the encoder, everybody is interested in a good free ac3 encoder.

But, please, use the Aften thread (http://forum.doom9.org/showthread.php?p=1341620#post1341620) for this.

[shon3i]

When using PEAQ metric aften is closer to DD, but still worse.Nothing than usual ABX comparing is not optimal for making decision which encoder is better, none of methods PEAQ or Spectogram like MokrySedeS posted. Because encoder made decision to cutoff certain frequency range to keep overall quality. That means audio still can be transparent to source, but if you look at spectogram and see that something missing, because forcing to not cutoff (leave frequency range) is not good.

Many people here have a bad decision, rather than listen, they watch the some values and graphs, which is completly wrong.

[Terrachild]

Abradoks,

very interesting.
Could you tell me what commercial encoder you tested. (Surcode?)

How did you measure signal to noise ratio, is there a free program you can point me to that can do that? What about measuring PEAQ?

You said use W=8, what setting is that. Im using "WAV to AC3 Encoder 4.1" where is that setting?

Thanks

[raziel666]

I have a certain DTS-MA audio track from a bluray disc of mine and I want to compress it to FLAC.

I own Sonic Cinemaster Audio Decoder 4.3 and have installed the .dll's of ArcSoft using the trial version as described in other posts in doom9.

Now the problem is this: when trying to encode to FLAC, the ArcSoft decoder gives a file twice the size of that from the Sonic Audio Decoder.

After examining the files with MediaInfo, I found out that the bitrate varied between these two decoders. The ArcSoft one produced a file with 2833kbps and the Sonic one with 1631kbps. Moreover, the file produced with ArcSoft is supposedly 14 minutes longer than the original duration and when I tried to play it back, it sounded slowed down and like crap... :(

Am I doing something wrong? Is it some bug that one of the decoders has?

Thanks in advance for your answers.

[rapscallion]

It's a True-HD Blu-ray track and I'm trying to create lossless wavs. (using eac3to_more) and I'm running into a strange problem.

First, when I create wavs, from a regular AC-3 track, Nero decoder (direct show) does the processing and maintains 24 bit.

However, when I try to do this to the HD track, ffmpeg/Libav somehow does the decoding and after the first pass Libav reports superfluouus zero bytes, eliminates them and drops down to 16 bits.

Is there a way create lossless wavs, maintaining 24bit, from True-HD ?

[Inspector.Gadget]

If eac3to using LAVC says a lossless codec is actually 16-bit, it is. 24-bit decoding for a lossy format like AC3 makes sense because presumably you're going to transcode it later and the additional accuracy is desirable.

[rapscallion]

Yes, however, at the first pass it's reported as 24 bit. And, as I stated, libav is dropping it down from 24 to 16 bit, NOT reporting that it originally is.
What is lavc btw, or did you mean libav?

My goal is to re encode the wavs to DTS-MA.

[Inspector.Gadget]

LAVC = libavcodec. The first pass calls it 24 bit because eac3to has to make sure that the 8 bits of padding lasts the length of the file and there's no actual 24-bit content in there. Once eac3to has verified this, it can discard the padding during decoding.

[rapscallion]

Ok, thanks, I got it now....just two last questions.

Is Nero docoder (from ver 7.10.1) not able to decode TrueHD and that's why eac3to is choosing libav/ffmpeg ?

When I re encode to DTS-MA will the resulting track be back to 24 bit or does it even matter ?

[Abradoks]

I have posted (http://forum.doom9.org/showthread.php?p=1341798#post1341798) comparison in aften thread.

Nothing than usual ABX comparing is not optimal for making decision which encoder is better, none of methods PEAQ or Spectogram like MokrySedeS posted. Because encoder made decision to cutoff certain frequency range to keep overall quality. That means audio still can be transparent to source, but if you look at spectogram and see that something missing, because forcing to not cutoff (leave frequency range) is not good.

Many people here have a bad decision, rather than listen, they watch the some values and graphs, which is completly wrong.
I'm not going to argue about subjective vs. objective tests. If you have enough resources to make ABX comparison, it'll be very interesting to see how they correlate with SNR and PEAQ.
Besides, my tests were started because somebody reported bad listening quality of AC3 track encoded with aften compared to DD encoder. It was suggested that default frequency cutoff causes such effect.

You said use W=8, what setting is that. Im using "WAV to AC3 Encoder 4.1" where is that setting?
You can control encoded frequency range through commandline option "-w" when using aften. I don't know if any GUI can do this.
You can find information about used encoders and measurement tools in the post I linked.

[Inspector.Gadget]

Is Nero docoder (from ver 7.10.1) not able to decode TrueHD and that's why eac3to is choosing libav/ffmpeg ?


I'm not sure, but both Nero and LAVC can decode TrueHD (the MLP part, which is what you're after) bit-perfect.

When I re encode to DTS-MA will the resulting track be back to 24 bit or does it even matter ?

If you're using the DTS HD Encoder Suite, then it will remain 16 bit. I'm not sure what Surcode would do.

[rapscallion]

OK, thanks IG.....and yes DTS MA Suite to encode.

Edit: Both DTS-MA and Surcode encodes result in 16 bit depth.

[TinTime]

I'm not sure, but both Nero and LAVC can decode TrueHD (the MLP part, which is what you're after) bit-perfect.

Nero's limited to 5.1 output though which is why eac3to uses libavcodec by default.

[rapscallion]

If eac3to using LAVC says a lossless codec is actually 16-bit, it is. 24-bit decoding for a lossy format like AC3 makes sense because presumably you're going to transcode it later and the additional accuracy is desirable.

Well, not only are you exactly right, but I found this info, for the audio track I'm processing, on Blu-raystats.com :

Run -Time2h 32m 13s
Language -English
Disc Language -English
Video -1080p 2.35:1
Video Encoding -VC-1 24 Mbps
Audio- Dolby True HD 5.1 16 bits 48 kHz 1505 kbps
Secondary Audio -DD AC3 5.1 640Kbps
Disc- BD50 40.72 GB
Region- A, B, C

I always thought that both DTS-MA and DDTrueHD were 24 bit. Live and learn.

[rapscallion]

Ok guys, I've been playing around with this and now I'm a bit confused.

First time extracting TrueHD to 6 wav files via eac3to_more.

When I do this directly, 2 passes are made, resulting in wav files that are ~800,000KB ea/16bit.

However ,if I process the TrueHD to an AC-3 file and THEN process the AC-3 to wav files, the size of the wavs are ~1,285,000KB ea/24bit.

My reasoning is that the larger wavs are lossless and I can use them to create a DTS-MA audio track ? And I don't know what the smaller wav files are (the 2 pass process) ??

Any explanation would be greatly appreciated.

Another question is, when creating the DTS file should the rear channels (5.1) be attenuated to -3db or should that be left alone ? From what I've read movie theater tracks are boosted by 3db for the rears so theaters then attenuate them by the same. But BD tracks are not boosted, so should be left alone? Is that correct ?

My head hurts !

[nurbs]

TrueHD is a lossless codec that will give you the same output as the source. Lossy codecs like AC-3 and AAC don't store the information at a fixed bitdepth, so you can either decode them to 16 bit or 24 bit. Since the latter is a more accurate representation of the files content eac3to will do that by default.

[rapscallion]

Thanks, I understand that but it doesn't answer my questions re the wav files. Which processing gave me the lossless wavs?

[nurbs]

Both of them in a way. If you decode the TrueHD you get exactly what was in it. If you decode the AC-3 you might not get exactly what's in it (assuming they can be decoded to an arbitrary bitdepth), but 24 bit is close enough as in no audible difference from the source.
Of course if you encoded the AC-3 from the TrueHD there is a loss during the AC-3 encoding.

[Inspector.Gadget]

If you're making a DTS-HD MA track, the you should ONLY go TrueHD -> WAVs -> DTS Encoder. Going TrueHD -> AC3-> WAVs-> DTS encoder is converting lossy to lossless whether you re-encode or take the TrueHD core, because either way you're discarding the lossless MLP data.

[rapscallion]

......which is the logic I was using from the start. Thanks.

However, what really puzzles me here are the substantially smaller wav files (and DTS-MA files- 2.5gb vs 3.8gb) when going from TrueHD -> WAVs -> DTS Encoder, then when going TrueHD -> AC3-> WAVs-> DTS encoder.

Logically, I would think all the files would be larger in the first lossless option. Additionally, the first option results in 16 bit wavs, the second in 24 bit. I think you guys can appreciate my reasoning, even if it's way off base.

From what you're saying, I should also use the first option if I want to just create a straight, non HD, DTS track @1536 ?

[Inspector.Gadget]

24-bit PCM (16 bit + 8 bits padding) is generated by decoding AC3 to 24 bit, for no quality gain (and in this case, quality loss versus MLP to WAV). Decoding straight to WAVs nets you only the real 16-bit data.

From what you're saying, I should also use the first option if I want to just create a straight, non HD, DTS track @1536 ?

Yes. Original to lossless to lossy is preferable to original to lossy to lossy.

[nurbs]

From what you're saying, I should also use the first option if I want to just create a straight, non HD, DTS track @1536 ? You should always use the first option.

About the filesize. IIRC wav filesize is simply bitdepth * sampling rate * duration, so with the same duration and sampling rate a 24 bit file is 50% larger than a 16 bit file. 0.8 GB *1.5 = 1.2 GB


24-bit PCM (16 bit + 8 bits padding)
AFAIK it's not padding, it's actual information, insofar as the 24 bit file is a more accurate representation of the AC-3 file than a 16-bit wav would be even if the source of the AC-3 was only 16 bit. Of course it probably won't make much difference.

[Thunderbolt8]

as it seems eac3to cannot remove gaps from truehd tracks, but for example it can do so when I convert the original truehd track from a movie BD to flac. but when I demux the truehd track and then convert it to flac afterwards, then the gap message suddenly disappeares. does this mean that the gap remains undetected here and untreated?

[honai]

as it seems eac3to cannot remove gaps from truehd tracks, but for example it can do so when I convert the original truehd track from a movie BD to flac. but when I demux the truehd track and then convert it to flac afterwards, then the gap message suddenly disappeares. does this mean that the gap remains undetected here and untreated?

Yes, gap detection means that eac3to looks for audio gaps in relation to the video stream (i.e. container timecodes), so once you demux audio there is no reference video stream (i.e. container timecodes) anymore.

[rapscallion]

In case you all missed this last question in an earlier post today :

"Another question is, when encoding a DTS file should the rear channels (5.1) be attenuated to -3db or should that be left alone (unchecked) ? From what I've read movie theater tracks are boosted by 3db for the rears so theaters then attenuate them by the same. But BD tracks are not boosted, so should be left alone? Is that correct ?"

Edit : Found this in the Surcde (which I don't use) manual :
7.2
The Attenuate Rear Channels 3 dB Option

The original standard for Surround Sound in movie theaters was to attenuate
the rear channels by 3 dB.

So, in mixing for theaters, studios have their rear channel monitors attenuated 3 dB.

Home theaters, on the other hand, have the rear channels at unity gain.
So, a mix that was made with the rear monitors attenuated 3 dB will
have rear channel levels that are 3 dB too high for home theater.

This option takes care of that difference. So, in general, you use this option
if creating a DVD-Video using a Surround master that was originally mixed for movie theaters

So, for tracks from BD discs everyone leaves the -3db option unchecked, right ?

[Snowknight26]

eac3to incorrectly reports the resolution for one of the playlists on a Blu-ray of mine:

C:\unzipped\eac3to>eac3to.exe E: 1)
M2TS, 2 video tracks, 8 audio tracks, 4 subtitle tracks, 1:36:07, 112.806p
1: Chapters, 35 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
5: DTS Master Audio, English, 2.0 channels, 24 bits, 48khz
(core: DTS, 2.0 channels, 24 bits, 1509kbps, 48khz)
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
7: AC3 EX, French, 5.1 channels, 640kbps, 48khz
8: AC3 EX, Spanish, 5.1 channels, 640kbps, 48khz
9: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
10: AC3 Surround, French, 2.0 channels, 192kbps, 48khz
11: AC3 Surround, Spanish, 2.0 channels, 192kbps, 48khz
12: Subtitle (PGS), English
13: Subtitle (PGS), French
14: Subtitle (PGS), Spanish
15: Subtitle (PGS), English

[honai]

Is that some CGI movie from Pixar?

[Snowknight26]

It is. Seamlessly branched, too.

[Abradoks]

When decoding dts through libav I get pcm that should have gain factor of 1.0605 to match the source. While arcsoft decoder creates proper wavs. Is it known issue?

[honai]

It is. Seamlessly branched, too.

Then take a look at the last few lines after muxing, eac3to 3.17 should be reporting the correct fps there, nothing to worry about.

[honai]

When decoding dts through libav I get pcm that should have gain factor of 1.0605 to match the source. While arcsoft decoder creates proper wavs. Is it known issue?

I believe you are the first to mention this.

[Thunderbolt8]

Yes, gap detection means that eac3to looks for audio gaps in relation to the video stream (i.e. container timecodes), so once you demux audio there is no reference video stream (i.e. container timecodes) anymore.is there any chance to save a gap file which then can be used when I process only the audio track alone later on?

[Snowknight26]

Then take a look at the last few lines after muxing, eac3to 3.17 should be reporting the correct fps there, nothing to worry about.

A cosmetic issue is still an issue.

[tebasuna51]

When decoding dts through libav I get pcm that should have gain factor of 1.0605 to match the source. While arcsoft decoder creates proper wavs. Is it known issue?
Welcome to lossy encoders/decoders.
Yes it is know, to avoid overflows, when convert from frequency domain to time domain, many decoders attenuate a little bit the output.

You can use the -normalize parameter if you want.
And, of course, the use of ArcSoft decoder is always recommended.

Decoding ac3 sometimes eac3to detect overflows and make a second pass attenuating the signal.

[Thunderbolt8]

as it seems eac3to cannot remove gaps from truehd tracks, but for example it can do so when I convert the original truehd track from a movie BD to flac. but when I demux the truehd track and then convert it to flac afterwards, then the gap message suddenly disappeares. does this mean that the gap remains undetected here and untreated?
for some strange reason the subtitle file for this movie is about 4 seconds out of sync after remuxing. the movie consists of 2 m2ts files, the first one only being 7-8 seconds long. still, no apparent relation to those 4 seconds of subtitle delay. the fps rate of this movie is listed as 26.744fps (even though the value added to the header is 24/1.001), maybe this affects the subtitles here?

[crasus]

v02 The video framerate is correct, but rather unusual.
a03 Extracting audio track number 3...
a03 Removing AC3 dialog normalization...
a03 Applying (E-)AC3 delay...
a03 A remaining delay of -13ms could not be fixed.

The BluRay has two audio tracks. An ac3 640kbps one and a DTS-HD one with a core of 1536kbps. I get a delay could not be fixed message on both. Is there any way to correct it? ArcSoft DTS is installed and fully functional with eac3to (latest version).

Thank you!

[Snowknight26]

No, because of the size of AC3/DTS frames. I can't remember if AC3 is 32ms per frame or not, but if you have an AC3 track that has a -40ms delay, adding 32ms would yield an unfixable -8ms delay unless you reencode the audio. Same goes for DTS.

Not like you'd be able to hear the difference with an -8ms delay anyway.

madshi, would it be possible to parallelize audio conversion and gap fixing? I noticed that when converting a track to AAC and extracting a DTS track that had gaps, the 2nd pass for the DTS gap fixing only started after the 2nd pass of the AAC encoding/gap fixing finished.

[crasus]

eac3to reports that both tracks have a 83ms delay on the BluRay. Should I try to encode the DTS-HD to an ac3 5.1 448kbps track?

[Snowknight26]

Why are you trying to reencode? If it's to remove that miniscule delay, I suggest you rethink what you're doing.

[dcmo]

I'm pretty new to this, but I have a pretty big issue with the sound quality which I am getting. I'm using eac3to, pcm2tsmu, and ts muxer to get my backup. When running an
eac3to test it shows I have the Arcsoft Decoder, Haali Matroska Muxer, Nero AAC Endoder, and the MkvToolnix as up to date or working fine. The picture comes out great but the sound is definitely lacking. The channels being mapped wrong on some of the movies is merely an annoyance, I can just switch the cables; but the poor quality coming out of the center channel is almost a deal breaker. I'm re-encoding them all as lossless pcm to stream to a PS3, for that matter they may still be lossless. However the center sounds boxy, or boomy; after searching thru this thread it could be it has a kind of mono sound to it. I don't know what to call it, but I do know that it pales in comparison to the disc. I've got a pretty nice little set-up and I don't want to lose anything quality wise. Is there something I'm doing wrong while backing these movies up. Any help is much appreciated on this end.

[Inspector.Gadget]

Post the exact command line you're using, including the output of eac3to and what it reports about the source and destination files.

[tebasuna51]

I'm pretty new to this, but I have a pretty big issue with the sound quality which I am getting. I'm using eac3to, pcm2tsmu, and ts muxer to get my backup.
Pcm2Tsmu now is obsolete, the last TsMuxer accept .w64 files to avoid the problem with .wav files greater than 4GB.

Use eac3to to output .w64 files instead .pcm+pcm2tsmu

I'm re-encoding them all as lossless pcm to stream to a PS3, for that matter they may still be lossless. However the center sounds boxy, or boomy; ...

I think PS3 only support PCM 2.0, if you have an audio equipment with 5.1 speakers use ac3 640 Kb/s instead PCM.

If you want only stereo sound you need use :
eac3to input.file output.w64 -down2 -normalize
to obtain a PCM 2.0 with all the channels mixed.

[KidKappa]

i'm having troubles getting eac3to to recognize my arcsoft dts decoders/encoders, i have TotalMedia Theatre (2.1.6.131) and eac3to (v3.17) and running on Windows 7 64bit, when I run the -test i get that arcsoft is not installed and and error saying that magcore.dll cannot be found. i've looked through this thread but alot of it is over my head. i noticed that the arcsoft common files are in the "program files (x86)" folder and not in the "program files" folder. Also the missing magcore.dll is in there as well. could someone please give me a nice step by step on how to get this working. thank you.

[dcmo]

Post the exact command line you're using, including the output of eac3to and what it reports about the source and destination files.

eac3to v3.17
command line: eac3to.exe E: 1) 2: C:\movie\HIC.h264 3: C:\movie\HIC.pcm
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 3 subtitle tracks, 1:42:50, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3, English, 2.0 channels, 224kbps, 48khz
6: AC3, English, 2.0 channels, 224kbps, 48khz
7: Subtitle (PGS), English
8: Subtitle (PGS), English
9: Subtitle (PGS), Spanish
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[v02] Extracting video track number 2...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Creating file "C:\movie\HIC.pcm"...
[v02] Creating file "C:\movie\HIC.h264"...
[a03] The original audio track has a constant bit depth of 24 bits.
[a03] Audio overlaps for 9ms at playtime 0:21:31. <WARNING>
[a03] Audio overlaps for 10ms at playtime 0:23:05. <WARNING>
[a03] Audio overlaps for 10ms at playtime 0:39:41. <WARNING>
[a03] Audio overlaps for 9ms at playtime 0:40:32. <WARNING>
[a03] Audio overlaps for 10ms at playtime 1:03:57. <WARNING>
[a03] Audio overlaps for 8ms at playtime 1:16:14. <WARNING>
[a03] Audio overlaps for 9ms at playtime 1:20:34. <WARNING>
[a03] Audio overlaps for 8ms at playtime 1:25:34. <WARNING>
[a03] Starting 2nd pass...
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Realizing RAW/PCM gaps...
[a03] Creating file "C:\movie\HIC.pcm"...
[a03] The processed audio track has a constant bit depth of 24 bits.
Video track 2 contains 147951 frames.
eac3to processing took 51 minutes, 59 seconds.
Done.

MUXOPT --no-pcr-on-video-pid --new-audio-pes --vbr --vbv-len=500
V_MPEG4/ISO/AVC, "C:\Movie\HIC.h264", fps=23.976, insertSEI, contSPS
A_LPCM, "C:\Movie\HICUPDATE.pcm"


SmartLabs tsMuxeR. Version 1.10.6 http://www.smlabs.net
Decoding H264 stream (track 1): Profile: High@4.1 Resolution: 1920:1080p Frame rate: 23.976
H.264 stream does not contain fps field. Muxing fps=23.976
Decoding LPCM stream (track 2): Bitrate: 9216Kbps Sample Rate: 48KHz Channels: 7.1 Bits per sample: 24bit
Processed 147951 video frames
Mux successful complete.
Muxing time: 8 min 43 sec


I've only backed up a few movies so far, this is the only one where the audio was overlapping. I used Clown BD to do this one the first time, said the blu-ray structure wasn't recognized when I tried to use Eac3to with it the first time. The log files indicated wav files with Clown BD so I re-ran it.

I think PS3 only support PCM 2.0, if you have an audio equipment with 5.1 speakers use ac3 640 Kb/s instead PCM.

If you want only stereo sound you need use :
eac3to input.file output.w64 -down2 -normalize
to obtain a PCM 2.0 with all the channels mixed.

I've got multiple channels working now out of the PS3 with this, would the other channels be active if the PS3 only supported 2.0. Downgrading my sound quality is unacceptable, I enjoy the sound more than I do the picture. There are times when I will just shut my eyes and let the sound sink in. If this can't work I will just rip to a back up disc with BD Rebuilder.

[Inspector.Gadget]

What about taking the DTS core? You'll have known surround compatibility at the cost of the lossless data that you probably can't ABX against the core. That said, tebasuna 51 knows more than I do about the PS3, so I'll defer to him.

[dcmo]

A little more follow-up. I have a 7.2 system set-up and was testing (receiver set to MCH PCM) out the new results this morning. I had to switch the center and the RF channel to get the dialog coming out of the center channel. All speakers were active except for the LS (for further clarification not the LBS speaker) speaker, which was completly dead. As I had suggested earlier about the sound not sounding right when coming out of the center (possibly being mono) I wanted to hear what the mono would actually sound like so I changed my selection on my receiver to mono. While listening to it in mono I don't believe it is quite mono sounding when it is set to MCH PCM, and more than that the dialog changed from the center channel back to the RF channel with everything coming out of the RF speaker. So to sum it all up:

MCH PCM: have to switch C and RF speakers to get dialog out of center.

Mono: With the C and RF speakers switched (have to switch them to get MCH PCM), the sound now comes out the RF.

[tebasuna51]

command line: eac3to.exe E: 1) 2: C:\movie\HIC.h264 3: C:\movie\HIC.pcm
...
[a03] Creating file "C:\movie\HIC.pcm"...
...

MUXOPT --no-pcr-on-video-pid --new-audio-pes --vbr --vbv-len=500
V_MPEG4/ISO/AVC, "C:\Movie\HIC.h264", fps=23.976, insertSEI, contSPS
A_LPCM, "C:\Movie\HICUPDATE.pcm"

How do you convert HIC.pcm to HICUPDATE.pcm?

Try with:
eac3to.exe E: 1) 2: C:\movie\HIC.h264 3: C:\movie\HIC.w64

TsMuxer 1.10.6 accept w64 like PCM container.

[dcmo]

How do you convert HIC.pcm to HICUPDATE.pcm?

Try with:
eac3to.exe E: 1) 2: C:\movie\HIC.h264 3: C:\movie\HIC.w64

TsMuxer 1.10.6 accept w64 like PCM container.

I've been converting with pcm2tsmu. My question now is is
.w64 going to be equivalant to DTS HD MA/PCM or is it a lower grade sound quality. I'm really not interested in a 2.0 solution or any loss in sound quality.

[nurbs]

It's going to be exactly the same as the source.

[tebasuna51]

I've been converting with pcm2tsmu.
Please put command line
My question now is is .w64 going to be equivalant to DTS HD MA/PCM or is it a lower grade sound quality.
PCM in WAV or W64 container and raw LPCM have absolutely the same uncompressed audio data. You can verify the sizes, only differents by a few bytes for the header.

[dcmo]

Been pretty busy working today, just realized what you meant by command line. I just assumed it was pretty universal on pcm2tsmu input. Here is what was input:
pcm2tsmu.exe C:\movie\hic.pcm C:\movie\hicupdated.pcm -i 24 -c 8 -s 48000

Getting ready to give w64 a try now, hope it comes thru for me.

[arty]

D:\util\eac3to>eac3to i:\temp\movie.mkv 3:d:\trans.flac
MKV, 1 video track, 2 audio tracks, 2:30:25, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
"movie h264 stream"
2: DTS, English, 5.1 channels, 24 bits, 1509kbps, 48khz
"dts core track"
3: FLAC, English, 5.1 channels, 2:30:25, 24 bits, 48khz
"FLAC created by eac3to3.17 + Sonic 4.3.0 filters"
a03 Extracting audio track number 3...
a03 Decoding FLAC...
a03 Encoding FLAC with libFlac...
a03 Creating file "d:\trans.flac"...
----------------------------------

is this "demux" method lossless? couldn't find any other way to demux a flac from mkv - mktoolnix creates flac inside ogg :(

[Snowknight26]

is this "demux" method lossless?
Of course it is, you're going from lossless -> lossless.

couldn't find any other way to demux a flac from mkv - mktoolnix creates flac inside ogg :(
You didn't look hard enough then. There's a switch to have it just write a raw FLAC track without doing an OggFLAC.

[arty]

Of course it is, you're going from lossless -> lossless.


You didn't look hard enough then. There's a switch to have it just write a raw FLAC track without doing an OggFLAC.

thank you!

ahh, the --no-ogg param, so true :)

[dcmo]

.w64 was a complete bust for me, TS Muxer locked up on two different movies on me (one with TrueHD and the other with DTS HD MA). Both times with this:

LPCM bad frame detected. Resync stream.

Settings for TS Muxer:

MUXOPT --no-pcr-on-video-pid --new-audio-pes --vbr --vbv-len=500
V_MPEG4/ISO/AVC, "C:\Movie\ppl.h264", fps=23.976, insertSEI, contSPS
A_LPCM, "C:\Movie\ppl.w64"

SmartLabs tsMuxeR. Version 1.10.6 http://www.smlabs.net
Decoding H264 stream (track 1): Profile: High@4.1 Resolution: 1920:1080p Frame rate: 23.976
H.264 stream does not contain fps field. Muxing fps=23.976
Decoding LPCM stream (track 2): Bitrate: 6912Kbps Sample Rate: 48KHz Channels: 5.1 Bits per sample: 24bit

Actually though it wasn't a complete bust, it did about 35% of the THD version and about 25% of the DTS before it locked up on me. Both played fine on the computer, haven't checked it out on the PS3 as of yet.

Okay, played it on the PS3 and it came out great for what was there, does anyone know what caused TS Muxer to stop muxing. Or rather why a bad frame was detected. How do I solve this problem.

On what was there, there didn't appear to be any mapping problems (I really like that). The sound sounded pretty much identical, I think I would need two audio systems to be able to check it out.
However the tv upstairs still didn't play it properly, it was like the center channel (main dialog) was missing although on the system downstairs it came thru properly. It's an older receiver (5 years maybe, no hdmi, no advanced audio). Should I swap cables around or do I just need a new receiver.

Thanks to all for you help!

[tebasuna51]

...
LPCM bad frame detected. Resync stream.
...

Seems a TsMuxer problem. LPCM (or WAV or W64) don't have frames, only audio data. The audio data can be correct (music, voices, ...) or incorrect (noise, wrong channels, ...) but don't exist a method to detect a bad frame.

[mikelebron]

Guys.. Is it possible for me to remux my MKVs that have FLAC for audio and change the AUDIO to LPCM? Reason why I am asking is I want to stream my MKVs to the PS3... Am I going about it the wrong way?

[Inspector.Gadget]

Mikelebron: When you create the Matroska file from the original Blu-ray, just output PCM (WAV) audio instead of FLAC. Nothing requires you to use FLAC when demuxing a disc with eac3to...

[dcmo]

Seems a TsMuxer problem. LPCM (or WAV or W64) don't have frames, only audio data. The audio data can be correct (music, voices, ...) or incorrect (noise, wrong channels, ...) but don't exist a method to detect a bad frame.

So I should take this to the TS Muxer section? I saw this problem described over there (in April), but there never was really an answer for it.

I was reading in the thread about first converting it to WAV, then to W64; do you think this would help.

If so how does one go about converting it from wav to w64; I know that eac3to.exe E: 1) 2: C:\movie\movie.h264 3: C:\movie\movie.wav would get it to a wav file (although I've read that wav has problems with over 4G files), but how do you then convert that wav file to a w64 file. I assume it would be a command line, what would that line look like?

I'm going to try to convert it to pcm tonight and see if TS Muxer still has problems with muxing it.

[tebasuna51]

...
I'm going to try to convert it to pcm tonight and see if TS Muxer still has problems with muxing it.
TsMuxer don't accept standard LPCM (.pcm from eac3o), for that I wrote Pcm2tsmu.

TsMuxer always accept .wav files but only <4GB
Last versions of TsMuxer say accept .w64 files (same data than .wav but with a header to support >4GB).

I don't know how solve your problem.
My recommendation is use AC3 640 Kb/s, only with very good audio equipment in a conditioned room you can listen some difference.

[dcmo]

I was going to use pcm2tsmu just to see if it was w64 that was causing the problems and not the discs. I'm sure it's w64, I just wanted to be sure.

After reading some more, what about splitting the file and rejoining them later (that way I could do it in a wav file).

I just can't downgrade the sound; if the choice is between streaming with downgraded sound or not streaming at all then not streaming at all is hands down the winner (it's really not even an option).

I will just go the re-build route and burn it to a physical disc. Anyway can I split them and rejoin them, would I be able to tell where it was happening when I watched it.

Thanks again.

[utenteanonimo64]

I have just discovered eac3to and I am really impressed. In a matter of days it has become a central part of my audio processing workflow. There is only one thing I am missing (or I haven't been able to find out): I would like an option to have the audio output from a bluray stream split into several files one for each chapter.
Is Madshi still supporting this software and adding new features? Or is there an automated way to achieve this?
I have seen that someone has written a GUI called "HDConcertRipper" for this purpose but it gives me all sorts of errors on most of my discs...

[tebasuna51]

Try a search in this thread with your title "splitting audio by chapter"

For instance: http://forum.doom9.org/showpost.php?p=1209593&postcount=6845

[deebo]

some weird feature with powershell in windows7 (works fine without the -quality switch):

PS D:\> eac3to 00002.m2ts 8:commentary.aac -quality=0.35
...
Track 2 is used for destination file ".35".
This audio conversion is not supported.

works fine with cmd.exe:


D:\>eac3to 00002.m2ts 8:commentary.aac -quality=0.35
...
a08 Extracting audio track number 8...
a08 Patching bitdepth to 24 bits...
a08 Decoding with libav/ffmpeg...
a08 Reducing depth from 64 to 32 bits...
a08 Encoding AAC <0.35> with NeroAacEnc...
---

[utenteanonimo64]

Try a search in this thread with your title "splitting audio by chapter"

For instance: http://forum.doom9.org/showpost.php?p=1209593&postcount=6845

Well I wish Madshi changed his mind.... personally it's the only feature I miss...

[AnryV]

eac3to can't decode 1.0 DTS-HD MA
eac3to v3.17
command line: eac3to fre.dtsma fre.wavs
------------------------------------------------------------------------------
DTS Master Audio, 1.0 channels, 24 bits, 48khz
(core: DTS, 1.0 channels, 24 bits, 768kbps, 48khz)
Decoding with ArcSoft DTS Decoder...
The ArcSoft DTS Decoder reported an error while decoding. <ERROR>
Aborted at file position 262144. <ERROR>

eac3to v3.17
command line: eac3to fre.dtsma fre.wavs -sonic
------------------------------------------------------------------------------
DTS Master Audio, 1.0 channels, 24 bits, 48khz
(core: DTS, 1.0 channels, 24 bits, 768kbps, 48khz)
Decoding with DirectShow (Sonic Audio Decoder)...
The WAV writer didn't receive the format information. <ERROR>
Aborted at file position 896126544. <ERROR>

[dcmo]

Actually I got it figured out. It wouldn't do a wav or w64 file, but it would do a pcm (with pcm2tsmu) file. After having checked that out, I converted the original audio to a wav and then to w64. Turned out perfect, sounds great. As far as the issue upstairs, picked up a cheap Insignia receiver (HDMI input with HD audio codecs) on sale at Best Buy which solved the sound issue upstairs.

[tebasuna51]

eac3to can't decode 1.0 DTS-HD MA

Please, can you upload a sample?.
3 MB is enough:

eac3to fre.dtsma sample.dtshd -3mb

[Inspector.Gadget]

AnryV, maybe i'm missing something, but could the error be the result of the wavs switch for a mono source. What happens when you output to audio.flac instead?

[AnryV]

Please, can you upload a sample?.
3 MB is enough:

eac3to fre.dtsma sample.dtshd -3mb

http://rapidshare.de/files/48730597/sample_Tonton.dtshd.html

This occurs not only with this sound.

[tebasuna51]

This occurs not only with this sound.

Yes, seems a bug in ArcSoft decoder (same error with output to flac or wav).
I don't have Sonic to test.

BTW, the core bitrate, 768 Kb/s, is more than enough for a mono audio. Must be dificult diference from original.

[AnryV]

Yes, seems a bug in ArcSoft

And Sonic?
eac3to v3.17
command line: eac3to fre.dtsma fre.wavs -sonic
------------------------------------------------------------------------------
DTS Master Audio, 1.0 channels, 24 bits, 48khz
(core: DTS, 1.0 channels, 24 bits, 768kbps, 48khz)
Decoding with DirectShow (Sonic Audio Decoder)...
The WAV writer didn't receive the format information. <ERROR>
Aborted at file position 896126544. <ERROR>

TMT3 play it normally. I think that this is a bug of eac.

[xkodi]

TMT3 play it normally. I think that this is a bug of eac.

and why you are so sure that TMT3 plays the DTS-HD MA extension rather than DTS core, which i'm sure eac3to can also extract and decode correctly.

[AnryV]

and why you are so sure ...
Not sure... now. :(

[dcmo]

Been trying to get Ice Age 3 done, but Arc Soft is responding with strange set-up. The completed file outputs the center out the right channel, so a couple questions please. When channel remapping which channels do -0,1,2,3,4,5 correspond to? I take this is for 5.1, what happens when you have a 7.1 track? I assume you run the -log to find out what the channels are listed at, is this a correct assumption?

[Thunderbolt8]

And Sonic?


TMT3 play it normally. I think that this is a bug of eac.
afaik its a bug of the arcsoft decoder, we've already had this example that it cannot handle dts-hd ma 1.0 files

[AnryV]

afaik its a bug of the arcsoft decoder, we've already had this example that it cannot handle dts-hd ma 1.0 files
What about Sonic decoder?

[asc28]

Is there any way to recalculate and add back dialog normalization to AC3 tracks, after eac3to has stripped them?

[edit] realized that many studios use an arbitrary figure, so might not be able to recalculate. But how about using a ReplayGain like technique to find the optimal dB to be added and then adding it to an already encoded AC3 track?

Also, I know this has been asked a couple times before, but no one seems to know how exactly to go about reinserting dialnorm metadata without re-encoding.

[Esurnir]

I'm going to sound dumb, but which product from arcsoft still include the DTS-HD filter decoder? Does TotalMedia Theater 3 work ? Any special editions? (platinum?)

[Thunderbolt8]

What about Sonic decoder?
dont know to be frank.

[IanD]

I've been converting a DTS HD MA and DD2.0 soundtrack to wav with Eac3to, using -core and -down2 for the first, in order to align them.

However neither of the resulting wav files are directly readable by Goldwave, Cooledit or CDwave as it seems they are "raw". No matter what bit depth, signature or sample rate I choose, I only get noise. Strangely I am able to play the wav files with Directshow no problem.

How do I make the wav files readable in the sound editor applications?

It's frankly annoying that the output isn't wav standard.

[kypec]

Try to add switch -simple and see if it helps your applications to recognize such stereo WAV files. If yes then you should blame those apps that they don't support standard WAV headers.

[tebasuna51]

GoldWave (from v5.02, 2004? ) read WAVE_FORMAT_EXTENSIBLE headers without problems.

[73ChargerFan]

I've been converting a DTS HD MA and DD2.0 soundtrack to wav with Eac3to, using -core and -down2 for the first, in order to align them.

However neither of the resulting wav files are directly readable
Perhaps the DTS core is being converted to a dts-wav file, instead of being decoded?

eac3to (DTS or DD file) filename.wavs should decode, one channel per file.

[IanD]

Using -simple seems to have done the trick and both converted DTS HD MA and DD2.0 source files are now recognised as stereo wav.

However, I'm experiencing problems trying to synchronise the tracks: the DTS HD MA soundtrack is identified by eac3to as 24fps (not 23.976fps), whilst the commentary track I'm trying to synchronise to it is from a PAL source. Using the switches -25.000 and -ChangeTo24.000 results in the commentary being about 500ms longer than the soundtrack when the beginnings are synchronised.

I thought it might be possible to use the source switch -24.99x to fine tune the commentary conversion, in case the PAL source is not quite 25fps, but eac3to refuses to accept anything other than -25.000

Any thoughts?

Unfortunately I don't have access to an NTSC commentary, only a PAL one. It's quite close at the moment but not close enough because the commentary also has parts of the soundtrack that need to be in-sync with the on-screen action.

[Vick]

Many thanks to Christopher Key for providing the HDCD decoder!

Hi madshi

Whether Chris has given you the HDCD decoder source code?
He does not respond to email (cjk32 cam ac uk) :confused:

Thnx
Vick
p.s. sorry for my poor english

[pcordes]

I'm an old-school GNU/Linux command line junkie. Running eac3to inside
screen (http://en.wikipedia.org/wiki/GNU_Screen), the output comes out like this:


$ alias eac3to='wine /usr/local/.../eac3to.exe'
$ eac3to foo.eac3
fixme:mixer:ALSA_MixerInit No master control found on HDA ATI HDMI, disabling mixer
fixme:reg:GetNativeSystemInfo (0x32fc80) using GetSyE-AC3, 5.1 channels, 0:31:59, 1536kbps, 48khz
# or, discarding wine's messages: eac3to 2>/dev/null
$ eac3to 2(1) Haali Matroska Muxer


I spent a while looking for --help output, until I eventually realized that its output was designed for a Windows command window, not a VT100 terminal emulator. It happens to work outside of screen, in just gnome-terminal, and in native xterm, though.

eac3to outputs a bunch of ^h (ascii 0x08) characters at the start of every line, and maybe under screen, reverse-line-wrap happens, and the cursor backspaces up to the previous line, so every line of text overwrites the one before. If I redirect the output to a file,

$ eac3to > eac3to.txt 2>/dev/null
$ file eac3to.txt
eac3to.txt: ASCII English text, with CRLF line terminators, with overstriking
$ hexdump -C -v eac3to.txt
00000000 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000010 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000020 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000030 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000040 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000050 20 20 20 20 20 20 20 20 20 20 20 20 20 20 0d 0a | ..|
00000060 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000070 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000080 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000090 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000000a0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000000b0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
000000c0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
000000d0 20 20 20 20 20 20 20 20 0d 0a 08 08 08 08 08 08 | ........|
000000e0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000000f0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000100 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000110 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000120 08 08 08 08 08 08 08 08 08 08 65 61 63 33 74 6f |..........eac3to|
00000130 20 76 33 2e 31 37 2c 20 66 72 65 65 77 61 72 65 | v3.17, freeware|
00000140 20 62 79 20 6d 61 64 73 68 69 2e 6e 65 74 20 20 | by madshi.net |
00000150 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
00000160 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
00000170 20 20 20 20 20 20 20 20 20 0d 0a 08 08 08 08 08 | .......|
00000180 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
00000190 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000001a0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000001b0 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 08 |................|
000001c0 08 08 08 08 08 08 08 08 08 08 08 20 20 20 20 20 |........... |
000001d0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
000001e0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
000001f0 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
00000200 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 | |
00000210 20 20 20 20 20 20 20 20 20 20 0d 0a 08 08 08 08 | ......|
...


But the overstriking doesn't have any effect anywhere: xterm or gnome-terminal. (it visually looks the same as regular text). In VT100, ^h is cursor-movement, not highlighting. If eac3to is doing that itself on purpose, please don't, at least under wine. The CRLF line endings are not a problem.

I'd also love to have an option to not play sound. I haven't tried just renaming the .wav files. Maybe you could add a -unix option, or check an environment variable, to turn off sound and not print funky text. It's harder to deal with options at the end, instead of as the first arguments (e.g. an alias doesn't work), but a wrapper script like

#!/bin/sh
wine /.../eac3to.exe "$@" -unix 2>/dev/null
# or use a wine cmdline arg to filter out the messages it always prints, instead of redirecting to /dev/null

would do the trick.

apparently, wine devs
don't want apps to detect wine (http://forum.winehq.org/viewtopic.php?p=25906&sid=4a6025570ec9cb97aa5e4d952cf6ad88), so the -unix option would be a nice feature. Or have a -nosound option, or a -quiet option. (OTOH, -quiet usually means no messages, not literally quiet.) I'm worried that -nosound would cause confusion, though, since someone might think it means exclude audio tracks. But it's -nosound, not -noaudio...

Anyway, that's my feature request, please and thank you.

[samepaul]

I encounter problem. During THD -> AAC encoding the warning "Clipping detected, a 2nd pass will be necessary." is written and then eac3to indeed performs 2nd pass, making final audio very quiet and practically useless. Is there way to ignore clippings and avoid "2nd pass"?
Also eac3to writes "Reducing depth from 64 to 32 bits". What does it mean? Looks like volume related issue as well... if so, is it possible to prevent this downgrading too?

[tebasuna51]

I encounter problem. During THD -> AAC encoding the warning "Clipping detected, a 2nd pass will be necessary." is written and then eac3to indeed performs 2nd pass, making final audio very quiet and practically useless. Is there way to ignore clippings and avoid "2nd pass"?

You can add the parameter:
-no2ndpass
but is a very little diferent volume.

Also eac3to writes "Reducing depth from 64 to 32 bits". What does it mean? Looks like volume related issue as well... if so, is it possible to prevent this downgrading too?

Is the size of the audio sample, trust in eac3to, NeroAacEnc don't support 64 bit float audio samples.

[samepaul]

You can add the parameter:
-no2ndpass
but is a very little diferent volume.


Thank you! It did the magic - volume remained good. Seems anti-clipping algorithm requires some improvements :)

[tebasuna51]

... Seems anti-clipping algorithm requires some improvements :)

Then there are big volume difference ...
Please, can you upload a sample?

[samepaul]

Seems you're right. I tried to notch the place where clipping occurs, cut it out and encoded with and without 2nd pass.
There was difference in volume, but not so dramatic as it was when whole track was encoded. So I don't think that these samples are good for debugging.

I think it's not completely fault of encoder. I guess the track has places with really hard overload, but when eac3to performs 2nd pass it lowers volume of whole track evenly. Of course, after this previously clipped parts (blows, shootings etc) sound better, but "normal" parts where people talk become indiscernible. If I have tool to detect exact places where eac3to detects maximal clipping I would prepare sample for you. But without this I'd have to submit whole 700mb track , which I don't know how to do.
Btw, I've mistaken when I wrote "TrueHD". It was actually AC3.

But nevertheless, regarding to volume optimization, I think if eac3to performs 2 passes it can normalize levels non-linearly.

[honai]

I think if eac3to performs 2 passes it can normalize levels non-linearly.

... which is called Dynamic Range Compression (DRC) and is ignored by eac3to by default. DRC means you lose fidelity, and that is certainly not desirable if you want to preserve the quality of the original audio track.

[samepaul]

... which is called Dynamic Range Compression (DRC) and is ignored by eac3to by default. DRC means you lose fidelity, and that is certainly not desirable if you want to preserve the quality of the original audio track.

Uhum... and making audio quiet so much that you hadly can distinguish what people say - this is what you call "high fidelity"? If so, I'd stick with normal fidelity ;)

1st of all, DRC does not lower fidelity more than any volume adjustment, which is done by eac3to anyway. And really negligible comparing to quality loss due to audiocompression.
2nd, what I propose is not really DRC. Range compression only performed around overloaded zones, leaving ~90% (depending on content) of audiotrack intact.

[IanD]

The switches -25.000 (source) and -ChangeTo24.000 imply that they are floating point values that "may" accept any floating point value, but in fact seem to be hardcoded to -23.976, -24.000 and -25.000 and nothing else is accepted.

Is there any way to define variable floating point values to finetune these parameters?

Is there an alternate way to perform non-standard resampling and time expansion/compression?

[tebasuna51]

...
Is there any way to define variable floating point values to finetune these parameters?
To make the job eac3to use SSRC libraries to change the audio samplerate.
And not all conversion are possible. You can know available values here: http://avisynth.org/mediawiki/SSRC

Is there an alternate way to perform non-standard resampling and time expansion/compression?
There are commercial software like TimeFactory (http://www.prosoniq.com/editing-products/timefactory-2/)

And also AviSynth methods like TimeStretch (http://avisynth.org/mediawiki/TimeStretch)

[samepaul]

I think I found bug.
eac3to 00001.m2ts 1: chapters.txt 3: audio1.m4a -no2ndpass 6: audio2.m4a -quality=0.6 -no2ndpass 9: english.sup
It starts, spawns 2 nero encoders, processes whole input and then one Nero finishes, but 2nd does not and stays idle. I mean really idle - 0% CPU and no byte read or written.
As result 1st m4a file is ok and 2nd is unreadable, although is not empty.
No problem occurs if only one track is recoded at a time.

[Foofaraw]

So, any plans to support wmv/a? I need to get some wma10 out of a wmv and down to something sensible, like say ac3 :)

[tebasuna51]

To convert wma audio, in wmv container, to ac3 you can try several methods:
- Foobar and Aften
- dshow2wav (http://forum.doom9.org/showthread.php?t=141441) and pipe the output to Aften.
- AviSynth methods (Soundout, BeHappy, bepipe, wavi) with DirectshowSource decoder.

For more detailed explanation please open a new thread.

[b66pak]

off topic

@tebasuna51 it is not so simple...most of the time you will get an audio oos...
_

[Midzuki]

b66pak wrote:

off topic

it is not so simple...most of the time you will get an audio oos...

That's true. To whom this may interest, here goes my "method":

1) copy the Windows Media Audio stream to a standalone ASF file (*.wma) with AsfBin ;

2a) convert to a .WAV file via Avisynth + BassAudioSource() ;

2b) OR... playback the intermediate WMA file with Winamp's Disk Writer plugin.

So far, it's worked fine for stereo Windows Media Audio,
I still haven't found problematic multichannel WMAs on which I should test this workaround.

:helpful:

[hoju3508]

Is there a way to preserve the HD audio when slowing it from 25fps to 23.976fps? Once it's slowed down through eac3to, the HD streams are gone. It only uses the core of the track.

:thanks:

[tebasuna51]

Use flac output, is also lossless, or buy a TrueHD/DTS-HD encoder. But change the samplerate isn't a lossless operation.

[Mark_A_W]

I would use FLAC, but leave the file alone and change the speed on the during playback with Reclock.

You don't lose anything permanently that way.

[hoju3508]

Thanks for the feedback.

Use flac output, is also lossless, or buy a TrueHD/DTS-HD encoder. But change the samplerate isn't a lossless operation.


I would use FLAC, but leave the file alone and change the speed on the during playback with Reclock.

You don't lose anything permanently that way.

[sucker]

suppose i have a 1h30min track and i want to cut it, so that i only have a 1h25min track, what would be the commandline to achieve this?

tia

[jmone]

Given we are seeing advances in Filters supporting HD codec playback (eg FFDSHOW can now decode most and is working at bitstreaming)....is it worth transcoding the Audio or just Remux the Original Audio when using MKV containers?

[jmone]

wrong thread - sorry

[Blue_MiSfit]

IMO, most stuff can safely be transcoded to AAC with perfect transparency. I do this most of the time.

If AC3 is the best quality audio source on a given movie, then I just keep it. Why reinvent the wheel?

If a movie has an incredibly good audio track, or is a musical performance, then I either keep it as-is (if it's TrueHD), or just transcode to FLAC.

Of course, I don't have a fancy receiver, or even a 5.1ch system, so I don't mind terribly :) Headphones suit me just fine!

~MiSfit

[MrVideo]

There is a situation, which I've hit a few times, that causes eac3to to hang in a loop of some sort, forever.

It hits a bad section of the video, and if it is bad enough, it will not get through it, nor will it error out. This is what is on the screen when it happens:

a03 0:43:17 The source file seems to be damaged (transport error indicated).
v01 0:43:17 The source file seems to be damaged (transport error indicated).
a05 0:43:17 The source file seems to be damaged (transport error indicated).
a04 0:43:17 The source file seems to be damaged (transport error indicated).
a03 0:43:17 The source file seems to be damaged (discontinuity).
v01 0:43:17 The source file seems to be damaged (discontinuity).
a05 0:43:17 The source file seems to be damaged (discontinuity).
a04 0:43:17 The source file seems to be damaged (discontinuity).
a03 0:43:17 The source file seems to be damaged (discontinuity).
v01 0:43:17 The source file seems to be damaged (discontinuity).
a05 0:43:17 The source file seems to be damaged (discontinuity).
a04 0:43:17 The source file seems to be damaged (discontinuity).
v01 0:43:17 The source file seems to be damaged (discontinuity).
a05 This track is not clean.
a03 This track is not clean.
a04 This track is not clean.

I run via the command line (Zshell script within cygwin).

An update: I was able to take what was completed and finish the job. I then looked at the output and the real nasty damage is at 58:00 minutes. There was no error output placed onto the screen, eac3to just went into a loop and hung at that point in the file.

[Carpo]

I have a little issue whereby the encode i have done is just a little bit (i would say 100meg or so) big to fit on a dvd9 disc, i know i could use 2 dvd5s but was thinking it would be easier to just drop the rate of the ac3, i have the ac3 on the drive (used tsmuxer to remove true-hd part) so its sitting as a 640kbps ac3 file.

To use eac3to i would just do

eac3to gran.torino.640.ac3 gran.torino.448.ac3 -448

is that correct? or would people suggest i copy the true-hd file from the bluray disc and use ac3 to do the lot manually skipping tsmuxer down converting to ac3?

[Snowknight26]

To get the best quality, use the TrueHD track and go straight to the 448kbps AC3.

[Carpo]

Thought as much, so its just the line above i need? or do i need to add the frame rate as well?

Does this program still require external decoders to do its thing or is it all done by eac3to?

[Snowknight26]

The line above would work. For TrueHD, replace the source file name with the TrueHD track. You don't have to worry about any kind of 'rates' unless you're changing the video.
eac3to can decode TrueHD and it can encode AC3 without any additional extras.

[Carpo]

thanks :) - i have to change the rate on the ac3 otherwise i cant get the backup i made to fit on a dvd9 - its slightly oversized, if i drop the rate from 640 to 448 it should fit nicely :)

[Snowknight26]

Bitrate, not rate.

[Carpo]

same thing for me, im a lazy typist ;)

[Abradoks]

I have tested Nero for AC3 decoding and it seems that DRC isn't removed correctly. Here is source (http://rghost.ru/716070), AC3 (DRC: None) (http://rghost.ru/716075), AC3 (DRC: Speech) (http://rghost.ru/716076). When comparing SNR of decoded files (I use CompAudio) Nero gets same SNR as libav on file encoded without DRC, but on second file Nero gets much worse SNR, while libav has nearly same numbers.
I'm not the only one who gets this issue, it was previously reported on another forum.

[samepaul]

I'm sorry if I overlooked - I tried to find answer, but with no luck so far... I wonder where sources of eac3to available? As long as I understand this project uses some LGPL code/libraries and thus fall under LGPL and would have it's sources published. I wanted actually contribute in further eac3to development.

[kypec]

I wonder where sources of eac3to available? As long as I understand this project uses some LGPL code/libraries and thus fall under LGPL and would have it's sources published. I wanted actually contribute in further eac3to development.
You have it misunderstood I guess. Quoting the wikipedia info on LGPL (http://en.wikipedia.org/wiki/LGPL):The main difference between the GPL and the LGPL is that the latter can be linked to (in the case of a library, 'used by') a non-(L)GPLed program, and regardless of whether it is free software or proprietary software.
eac3to is free yet proprietary software (until now) and its sources are not publicly available. It uses a lot of LGPL libraries though which are bundled in the download package.

[samepaul]

hm.. i rather misunderstood the "linked library" concept. But yes, even so it's not "derivative work" as it's defined in the LGPL.
Anyway, I'm not pursuing any legal issues anyway. I just wanted to contribute and help with bugfixes, since I consider this project an excellent multimedia integration effort. Actually the best among all I ever seen on Doom9.

[madshi]

The whole point of LGPL is that closed source projects like eac3to can use LGPL libraries without having to go open source. That's the key difference between LGPL and GPL.

Thanks for the offer to help with development, and for the nice feedback, I appreciate that. I'd rather keep eac3to closed source, though, and do the bug fixing myself. I'm not aware of any major bugs (except maybe the Nero DRC bug, if that report is actually correct). So I'm not in a hurry to release a new version. Too busy with other stuff right now...

If you have some specific "big" new features you want to add to eac3to, you could either:
(1) Write some Delphi (!!) code and send it to me for integration into eac3to.
(2) Or you could create a little standard win32 dll, which eac3to would then link to.

[Carpo]

will you be adding support for nero 9 - or are you going to leave it with 7.x - a version you know works :p

[brasch]

On both Leaving Las Vegas and 3 days of the condor I can't choose any subtitles, even though they are there. Anyone know a fix for the problem?

[mikelebron]

Question... I noticed in some movies (so far only trueHD) once in a while at the same spot in a movie I get a little bit of static... What is that? How can i prevent it?

UPDATE: I tried redoing the file.. this time putting -keepDialNorm bu this did not help.... in the EXACT spot where the crackling was had the exact same issue... any other ideas? BTW: Its star trek first contact.. the scene is at the end where the vulcans are landing on earth for the first time... The crackle happens when the vulcan ship touches down..

[Carpo]

could this be an issue with the actual disc? or does this happen only after you have copied it to the drive?

[mikelebron]

Played right from the disc using all the same player (MPC-HC), codecs (FFDSHOW video and audio) and renderer (reclock) NO ISSUES....

what I did notice was the volume (level) was MUCH MUCH higher from the eac3to ripped version... Is this normal? I suspect in the audio chain (renderer, driver, windows mixer, etc.) is being over modulated... is there a way to have the volume level unmolested? (I know I can reduce it by -3db but I prefer it to just be like whats on the disc..)

could this be an issue with the actual disc? or does this happen only after you have copied it to the drive?

[Blue_MiSfit]

It's obviously not an issue with the disc.

Do you hear the crackling from the TrueHD file, simply extracted with eac3to, or is this in a transcoded file from a TrueHD source?

I'd try demuxing the TrueHD with TSMuxeR, and comparing the results (though I can't imagine there will be any).

Curious problem..

~MiSfit

[tebasuna51]

Played right from the disc using all the same player (MPC-HC),...

what I did notice was the volume (level) was MUCH MUCH higher from the eac3to ripped version... Is this normal? I suspect in the audio chain (renderer, driver, windows mixer, etc.) is being over modulated...
Maybe is a problem with TrueHD libav decoder, please put the log file at conversion.

If you can extract (in the source) the problematic fragment and upload to analyze ...

[umaximus]

Right, I have bunch of trueHD soundtracks that were demuxed in past with eac3to. Since I got a capable HD receiver I want to mux HD audio back. Doing this with TsMuxeR I have no problems with DTS-HD MA, but TsMuxer wont accept trueHD streams and is giving an error of 'Unsupported format' (cant detect stream type) on every single trueHD stream that I have (+100). Im sure there is difference in header info of how eac3to is demuxing trueHD but I have no clue how to 'repair' it.

[mikelebron]

Whats the best way for me to cut that one scene out to upload it?

Maybe is a problem with TrueHD libav decoder, please put the log file at conversion.

If you can extract (in the source) the problematic fragment and upload to analyze ...

[mikelebron]

Ok.. I used Tsmuxer to see if I had the same issue and it was there... same static while I hear the audio(not crackle)... that lead me to try again in MPC-HC and play the movie right from the disc. Low and behold... it was there from the disc. Sorry about that guys.. I dont know how I missed it. It must be something then with ffdshow decoder

[stax76]

I have a problem capturing console output, maybe somebody can explain me what's going on and how to fix it, the screen shot shows the problem:

http://i.imagehost.org/t/0466/aaa.jpg (http://i.imagehost.org/view/0466/aaa)

[honai]

Your console runs in ANSI mode, but eac3to probably assumes a Unicode codetable (which is the default for most Windows installations).

[Yraen]

I have a problem capturing console output, maybe somebody can explain me what's going on and how to fix it, the screen shot shows the problem:

Just a thought, are you trying to read the STDERR? Try just STDOUT if you are.

[stax76]

I'm using simple .NET code which works for all other applications, .NET and WinAPI are both pretty much unicode if you don't count the legacy stuff, the .NET lib is pretty powerful so maybe there is a simple solution, if not interfacing with native code would be no problem for me. I assume it has something to do with the colors eac3to outputs which is pretty cool if you just use the command shell. Both console and text encoding are not areas I have much experience. I've tried stderr first since it uses the same code running BeSweet and that use stderr, eac3to did however not write anything to stderr if I didn't miss something, I'm reading eac3to output from stdout.

[stax76]

It's some control chars appearing right in front of colored output, happens also when you make a bat file and redirect the output to a text file using: eac3to.exe test.wav test.ac3 > test.txt

Is there a switch to disable the colors? Otherwise I need to try trimming these control chars.

[Yraen]

I need to get around to trying to learn .Net. I know from autoit, when trying to read the stderr along with stdout I got the exact same result as you. When I went to stdout only I got the results I was looking for.

Edit: I'm not aware of any switches to turn off the colors.

[stax76]

I tried both and it only wrote something to stdout. I've checked the WinAPI console documentation and the eac3to documentation, seems I don't have luck here so I have to try to trim the 'garbage'. Thanks for the help.

[stax76]

If you look at the screenshot I've posted you can see BS BS BS BS etc., it means Backspace! I've checked it with a hex editor, where it was 08 08 08 so I went to ASCII at wikipedia and it said 08 is Backspace. I'm such a noob when it comes to binary data and console. :)

[Abradoks]

I have a problem capturing console output, maybe somebody can explain me what's going on and how to fix it, the screen shot shows the problem:

http://i.imagehost.org/t/0466/aaa.jpg (http://i.imagehost.org/view/0466/aaa)
It was already discussed a few times (1 (http://forum.doom9.org/showthread.php?p=1333829#post1333829), 2 (http://forum.doom9.org/showthread.php?p=1348250#post1348250)).
Adding -simpleoutput switch into eac3to looks like the best solution.

[mrr19121970]

It's really OT, but quite straightforward. See here (http://msdn.microsoft.com/en-us/library/ms682073%28VS.85%29.aspx). AttachConsole, CreateFile, GetConsoleScreenBufferInfo, ReadConsoleOutputCharacter & FreeConsole when you're done.

[stax76]

You receive lines from a callback using .NET, this lines can be cleaned with simple string processing.

[honai]

Your screenshot shows that your console assumes ANSI (in the lower right corner), not Unicode. ;)

[stax76]

From a text file you cannot make a relation to a console, this text file was a .NET string object which is UTF16 converted to ANSI when saving to file. I should save it better as UTF8. For the backspace (0x08) control char issue text encoding isn't relevant. eac3to is the only console application I know writing backspace to the console, other applications probably mostly write/read lines but since I never did a console application I cannot tell sure. I can handle it, it took me a while but I've learned something.

[Roscoe62]

I just bought the Clone Wars Season one box set.

After ripping with AnyDVD HD, I use eac3to to select one of the episodes and rip out the video, audio & chapter data and process further from there.

However, eac3to is only showing 1 "episode" (for lack of a better term) which I don't believe is actually an episode, and there's only video - duration 15:02. No audio, no chapters.

I know there are 7 separate episodes on this disc but, for some reason, eac3to is not finding them.

Anyone else struck this yet and, if so, found a way around it?

[73ChargerFan]

Try BDInfo, it will decode all MPLS files and give you the track lengths. I'm guessing you'll find seven 20 minute playlists. BDInfo does make mistakes sometimes, though.

[Killroy™]

I just bought the Clone Wars Season one box set.

After ripping with AnyDVD HD, I use eac3to to select one of the episodes and rip out the video, audio & chapter data and process further from there.

However, eac3to is only showing 1 "episode" (for lack of a better term) which I don't believe is actually an episode, and there's only video - duration 15:02. No audio, no chapters.

I know there are 7 separate episodes on this disc but, for some reason, eac3to is not finding them.

Anyone else struck this yet and, if so, found a way around it?

Good luck with this. I found the same thing until I found sort of a solution:

I had to hunt down the .mpls that showed the actual playlist and found that the "Play All" is listed on 00020.mpls but I found another surprise:

c:\eac3to317>eac3to f:\swcwdisc1\bdmv\playlist\00200.mpls
1) 00200.mpls, 2:41:03
[95+96+97+96+98+96+99+96+101+96+102+96+103+96].m2ts
- Chapters, 35 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48khz
- AC3, French, multi-channel, 48khz
- AC3, German, multi-channel, 48khz
- AC3, Spanish, multi-channel, 48khz
- AC3, Spanish, multi-channel, 48khz
- AC3, Japanese, multi-channel, 48khzDo you see that repeating 00096.m2ts file? Well, that little sucker is a 5 minute copyright placecard...yes, 5 minutes repeated after each episode.

Even if you decide to leave it in place and just skip it with the "Next Chapter" button (if you include chapters) you will find that those 5 minute files really screw up the synch for each episode after it. The first episode is fine but later on it gets worse and worse.

The only way to do it is to remux each individual episode as its own file. At least you know the correct order by looking at the mpls file.

[73ChargerFan]

The following should work.

eac3to f:\swcwdisc1\bdmv\stream\00095.m2ts+f:\swcwdisc1\bdmv\stream\00097.m2ts+f:\swcwdisc1\bdmv\stream\00098.m2ts+f:\swcwdisc1\bdmv\stream\00099.m2ts+f:\swcwdisc1\bdmv\stream\00101.m2ts+f:\swcwdisc1\bdmv\stream\00102.m2ts+f:\swcwdisc1\bdmv\stream\00103.m2ts -demux

To get a chapter list, create one manually by adding the length of each successive episode to the prior chapter time.

[Roscoe62]

The following should work.

eac3to f:\swcwdisc1\bdmv\stream\00095.m2ts+f:\swcwdisc1\bdmv\stream\00097.m2ts+f:\swcwdisc1\bdmv\stream\00098.m2ts+f:\swcwdisc1\bdmv\stream\00099.m2ts+f:\swcwdisc1\bdmv\stream\00101.m2ts+f:\swcwdisc1\bdmv\stream\00102.m2ts+f:\swcwdisc1\bdmv\stream\00103.m2ts -demux

To get a chapter list, create one manually by adding the length of each successive episode to the prior chapter time.

Thanks for this. I'm just going to do this individually for each episode, but so far this procedure is working fine. Thanks again! :)

[umaximus]

Is it possible that TrueHD track doesn't have audio gaps/overlaps on seamless branching disc while AC3 has?

Pineapple Express

eac3to v3.17
command line: eac3to 1) 2: video.h264 3: audio.wav 5: audio.ac3
------------------------------------------------------------------------------
M2TS, 1 video track, 7 audio tracks, 15 subtitle tracks, 1:57:26, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB)
4: TrueHD/AC3, French, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB)
5: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Extracting TrueHD stream...
[a03] Removing TrueHD dialog normalization...
[a05] Extracting audio track number 5...
[a03] Decoding with libav/ffmpeg...
[a05] Removing AC3 dialog normalization...
[a03] Writing WAV...
[a03] Creating file "audio.wav"...
[v02] Creating file "video.h264"...
[a05] Creating file "audio.ac3"...
[a03] [libav] End of stream indicated <WARNING>
[a03] The original audio track has a constant bit depth of 16 bits.
[a03] Caution: The WAV file is bigger than 4GB. <WARNING>
[a03] Some WAV readers might not be able to handle this file correctly. <WARNING>
[a05] Audio overlaps for 7ms at playtime 0:10:27. <WARNING>
[a05] Audio overlaps for 8ms at playtime 0:27:14. <WARNING>
[a05] Audio overlaps for 10ms at playtime 0:32:23. <WARNING>
[a05] Audio overlaps for 18ms at playtime 0:44:12. <WARNING>
[a05] Audio overlaps for 31ms at playtime 0:44:54. <WARNING>
[a05] Audio overlaps for 28ms at playtime 0:56:01. <WARNING>
[a05] Audio overlaps for 9ms at playtime 1:02:00. <WARNING>
[a05] Audio overlaps for 15ms at playtime 1:10:16. <WARNING>
[a05] Audio overlaps for 14ms at playtime 1:12:12. <WARNING>
[a05] Audio overlaps for 32ms at playtime 1:43:47. <WARNING>
[a05] Audio overlaps for 25ms at playtime 1:52:10. <WARNING>
[a03] Superfluous zero bytes detected, will be stripped in 2nd pass.
[a03] Starting 2nd pass...
[a03] Reading WAV...
[a03] Stripping zero bytes...
[a03] Writing WAV...
[a03] Creating file "audio.wav"...
[a03] Caution: The WAV file is bigger than 2GB. <WARNING>
[a03] Some WAV readers might not be able to handle this file correctly. <WARNING>
[a05] Starting 2nd pass...
[a05] Realizing (E-)AC3 gaps...
[a05] Creating file "audio.ac3"...
Video track 2 contains 168949 frames.
eac3to processing took 47 minutes, 28 seconds.
Done.

[nautilus7]

Yes, that's very usual. TrueHD frames are ~3ms, while AC3 ones are 32ms long. IIRC, eac3to won't report any overlaps if they are too short.

[wolfbane5]

I was checking out We Were Soldiers and noticed that it fronts a DTS-ES audio track. When I used the -down6 option, it created the .wavs but said that Surcode wasn't installed. It is installed, however, I've encountered this problem before and have simply ignored it since I've been unable to find a solution to it. My question is: does the -down6 option mix/rewrite the channels so that the BC .wav track is somehow included in the other 6 channels or does it simply ignore the BC .wav track created when you convert the .dts into .wavs? If it's the latter, then I can simply take the long route and create the .wavs from the .dts, ignoring the BC .wav track and then use Surcode to create a 5.1 dts audio track.

Thanks for the help in advance.

[nautilus7]

From 1st post:

-down6 --> downmix 7 or 8 channels to 6 channels

[wolfbane5]

Alright, so if that's the case, then I need to use -down6. But this leads to my problem of eac3to thinking Surcode isn't installed. Is there a way to use the Arcsoft DTS Decoder manually?

[TinTime]

If you already have a DTS track why decode it and encode to DTS again? Why not just keep it?

[SomeJoe]

1. Why the need to convert DTS-ES into something else? The DTS-ES stream is fully compatible with any DTS receiver, even if that receiver is 5.1 or doesn't recognize the ES extensions.

2. DTS-ES can come in two forms:

A) DTS-ES 5.1 Matrix, which derives the back center channel from matrix encoding on the Left Surround/Right Surround channels. This is the same way that Dolby Digital EX works. There is no difference between a DTS-ES 5.1 Matrix stream and a standard DTS 5.1 stream -- both contain 5.1 discrete channels of audio.

B) DTS-ES 6.1 Discrete, which actually has a discrete back center channel in the DTS stream. This is not nearly as common as the DTS-ES 5.1 Matrix described above. But this stream will still play back properly on 5.1 equipment, just without the back center channel.

If your current stream is DTS-ES 5.1 Matrix (likely), then there is no need for -down6, since the stream is already 5.1 channels.

[raquete]

The DTS-ES stream is fully compatible with any DTS receiver, even if that receiver is 5.1 or doesn't recognize the ES extensions. seems that have space for my fast question.
i have DTS receiver 6.1 and DTS-ES encoder but how to author & burn in medias(what 'format' to chose) ?
(with or without video, doesn't matter, the target is audio)

Happy New Year ! :)

[wolfbane5]

SomeJoe,

I was converting the DTS-ES into DTS since I don't have a receiver and was planning on watching the movie on my computer, which has 5.1 audio.

As for Matrix or Discrete, MediaInfo is telling me this:

General
Format : DTS
Format/Info : Digital Theater Systems
File size : 1.46 GiB
Duration : 2h 18mn
Overall bit rate : 1 510 Kbps

Audio
Format : DTS
Format/Info : Digital Theater Systems
Format profile : ES
Duration : 2h 18mn
Bit rate mode : Constant
Bit rate : 1 510 Kbps
Channel(s) : 7 channels
Channel positions : Front: L C R, Rear: L C R, LFE
Sampling rate : 48.0 KHz
Resolution : 24 bits
Stream size : 1.46 GiB (100%)

Am I correct in assuming that's DTS 6.1 Discrete?

[tebasuna51]

You only need eac3to to know this. Check the DTS with eac3to:

DTS-ES, 5.1 channels, 0:00:22, 16 bits, 1510kbps, 48khz (Matrix)
DTS-ES, 6.1 channels, 0:00:22, 16 bits, 1510kbps, 48khz (Discrete)

[tebasuna51]

i have DTS receiver 6.1 and DTS-ES encoder but how to author & burn in medias(what 'format' to chose) ?
The same than standard DTS.

To burn CD or play with some players (WD TV Live for instance) you need dtswav.

To mix with video or play on PC or some players (Xtreamer) you can use the .cpt (compact format) with the extension renamed to .dts.

Maybe some authoring program need the .dts (padded)

[ACrowley]

one question :

I never thought about but can ec3to outout 32bit float .wav and .wavs from any Source ?
Maybe -down32 works as a override ?

EDIT: no...doesnt work.

[TinTime]

You can try using -full.

However I don't think eac3to will ever output at a higher bit depth (i.e. pad with zeroes) than is necessary.

So with the Nero decoder...
eac3to input.ac3 output.wav -full
...will result in 24bit output but...
eac3to input.ac3 output.wav -25.000 -changeTo24.000 -full
...will result in 64bit float output.

[raquete]

The same than standard DTS.

To burn CD or play with some players (WD TV Live for instance) you need dtswav.

To mix with video or play on PC or some players (Xtreamer) you can use the .cpt (compact format) with the extension renamed to .dts.

Maybe some authoring program need the .dts (padded) very clever with full details.

thank you so much! :)

[Thunderbolt8]

what does it mean again when I have a 7.1 track with a "strange setup" note? anything I have to look out for? and what was it again with a DTS-HD MA track with -2db dialnorm & eac3to saying "decoding this track with arcsoft resulsts in low volume" (its the same track in both cases)

guess I have a problem now, because arcsoft results in low volume and sonic can only do 5.1?

[xkodi]

Arcsoft can't do bit-perfect decode of DTS-HD MA 7.1 "strange setup" files, while Sonic can but only 5.1 channels, look at this posts and all links in it to understand more:

http://forum.doom9.org/showthread.php?p=1179218#post1179218

[Thunderbolt8]

hm so whats the standing regarding this problem, is madshi willing to do this patch so that arcsoft can properly recognize the channel order?

and does the low volume decoding problem result from this strange channel order or from those -2db dialnorm value?

additionally, the acrsoft decoder says that track has constant 24-bit, while sonic says the 5.1 track is only 16-bit?

[SpaceAgeHero]

Hey folks,

I am having some troubles getting the Arcsoft DTS Decoder to work with eac3to.
Recently I installed Arcsoft Total Media Theatre and updated to 3.0.1.160.
I don't know what version of the dts decoder is included in this package.
Eac3to simply won't recognize the decoder. I'm currently running Windows 7 x64 Ultimate.
I also tried adding "c:\Program Files (x86)\ArcSoft\TotalMedia Theatre 3\" and/or "c:\Program Files (x86)\ArcSoft\TotalMedia Theatre 3\Codec\" to my environment path. c:\Program Files\Common Files\ArcSoft is not there!

Perhaps someone can help me out via E-Mail or private messages.
I do not want to bloat this thread with this issue.
Thank you!

[Leiw]

great program :) works fine for all my conversion needs

[Capsbackup]

@SpaceAgeHero;
Try this thread out, should help you.
http://forum.doom9.org/showthread.php?t=148324&page=2

[PowerGamer]

I have "Pixar Short Films Collection" Bluray. "Mike's New Car" movie on this bluray has the following tracks:

eac3to.exe "PIX1EGD1\BDMV\PLAYLIST\00008.mpls"
1) 00008.mpls, 00006.m2ts, 0:03:49
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, multi-channel, 48khz
...

I demuxed video and audio tracks with eac3to, tsmuxer and xport. Audio track in .pcm format created by eac3to and xport are bitwise identical. But the same audio track in .wav format produced by eac3to and tsmuxer are different.

Command lines used:

eac3to vs. xport:
eac3to.exe "PIX1EGD1\BDMV\PLAYLIST\00008.mpls" -demux
xport.exe -h "PIX1EGD1\BDMV\STREAM\00006.m2ts" 1 1 1

eac3to vs. tsmuxer:
eac3to.exe "PIX1EGD1\BDMV\PLAYLIST\00008.mpls" 2: audio_eac3to.wav
tsMuxeR.exe 1.meta .\

1.meta file content:
MUXOPT --demux
A_LPCM, "PIX1EGD1\BDMV\STREAM\00006.m2ts", track=4352, lang=eng, mplsFile=00008

Upon further analysis of .wav files created by eac3to and tsmuxer I discovered that these two .wav files contain different value in the dwChannelMask field in the .wav file header: tsmuxer .wav file has 0x0000003F and eac3to .wav file has 0x0000060F. Otherwise headers of both .wav files contain bitwise identical information. In the audio data contained within these two .wav files there were about 880 miscompared bytes (see wav_diff.txt attached to this post).

According to http://www.microsoft.com/whdc/device/audio/multichaud.mspx:

dwChannelMask=0x0000003F (tsmuxer):
FRONT_LEFT, FRONT_RIGHT, FRONT_CENTER, LOW_FREQUENCY, BACK_LEFT, BACK_RIGHT

dwChannelMask=0x0000060F (eac3to):
FRONT_LEFT, FRONT_RIGHT, FRONT_CENTER, LOW_FREQUENCY, SIDE_LEFT, SIDE_RIGHT.

My questions are:
1. Is it correct to say that eac3to or tsmuxer performed some kind of transformation of audio signal according to different dwChannelMask value which caused miscompared bytes in .wav files?
2. Which .wav file (produced by eac3to or tsmuxer) is "better" or "more identical to the source" and why?

[Abradoks]

neroAacEnc expects floating-point number as "-q" parameter while eac3to accepts only two decimal places. It is important because bitrate may vary significantly over 0.01, e.g. q=0.37 -> 240kbps, q=0.38 -> 370kbps.

[tebasuna51]

...
My questions are:
1. Is it correct to say that eac3to or tsmuxer performed some kind of transformation of audio signal according to different dwChannelMask value which caused miscompared bytes in .wav files?
2. Which .wav file (produced by eac3to or tsmuxer) is "better" or "more identical to the source" and why?
I can't see your wav_diff.txt (Attachements Pending Approval), use http://pastebin.com/ to upload txt files, but:

a) The Channel Mask 0x0000060F is the recommended by M$ for wav 5.1, after XP SP2, but the old 0x0000003F must be also compatible for 5.1. Then:
0x0000003F old, but compatible with old and new soft.
0x0000060F new, but can be a problem with old soft.

b) The audio data bit difference between eac3to and TsMuxer is a know, and reported, bug in TsMuxer extraction. Don't use TsMuxer to extract only to mux.

To answer your questions:
1) The channelmask are both valids and don't modify the audio data, the problem is the TsMuxer's bug.
2) Of course the eac3to wav is the "exact" audio source.

[AnryV]

eac3to v3.17
command line: 00000_1_02.dtshd dtshd.wavs
------------------------------------------------------------------------------
DTS Master Audio, 7.1 channels, 16 bits, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
Decoding with ArcSoft DTS Decoder...
Writing WAVs...
Creating file "dtshd.C.wav"...
Creating file "dtshd.R.wav"...
Creating file "dtshd.SR.wav"...
Creating file "dtshd.LFE.wav"...
Creating file "dtshd.L.wav"...
Creating file "dtshd.SL.wav"...
Creating file "dtshd.BR.wav"...
Creating file "dtshd.BL.wav"...


00000_1_02.dtshd has the following scheme of channells (according to Scenaris and BDReauthor):
C,L,R,LFE,Lsr,Rsr,Lss,Rss

To what channels in this scheme do correspond eac3to's channels SL,SR,BL, BR?

[tebasuna51]

00000_1_02.dtshd has the following scheme of channells (according to Scenaris and BDReauthor):
C,L,R,LFE,Lsr,Rsr,Lss,Rss

To what channels in this scheme do correspond eac3to's channels SL,SR,BL, BR?
SL (Side Left) is Lss (Left Surround Side?), SR is Rss.
BL (Back Left) is Lsr (Left Surround Rear?), BR is Rsr.

[PowerGamer]

I can't see your wav_diff.txt (Attachements Pending Approval), use http://pastebin.com/ to upload txt files, but:

a) The Channel Mask 0x0000060F is the recommended by M$ for wav 5.1, after XP SP2, but the old 0x0000003F must be also compatible for 5.1. Then:
0x0000003F old, but compatible with old and new soft.
0x0000060F new, but can be a problem with old soft.

b) The audio data bit difference between eac3to and TsMuxer is a know, and reported, bug in TsMuxer extraction. Don't use TsMuxer to extract only to mux.

To answer your questions:
1) The channelmask are both valids and don't modify the audio data, the problem is the TsMuxer's bug.
2) Of course the eac3to wav is the "exact" audio source.

Tebasuna51, thanks for your explanations. Meanwhile I tryed to demux that audio track using Blu-Ray Demuxer Pro (http://dvd-logic.com/bddemuxerpro.php) and much to my dismay found out that it produces exactly the same .pcm file as eac3to/xport but completely different .wav file that practically in every byte (apart from header which is the same as tsmuxer's .wav file) differs from both eac3to and tsmuxer versions. Still all .wav files have exactly the same size and play fine in media player.

Once again it is unclear which .wav output is better: eac3to or Blu-Ray Demuxer Pro? And what is the difference between .pcm and .wav files and what kind of transformation is involved when converting from .pcm to .wav?

[tebasuna51]

Once again it is unclear which .wav output is better: eac3to or Blu-Ray Demuxer Pro?
I never support commercial soft.
If the wav audio data output is different (the header may be different), then the Blu-Ray Demuxer Pro is wrong because the eac3to output is exact (not better, is exact).

And what is the difference between .pcm and .wav files and what kind of transformation is involved when converting from .pcm to .wav?
The audio data is stored with different byte order (Big endian, Litle Endian) and the audio channels are stored also with different order.
The wav file have a header, the pcm is only audio data without header.

[PowerGamer]

The audio data is stored with different byte order (Big endian, Litle Endian) and the audio channels are stored also with different order.
The wav file have a header, the pcm is only audio data without header.

Taking that into account (thanks again for explaining) I found out the following (EAC - channel data in .wav file produced by eac3to, BDR - same in Blu-ray Demuxer .wav, PCM - corresponding data in .pcm file):

PCM EAC BDR
0100 0001 0001
0200 0002 0002
0000 0000 0000
ffff 0000 ffff
0000 ffff 0000
0000 0000 0000

ffff ffff ffff
ffff ffff ffff
0000 0000 0000
0100 0001 0001
0000 0001 0000
0100 0000 0001

0000 0000 0000
0000 0000 0000
0000 0000 0000
0000 fffe 0000
0000 0000 0000
feff 0000 fffe

0100 0001 0001
0000 0000 0000
0100 0001 0001
ffff 0003 ffff
0000 ffff 0000
0300 0000 0003

ffff ffff ffff
0100 0001 0001
feff fffe fffe
0000 fffc 0000
0000 0000 0000
fcff 0000 fffc

So apart from Big-Endian to Little-Endian convertion (done by each program) eac3to takes 6th channel from .pcm writes it at position of 3rd channel and pushes one position below 4th and 5th channels. Can someone please explain what is the meaning of this channel rearrangement? (Headers of eac3to and Blu-ray Demuxer .wav files differ only by dwChannelMask: 0x0000060F (eac3to), 0x0000003F (Blu-ray Demuxer)).

[tebasuna51]

So apart from Big-Endian to Little-Endian convertion (done by each program) eac3to takes 6th channel from .pcm writes it at position of 3rd channel and pushes one position below 4th and 5th channels. Can someone please explain what is the meaning of this channel rearrangement? (Headers of eac3to and Blu-ray Demuxer .wav files differ only by dwChannelMask: 0x0000060F (eac3to), 0x0000003F (Blu-ray Demuxer)).
Like I say before the audio data in .pcm have different channel order than in wav container:

pcm: FL, FR, FC, BL, BR, LF
wav: FL, FR, FC, LF, BL, BR
then the LF channel in PCM (6th) must go to 4th position, BL (4th) to 5th and BR (5th) to 6th.

Seems BDR don't make the channel arrangement.
Is easy to verify the problem, if you play the BDR.wav with a multichannel capable player must listen the Low Frequency channel in the Back Right speaker.

[Stephen R. Savage]

Is there any way to implement audio cutting (by timecodes) in eac3to? A sample-accurate way to cut and join audio clips without intermediary files or pipes would be great. Perhaps something like "eac3to in.m2ts 2:out.flac -remove 00:00.000-01:30.000,20:30.000,22:00.000"

[TinTime]

@PowerGamer

Specifically the channels must be interleaved in this order (http://www.microsoft.com/whdc/device/audio/multichaud.mspx#EKLAC) for wav pcm.

[tebasuna51]

Is there any way to implement audio cutting (by timecodes) in eac3to? A sample-accurate way to cut and join audio clips without intermediary files or pipes would be great. Perhaps something like "eac3to in.m2ts 2:out.flac -remove 00:00.000-01:30.000,20:30.000,22:00.000"

Something like that:

v2.84
...
* new option for removing or looping audio data, e.g. "-edit=0:20:47,-100ms"

[umaximus]

I got loads of lossless check failed errors on Animatrix's TrueHD track while extracting/transcoding. Are this safe to overlook? How much of ms is failed byte?

eac3to v3.17
command line: eac3to 1) 2: video.vc1 4: audio.wavs
------------------------------------------------------------------------------
M2TS, 1 video track, 8 audio tracks, 18 subtitle tracks, 1:40:50, 24p /1.001
1: Chapters, 10 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
4: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
5: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: AC3, Italian, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
7: AC3, Japanese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
8: TrueHD/AC3, Japanese, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
9: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
10: AC3, Portuguese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
11: Subtitle (PGS), English
12: Subtitle (PGS), French
13: Subtitle (PGS), Italian
14: Subtitle (PGS), Italian
15: Subtitle (PGS), Dutch
16: Subtitle (PGS), Spanish
17: Subtitle (PGS), Portuguese
18: Subtitle (PGS), Japanese
19: Subtitle (PGS), English
20: Subtitle (PGS), Japanese
21: Subtitle (PGS), French
22: Subtitle (PGS), Italian
23: Subtitle (PGS), Italian
24: Subtitle (PGS), Dutch
25: Subtitle (PGS), Spanish
26: Subtitle (PGS), Portuguese
27: Subtitle (PGS), Japanese
28: Subtitle (PGS), English
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Writing WAVs...
[v02] Creating file "video.vc1"...
[a04] Creating file "audio.R.wav"...
[a04] Creating file "audio.L.wav"...
[a04] Creating file "audio.LFE.wav"...
[a04] Creating file "audio.SR.wav"...
[a04] Creating file "audio.C.wav"...
[a04] Creating file "audio.SL.wav"...
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated fe <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 8c <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 68 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated f5 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 73 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated e4 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 2e <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 26 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated b7 <WARNING>
[v02] Video overlaps for 7 frames at playtime 0:53:49. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:06:51. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:33:01. <WARNING>
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] Audio overlaps for 43ms at playtime 0:18:49. <WARNING>
[a04] Audio overlaps for 43ms at playtime 0:28:12. <WARNING>
[a04] Audio overlaps for 43ms at playtime 0:45:06. <WARNING>
[a04] Audio overlaps for 42ms at playtime 0:53:50. <WARNING>
[a04] Audio overlaps for 42ms at playtime 1:06:54. <WARNING>
[a04] Audio overlaps for 43ms at playtime 1:33:01. <WARNING>
[a04] Superfluous zero bytes detected, will be stripped in 2nd pass.
[a04] Starting 2nd pass...
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Reducing depth from 24 to 16 bits...
[a04] Writing WAVs...
[a04] Realizing RAW/PCM gaps...
[a04] Creating file "audio.R.wav"...
[a04] Creating file "audio.L.wav"...
[a04] Creating file "audio.SR.wav"...
[a04] Creating file "audio.SL.wav"...
[a04] Creating file "audio.C.wav"...
[a04] Creating file "audio.LFE.wav"...
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated fe <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 8c <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 68 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated f5 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 73 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated e4 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 2e <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 26 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated b7 <WARNING>
[a04] The processed audio track has a constant bit depth of 16 bits.
Video track 2 contains 145067 frames.
eac3to processing took 3 hours, 23 minutes.
Done.

[AnryV]

I think that eac3to (or ArcSoft decoder ??) uncorrectly decode 7.1 - L,R,C,LFE,Ls,Rs,Lsr,Rsr scheme.


eac3to v3.17
command line: eac3to Ls_for_eac3to.dtshd Ls.wavs
------------------------------------------------------------------------------
DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48khz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
CAUTION: Decoding this track with ArcSoft results in low volume. <WARNING>
Decoding with ArcSoft DTS Decoder...
Writing WAVs...
The original audio track has a constant bit depth of 24 bits.
Creating file "Ls.L.wav"...
Creating file "Ls.R.wav"...
Creating file "Ls.C.wav"...
Creating file "Ls.LFE.wav"...
Creating file "Ls.BL.wav"...
Creating file "Ls.BR.wav"...
Creating file "Ls.SL.wav"...
Creating file "Ls.SR.wav"...
eac3to processing took 4 seconds.
Done.


1. Ls.dtshd - is the file encoded by DTS-HD Encoder Suite from wavs from <Original>. Scheme - 7.1 - L,R,C,LFE,Ls,Rs,Lsr,Rsr
2. Ls_for_eac3to.dtshd - the same file but without header which don't recognized by eac3to
3. <Original> dir - contains original 8 wavs (1 second duration)
4. <eac3to> dir - contains 8 wavs decoded from Ls_for_eac3to.dtshd by eac3to
5. <StreamPlayer> dir - contains 8 wavs decoded from Ls.dtshd by DTS-HD StreamPlayer

As result - <StreamPlayer> fully identical <Original> but BL and BR files from <eac3to> contains not Ls and Rs from <Original> but mixed Ls+Lsr and Rs+Rsr

Sample
http://multi-up.com/201196

[Thunderbolt8]

look at the link of post #9676

[AnryV]

look at the link of post #9676
Problem, as I do understand, was not solved?

[umaximus]

I have a little question. In the past I have demux the PCM tracks from blu-rays and save them. As I can see from the logs, eac3to have remapped the channels already, but now when I want to transcode PCM to WAVS (for use with DTS-HD mas suite) eac3to is again remapping them which I think is not right. Is there a switch to turn off remapping of the channels?

[Thunderbolt8]

Problem, as I do understand, was not solved?no, apparently there havent been enough discs yet with this problem to justify the work

[bassgoonist]

I have some DVD-A disc in 5.0. eac3to seems to have no idea what to do with these. It won't downmix, it won't do DTS or AC3. If I open it up in audacity and insert an empty track where the LFE would be it works just fine. Is there anyway eac3to could have an option to do this?

[Tiziano]

Sorry, perhaps I'm very dum but I need a help. Sorry if I'm disturbing you continuously

I tried to convert a 7.1 ( 8 channels ) interleaved WAV file into an e-AC3 file ... I mean: it should be Dolby Digital Plus standard ... but I was not able to do it.

I'm wondered if anybody can help me giving the correct settings or command line in order to get this .WAV correctly converted ...

I thank you very much

[utenteanonimo64]

I have some DVD-A disc in 5.0. eac3to seems to have no idea what to do with these. It won't downmix, it won't do DTS or AC3. If I open it up in audacity and insert an empty track where the LFE would be it works just fine. Is there anyway eac3to could have an option to do this?

Why don't you use DVD-A Explorer? You can extract the multichannel data or downmix to 2.0. Once you have the WAV files then eac3to will do whatever you want with them...

[tebasuna51]

...
I tried to convert a 7.1 ( 8 channels ) interleaved WAV file into an e-AC3 file ... I mean: it should be Dolby Digital Plus standard ... but I was not able to do it.
...

Aften, the ac3 encoder included with eac3to, isn't a Dolby Digital Plus encoder, you can obtain only 5.1 standard ac3.

[tebasuna51]

I have some DVD-A disc in 5.0. eac3to seems to have no idea what to do with these. It won't downmix, it won't do DTS or AC3. If I open it up in audacity and insert an empty track where the LFE would be it works just fine. Is there anyway eac3to could have an option to do this?

To convert a wav file 5.0 you can use WavToAc3Enc or eac3to with external Aften.exe encoder:

eac3to source50 stdout.wav | Aften -pad 0 -readtoeof 1 -exps 32 -s 1 -b 640 - output.ac3

where:
-exps 32 -s 1 -b 640
is the best quality (and slow encode) than Aften can offer.

[deathlord]

Hi

Can I have eac3to use the source filename for the destination?
I mean in the command line, like eac3to source.wav dest.wav

[twazerty]

@madshi:

Awhile back I saw clipping audio when transcoding MP2->AC3. You mentioned it was due to internal bit depth restrictions in libavcodec, which could probably be overcome with a more recent build of libavcodec. Did you ever integrate this?

Noticed that the clipping problem isn't fixed yet. Can we expect a fix?

[blackcell]

For some unknown reason, Eac3to is unable to extract the BD disc "The Express". It gets nearly finished then aborts.

Log file:
eac3to v3.17
command line: "C:\Tools\RipBot\Tools\eac3to\eac3to.exe" "D:\BDMV\STREAM" 1) 1: "C:\tmp\The Express - Chapters.txt" 2: "C:\tmp\The Express 1080p VC-1.mkv" 3: "C:\tmp\The Express - 3 DTS Master.flac" 6: "C:\tmp\The Express - 6 English Subtitle.sup" -log="C:\tmp\The Express - Log.txt"
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 5 subtitle tracks, 2:09:42, 24p /1.001
1: Chapters, 20 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: DTS, Spanish, 5.1 channels, 24 bits, 768kbps, 48khz
5: AC3 Surround, French, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: Subtitle (PGS), English
7: Subtitle (PGS), Spanish
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), French
Creating file "C:\tmp\The Express - Chapters.txt"...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[s06] Extracting subtitle track number 6...
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "C:\tmp\The Express - 3 DTS Master.flac"...
[s06] Creating file "C:\tmp\The Express - 6 English Subtitle.sup"...
Reading the source file failed. <ERROR>
Aborted at file position 24368906240. <ERROR>

Anyone else seen this issue with other BD's? Any workarounds?

[flyingernst]

Hi,

for a hardwaretest I need a 6.1 Flac (in mkv container with video).
Is there any possibility to get a 100mb sample?

I tried to make a own one, but my mit dts-es 6.1 file got decoded to 5.1 flac, I think because I only use the dts-lib that comes with eac3to

thank you very much

greetings, flyingernst

PS: I tried a 5(6).1 ac3-EX File to, but eac3to don´t finds my nero decoder (BD/HD-DVD Plugin is installed, has a key)

[Inspector.Gadget]

flyingernst, you can make a 6.1 AVS script with Avisynth's SoundOut(), which has its own FLAC export plugin.

[setarip_old]

@blackcell

Hi!

I believe that error would be indicative of a disc that's either dirty or physically damaged. Try cleaning the disc...

[jasonwc]

I'm attempting to convert a DTS-HD MA 7.1 24/96 track to FLAC, but eac3to reports a discontinuity error, and aborts. I know that another individual attempted to convert the same file, and received an identical error. I have done a hash check to ensure that the file is not corrupted on my end. Apparently, the discontinuity is only 11 ms, so I am attempting to override the error, and encode to FLAC anyway.

I found this option that seems promising:

"eac3to v2.34 released

http://madshi.net/eac3to.zip

Code:

* TS/M2TS: new "-ignoreDiscon" option makes eac3to ignore discontinuity errors

As usual eac3to demands the source files to be ultra clean. If eac3to finds any stream errors it will at once stop processing. Since it happens rather often that there are some discontinuity errors in the stream I've now added a new option called "-ignoreDiscon". This way you can force eac3to to continue processing despite of eventual discontinuity errors. I have to warn you, though: When there are discontinuity errors, there's a good chance that there will be artifacts in the video/audio. So use this option with care..."

However, I can't seem to get the option to work. I just get "Command Line paramater "ignoreDiscon" is unknown" when I attempt to demux the m2ts or encode the DTS-HD MA track to FLAC.

I can extract the 24/48 DTS core, but thus far, I've had no luck getting the lossless dtsma track.

Sorry for my ignorance on this topic. I'm new to HD audio. While I haven't had an issue with other disks, this one appears to be corrupt and is therefore giving me issues.

[setarip_old]

@jasonwc

Hi!

If you look at Post #1 of this thread, you'll see that the latest version is v.3.17 - Perhaps your problem no longer exists...

[jasonwc]

@jasonwc

Hi!

If you look at Post #1 of this thread, you'll see that the latest version is v.3.17 - Perhaps your problem no longer exists...

Sorry, I should have been more clear. I am using eac3to 3.17. I was pasting from the change log that added the feature to which I referred.

[tebasuna51]

Read also in the changelog:

v2.45
...
* removed "-ignoreDiscon" from help; hint is shown when a discontinuity occurs

[jasonwc]

I saw that, but I get no hint. It just fails.


eac3to v3.17
command line: eac3to "J:\VIVALDI\BDMV\PLAYLIST\00000.mpls" 3: "G:\vivaldi.flac"
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 1:11:53, 60i /1.001
1: Chapters, 31 chapters
2: MPEG2, 1080i60 /1.001 (16:9)
3: DTS Master Audio, , 7.1 channels, 24 bits, 96khz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "G:\vivaldi.flac"...
[a03] [0:36:07] The source file seems to be damaged (discontinuity). <WARNING>
[a03] The ArcSoft DTS Decoder reported an error while decoding. <ERROR>
Aborted at file position 3571449856. <ERROR>

[Snowknight26]

Because eac3to can't force the ArcSoft DTS decoder to ignore it. eac3to ignored it, hence the warning.

[umaximus]

Ok, I have found my problem with trueHD tracks. Im using Pioneer receiver and when I was demuxing my trueHD tracks I didnt use -keepdialnorm switch. Now I have over 50 unplayable tracks that I cant use. Is there any software that could re-apply dialnorm?

[Xorp]

Did eac3to mess up the channel mapping when converting to flac:

before: 4: RAW/PCM, Japanese, 3/1 channels, 16 bits, 48khz

after: FLAC, 2/2 channels, 1:02:46, 16 bits, 771kbps, 48khz

Shouldn't it keep it as 3/1? I get also "a04 The AC3 encoder received a non-supported data format (pcm, 4, 16, -)." when trying to convert the both original pcm and the flac to ac3.

[Snowknight26]

FLAC has no 3/1 channel configuration so the closest one (2/2) is used.

eac3to's Aften implementation doesn't support strange channel configurations. Use eac3to's stdout.wav feature and pipe it to Aften. Example here (http://forum.doom9.org/showthread.php?p=1283462#post1283462).

[deathlord]

I saw that, but I get no hint. It just fails.

You could try demuxing the track using xport or tsmuxer, then convert using eac3to. It's quite likely the error will occur again, but maybe it's still worth a try?

[b66pak]

or you can use "-libav" to bypass the arcsoft decoder...
_

[Snowknight26]

...which would only decode the core.

[deathlord]

...which would only decode the core.

...or -sonic, which would decode only 5.1.
But the best bet in such situations is probably reripping the disc, the error might be gone.

[yesgrey]

When I used XP, I was able to run eac3to while I was doing other things. Now, with W7 RC1, I cannot, my computer stalls. I have to start Task Manager and change eac3to's process priority to low, and then it works a little better. Though, I still feel that with XP my computer did not stall at all.
Can any of W7 users share their experience? It might be some problem with multi-tasking handling that only occurs in RC1...

[Snowknight26]

Your HDD is probably being thrashed.

[newday_99]

Feature Request:

Don't suppose you could extract 'S_HDMV/PGS' subtitles to '.SUP' files

(eac3to title00.mkv movie.mkv)

like mkvextract (mkvtoolnix-3.1.0) can now demux/convert ????

(mkvextract tracks title00.mkv 3:track3.sup)

Thanks.

[yesgrey]

Your HDD is probably being thrashed.
Sorry, but I did not understand.:o
Can you elaborate a little more?

[datman]

I just started using this tool with anotherEAC3to and first off it works really good. I only wish I had the wits to use it before a GUI was developed.

I was able to get it to use Archsoft's DTS decoders but not Nero 7's AC3 decoders. I'm not sure how to get it to use the decoder?


Is this still one of the better ways to go?

[consultant]

Forgive me if this has been discussed (I did a search for TrueHD and FLAC and got 367 posts). I'm using ripbot to shrink buy blu-ray rips down. I recently started ripping my Star Trek movies. They all have TrueHD audio. I selected the option to convert to FLAC. The movie plays in VLC Media Player and it shows there is indeed an audio track but there is no sound. So I used USEAC3To to demux and convert the original mt2s from the disc. I tried to play the resulting .FLAC file and it won't play. So I'm just extracting the AC3 core now. Bummer, I would have preferred the TrueHD but in FLAC version but I guess that isn't possible. The AC3 is virtually transparent anyway. Just wondering if this is confirmed limitation and if it is a bug, or it is just not possible in a free program.

[datman]

Forgive me if this has been discussed (I did a search for TrueHD and FLAC and got 367 posts). I'm using ripbot to shrink buy blu-ray rips down. I recently started ripping my Star Trek movies. They all have TrueHD audio. I selected the option to convert to FLAC. The movie plays in VLC Media Player and it shows there is indeed an audio track but there is no sound. So I used USEAC3To to demux and convert the original mt2s from the disc. I tried to play the resulting .FLAC file and it won't play. So I'm just extracting the AC3 core now. Bummer, I would have preferred the TrueHD but in FLAC version but I guess that isn't possible. The AC3 is virtually transparent anyway. Just wondering if this is confirmed limitation and if it is a bug, or it is just not possible in a free program.

anotherAC3to GUI will do it http://www.curtpalme.com/forum/viewtopic.php?t=17002&sid=6aa1e7a252df759c18ed801496533911
uses free decoders too

[tebasuna51]

TrueHD to FLAC? Possible?
Yes.
Please put your log file:logfile:

[Beastie Boy]

Your HDD is probably being thrashed.

Using W7 Resource Monitor, I have noticed that eac3to seems to continually write to and read from the swap file, regardless off how much RAM is available. If your swap file is on a drive that you are using, then this causes a bottleneck.

Cheers, Beastie.

[TinTime]

The movie plays in VLC Media Player and it shows there is indeed an audio track but there is no sound.

Have you tried a different media player? Try MPC with or without the madFlac decoder, or try playing the flac file in foobar.

[mikelebron]

All my TrueHD movies are FLAC.. if your using RipBot then your doing something wrong as Ripbot will convert all HD audio formats to FLAC as it uses eac3to under the covers to demux the video and audio (then re-encodes to flac for the audio)...

Ask in the RipBot Forum to see what you are doing wrong...

If you dont car about compressing the video then just use eac3to command line.. its a multi step process if your dealing with a branching movie..


If you dont want to learn the eac3to command line then use
for mkv

HD DVD/Blu-Ray Stream Extractor - place in directory where eac3to is located. You will need to use mkvmerge (alsno known as mkvtoolnix) after the demux.
http://forum.doom9.org/showthread.php?t=141829


for m2ts, etc. (NOT MKV) - self containing solution
Clown BD:
http://forum.slysoft.com/showthread.php?t=25818

MKV has slightly better file size and allows chapters.. m2ts typically is more fogiving when steaming and can re-assembled into a bluray structure for an ISO.

Forgive me if this has been discussed (I did a search for TrueHD and FLAC and got 367 posts). I'm using ripbot to shrink buy blu-ray rips down. I recently started ripping my Star Trek movies. They all have TrueHD audio. I selected the option to convert to FLAC. The movie plays in VLC Media Player and it shows there is indeed an audio track but there is no sound. So I used USEAC3To to demux and convert the original mt2s from the disc. I tried to play the resulting .FLAC file and it won't play. So I'm just extracting the AC3 core now. Bummer, I would have preferred the TrueHD but in FLAC version but I guess that isn't possible. The AC3 is virtually transparent anyway. Just wondering if this is confirmed limitation and if it is a bug, or it is just not possible in a free program.

[consultant]

I don't see how I could be doing something wrong since there are only one option in RipBot, check the FLAC box or don't check it. lol

The fact that I demuxed the m2ts and converted the TrueHD to FLAC using USEac3to (select the m2ts, set command line for the TrueHD (3) track to FLAC) and the FLAC file still wouldn't play is indicating to me Eac3to may have an issue with the TrueHD encoding on the Star Trek original movies.

What is interesting is that I can convert the TrueHD to WAV and it plays fine. For kicks, I tried converting the WAV back to FLAC and it won't play. I tried playing it in MediaMonkey also just to make sure it wasn't VLC having the problem.

[stax76]

HD DVD/Blu-Ray Stream Extractor appear to be very buggy with nobody having an interest to fix the bugs.

[consultant]

MPC has killed two birds with one stone. It has less stuttering issues than VLC AND it plays the FLAC audio track correctly! Hurray!

Thanks for your help.

P.S. AnotherAC3To too buggy for me. Couldn't handle directories with spaces in the names for one. Got exception error, etc.

[mikelebron]

Using VLC.. yea that player is garbage... MPC is much better player...

MPC has killed two birds with one stone. It has less stuttering issues than VLC AND it plays the FLAC audio track correctly! Hurray!

Thanks for your help.

P.S. AnotherAC3To too buggy for me. Couldn't handle directories with spaces in the names for one. Got exception error, etc.

[MuLTiTaSK]

UsEac3to: a different GUI for eac3to (bit.ly/cgukqK)

[yesgrey]

Using W7 Resource Monitor, I have noticed that eac3to seems to continually write to and read from the swap file, regardless off how much RAM is available. If your swap file is on a drive that you are using, then this causes a bottleneck.
Thanks for your post, but with my 4GB memory I am running W7 without any swap file, and I also disabled the superfetch service. With XP I also don't use any swap file, but the usage is a lot better...

[Blue_MiSfit]

Erm.. you do know that running any Windows OS without the swap file enabled is generally a bad idea, right? I would suggest keeping it turned on, unless you have solid reproducible results to the contrary (and all your software works without issues)

~MiSfit

[raquete]

yesgrey3,
in my understanding Win7(x86) don't recognize 4Gb Ram but only round 3,25Gb.
even with XP or Win7 is good adjust the swap file....to tell the true, i don't know how to deal/adjust swap file in Win7.

[yesgrey]

Erm.. you do know that running any Windows OS without the swap file enabled is generally a bad idea, right? I would suggest keeping it turned on, unless you have solid reproducible results to the contrary (and all your software works without issues)
I don't have any problems working without the swap file. Only once when I was performing some tests with an application I'm developing and I was allocating a lot of memory. I enabled the swap file and it worked ok.

yesgrey3,
in my understanding Win7(x86) don't recognize 4Gb Ram but only round 3,25Gb.
even with XP or Win7 is good adjust the swap file....to tell the true, i don't know how to deal/adjust swap file in Win7.
I'm Using W7(x64), so it's really 4GB.;)
Adjusting the swap file in W7 is exactly the same as it is in XP.
Control Panel->System->Advanced Settings->Performance->Advanced->Virtual Memory

I think that the problem I'm having should be some kind of bug in W7 RC1 (probably it was already fixed on the retail version), because I already had some problems with my application cr3dlut.
To avoid computer stalling, I have set it's priority to "BELOW_NORMAL_PRIORITY_CLASS". With XP it worked fine. When I tried it in W7, it didn't. I had to set the priority to "IDLE_PRIORITY_CLASS" to avoid the stalling.
That's what I have to do now with eac3to, I have to go to the task manager and set it manually.:(
OTOH, they process handling in W7 is different from XP's, so it could also be something related to that... that's why I asked here if anyone else with W7 retail are getting the same results when using eac3to...

[raquete]

i'm newby in W7 and i'm installing/adjusting in 4 pcs at the same time, thanks for your hints.

i only use pc for audio and never could 'stand' XP, i still use Win2000 and with the rights adjusts in swap file 'everything' run fine, even with some programs that need XP as min system...don't ask me why but i can put some forbidden programs for Win2000 working fine here.
is hard to left what works...
for example: after install W7 with roung 9.3Gb in hd, the system don't recognize my wireless...gee, 9,3Gb around and still need a single dll to put a single wireless working...i can't stand news systems...lol.
my bootalble dvd of Linux Ubuntu recognize the wireless and all cards in my pcs without install anything in hd, how can it be? bootable Ubuntu free versus W7 Ultimate ~U$350,00(here) installing 9,3GB.

in the other side my nephew is video designer of one great tv channel and he told me that using W7(x64) have some problems, some programs don't run and he need to use 2 pcs: one with x86 and other with x64...and with 12GB Ram in each!
then, if you have 2 pcs test one with x86, maybe yours issues can end.

[yesgrey]

i only use pc for audio and never could 'stand' XP, i still use Win2000
I also cannot stand XP GUI. Those very highly saturated colors are irritating. The trick is using the classic GUI theme, which will give you the Win2000 GUI in XP.;)

if you have 2 pcs test one with x86, maybe yours issues can end.
Thanks, but I don't think this is a x86 vs x64 problem. Also, I will not go back to x86, I want a 64bit OS. It's preferable for software developing purposes.

[raquete]

I also cannot stand XP GUI. Those very highly saturated colors are irritating. The trick is using the classic GUI theme, which will give you the Win2000 GUI in XP.;)
:eek: oh...classic GUI theme? really?
i never heard and knew about that possibility. :cool:
can you pm me with the hints to do that please?

cheers!

[BLKMGK]

Has anyone used eac3to to create branching Matroska files? I'm led to believe it's possible using eac3to by a posting to the MakeMKV support forum (http://www.makemkv.com/forum2/viewtopic.php?f=1&t=897) but I cannot figure out how to do it. I've asked elsewhere in the Matroska forums (http://forum.doom9.org/showthread.php?t=72700) for help as well. Currently I rip a BD that has multiple versions multiple times but this really seems silly. MKV apparently supports the ability to branch much like BD does and I'd like to try and take advantage of that as well as urge the XBMC developers to support it. However if it cannot be easily built I won't be able to provide examples nor will anyone be able to use it. :(

Thoughts?

[Snowknight26]

eac3to would have nothing to do with that.

[setarip_old]

@BLKMGK

Hi!

Perhaps you're referring to this post:

http://forum.doom9.org/showpost.php?p=1129751&postcount=4465

But I believe it refers to BD...

[yesgrey]

:eek: oh...classic GUI theme? really?
i never heard and knew about that possibility. :cool:
can you pm me with the hints to do that please?

I've created a thread (http://forum.doom9.org/showthread.php?t=152748) with instructions for both XP and 7, in case other people also do not know how to do it.

[BLKMGK]

eac3to would have nothing to do with that.

Actually yes it would since eac3to is capable of pulling out specific sections and is capable of ripping the various branches. If you had followed the post I linked on the MakeMKV forums you'd have noted that the author specifically mentioned eac3to as one of the tools he used to facilitate the process but provided no details on his process. eac3to builds MKV files, I use it pretty much weekly to do so, I am simply asking if anyone has used it to pull out branches and facilitate creating a file that allowed the various branches to live in one file....

@BLKMGK
Hi!

Perhaps you're referring to this post:

http://forum.doom9.org/showpost.php?p=1129751&postcount=4465

But I believe it refers to BD...

No, this refers to the ability eac3to possesses to follow a specific branch in a BD structure. This is pretty handy if you've got a BD with multiple versions and want just one. What I want is all of them but without the overhead of storing the redundant parts - just like a BD does it...

[n3rdy]

Hello,

In case I would ask a stupid question... please forgive me.
When using eac3to to demux my HDTV records it says

"a remaining delay of -11ms seconds could not be fixed"

So in order to fix that delay in my re-encode, do i have to type :
DelayAudio(-0.011) or DelayAudio(0.011)

(which obviously is a part of my avisynth script)


Thank You !

[tebasuna51]

If you need recode the audio you can use DelayAudio(-0.011), but you can preserve the original audio without recode, -11ms isn't a problem.

[zeropc]

question: i'd like to cut a 22sec and 160ms at the beginning from a dts-hd file. what cmd would i need to use to achieve this with eac3to?
i'm not sure if the delay option is the one i need to use.

thanks for the help :)

[xxx666yyy777]

I normally use v3.14 because latest version has some type of problem where the last line of output to stdout is not present, so my front end cannot parse certain data.

But I have also tried v3.16 and get the same issue.

Apparently, my AV receiver is highly sensitive to a TrueHD stream. Unless -keepDialnorm is used, TrueHD output is very distorted.

It is no big deal, I have dialogue normalization turned off on the AV receiver anyway.

Does anyone know, whether the TrueHD/DialNorm issue still persists with the new Pioneer Receivers (1019 series for example)?

Thx

[stax76]

There is a new GUI in StaxRip:

Download: http://www.stax76.bplaced.net/files/applications/StaxRip/StaxRip_1.1.5.1.7z

http://f.imagehost.org/0756/Source_Files-2010-02-15_04_33_31.png (http://f.imagehost.org/view/0756/Source_Files-2010-02-15_04_33_31)

http://i.imagehost.org/0810/StaxRip-2010-02-15_04_34_06.png (http://i.imagehost.org/view/0810/StaxRip-2010-02-15_04_34_06)

http://f.imagehost.org/0415/SS-2010-02-15_04_38_14.png (http://f.imagehost.org/view/0415/SS-2010-02-15_04_38_14)

[Snowknight26]

I hope you're only demonstrating your GUI's capabilities and not actually trying to output DTS-HD with those options.

[MuLTiTaSK]

@Snowknight26

stax76 is just demonstrating eac3to switches he did a great job creating a GUI for StaxRip check it out -> StaxRip 1.1.5.1 (bit.ly/cu2A2W)
http://img192.imageshack.us/img192/7747/ss20100217232155.png
http://img192.imageshack.us/img192/8356/ss20100217232121.png

[raquete]

using this .cue to burn with EACopy

REM GENRE Alternative
REM DATE 2005
REM DISCID EA12B813
REM COMMENT "ExactAudioCopy v0.99pb5"
PERFORMER "Coldplay"
TITLE "Greatest Hits (CD 1)"
FILE "Coldplay - Greatest Hits (CD 1).wav" WAVE
TRACK 01 AUDIO
TITLE "Square One"
PERFORMER "Coldplay"
...etc

i get 'error' on line 7(file type not supported) if i decompress .flac to .wav with eac3to
(eac3to "Coldplay - Greatest Hits (CD 1).flac" "Coldplay - Greatest Hits (CD 1).wav")

but when i decompress the .flac to .wav with adobe audition and using the same .cue, EACopy don't give error message and burn the media.

what can be?

[nurbs]

Sampling rate or bitdepth? Or maybe eac3to writes the wrong kind of wav (w64) by default. Would be helpful if you posted what eac3to has to say about the file.

[raquete]

Sampling rate or bitdepth? Or maybe eac3to writes the wrong kind of wav (w64) by default. Would be helpful if you posted what eac3to has to say about the file.

.flac Sampling rate 44100, bitdepth 16
command line to decompress with eac3to:
eac3to "Coldplay - Greatest Hits (CD 1).flac" "Coldplay - Greatest Hits (CD 1).wav"

loading the same .flac in adobe audition and saving as .wav 44100,16 works.

eac3to don't say anything against the .flac and about the .wav in the log, just decompress without advertences.

the only difference is in size of the .wav:
eac3to: 845.466.452 bytes
audition: 845.466.564 bytes

[Snowknight26]

Try with -simple.

[raquete]

Try with -simple.

i'm sorry, i don't understood...please, explain me what to do.

i forgot to post, here the eac3to log file:

eac3to v3.17
command line: eac3to "Coldplay - Greatest Hits (CD 1).flac" "Coldplay - Greatest Hits (CD 1).wav"
------------------------------------------------------------------------------
FLAC, 2.0 channels, 1:19:53, 16 bits, 864kbps, 44.1khz
Decoding FLAC...
Writing WAV...
Creating file "Coldplay - Greatest Hits (CD 1).wav"...
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 2 minutes, 2 seconds.
Done.

[nurbs]

He means try:
eac3to "Coldplay - Greatest Hits (CD 1).flac" "Coldplay - Greatest Hits (CD 1).wav" -simple

[raquete]

He means try:
eac3to "Coldplay - Greatest Hits (CD 1).flac" "Coldplay - Greatest Hits (CD 1).wav" -simple

work, now EACopy get the file without problems,
:thanks: thank you both but...
i don't saw this option in eac3to help.

why is needed the option -simple?

[setarip_old]

@raquete

See this earlier post:

http://forum.doom9.org/showpost.php?p=1347703&postcount=9592

[raquete]

@ setarip_old

now is clever.
this option could be in the help file and be used as 'default'.

:thanks:

[setarip_old]

@raquete
this option could be in the help file and be used as 'default'.True - but it seems that the author is no longer supporting this program...

[raquete]

@raquete
True - but it seems that the author is no longer supporting this program...

a pity and :confused: i don't understand why he don't support as the program is great, full of features, fast(etc) and is used 'around the world' for lots of people in lots of forums.
i know some GUIs where eac3to was chosed to be used and is used...because works.
always i see questions and answers here and around about eac3to then, is still running, is good and deserve support but we can't discuss the developer tastes...

do you know the complete list of options/features?

[b66pak]

why is eac3to not respecting the M$ WAVE_FORMAT_EXTENSIBLE ChannelMask (http://www.microsoft.com/whdc/device/audio/multichaud.mspx)?

http://www.microsoft.com/whdc/device/audio/multichaud.mspx#E4C

http://www.microsoft.com/whdc/device/audio/multichaud.mspx#EKLAC


after decoding ac3/dts 5.1 channels to .wav i get the wrong channel order!

original file (.dts):

General
Complete name : audio.dts
Format : DTS
Format/Info : Digital Theater Systems
File size : 16.9 MiB
Duration : 1mn 33s
Overall bit rate : 1 510 Kbps

Audio
Format : DTS
Format/Info : Digital Theater Systems
Duration : 1mn 33s
Bit rate mode : Constant
Bit rate : 1 510 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Surround: L R, LFE
Sampling rate : 48.0 KHz
Resolution : 24 bits
Stream size : 16.9 MiB (100%)




the decoded .wav:

General
Complete name : audio.eac3to.wav
Format : Wave
File size : 77.3 MiB
Duration : 1mn 33s
Overall bit rate : 6 912 Kbps

Audio
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Unsigned
Codec ID : 00001000-0000-0100-8000-00AA00389B71
Codec ID/Hint : Microsoft
Duration : 1mn 33s
Bit rate mode : Constant
Bit rate : 6 912 Kbps
Channel(s) : 6 channels
Channel positions : Front: L, C, R, Middle: L, R, LFE should be Front: L, C, R, Surround: L, R, LFE
Sampling rate : 48.0 KHz
Resolution : 24 bits
Stream size : 77.3 MiB (100%)




File ........: audio.eac3to.wav
Size ........: 81017924 bytes

---------------------------------------------- Header Info
ChunkID .....: RIFF
RiffLength ..: 81017916
Container ...: WAVE
SubchunkID ..: fmt (Length: 40)
AudioFormat .: 65534 (WAVE_FORMAT_EXTENSIBLE)
NumChannels .: 6
SampleRate ..: 48000
ByteRate ....: 864000
BlockAlign ..: 18
BitsPerSample: 24
ValidBitsPS .: 24
MaskChannels : 1551 (FL FR FC LF SL SR) should be 63 (FL FR FC LF BL BR)
SubType .....: 1 (Integer)
SubchunkID ..: data (Length: 81017856)
Offset data .: 68
Duration ....: 93.77067 sec., (0h. 1m. 33.77067s.)
------------------------------------------------- End Info
_

[Snowknight26]

It's hard to tell if you really care or not when you say 'M$.'

[b66pak]

sorry...Microsoft WAVE_FORMAT_EXTENSIBLE ChannelMask...

a quick fix for the people who need it (for a 5.1 channel 24 bit@48000Hz .dts):

eac3to audio.dts stdout.raw | wavfix - audiofixed.wav -ignorelength -m 63 -i 24 -c 6 -s 48000

the result:

Front: L, C, R, Surround: L, R, LFE
MaskChannels : 63 (FL FR FC LF BL BR)

[Snowknight26]

Actually, a ChannelMask of 0x0000060F is recommended by Microsoft post XP SP2 for 5.1 channel WAVs. 0x0000003F is the old ChannelMask and is also compatible. Seems like it's merely a cosmetic difference.

[b66pak]

Actually, a ChannelMask of 0x0000060F is recommended by Microsoft post XP SP2 for 5.1 channel WAVs. 0x0000003F is the old ChannelMask and is also compatible. Seems like it's merely a cosmetic difference.


i am asking because i use apple's aacencoder (qtaacenc (http://tmkk.hp.infoseek.co.jp/qtaacenc/)) and with ChannelMask 1551 (0x0000060F) i get empty sorround channels (SL SR)...
_

L.E. this is now fixed in the new version of qtaacenc...
_

[Biggiesized]

Whenever I try to run an .mlp file through eac3to, I get an error claiming the program cannot determine the source file. I thought eac3to could decode the MLP file format?

[tebasuna51]

@Biggiesized
Work with all my mlp files.
Upload a sample to test.

[Biggiesized]

@Biggiesized
Work with all my mlp files.
Upload a sample to test.
http://rapidshare.com/files/354338838/track-01-02_1_-01-_L-R_-24-96000.mlp

[mrr19121970]

It's a longshot, I know as Madshi isn't active currently on this project. I'm in the process of converting my HD-DVDs (again) to Blu-Ray. This time I'm going the path E-AC3 to DTS-MA using "DTS-HD Master Audio Suite v2.0 (http://store.dts.com/)". Is there any chance that you can implement a similar 'hooking' to this application as you do with SurCode?

[Inspector.Gadget]

Why go lossy -> lossless? Makes no sense.

[TinTime]

Doesn't it make more sense from a quality point of view to convert it to a lossless format rather than another lossy one?

I mean, given that HD DVD DD+ has to be converted to something else for Blu-ray compatibility.

[Inspector.Gadget]

In a strictly technical sense, sure, but from the perceptual side? I don't transcode audio where I can get away with it, and like a lot of people I'd rather have lossless if the source itself is lossless, but once you're transcoding anyway, I doubt you the vast majority of people can tell the difference between DTS-MA and Dolby Digital @ 640 kbps.

[tebasuna51]

http://rapidshare.com/files/354338838/track-01-02_1_-01-_L-R_-24-96000.mlp
Yes eac3to don't recognize the track, ffmpeg can't decode it (lot of errors) and Foobar2000 crash when try play the file.

Can you play this file with any player?

[Biggiesized]

No, hence my problem. I get a strange, loud glitching noise when I try to play it in foobar2000 with the proper DVD-Audio component installed.

[mrr19121970]

Doesn't it make more sense from a quality point of view to convert it to a lossless format rather than another lossy one?

I mean, given that HD DVD DD+ has to be converted to something else for Blu-ray compatibility.

What would be a comparible DTS conversion for Blu-Ray format (as HD-DVD e-AC3 is not compatible)? DTS-HD I suppose is best (and not DTS-HA Master Audio as I though before).

Thanks for the reply.

[Inspector.Gadget]

DTS-MA is lossless. What I was saying to TinTime, though, was that transcoding from a lossy format to another lossy format seems more sensible to me provided that the destination file is transparent to the ear because there's no intrinsic value in retaining a lossless-from-the-master-copy format anymore. On the other hand, TinTime made the point that lossy->lossless will retain the mathematical quality of the original. This is essentially a philosophical debate, because you almost certainly won't be able to hear the difference between DTS-MA and 640kbps AC3.

[Snowknight26]

TinTime made the point that lossy->lossless will retain the mathematical quality of the original.

It won't though. Lossy to lossless is a lossy process.

[TinTime]

It won't though. Lossy to lossless is a lossy process.

...yes but the lossy source has to be decoded at some point before you can listen to it anyway. Whether that's at playback or offline beforehand doesn't really matter.

I'll approach the question differently.

mrr19121970 said that this is being done for Blu-ray compatibility. Presumably that means burning the result to a BD. In that case the audio codec choice should be straightforward - choose the highest quality you can that will fit on the blank disc. If there's room for DTS-HD MA then why not use it?

[Inspector.Gadget]

It won't though. Lossy to lossless is a lossy process.

You're of course correct. I should have said will retain MORE of the mathematical quality of the original because there's no data discarded for compression purposes.

[mrr19121970]

I'll just go E-AC3 to PCM. It's quick & easy. Space is not an issue.

On a side note, I noticed the following:

Converting to DTSHD-MA saved about 40% in space over the .WAV produced by eac3to. For some strange reason converting to DTSHD actually produced files larger than DTSHD-MA.

[jj666]

DTS-HD HR is CBR, DTS-HD MA is VBR, I suppose the compression is better that's all.

Cheers,

-jj-

[deado]

With the Lord of the Rings trilogy just announced as DTS-HD MA 6.1, I'm guessing it will be just like the Harry Potters and The Number 23 where it has a DTS-ES core and the MA information is not extracted as 6.1 by eac3to, but only played properly when decoded by a player such as the PS3.

Is there any way that eac3to can convert such tracks to 6.1 FLAC? Or are we simply limited to 5.1 for those?

[Inspector.Gadget]

deado - are you using the Arcsoft DTS decoder?

[deado]

deado - are you using the Arcsoft DTS decoder?

Yes I am, version 1.1.0.7. I just tried with Terminator 2 Skynet and eac3to only detects 5.1 channels, not 6.1. So there seems to be no way to properly extract the 6.1?

[Snowknight26]

If eac3to detects it as having 5.1 channels, it's 5.1 channel audio. The ArcSoft decoder isn't used until you actually start decoding the audio.

[deado]

Hmm well that's false advertising stating 6.1 all over the back cover... that goes for both Lionsgate (T2 Skynet) and Warner (HP 1& 2 ultimate, Number 23, LOTR trilogy).

They should say 5.1 ES not 6.1. X-men 3 seems to be pretty much the only BD which is marked as 6.1 which is actually encoded as such.

Here I thought I might have been able to watch LOTR trilogy in 8-channel FLAC, rather than 6-channel FLAC :(

[tebasuna51]

Hmm well that's false advertising stating 6.1 all over the back cover... that goes for both Lionsgate (T2 Skynet) and Warner (HP 1& 2 ultimate, Number 23, LOTR trilogy).

They should say 5.1 ES not 6.1. X-men 3 seems to be pretty much the only BD which is marked as 6.1 which is actually encoded as such.

Here I thought I might have been able to watch LOTR trilogy in 8-channel FLAC, rather than 6-channel FLAC :(
There are two kinds of DTS-ES, and eac3to report like:

1) DTS-ES, 6.1 channels
Called DTS-ES discrete, and the Back Center is encoded digitally like a independent channel.
ArcSoft decode this like 6.1 channels

2) DTS-ES, 5.1 channels
Here the Back Center channel is encoded in Back Left/ Back Right channels in analog mode. This is DTS ES-matrixed.
When you send this stream (SPDIF or HDMI) to a receiver DTS-ES compliant you can obtain a 6.1 output.
ArcSoft decode this like 5.1 wav, but the Back Center channel is still matrixed in Back channels.
When you send this wav (or Flac decoded) to a receiver like PCM 5.1 (only by HDMI) you can obtain a phantom Back Center channel.

[deado]

There are two kinds of DTS-ES, and eac3to report like:

1) DTS-ES, 6.1 channels
Called DTS-ES discrete, and the Back Center is encoded digitally like a independent channel.
ArcSoft decode this like 6.1 channels

2) DTS-ES, 5.1 channels
Here the Back Center channel is encoded in Back Left/ Back Right channels in analog mode. This is DTS ES-matrixed.
When you send this stream (SPDIF or HDMI) to a receiver DTS-ES compliant you can obtain a 6.1 output.
ArcSoft decode this like 5.1 wav, but the Back Center channel is still matrixed in Back channels.
When you send this wav (or Flac decoded) to a receiver like PCM 5.1 (only by HDMI) you can obtain a phantom Back Center channel.

Hm, damn well I'm just outputting over analog with madFLAC -> ffdshow -> Reclock WASAPI so is there anything in ffdshow that can do the same thing as receivers do to obtain that rear channel?

[tebasuna51]

Hm, damn well I'm just outputting over analog with madFLAC -> ffdshow -> Reclock WASAPI so is there anything in ffdshow that can do the same thing as receivers do to obtain that rear channel?
over analog? I don't know sound cards with 6.1 output (only 5.1 or 7.1)

AFAIK ffdshow can't extract the back center channel.

[deado]

over analog? I don't know sound cards with 6.1 output (only 5.1 or 7.1)

AFAIK ffdshow can't extract the back center channel.

It's 7.1... but I was using the eac3to -double7 command to double the rear channel on the couple of 7-channel FLAC files I have (X-Men 3 and Crash Australian version), to get 8-channel FLAC.

But anything which isn't originally discrete 7-channel and only 6-channel with ES it seems I'm out of luck :(

[Snowknight26]

Hmm, after audio overlaps are detected, it seems that eac3to no longer runs audio encoding concurrently. Say extract and convert two DTS-HD MA tracks to FLAC and both of them have audio overlaps, you'd expect eac3to to run the 2nd pass, fix the overlaps, and do the conversion of both at the same time, but the 2nd track's conversion begins only after the first one has finished. :|

[utenteanonimo64]

Sorry if this has been covered already but this thread is overwhelming to search.


eac3to v3.17
command line: "c:\program files (x86)\eac3to\eac3to.exe" track01.dts "01 - .flac"
------------------------------------------------------------------------------
DTS-96/24, 5.1 channels, 0:03:12, 24 bits, 1510kbps, 96khz
The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
Decoding with libav/ffmpeg...
Remapping channels...
Reducing depth from 64 to 24 bits...
Encoding FLAC with libFlac...
Creating file "01 - .flac"...
eac3to processing took 24 seconds.
Done.


The FLAC file I get is 48KHz not 96KHz. Is this normal? Why? (I suppose because only the core DTS is decoded). Do I need one of the external decoders to decode the full DTS stream and maintain 96KHz?

Thanks.

Appendix: I have downloaded and installed a trial copy of ArcSoft TotalMedia. eac3to can't find the DTS decoder even after adding the ArcSoft decoder directory to the Windows PATH variable (which by the way it is not what the first post says but it's "C:\Program Files (x86)\ArcSoft\TotalMedia Theatre 3\Codec".
Any help on this additional problem?

[TinTime]

The FLAC file I get is 48KHz not 96KHz. Is this normal? Why? (I suppose because only the core DTS is decoded). Do I need one of the external decoders to decode the full DTS stream and maintain 96KHz?

Yes. You need to use the ArcSoft or Sonic decoder for full 96/24 decoding.

[utenteanonimo64]

Yes. You need to use the ArcSoft or Sonic decoder for full 96/24 decoding.

OK Thanks. So now I know I need to make this ArcSoft decoder work. Any suggestions? I am using Windows 7 64bit and I installed the trial version of TotalMedia Theatre 3. As far as I can see the DTS decoder is in this folder:

C:\Program Files (x86)\ArcSoft\TotalMedia Theatre 3\Codec

and

dtsdecoderdll.dll is version 1.1.0.7 which according to madshi's instructions should be compatible with eac3to.

As I said I have added the above directory to PATH enviroment variable and I have also copied all dts dlls into the eac3to directory. But if I run "eac3to -test" eac3to cannot find any external decoder. Is it because TMT is a trial version?

Any help is appreciated.

[TinTime]

Can't help you there I'm afraid - it just worked on my system (WinXP) without any fiddling about.

Try looking through this thread (http://forum.doom9.org/showthread.php?t=148324).

[b66pak]

look at this:

http://forum.doom9.org/showthread.php?t=148324
_

[utenteanonimo64]

look at this:

http://forum.doom9.org/showthread.php?t=148324
_

Thanks! I followed the instructions in this message:

http://forum.doom9.org/showthread.php?p=1255699#post1255699

and it worked.

:thanks:

[madshi]

A while back you mentioned that you were working or had implemented a plugin system for eac3to with support for plugging custom filters. Would that suit my needs (I'm a programmer myself)? Any documentation for it?
It's not ready yet.

i have the Freedom bluray set and i would like to remux the english versions but eac3to can't do it correctly, as the playlists use only parts of certain m2ts files. it's the only bluray where i've seen such playlists.
I was aware of that this in theory is possible, but I haven't seen such a Blu-Ray myself yet. Currently eac3to does not support this kind of Blu-Ray. I'd have to have such a full Blu-Ray myself, then I could add support for that. But honestly, just one Blu-Ray doing this is not enough to motivate me to work on implementing such a complicated thing.

any chance to update the lavc lib ? there are EAC3 Spectral & AAC LATM patches floating around in the FFmpeg-dev ML...
I should really do that, but I'm a bit afraid, because it will be quite a bit of work and I'll need to retest all libav decoders etc. Will come sooner or later, but not right now...

A aesthetic bug I noticed is that eac3to reports the dialnorm setting different between Dolby and DTS. For a TrueHD track it might say -27dB (the DN setting), but for a DTS-MA track it says -4dB (the DN offset). It's unlikely the DTS-MA track was actually set 27dB lower than standard.
I've researched this. The AC3 and DTS specs are very kinda the opposite of each other in terms of dialnorm:

AC3: The dialnorm value tells us how much headroom there is between 100% volume and speech volume.
DTS: The dialnorm value tells us how much the decoder should lower volume.

Because of that a -27dB AC3 dialnorm value equals a -4dB DTS dialnorm value. Now of course eac3to could change the display of the dialnorm values, so that AC3 and DTS match. But I'm not sure if I should do that? What do you guys think?

I was little bit inaccurate, libsox is library under LGPL, so it can be used
Oh, didn't know that - thanks! However, right now I don't feel the need to use sox, as eac3to's current resampling algorithm seems to work pretty well...

Any News about the libav Lossless check failed on TrueHD Tracks with libavcodec ?
What news do you expect? Do a search in this thread. All has been explained.

What is the status of the PGS subtitle issue regarding Decoder Time Stamp? Thanks.
Ehm, not fixed yet... :(

i noticed that eac3to isn't capable of removing dialnorm from dts master tracks. is this a known issue?
eac3to can remove dialnorm from the core. However, eac3to at this point in time can not modify the DTS-HD data blocks, because there's no full DTS-HD specification available. I wouldn't know what to change and where. To my best knowledge, the Sonic DTS decoder ignores dialnorm, anyway. Not sure about ArcSoft, though...

Due to Miramax giving lossless honours to the English dub only on their Hero blu-ray, I've remuxed the original Mandarin DTS-ES track from the R3 DVD with an x264 encode of the blu-ray video. However, I needed to add 7900ms of delay for the A/V to sync up and when I put the DTS track through eac3to for this the resulting track has 7900ms of 'noise' at the beginning instead of silence.
eac3to can not "create" new silent DTS frames which are needed for this delay. Instead eac3to simply repeats the very first frame of the audio track as often as necessary to achieve the wanted delay. Normally a DVD audio track begins silent, so this usually works. If your DVD audio track begins non-silent, then eac3to's way of delaying isn't silent, but results in ugly noise, sadly. If there was a free DTS encoder available, I could solve this, but there isn't. For AC3 I'm using Aften to do the audio delay properly and guaranteed-noise-free.

delaycut can do this better, but since it doesn't have a DTS encoder, either, it simply uses a default silent DTS frame - which doesn't have the DTS-ES flag set, of course.

I'm an old-school GNU/Linux command line junkie. Running eac3to inside
screen (http://en.wikipedia.org/wiki/GNU_Screen), the output comes out like this: [...]
eac3to was not written with Linux in mind. I've zero knowledge about Linux. But I thought Linux was quite good at emulating Windows? If so, why can't it even simulate colored command line output? eac3to isn't doing anything special, it's using the official win32 APIs for colored command line output. IMHO you should ask the Linux guys (wine?) to fix the problem.

I'd also love to have an option to not play sound.
Just delete or rename the wav files.

apparently, wine devs
don't want apps to detect wine (http://forum.winehq.org/viewtopic.php?p=25906&sid=4a6025570ec9cb97aa5e4d952cf6ad88)
Great! So ask them to fix the problem with colored command line output. Done. Next?

I think I found bug.
eac3to 00001.m2ts 1: chapters.txt 3: audio1.m4a -no2ndpass 6: audio2.m4a -quality=0.6 -no2ndpass 9: english.sup
It starts, spawns 2 nero encoders, processes whole input and then one Nero finishes, but 2nd does not and stays idle.
I'll add that to my to do list.

[madshi]

will you be adding support for nero 9
Maybe, sooner or later.

Right, I have bunch of trueHD soundtracks that were demuxed in past with eac3to. Since I got a capable HD receiver I want to mux HD audio back. Doing this with TsMuxeR I have no problems with DTS-HD MA, but TsMuxer wont accept trueHD streams and is giving an error of 'Unsupported format' (cant detect stream type) on every single trueHD stream that I have (+100). Im sure there is difference in header info of how eac3to is demuxing trueHD but I have no clue how to 'repair' it.
I've no clue, either. eac3to demuxes TrueHD tracks the way it's supposed to. If tsMuxeR doesn't accept these files, then it's a problem with tsMuxeR. You could try converting your TrueHD tracks to TrueHD+AC3, maybe tsMuxeR wants that?

eac3to source.thd dest.thd+ac3

hm so whats the standing regarding this problem, is madshi willing to do this patch so that arcsoft can properly recognize the channel order?
I think that eac3to (or ArcSoft decoder ??) uncorrectly decode 7.1 - L,R,C,LFE,Ls,Rs,Lsr,Rsr scheme.
Problem, as I do understand, was not solved?
This is a problem in the ArcSoft decoder and not in eac3to. It would be theoretically possible to workaround the problem by manipulating the DTS-HD tracks in such a way that they're more pleasing to the ArcSoft decoder, but it would be a hell of a lot of work for me. And I don't really see why I'm always asked to fix problems in the software of other people?

I have a little question. In the past I have demux the PCM tracks from blu-rays and save them. As I can see from the logs, eac3to have remapped the channels already, but now when I want to transcode PCM to WAVS (for use with DTS-HD mas suite) eac3to is again remapping them which I think is not right. Is there a switch to turn off remapping of the channels?
Why don't you just trust eac3to? I've not heard of any channel remapping bugs for months/years.

Awhile back I saw clipping audio when transcoding MP2->AC3. You mentioned it was due to internal bit depth restrictions in libavcodec, which could probably be overcome with a more recent build of libavcodec. Did you ever integrate this?
Noticed that the clipping problem isn't fixed yet. Can we expect a fix?
Yes, sooner or later, but not right now.

Ok, I have found my problem with trueHD tracks. Im using Pioneer receiver and when I was demuxing my trueHD tracks I didnt use -keepdialnorm switch. Now I have over 50 unplayable tracks that I cant use. Is there any software that could re-apply dialnorm?
Yes, the upcoming eac3to v3.18 will do that.

When I used XP, I was able to run eac3to while I was doing other things. Now, with W7 RC1, I cannot, my computer stalls. I have to start Task Manager and change eac3to's process priority to low, and then it works a little better. Though, I still feel that with XP my computer did not stall at all.
Can any of W7 users share their experience? It might be some problem with multi-tasking handling that only occurs in RC1...
Don't know what problem that could be. You can use the "-lowPriority" switch to avoid having to use the task manager.

Using W7 Resource Monitor, I have noticed that eac3to seems to continually write to and read from the swap file
eac3to does not do such a thing. There isn't even a Windows API available to access the swap file. The swap file is exclusively controlled by the OS.

question: i'd like to cut a 22sec and 160ms at the beginning from a dts-hd file. what cmd would i need to use to achieve this with eac3to?
i'm not sure if the delay option is the one i need to use.
You could use the edit command. Something like this:

eac3to source.dts dest.dts -edit=0:00:00,-22160ms

If that doesn't work, try "-edit=0:00:00.050,-22160ms".

Apparently, my AV receiver is highly sensitive to a TrueHD stream. Unless -keepDialnorm is used, TrueHD output is very distorted.
Does anyone know, whether the TrueHD/DialNorm issue still persists with the new Pioneer Receivers (1019 series for example)?
No idea, but you'll be able to work around the problem with the next eac3to v3.18.

this option could be in the help file and be used as 'default'.
It is not default because "simple" wav files are outdated. Some old applications need them, so that's what the eac3to option is for. But some other applications don't like them and require the new format. So whatever default option I choose, there will be problems for some people. As a result I chose the newer & better format as default.

True - but it seems that the author is no longer supporting this program...
Huh?

This time I'm going the path E-AC3 to DTS-MA using "DTS-HD Master Audio Suite v2.0 (http://store.dts.com/)". Is there any chance that you can implement a similar 'hooking' to this application as you do with SurCode?
Maybe, sooner or later...

Hmm, after audio overlaps are detected, it seems that eac3to no longer runs audio encoding concurrently. Say extract and convert two DTS-HD MA tracks to FLAC and both of them have audio overlaps, you'd expect eac3to to run the 2nd pass, fix the overlaps, and do the conversion of both at the same time, but the 2nd track's conversion begins only after the first one has finished. :|
True. eac3to does not support reading two different source files at the same time. That's why all 2nd pass operations are run separately.

[madshi]

eac3to v3.18 released

http://madshi.net/eac3to.zip

* TrueHD dialnorm removal now uses "-31db" instead of "-0db"
* TrueHD frames with -0db dialnorm are now automatically patched to -31db
* added MKV demuxing support for TrueHD tracks
* added MKV demuxing support for Blu-Ray PGS subtitle tracks
* added MKV demuxing support for (MS ACM) extensible WAV tracks
For those people who have problems with the receiver disliking TrueHD tracks demuxed by older eac3to versions, please "fix" your TrueHD tracks by simply running them through v3.18 like this:

eac3to old.thd new.thd
Please let me know, if the problem is gone. Thanks.

[jj666]

Many thanks for the new version :-)

Cheers,

-jj-

[Thunderbolt8]

yes, thanks for the update!

so just to clarify, the dialnorm 0db -> -31db change is only to fix the thd tracks for those people with receiver which don't accept the old thd tracks? theres no real difference is volume between the old and new setting and I won't have to redo all my flac tracks (or thd tracks when only used inside mkv, but not with a receiver), right?

[TinTime]

Thanks very much madshi!

In response to your question on dialnorm reporting...

I've researched this. The AC3 and DTS specs are very kinda the opposite of each other in terms of dialnorm:

AC3: The dialnorm value tells us how much headroom there is between 100% volume and speech volume.
DTS: The dialnorm value tells us how much the decoder should lower volume.

Because of that a -27dB AC3 dialnorm value equals a -4dB DTS dialnorm value. Now of course eac3to could change the display of the dialnorm values, so that AC3 and DTS match. But I'm not sure if I should do that? What do you guys think?

My feeling is that eac3to should just do as it does now and report the actual values, rather than interpreting the info for the sake of consistency.

However it really doesn't bother me one way or the other. If you get an overwhelming response to, say, change the reporting of the DTS dialnorm value then I think I could learn to live with it :)

One thought. If you changed eac3to so that it reported one of the values differently then I predict a few posts along the lines of...

"My amp/mediainfo/etc. says that dialnorm is set to -4 but eac3to reports -27. Why the discrepancy? Which is correct?"

[setarip_old]

@madshi

My remark was based on your post back in May:

http://forum.doom9.org/showpost.php?p=1282194&postcount=8835

[madshi]

so just to clarify, the dialnorm 0db -> -31db change is only to fix the thd tracks for those people with receiver which don't accept the old thd tracks? theres no real difference is volume between the old and new setting and I won't have to redo all my flac tracks (or thd tracks when only used inside mkv, but not with a receiver), right?
Right.

In response to your question on dialnorm reporting...

My feeling is that eac3to should just do as it does now and report the actual values, rather than interpreting the info for the sake of consistency.
I'm not 100% sure, but I tend to agree.

My remark was based on your post back in May:

http://forum.doom9.org/showpost.php?p=1282194&postcount=8835
Your interpretation of my post is really weird.

[nurbs]

I have a small problem with eac3to. I have a blu-ray with the following audio track

3: DTS Hi-Res, English, 7.1 channels, 24 bits, 2814kbps, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
When I try to convert it to aac with one of the following command lines

eac3to "D:\BR\LEON" 1) 3:"D:\LEON\test.aac" -core -down2 -normalize
or
eac3to "D:\BR\LEON" 1) 3:"D:\LEON\test.aac" -down2 -normalize
I get

a03 The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
a03 The libav DTS decoder doesn't decode the full DTS-HD information.
a03 Can't downmix this channel configuration ($60f).
a03 Internal error - unknown audio output format!
however first extracting the the track to .dts and then converting the dts to aac with the same parameters works fine.

[TinTime]

Are you mixing up track 3 and track 4?

[nurbs]

I did in the post, but track 3 and track 4 are the same track (same properties) with different languages. I'll correct that to avoid confusion. I wanted to convert them with the same command line which gave the same result.

[b66pak]

@madshi do you plan to add the ability to cut audio (besplit style)? also any chance to support muxing h264 to .mp4 too?
_

[madshi]

I have a small problem with eac3to. I have a blu-ray with the following audio track [...]
Hmmmm... I can reproduce that. Have put it on my to do list. But since you can work around it, I've sorted it in as "not so important".

do you plan to add the ability to cut audio (besplit style)?
Not sure what you mean. Can you clarify?

also any chance to support muxing h264 to .mp4 too?
Edit: Not possible (by using the Haali Matroska Muxer, at least), it seems. There doesn't seem to be a way to switch the Haali Matroska Muxer to MP4 from within eac3to.

[b66pak]

Not sure what you mean. Can you clarify?

let's say i have 60 min audio and i want to split in in three parts: from min 0 to 15, from 25 to 35 and from 45 to end...
_

L.E. in other words the ability to split an audio file at the time you want...also the ability to join (append) audio files would be nice...
_

[Xorp]

Thanks for the update

* added MKV demuxing support for Blu-Ray PGS subtitle tracks How are people muxing PGS subs into mkv? I heard MakeMKV can do it, but it's not freeware.

[Snowknight26]

For consistency's sake, 'khz' should be capitalized.

[Midzuki]

For consistency's sake, 'khz' should be capitalized.

kHz alright. :)

[xxx666yyy777]

eac3to v3.18 released

http://madshi.net/eac3to.zip

* TrueHD dialnorm removal now uses "-31db" instead of "-0db"
* TrueHD frames with -0db dialnorm are now automatically patched to -31db
* added MKV demuxing support for TrueHD tracks
* added MKV demuxing support for Blu-Ray PGS subtitle tracks
* added MKV demuxing support for (MS ACM) extensible WAV tracks
For those people who have problems with the receiver disliking TrueHD tracks demuxed by older eac3to versions, please "fix" your TrueHD tracks by simply running them through v3.18 like this:

eac3to old.thd new.thd
Please let me know, if the problem is gone. Thanks.

Nice! Thanks a lot!!!

Cheers.

[raquete]

this option could be in the help file and be used as 'default'.

It is not default because "simple" wav files are outdated. Some old applications need them, so that's what the eac3to option is for. But some other applications don't like them and require the new format. So whatever default option I choose, there will be problems for some people. As a result I chose the newer & better format as default.

is very clever!

eac3to v3.18 released

http://madshi.net/eac3to.zip

* TrueHD dialnorm removal now uses "-31db" instead of "-0db"
* TrueHD frames with -0db dialnorm are now automatically patched to -31db
* added MKV demuxing support for TrueHD tracks
* added MKV demuxing support for Blu-Ray PGS subtitle tracks
* added MKV demuxing support for (MS ACM) extensible WAV tracks
For those people who have problems with the receiver disliking TrueHD tracks demuxed by older eac3to versions, please "fix" your TrueHD tracks by simply running them through v3.18 like this:

eac3to old.thd new.thd
Please let me know, if the problem is gone. Thanks.

very cool, thank you so much!

[rapscallion]

I was playing around with some movie tracks yesterday and am just curious.

Original track was DD True HD 5.1 and when I select "wavs" as the output I get exactly the same wav file sizes as when I process the ac-3 core.

Apparently eac doesn't extract the lossless wavs, just the core ?? Or is their a way to extract the True HD wavs?

[Snowknight26]

WAV has a bit-depth, AC3 doesn't, so when it's decoded, it's decoded (from 64-bit floating point) to 24-bit WAV, just like the TrueHD is.

[Blue_MiSfit]

Exactly.

They both decode to 6x mono 24 bit 48 KHz WAVs of identical length, which will therefore always be the same exact size.

~MiSfit

[rapscallion]

OK, so if I'm understanding you both, supposedly I could reverse the process and create DD True HD again, or DTS MA, from those wavs ?

Or, to put it another way, any standard DD 5.1 ac-3 or DTS @ 1536 track can be made into a HD/MA track by extracting the wavs and reprocessing to HD ? If this is true, that's great !

[Snowknight26]

You can do that but there's absolutely no point in doing that. You'd be doing a lossy -> lossless conversion resulting in a bigger file than the original source. Remember, you can't make something from nothing.

[rapscallion]

But I thought the newly created/extracted wavs were lossless ?

For instance, you can extract the audio wavs from an audio cd, convert them to MP3. Then you can reverse the process back to wavs again and you're back to lossless, no ? Or is my reasoning way off ?

Also, it would appear that DTS MA has 24 bit resolution , where DD True HD does not.

Created from the same wavs :

General
Complete name : D:\Blu-ray movies\Star Trek\star.ma.dts
Format : DTS
Format/Info : Digital Theater Systems
Format profile : MA
File size : 3.56 GiB
Audio
Format : DTS
Format/Info : Digital Theater Systems
Format profile : MA
Bit rate mode : Variable
Channel(s) : 6 channels
Channel positions : Front: L C R, Surround: L R, LFE
Sampling rate : 48.0 KHz
Resolution : 24 bits

General
Complete name : D:\Blu-ray movies\Star Trek\true.track_4352.ac3
Format : AC-3
Format/Info : Audio Coding 3
Format profile : TrueHD
File size : 3.76 GiB

Audio
Format : AC-3
Format/Info : Audio Coding 3
Format profile : TrueHD
Bit rate mode : Variable
Channel(s) : 6 channels
Channel positions : Front: L C R, Surround: L R, LFE
Sampling rate : 48.0 KHz

[Snowknight26]

No, you can't get the original result back, but this isn't really relevant to eac3to so I suggest you make a new thread or search the forums (or hydrogenaudio.org).

[rapscallion]

You're right although the op was relevant to how eacto extracted wavs from an HD source.

Practically speaking I was just curious in the next steps.Thanks

[Fiffy]

I'm trying to demux the Blade Runner "International Cut" from disc 3 of the 5-disc HD-DVD set. This disc contains 3 different versions of the movie and uses seamless branching (I think it's actually the only existing HD-DVD that uses branching).

Unfortunately, after demuxing with eac3to audio and video are out of sync, regardless of whether a gap description file is used or not. Does eac3to not support seamless branching on HD-DVD?

[laserfan]

What did eac3to's log say about it?

[Fiffy]

What did eac3to's log say about it?Log from first run (without gap description file):
eac3to v3.18
command line: eac3to h: 1) 2: chapts.txt 3: br.mkv 7: br.flac -down16 -no2ndpass 10: subs.sup
------------------------------------------------------------------------------
EVO, 1 video track, 6 audio tracks, 6 subtitle tracks, 1:57:29
1: Joined EVO file
2: Chapters, 35 chapters
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: E-AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB, -21ms
5: E-AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -21ms
6: E-AC3 Surround, French, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -21ms
7: E-AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB, -18ms
8: E-AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
9: E-AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -7ms
10: Subtitle (DVD), English
11: Subtitle (DVD), French
12: Subtitle (DVD), Spanish
13: Subtitle (DVD)
14: Subtitle (DVD)
15: Subtitle (DVD)
Creating file "chapts.txt"...
[a07] The Nero decoder doesn't seem to work, will use libav instead.
[v03] Extracting video track number 3...
[a07] Extracting audio track number 7...
[v03] Writing new framerate "24fps /1.001" to bitstream.
[a07] Removing E-AC3 dialog normalization...
[v03] Removing VC-1 pulldown...
[a07] Decoding with libav/ffmpeg...
[v03] Muxing video to Matroska...
[a07] Remapping channels...
[a07] Applying RAW/PCM delay...
[a07] Reducing depth from 64 to 16 bits...
[a07] Encoding FLAC with libFlac...
[a07] Creating file "br.flac"...
[s10] Extracting subtitle track number 10...
[s10] Creating file "subs.sup"...
[a07] Clipping detected, a 2nd pass will be necessary. <WARNING>
[v03] Video overlaps for 705 frames at playtime 1:40:02. <WARNING>
[v03] Video overlaps for 1 frames at playtime 1:40:32. <WARNING>
[v03] Video overlaps for 7 frames at playtime 1:52:24. <WARNING>
[v03] Video overlaps for 6 frames at playtime 1:52:25. <WARNING>
[v03] The MKV file was created without making use of the gap/overlap information. <WARNING>
[v03] Please check whether audio is in sync. If it is in sync everything is fine. <WARNING>
[v03] Otherwise ask eac3to to repeat the muxing. It will then automatically make <WARNING>
[v03] use of the detailed gap/overlap information. <WARNING>
[a07] Audio overlaps for 4192ms at playtime 1:39:58. <WARNING>
The 2nd pass is skipped by request. <WARNING>
Added fps value (24 /1.001) to MKV header.
Video track 3 contains 169079 frames.
Subtitle track 10 contains 875 captions.
eac3to processing took 37 minutes, 28 seconds.
Done.

Log from second run (with gap description file generated by first run):

eac3to v3.18
command line: eac3to h: 1) 2: chapts.txt 3: br.mkv 7: br.flac -down16 -no2ndpass 10: subs.sup
------------------------------------------------------------------------------
EVO, 1 video track, 6 audio tracks, 6 subtitle tracks, 1:57:29
1: Joined EVO file
2: Chapters, 35 chapters
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: E-AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB, -21ms
5: E-AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -21ms
6: E-AC3 Surround, French, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -21ms
7: E-AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB, -18ms
8: E-AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
9: E-AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -7ms
10: Subtitle (DVD), English
11: Subtitle (DVD), French
12: Subtitle (DVD), Spanish
13: Subtitle (DVD)
14: Subtitle (DVD)
15: Subtitle (DVD)
Creating file "chapts.txt"...
[v03] Video gap description file detected, will be used for muxing...
[a07] The Nero decoder doesn't seem to work, will use libav instead.
[a07] Extracting audio track number 7...
[v03] Extracting video track number 3...
[a07] Removing E-AC3 dialog normalization...
[v03] Writing new framerate "24fps /1.001" to bitstream.
[a07] Decoding with libav/ffmpeg...
[v03] Removing VC-1 pulldown...
[a07] Remapping channels...
[v03] Muxing video to Matroska...
[a07] Applying RAW/PCM delay...
[a07] Reducing depth from 64 to 16 bits...
[a07] Encoding FLAC with libFlac...
[a07] Creating file "br.flac"...
[s10] Extracting subtitle track number 10...
[s10] Creating file "subs.sup"...
[a07] Clipping detected, a 2nd pass will be necessary. <WARNING>
[a07] Audio overlaps for 4192ms at playtime 1:39:58. <WARNING>
The 2nd pass is skipped by request. <WARNING>
Added fps value (24 /1.001) to MKV header.
Video track 3 contains 169079 frames.
Subtitle track 10 contains 875 captions.
eac3to processing took 37 minutes, 10 seconds.
Done.

[laserfan]

And why do you have -no2ndpass in your eac3to command line? :confused:

[Fiffy]

And why do you have -no2ndpass in your eac3to command line? :confused:Because the clipping is insignificant and does not warrant a second audio transcoding at lower volume. But this has nothing to do with the topic at hand.

[laserfan]

Ok, whatever you say, but the 2nd pass is where the "magic" of gap repair is applied. Try it.

[Fiffy]

Ok, whatever you say, but the 2nd pass is where the "magic" of gap repair is applied. Try it.No, it is not. The 2nd pass just transcodes the audio once more with different parameters to eliminate the clipping. It has nothing to do with gap repair. The asynchronicity also occurs if the option is not used or the audio is just extracted without transcoding.

[zeropc]

eac3to v3.18 released


[code]* TrueHD dialnorm removal now uses "-31db" instead of "-0db"
* TrueHD frames with -0db dialnorm are now automatically patched to -31db

what does that "-31db" instead of "-0db" mean for thd to wavs?

i use wavs to create a new dts-hd ma and i wonder if this new option would change the audio level or does it not matter at all?

[slk486]

Hi madshi

The Denon receivers also had problems with a -0db dialnorm, but ran 4 TrueHD tracks through the new eac3to version, and can happily announce, that it fixed the problem with low (-31db) audio and shifts in dialnorm.

So, thanks!!!

[madshi]

let's say i have 60 min audio and i want to split in in three parts: from min 0 to 15, from 25 to 35 and from 45 to end...

L.E. in other words the ability to split an audio file at the time you want...also the ability to join (append) audio files would be nice...
Joining works for all audio types which have no separate header at the beginning (or end) of the file, like e.g. AC3, E-AC3, DTS, MP2, PCM etc.

Full splitting/joining support is not planned right now. I've so many other things to do...

For consistency's sake, 'khz' should be capitalized.
kHz alright. :)
Ok, will change that to kHz.

I'm trying to demux the Blade Runner "International Cut" from disc 3 of the 5-disc HD-DVD set. This disc contains 3 different versions of the movie and uses seamless branching (I think it's actually the only existing HD-DVD that uses branching).

Unfortunately, after demuxing with eac3to audio and video are out of sync, regardless of whether a gap description file is used or not. Does eac3to not support seamless branching on HD-DVD?
I've not seen this disc and since it's the only such disc, I'm not planning to do anything about it. Not sure why there should be problems, though. In theory it should work. Maybe the audio can be made sync with a fixed delay? Or does it go out of sync gradually throughout the runtime of the movie?

what does that "-31db" instead of "-0db" mean for thd to wavs?

i use wavs to create a new dts-hd ma and i wonder if this new option would change the audio level or does it not matter at all?
Don't worry. The fix I made is only to satisfy some receivers which don't fully follow the Dolby specifications. It's only important for people who want to bitstream TrueHD to their receiver via HDMI. If you convert to WAV(s) on your PC, there's nothing to worry about. The older versions did that correctly.

The Denon receivers also had problems with a -0db dialnorm, but ran 4 TrueHD tracks through the new eac3to version, and can happily announce, that it fixed the problem with low (-31db) audio and shifts in dialnorm.
Glad to hear that - thanks for the feedback! :)

[Fiffy]

I've not seen this disc and since it's the only such disc, I'm not planning to do anything about it.That's understandable. The tool has worked great for all my other HD DVDs, so I'm not complaining. ;)
Not sure why there should be problems, though. In theory it should work. Maybe the audio can be made sync with a fixed delay? Or does it go out of sync gradually throughout the runtime of the movie?It starts out fine and then gets more and more out of sync. I think it may get worse after every branch point, but I'd have to check more closely to be sure.

[zeropc]

Don't worry. The fix I made is only to satisfy some receivers which don't fully follow the Dolby specifications. It's only important for people who want to bitstream TrueHD to their receiver via HDMI. If you convert to WAV(s) on your PC, there's nothing to worry about. The older versions did that correctly.

thanks for the clarification :)

[utenteanonimo64]

Full splitting/joining support is not planned right now. I've so many other things to do...


Hi Madshi thanks for this new version.
In case you choose your development priorities on the basis of how many people want a certain feature, here is another user who would like to see an easy way to split audio by chapter!

[STaRGaZeR]

eac3to v3.18 released

http://madshi.net/eac3to.zip

* TrueHD dialnorm removal now uses "-31db" instead of "-0db"
* TrueHD frames with -0db dialnorm are now automatically patched to -31db
* added MKV demuxing support for TrueHD tracks
* added MKV demuxing support for Blu-Ray PGS subtitle tracks
* added MKV demuxing support for (MS ACM) extensible WAV tracks
For those people who have problems with the receiver disliking TrueHD tracks demuxed by older eac3to versions, please "fix" your TrueHD tracks by simply running them through v3.18 like this:

eac3to old.thd new.thd
Please let me know, if the problem is gone. Thanks.

Guys, I've always demuxed TrueHD+AC3 tracks with -keepDialnorm and then muxed it to Matroska, I understand I won't have any receiver issues with this MKVs if I use for example ffdshow's bitstream ability?

[madshi]

Guys, I've always demuxed TrueHD+AC3 tracks with -keepDialnorm and then muxed it to Matroska, I understand I won't have any receiver issues with this MKVs if I use for example ffdshow's bitstream ability?
There should be no problems with your MKVs, except of course, that your receiver will apply dialnorm (audio track volume is lowered by your receiver by using digital processing).

[STaRGaZeR]

There should be no problems with your MKVs, except of course, that your receiver will apply dialnorm (audio track volume is lowered by your receiver by using digital processing).

Of course, but that's intented, I want the THD tracks untouched, and with -keepDialnorm that's exactly what I've achieved, right? But after reading all the problems with eac3to remuxed tracks I feared the worst. Dialog normalization can be disabled in the receiver if needed/wanted.

[madshi]

Dialog normalization can be disabled in the receiver if needed/wanted.
No, normally it can't. You can disable DRC, but most (all?) receivers don't allow you to disable dialnorm, at least AFAIK. A while ago the Dolby license even explicitly forbid receiver manufacturers to offer an option to disable dialnorm. I know that Dolby considered changing their license terms (based on AVSForum feedback), but I don't know if they did in the end.

[b66pak]

@madshi could you add downmix to stereo (beside the current downmix to dpl2) to your tool...dpl2 is not at all suited for headphones...
_

[hewwra]

I'm trying to remux some blurays using clown bd and it works most of the time however I have problem with 2 movies, kill bill 2 and sin city.
There are no errors in clown bd however there are 2 warnings in the logs.
Sin City: [a03] This track begins with a non-major frame. <WARNING>
Kill Bill 2: [a03] Caution: The WAV file is bigger than 4GB. <WARNING> [a03] Some WAV readers might not be able to handle this file correctly. <WARNING>
I'm using MPC HC with internal ffmpeg filter to play blurays and both movies work just fine when they are in full bluray structure on my hd but when I try to play them after I've used clown bd there is no audio or video and the screen is just black and the time stays at "00:00:00".
So does anyone have a clue where the remux goes wrong?

[Bilkoff]

I have some problems with "Halo Legends" blu-ray, with some audios. MediaInfo says they have 6 channels, but eac3to extracts them like stereo audios, and the duration is 3x.
The .mpls has different files, some warning of small overlaps are in the log. This happens with gaps files and without. There is a real stereo audio extracted correctly.
Can this be corrected remapping the channels audio? Witch command should I use, -0,1,2,3,4,5?

[Killroy™]

I have some problems with "Halo Legends" blu-ray, with some audios. MediaInfo says they have 6 channels, but eac3to extracts them like stereo audios, and the duration is 3x.
The .mpls has different files, some warning of small overlaps are in the log. This happens with gaps files and without. There is a real stereo audio extracted correctly.
Can this be corrected remapping the channels audio? Witch command should I use, -0,1,2,3,4,5?

Halo legends has a weird mix of 5.1 and 2.0 in the playlist that will screw up the remux. You need to delete that one track or find a replacement for it.

[mikelebron]

Maybe you guys will know the answer.... right now I just upgraded my TMT 3 to 3.0.170 and as may be aware that it breaks eac3tos ability to use the DTS filter... Now I have the original files from my original install of TMT 3 which work. Can they co-exist on the same machine if I register them in a different directory?

[lzep]

Hi,
I've been using this just fine for about a year or so now for converting dts files in an mkv container (from mkvmerge) to ac3 so I can play them back on my WD media player.

I'm having this weird problem now where the file converts from dts to ac3 and right after it's finished, my computer crashes and reboots with a "windows has recovered from a serious error" message. So the conversion is working but it's crashing my comp every time. Is there a log file I can look at or drivers/software I should reinstall and/or update? Anyone have any ideas?

Thanks!

[setarip_old]

Hi!

Perform a "System Restore" back to the last time things were working properly...

[VAMET]

Dear Friends

First of all, hello to everybody.

I would like to know, what such information means during eac3to process:

Case 1:
[a07] Audio overlaps for 25ms at playtime 0:11:08. <WARNING>

Case 2:
[a08] A remaining delay of -9ms could not be fixed.

Case 3:
[a05] Reading RAW/PCM...
[a05] Swapping endian...
[a05] Remapping channels...
[a05] The original audio track has a constant bit depth of 16 bits.

Case 4:
[v02] [1:13:34] The source file seems to be damaged (sync byte missing). <WARNING>
[a03] [1:13:33] The source file seems to be damaged (sync byte missing). <WARNING>

My question is this, beacuse the original disc BD 1:1 has got errors, scratches or something has gone wrong, while copying to HDD via AnyDVD HD?
Which of above cases are the disc faults and should I copy them once more to get rid off?

I would be glad for reply.
Thank you in advance.
Best regards.

Sincerely

[nautilus7]

Case 1:
In seamless braching BDs main movie is split among several .m2ts files. This looks like this:
http://img361.imageshack.us/img361/359/seamlessbranchingaudioo.png

Case 2:
Audio consists of frames of fixed duration (duration depends on audio format). Delay adjustment is done by adding or removing a number of audio frames. If you multiply (audio frame duration) x (number of frames) you don't always result (in fact you never result) with a value that matches the exact delay needed.

Case 3:
Some modifications are done when converting from one format to another. Nothing to worry about.

Case 4:
What is says. It seems like your data are corrupted. Try re-ripping the disc. This is the only thing you have to worry about. If you still get the same message after ripping the disc again, see if you get any errors (corrupted picture, audio/video de-sync) during playback If you can't find a flaw at these points, then everything should be ok.

[VAMET]

Dear nautilus7

Thank you very much for your reply and great explanation.

In Case 3, I have RAW/PCM it's in Casino Royale, you said:
Some modifications are done when converting from one format to another..
I would like to have no conversion. I used .pcm, should I use another format (extension) in eac3to process, to have unconverted RAW/PCM track from BDMV folder structure? Which one should I enter in eac3to process.

Dolby TrueHD = thd+ac3
DTS HD-MA = dtsma
DTS HD High Resolution = dtshr
Dolby Digital = ac3
DTS = dts
RAW/PCM = ??? I used .pcm and conversion is done, I want uncoverted

I thought that for RAW/PCM I should use .pcm.

Best regards.

Sincerely

[tebasuna51]

I thought that for RAW/PCM I should use .pcm.

Nope, .pcm file is audio uncompressed without headers.
Is better use .wav or .w64 (if is >4GB) with the same uncompressed audio data + header.

edit: .wav if you want use after MkvMerge, .w64 if TsMuxer.

[spork985]

http://img63.imageshack.us/img63/3817/71004044.png

Can someone help me with this? I have installed latest versions of eac3to, mkvtoolnix, haali, arcsoft dts decoder, and c++ redistributable. Operating system is windows xp professional sp3 x86. Am I missing any softwares? I have 3 different blurays, all of them do the same thing. I have used eas3to with around 30ish blurays on my laptop (other computer) and never have had a problem. I'm knowledgeable with video and containers and how stuff works, I just can't figure out why eac3to is crashing. Thanks!

It is crashing regardless of the file... Whether it be
eac3to "C:\file.ac3"
eac3to "C:\file.ts"
eac3to "C:\file.m2ts"

[Inspector.Gadget]

You apparently know that nobody can help you with that source, hence obliterating the path name. Yet you ask for help anyway. Please don't ask other users to violate Rule 6.

[spork985]

You apparently know that nobody can help you with that source, hence obliterating the path name. Yet you ask for help anyway. Please don't ask other users to violate Rule 6.

It happens with ANY FILE, that is just the example I used. I don't like the path name. I own AnyDVD and I own the Bluray disc and I copied it to my hard drive as the one legal backup allowed by US law. It happens wit ANY FILE not just this particular one, I need help on the error not the content.

Perhaps these will help.

http://thumbnails16.imagebam.com/7337/04ccf373362574.gif (http://www.imagebam.com/image/04ccf373362574) http://thumbnails3.imagebam.com/7337/2574ab73362579.gif (http://www.imagebam.com/image/2574ab73362579)

[madshi]

@madshi could you add downmix to stereo (beside the current downmix to dpl2) to your tool...dpl2 is not at all suited for headphones...
_
Is there a standard for mixing 5.1 to stereo? If so, what is the downmixing matrix?

I'm trying to remux some blurays using clown bd and it works most of the time however I have problem with 2 movies, kill bill 2 and sin city.
There are no errors in clown bd however there are 2 warnings in the logs.
Sin City: [a03] This track begins with a non-major frame. <WARNING>
Kill Bill 2: [a03] Caution: The WAV file is bigger than 4GB. <WARNING> [a03] Some WAV readers might not be able to handle this file correctly. <WARNING>
I'm using MPC HC with internal ffmpeg filter to play blurays and both movies work just fine when they are in full bluray structure on my hd but when I try to play them after I've used clown bd there is no audio or video and the screen is just black and the time stays at "00:00:00".
So does anyone have a clue where the remux goes wrong?
I don't know what clown bd does. Does it put all audio tracks into the MKV? In which format? Maybe you should ask the clown bd author for help? My first guess would be that either clown bd does something wrong, or that your DirectShow filter chain needs some tweaking.

Hi,
I've been using this just fine for about a year or so now for converting dts files in an mkv container (from mkvmerge) to ac3 so I can play them back on my WD media player.

I'm having this weird problem now where the file converts from dts to ac3 and right after it's finished, my computer crashes and reboots with a "windows has recovered from a serious error" message. So the conversion is working but it's crashing my comp every time.
That's weird. Try deleting the eac3to folder and redownloading it. Try deleting the two WAV files from the eac3to folder. Does that help? If not, you may have to follow setarip_old's suggestion or even reinstall the OS. I don't think this is an eac3to issue.

I would like to have no conversion. I used .pcm, should I use another format (extension) in eac3to process, to have unconverted RAW/PCM track from BDMV folder structure?
What is your final aim? You can't play .pcm files.

It happens with ANY FILE [...].

Perhaps these will help.
Looks like the FLAC decoders crashes. Don't be confused. When eac3to tries to detect the source's file format, it calls the FLAC decoder to check whether it's a FLAC file. Even if it's not. Well, anyway. My first suggestion would be to delete the eac3to folder and redownload it. My best guess is that the libFlac.dll file is corrupted on your PC.

[b66pak]

Is there a standard for mixing 5.1 to stereo? If so, what is the downmixing matrix?

avisynth script for 5.1 to Stereo NO LFE:
function Dmix5.1Stereo(clip a) # 5.1 Channels L,R,C,NO-LFE,SL,SR -> Stereo
{
flr = GetChannel(a, 1, 2)
fcc = GetChannel(a, 3, 3)
lrc = MixAudio(flr, fcc, 0.3694, 0.2612)
blr = GetChannel(a, 4, 5)
return MixAudio(lrc, blr, 1.0, 0.3694)
}

sox remix line for 5.1 to Stereo NO LFE:
remix -m 1v0.3694,3v0.2612,5v0.3694 2v0.3694,3v0.2612,6v0.3694

avisynth script for 5.1 to Stereo with LFE:
function Dmix5.1Stereo(clip a) # 5.1 Channels L,R,C,LFE,SL,SR -> Stereo
{
flr = GetChannel(a, 1, 2)
fcc = GetChannel(a, 3)
lfe = GetChannel(a, 4)
lfc = MixAudio(fcc, lfe, 0.2071, 0.2071)
mix = MergeChannels(lfc, lfc)
lrc = MixAudio(flr, mix, 0.2929, 1.0)
blr = GetChannel(a, 5, 6)
return MixAudio(lrc, blr, 1.0, 0.2929)
}

sox remix line for 5.1 to Stereo with LFE:
remix -m 1v0.2929,3v0.2071,4v0.2071,5v0.2929 2v0.2929,3v0.2071,4v0.2071,6v0.2929
_

[spork985]

Looks like the FLAC decoders crashes. Don't be confused. When eac3to tries to detect the source's file format, it calls the FLAC decoder to check whether it's a FLAC file. Even if it's not. Well, anyway. My first suggestion would be to delete the eac3to folder and redownload it. My best guess is that the libFlac.dll file is corrupted on your PC.
I redownloaded and tried it and it does the same thing. I hashed all the files with md5 and they match the hashes of the fiels on the computer that does work.

[madshi]

avisynth script for 5.1 to Stereo NO LFE:
sox remix line for 5.1 to Stereo NO LFE:
avisynth script for 5.1 to Stereo with LFE:
sox remix line for 5.1 to Stereo with LFE:
avisynth and sox seem to use the same matrixes? But is this a standard matrix? Or did just somebody say "this sounds ok" and everyone else is copying the parameters?

I redownloaded and tried it and it does the same thing. I hashed all the files with md5 and they match the hashes of the fiels on the computer that does work.
Hmmmm... Maybe your PC doesn't like the libFlac.dll version for whatever reason. You could try replacing the libFlac.dll with an older version. eac3to ships with libFlac 1.2.1. You can get older versions here in the "flac-win" section:

http://sourceforge.net/projects/flac/files/

You need to download the "flac-x.x.x-devel-win.zip" file. That's the only one where the libFlac.dll file is in.

[tebasuna51]

avisynth and sox seem to use the same matrixes? But is this a standard matrix? Or did just somebody say "this sounds ok" and everyone else is copying the parameters?

Is the standard matrix:

fl' = FL + sqr(2)/2 x FC + BL
fr' = FR + 0.7071 x FC + BR

now we have the same acustic power for FC when add the part in FL and FR.

If all channels have a peak at same time we can have overflow reaching values until 2.7071, then we need divide the coefficients by 2.7071:
1/2.7071 = 0.3694
0.7071/2.7071 = 0.2612

fl' = 0.3694 x FL + 0.2612 x FC + 0.3694 x BL
fr' = 0.3694 x FR + 0.2612 x FC + 0.3694 x BR

Now the max value is:
0.3694 + 0.2612 + 0.3694 = 1
After the downmix is recommended a Normalize to avoid low volume.

[madshi]

Is the standard matrix:

fl' = FL + sqr(2)/2 x FC + BL
fr' = FR + 0.7071 x FC + BR

now we have the same acustic power for FC when add the part in FL and FR.
Ok, thanks, that makes a lot of sense.

Now the big question is: How should I make this available in eac3to? Right now "-down2" invokes Dolby ProLogic II. How should I name the options for Dolby ProLogic II and for this other downmixing matrix? Maybe I should use "-down2" for the normal matrix and "-downDpl2" for Dolby ProLogic II? Any other suggestions?

If all channels have a peak at same time we can have overflow reaching values until 2.7071, then we need divide the coefficients by 2.7071:
1/2.7071 = 0.3694
0.7071/2.7071 = 0.2612

fl' = 0.3694 x FL + 0.2612 x FC + 0.3694 x BL
fr' = 0.3694 x FR + 0.2612 x FC + 0.3694 x BR

Now the max value is:
0.3694 + 0.2612 + 0.3694 = 1
After the downmix is recommended a Normalize to avoid low volume.
I'll simply skip the 2.7071 division. If there's overflow/clipping, eac3to will detect that automatically and fix it in a 2nd run, resulting in normalized audio.

[jpsdr]

Hello.
Question about use. If i want to extract the raw video track from an .mts/.m2ts file, to get an untouched raw video data stream, to be able to use it later in authoring SW like Scenarist.
Is the following enough : eac3to file.mts -demux
or, is there others options to add ? (Maybe to extract only video if possible).
If i want the results file(s) to be put on another place than the source file, what the command line should be ?
Thanks.

[tebasuna51]

Now the big question is: How should I make this available in eac3to? Right now "-down2" invokes Dolby ProLogic II. How should I name the options for Dolby ProLogic II and for this other downmixing matrix? Maybe I should use "-down2" for the normal matrix and "-downDpl2" for Dolby ProLogic II? Any other suggestions?
To be backward compatible I suggest use a new parameter "-downstereo" for the new mix and let "-down2" with the Dolby ProLogic II behaviour.

I'll simply skip the 2.7071 division. If there's overflow/clipping, eac3to will detect that automatically and fix it in a 2nd run, resulting in normalized audio.
You are right, maybe you can use the same procedure for dpl2 with:

fl' = FL + 0.7071 x FC + 0.866 x BL + 0.5 x BR
fr' = FR + 0.7071 x FC - 0.5 x BL - 0.866 x BR

[Midzuki]

Let's say, I create an MKV which contains DTS audio, encoded from 16-bit .WAVs. When I demux the MKV with eac3to, this always "patches" the demuxed DTS to "24-bits". What is the point of this behavior? :confused: :confused: :confused:

[SomeJoe]

Let's say, I create an MKV which contains DTS audio, encoded from 16-bit .WAVs. When I demux the MKV with eac3to, this always "patches" the demuxed DTS to "24-bits". What is the point of this behavior? :confused: :confused: :confused:

DTS and Dolby Digital store audio information in the frequency domain. As such, they do not have a "bit depth". When PCM audio (time-domain) is recreated from them during decoding, the audio may in fact decode to values that have data into the 8 additional bits. Those bits may not be accurate with respect to the source, but are accurate with respect to the frequency-domain representation in the compressed audio.

Remember that a 16-bit PCM source is itself an approximation of an analog waveform, and as such is inaccurate because of the 16-bit quantization.

[tebasuna51]

Let's say, I create an MKV which contains DTS audio, encoded from 16-bit .WAVs. When I demux the MKV with eac3to, this always "patches" the demuxed DTS to "24-bits". What is the point of this behavior? :confused: :confused: :confused:

This is because standard DTS isn't lossless. The internal precission of the samples, in frequency domain, is equivalent (not equal) to 24 bits in time domain (no matter what is the source bitdepth).

Then decode to 24 bits is always better than decode to 16 bits, BTW if you want only 16 bits you can add -down16.

edit: I don't see the SomeJoe post with, more or less, the same thing.

[b66pak]

@madshi in my opinion we should have:


-down2stereo
-down2dpl
-down2dpl2
-lfe


the present "-down2" will be equal to "-down2dpl2" for backward compatibility...

also for everybody to be happy i think you should also give full access to the mixer with something like:


-down2custom 1v0.3694,3v0.2612,5v0.3694 2v0.3694,3v0.2612,6v0.3694


this will also solve the present eac3to inability to downmix anything but 5.1...

for example a 3ch (FL,FR,FC) to stereo will luke like:


-down2custom 1v0.5858,3v0.4142 2v0.5858,3v0.4142

_

[tebasuna51]

...
also for everybody to be happy i think you should also give full access to the mixer with something like:


-down2custom 1v0.3694,3v0.2612,5v0.3694 2v0.3694,3v0.2612,6v0.3694

...

That's could be interesting.

A 'Quadro' audio (2/2.0 : FL,FR,BL,BR) can be downmixed to Dpl2 with:

-down2custom 1v1.0000,3v0.8660,4v0.5000 2v1.000,3v-0.5000,4v-0.8660

(the sample is for remember accept the '-' sign)

[spork985]

Hmmmm... Maybe your PC doesn't like the libFlac.dll version for whatever reason. You could try replacing the libFlac.dll with an older version. eac3to ships with libFlac 1.2.1. You can get older versions here in the "flac-win" section:

http://sourceforge.net/projects/flac/files/

You need to download the "flac-x.x.x-devel-win.zip" file. That's the only one where the libFlac.dll file is in.

FLAC 1.1.4 worked, flac 1.2.0 would not. Thank you very much, I have been going crazy about this for a few days now!

[Midzuki]

tebasuna51 wrote:

BTW if you want only 16 bits you can add -down16.

Doesn't work. MediaInfo says the demuxed DTS still "is" 24-bits. Fortunately, there are workarounds — namely, TSMuxer and AVI-Mux GUI.

SomeJoe wrote:

Remember that a 16-bit PCM source is itself an approximation of an analog waveform, and as such is inaccurate because of the 16-bit quantization.

Even a 64-bit @ 768kHz PCM file will be "just an approximation". Remember, there is no such thing as a "perfect" Analog-To-Digital conversion. :)

Anyway, thanks for the accurate answers.
Actually, I was expecting to receive an answer from madshi himself, but it seems he really is "much busier" than I thought. ;)

[Snowknight26]

It doesn't matter what MediaInfo says about lossy tracks regarding bitness. It's not 16-bit and it's not 24-bit.

[raquete]

It doesn't matter what MediaInfo says about lossy tracks regarding bitness. It's not 16-bit and it's not 24-bit.

all right, can be but...
i was sending pm to Midzuki telling that i have few dvds where olaying DTS from video_ts folder my receiver/decoder inform in the display: 24b..this info stay there until the dvd end!
others dvds don't inform and i don't know how to encode in 24b.
i need lessons and explanations about this informations. :confused: how and what program can be used to do that?!?
Porcupine Tree is one of my DVDs that give 24b info, i have others!

@ Midzuki
i could not resisit to post here too but i'm waiting yours infos too. :)

[Midzuki]

It doesn't matter what MediaInfo says about lossy tracks regarding bitness. It's not 16-bit and it's not 24-bit.

I already knew that, since DTS "Digital Surround" is a lossy format. BUT, a binary file-comparison between a 16-bit DTS and a "patched-to-24-bits" DTS shows they are different beasts (on a "frame-by-frame" level, I think).

[Midzuki]

4 raquete: lossy "24-bit DTS" requires 24-bit sources
@ 96kHz or higher.

Also, take a look at your PM inbox. :rolleyes:

[raquete]

4 raquete: lossy "24-bit DTS" requires 24-bit sources
@ 96kHz or higher.

Also, take a look at your PM inbox. :rolleyes:
great, i got the lesson Mid.

thanks so much! :)

[tebasuna51]

One more time, AC3/DTS standard (lossy) don't have bitdepth. The parameter quality for this files is the bitrate, forget the bitdepth.

Only DTS have a value in the header than say the bitdepth of the source wav files used to create the dts. And Surcode put always 24 bits also when source wavs is 16 bits, then is a useless info. I say to madshi don't put this info (sometimes wrong) with dts files because confuse the users, like you can see.

AC3/DTS files must be decoded to, at least, 24 bit to obtain the best aproach to the source, no matter the original bitdepth.

[raquete]

ok tebasuna,
i was curious based on some informations from my decoder like i posted here:
http://forum.doom9.org/showthread.php?p=1386384#post1386384

to tell the true i was not only confused, i was really lost with that infos. :)

thanks so much!

[Midzuki]

Only DTS have a value in the header than say the bitdepth of the source wav files used to create the dts. And Surcode put always 24 bits also when source wavs is 16 bits, then is a useless info.

AC3/DTS files must be decoded to, at least, 24 bit to obtain the best aproach to the source, no matter the original bitdepth.

So, let's just hope there don't exist too many
" 'dumb' DTS decoders ". :)

[madshi]

To be backward compatible I suggest use a new parameter "-downstereo" for the new mix and let "-down2" with the Dolby ProLogic II behaviour.
in my opinion we should have:


-down2stereo
-down2dpl
-down2dpl2
-lfe


the present "-down2" will be equal to "-down2dpl2" for backward compatibility...
Hmmmmm... I think I like b66pak's idea of appending a qualifier to "-down2", however, I don't really like the term "stereo" here, especially because "-down2stereo" is kind of double information. Do you guys have a good idea how to term the "straight" downmixing matrix? Maybe "-down2straight"? Or "-down2simple"? Or something else?

also for everybody to be happy i think you should also give full access to the mixer
That's could be interesting.
I'll think about that. Don't know right now how difficult it would be to implement...

You are right, maybe you can use the same procedure for dpl2 with:

fl' = FL + 0.7071 x FC + 0.866 x BL + 0.5 x BR
fr' = FR + 0.7071 x FC - 0.5 x BL - 0.866 x BR
Yeah, good thinking, will do that!

If i want to extract the raw video track from an .mts/.m2ts file, to get an untouched raw video data stream, to be able to use it later in authoring SW like Scenarist.
Is the following enough : eac3to file.mts -demux
or, is there others options to add ? (Maybe to extract only video if possible).
If i want the results file(s) to be put on another place than the source file, what the command line should be ?
"-demux" is ok, but it will demux all video, audio and subtitle tracks. You can do "eac3to file.mts c:\whatever\video.h264", if the video track is h264 (use .mpeg2 or .vc1 for other video types). That will demux only the video track, and this way you can also choose the target file path.

Let's say, I create an MKV which contains DTS audio, encoded from 16-bit .WAVs. When I demux the MKV with eac3to, this always "patches" the demuxed DTS to "24-bits". What is the point of this behavior? :confused: :confused: :confused:
Two facts:

(1) Each DTS frame contains a header field which informs us about whether the original source was 16bit or 24bit. In theory this header field is only for our information and should not be used during decoding (except for DTS-HD MA where due to being lossless the final output should have the same bitdepth as the original source). Because this header field is only for our information, and should not really be used during decoding, it should not *harm* to modify it (again: the exception is DTS-HD MA).

(2) There are some decoders which always limit their output to the bitdepth of the original audio source - even for lossy DTS. That means, if the header field says that the original audio data was only 16bit, then the decoder only outputs 16bit, too. This is *BAD* because lossy formats decode always to higher than 16bit (doesn't matter which bitdepth the original source had), and if the decoder downconverts that to 16bit, we lose quality.

Now combine the two facts above and you will see that what eac3to does, shouldn't ever harm, but it will sometimes (with stupid decoders) help to get higher quality playback.

Got it?

FLAC 1.1.4 worked, flac 1.2.0 would not. Thank you very much, I have been going crazy about this for a few days now!
<shameless self promotion> I wish all software had such exact crash reporting as eac3to has. I think with most other software you'd be stuck and never know why it didn't work on your PC. </shameless self promotion>

[Midzuki]

Originally posted by madshi
Got it?

Yes, sir:

1) there is a certain DTS encoder that is not, let's say,
"as smart as it should be" :D ;

2) In the "dictionary of advertising and marketing" :) ,
"24-bit DTS" is a misnomer for *96kHz* :devil:

:thanks:

[madshi]

there is a certain DTS-HD encoder that is not, let's say, "as smart as it should be" :D ;
Not sure what you're saying here...

In the "dictionary of advertising and marketing" :) ,
"24-bit DTS" is a misnomer for *96kHz* :devil:
From a technical point of view 24bit and 96kHz have *nothing* to do with each other. Totally separate things. Don't let yourself be confused by the DTS marketing department...

[tebasuna51]

Hmmmmm... I think I like b66pak's idea of appending a qualifier to "-down2", however, I don't really like the term "stereo" here, especially because "-down2stereo" is kind of double information. Do you guys have a good idea how to term the "straight" downmixing matrix? Maybe "-down2straight"? Or "-down2simple"? Or something else?

No problem, my vote for "-down2simple", "straight" is more dificult to spell, for no english people, and can cause typos.

Related with this kind of downmix uses, maybe you can add support for Lame.exe encoder (like NeroAacEnc or with dll).
Many people want mp3 output for compatibility issues (device or container avi related).

We can use parameters like -192 to CBR encode (Lame -b 192). Valid values:
[32,40,48,56,64,80,]96,112,128,160,192,224,256,320
Maybe we can forget low values, not recommended at all.

For VBR we can use the same parameter than for NeroAacEnc, with this conversion:
-quality=1.0 -> -b 320 (high possible with Lame
-quality=0.9.. -> -V 0
-quality=0.8.. -> -V 1
-quality=0.7.. -> -V 2
-quality=0.6.. -> -V 3
-quality=0.5.. -> -V 4 (default VBR)
-quality=0.4.. -> -V 5
-quality=0.3.. -> -V 6
-quality=0.2.. -> -V 7
-quality=0.1.. -> -V 8
-quality=0.0.. -> -V 9
To avoid the strange behaviour of Lame parameter (high V less quality)

I know this can be do with stdout.wav | Lame ... (like my GUI UsEac3to do) but other GUI's (Clown_BD, HdBrStreamExtractor-MeGUI, ...) can't convert to mp3.

[Midzuki]

Originally Posted by Midzuki
there is a certain DTS-HD encoder that is not, let's say, "as smart as it should be" :D
Not sure what you're saying here...

tebasuna51 had written:

And Surcode put always 24 bits also when source wavs is 16 bits,

Conclusion: I was not talking about Surcode.

[yesgrey]

Hmmmmm... I think I like b66pak's idea of appending a qualifier to "-down2", however, I don't really like the term "stereo" here, especially because "-down2stereo" is kind of double information. Do you guys have a good idea how to term the "straight" downmixing matrix? Maybe "-down2straight"? Or "-down2simple"? Or something else?

Here is my suggestion...
Changing the "2" to "To", like this:

-downTo6 (previously down6)
-downTo2
-downTodpl
-downTodpl2
-downTo "some specification to decide" (custom mix)

This would have the benefit of not using the same command for downmixing channels as for reducing the bit depth (down16...23).
Another idea is using "mixTo" instead of "downTo".

[yesgrey]

also for everybody to be happy i think you should also give full access to the mixer with something like:

-down2custom 1v0.3694,3v0.2612,5v0.3694 2v0.3694,3v0.2612,6v0.3694


I also like the full access to the mixer idea, and I would like if you would also add another option in case you decide to go for it.
Using the custom mix only for scanning the file and then output the peak value in dB.
Why I need this?
I always use the hardware mixer of my soundcard to mix from 5.1 to 4.0, and with that info I could set the right values to avoid any clipping without lowering the volume too much. Currently I have it set to safe values, which give me low volume in a lot of movies...

[Midzuki]

Example:

--- create a stereo DTS file @ 48kHz @ 384kbps;

--- run "eac3to input.dts output.wav";

--- play "output.wav" in your favorite player;

*giggles*

OTOH,

"ffmpeg -i input.dts output.wav" works properly.

[nurbs]

Try "eac3to input.dts output.wav -simple"

[b66pak]

i am ok with "-down2simple" too...
_

[Midzuki]

Try "eac3to input.dts output.wav -simple"

Doesn't work.

[VAMET]

Dear Friends

eac3to is a very good application to demux original Blu-Ray Disc.
I used it very often and then my movies I watch on Dune HD Base 3.0.

Is there any possibility to demux with eac3to, but with such flag, option, to have subtitles ready for remux with tsMuxeR with final result: specified subtitles always on during movie playback?

I have audio English and Polish subtitles, but when I watch such movie after eac3to and tsMuxeR process, I need to choose manualy my Polish subtitles.

There is no option to automaticaly choose specified subtitles on Dune HD Base 3.0. So I would like to create BDMV folder structure, but with subtitles, which are always on from the beginning of the movie.

Is it possible?

I will be glad for any help.
Thank you.
Best regards.

Sincerely

[setarip_old]

@VAMET

Hi!

Try "multiAVCHD"...

[Adub]

Okay, I seem to have encountered a bug:

Source: Inglorious Bastards Blu-ray, US.
Problem:
Eac3to appears to hang, or "deadlock", with no CPU usage after finding clipping in the DTS-MA audio track that is being converted.

However, it only does this when the "resampleto48000" commandline flag is present. In addition, eac3to was complaining about the resampling and told me to use r8brain, if that is any help.

Example:
Microsoft Windows [Version 6.1.7600]
Copyright (c) 2009 Microsoft Corporation. All rights reserved.

C:\Users\Adub>"C:\multiAVCHD\tools\eac3to\eac3to.exe" "D:\Multimedia\BluTemp\ING
_BASTERDS\BDMV\STREAM\00010.m2ts" 3: "c:\00010.m2ts.t2.ac3" -resampleTo48000 -3
84 -r8brain -progressnumbers
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M2TS, 1 video track, 3 audio tracks, 5 subtitle tracks, 2:33:00, 24p /1.001
1: Chapters, 34 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: DTS, Spanish, 5.1 channels, 24 bits, 768kbps, 48khz
5: DTS, French, 5.1 channels, 24 bits, 768kbps, 48khz
6: Subtitle (PGS), English
7: Subtitle (PGS), Spanish
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), French
a03 The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
a03 The libav DTS decoder doesn't decode the full DTS-HD information.
a03 Extracting audio track number 3...
a03 Extracting DTS core...
a03 Decoding with libav/ffmpeg...
a03 Remapping channels...
a03 Resampling to 48khz...
a03 Encoding AC3 <384kbps> with libAften...
a03 Creating file "c:\00010.m2ts.t2.ac3"...
process: 1%
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a03 Clipping detected, a 2nd pass will be necessary.


If there is anything that you need from me, please, let me know.

[Jynx980]

I can't get eac3to to recognize the Monsters Inc. Bonus disc on Blu Ray. I get "HD DVD / Blu-Ray disc structure not found." from either a rip or actual disc in drive. It's the regular commercial disc, plays fine, just cant get any info on it with eac3to. This is the first time I have had a problem with this. Suggestions?

[NanoBot]

Are you sure that the bonus disc is a Bluray ? Amazon.com tells me "DVD Of Film Plus Original DVD Bonus", so the bonus disk might be a DVD, which of course can't be prcessed with eac3to.

[JnZ]

I'm wondering if would be posible support user defined FPS conversion.
For example I have audio from 23.976fps version, and video 25.050fps, so I need use something like this commandline:
eac3to source.mp3 dest.wav -23.976 -changeTo25.050

It will be nice to support reading these user defined float values.
:thanks:

Anyway thanks for superrior software.

[Jynx980]

It's Blu-Ray. File Size is about 21GB. The one on Amazon has 4 discs, two blu-ray and two dvd. It has the same structure as other blu-rays; BMDV, CERTIFICATE, PLAYLIST, STREAM... Not sure how it's different than any other one.

[tebasuna51]

@madshi, feature request if is easy:

When a gain is applied by request (+3dB) eac3to send the message:
Applying +3dB gain...

Is possible send a equivalent message when a 2nd pass for 'Clipping detected', or -normalize parameter, apply a gain?
The user then know the amount of gain needed.
The value can be in dB, % or coefficient, the more easy way for you.

Thanks.

[liquidator87]

When using a command line like this:

eac3to.exe source dest -slowdown -295ms

is the delay applied before or after the framerate conversion?

Looking at the program output it seems before, just to be sure :)

[billynews]

Actually I have some problems with "There's Something About Mary". It's a Seamless-Branching Blu-Ray, so I try to demux all streams, because I want to play it on Popcorn Hour.
But I'm always getting an error:

[v02] Writing the destination file failed. <ERROR>
Aborted at file position 18752526336. <ERROR>

I've ripped the disc without problems to a harddisk, so it's not a "bad disc".
I've tried different playlists (Kino-Version, extended Version), but eac3to always fails.

Is it a problem with the content of the disc? Or is it a problem with eac3to?


eac3to v3.18
command line: eac3to.exe BDMV 4) -demux
------------------------------------------------------------------------------
M2TS, 1 video track, 7 audio tracks, 19 subtitle tracks, 1:59:05, 24p /1.001
1: Chapters, 30 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: DTS, Japanese, 5.1 channels, 24 bits, 768kbps, 48khz
5: DTS, German, 5.1 channels, 24 bits, 768kbps, 48khz
6: DTS, French, 5.1 channels, 24 bits, 768kbps, 48khz
7: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
8: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
9: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
10: Subtitle (PGS), Japanese
11: Subtitle (PGS), English
12: Subtitle (PGS), Japanese
13: Subtitle (PGS), German
14: Subtitle (PGS), French
15: Subtitle (PGS), Danish
16: Subtitle (PGS), Dutch
17: Subtitle (PGS), Finnish
18: Subtitle (PGS), Norwegian
19: Subtitle (PGS), Swedish
20: Subtitle (PGS), German
21: Subtitle (PGS), German
22: Subtitle (PGS), French
23: Subtitle (PGS), French
24: Subtitle (PGS), Dutch
25: Subtitle (PGS), Dutch
26: Subtitle (PGS), German
27: Subtitle (PGS), French
28: Subtitle (PGS), Dutch
Creating file "00086 - Chapters.txt"...
[a07] Extracting audio track number 7...
[s15] Extracting subtitle track number 15...
[a08] Extracting audio track number 8...
[s12] Extracting subtitle track number 12...
[a07] Removing AC3 dialog normalization...
[a05] Extracting audio track number 5...
[s19] Extracting subtitle track number 19...
[s21] Extracting subtitle track number 21...
[a08] Removing AC3 dialog normalization...
[a06] Extracting audio track number 6...
[s18] Extracting subtitle track number 18...
[s20] Extracting subtitle track number 20...
[s13] Extracting subtitle track number 13...
[a03] Extracting audio track number 3...
[s10] Extracting subtitle track number 10...
[s11] Extracting subtitle track number 11...
[s17] Extracting subtitle track number 17...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a09] Extracting audio track number 9...
[s14] Extracting subtitle track number 14...
[s26] Extracting subtitle track number 26...
[s22] Extracting subtitle track number 22...
[s16] Extracting subtitle track number 16...
[a09] Removing AC3 dialog normalization...
[s27] Extracting subtitle track number 27...
[s23] Extracting subtitle track number 23...
[s25] Extracting subtitle track number 25...
[s28] Extracting subtitle track number 28...
[s24] Extracting subtitle track number 24...
[v02] Creating file "00086 - 2 - h264, 1080p24.h264"...
[a03] Creating file "00086 - 3 - DTS Master Audio, English, 5.1 channels, 24 bits, 48khz.dtsma"...
[a04] Creating file "00086 - 4 - DTS, Japanese, 5.1 channels, 24 bits, 768kbps, 48khz.dts"...
[a06] Creating file "00086 - 6 - DTS, French, 5.1 channels, 24 bits, 768kbps, 48khz.dts"...
[a05] Creating file "00086 - 5 - DTS, German, 5.1 channels, 24 bits, 768kbps, 48khz.dts"...
[a08] Creating file "00086 - 8 - AC3, English, 2.0 channels, 224kbps, 48khz.ac3"...
[a07] Creating file "00086 - 7 - AC3, English, 2.0 channels, 224kbps, 48khz.ac3"...
[a09] Creating file "00086 - 9 - AC3, English, 2.0 channels, 224kbps, 48khz.ac3"...
[s23] Creating file "00086 - 23 - Subtitle (PGS), French.sup"...
[s21] Creating file "00086 - 21 - Subtitle (PGS), German.sup"...
[s20] Creating file "00086 - 20 - Subtitle (PGS), German.sup"...
[s25] Creating file "00086 - 25 - Subtitle (PGS), Dutch.sup"...
[s24] Creating file "00086 - 24 - Subtitle (PGS), Dutch.sup"...
[s22] Creating file "00086 - 22 - Subtitle (PGS), French.sup"...
[s11] Creating file "00086 - 11 - Subtitle (PGS), English.sup"...
[s13] Creating file "00086 - 13 - Subtitle (PGS), German.sup"...
[s14] Creating file "00086 - 14 - Subtitle (PGS), French.sup"...
[s19] Creating file "00086 - 19 - Subtitle (PGS), Swedish.sup"...
[s15] Creating file "00086 - 15 - Subtitle (PGS), Danish.sup"...
[s17] Creating file "00086 - 17 - Subtitle (PGS), Finnish.sup"...
[s18] Creating file "00086 - 18 - Subtitle (PGS), Norwegian.sup"...
[s10] Creating file "00086 - 10 - Subtitle (PGS), Japanese.sup"...
[s16] Creating file "00086 - 16 - Subtitle (PGS), Dutch.sup"...
[s26] Creating file "00086 - 26 - Subtitle (PGS), German.sup"...
[s27] Creating file "00086 - 27 - Subtitle (PGS), French.sup"...
[s28] Creating file "00086 - 28 - Subtitle (PGS), Dutch.sup"...
[v02] Writing the destination file failed. <ERROR>
Aborted at file position 18752526336. <ERROR>

[nautilus7]

It sounds like there isn't enough free space in your hard disc.

[billynews]

It sounds like there isn't enough free space in your hard disc.

no, it's not that easy :p

And I don't think it's a problem with the hd. I've tried the whole procedure (ripping, demuxing) on different harddisks.

Strange - right now I've found out, demuxing every stream separately is working :)

[popr]

Perhaps this has been asked before but with so much posts an answer is hard to find and also asking again wouldn't hurt :)

Is there any chance to share the source code for eac3to? Ideally into an open git repository such that updates could be offered by anybody interested in adding features or fixing issues.

I'd like to be able to add muxing into mkv with unprocessed streams for everything i.e. audio+video+subtitles for dvd, hd-dvd and blu-ray. Even though wine deals with eac3to.exe nicely, a native linux port would be also great :-)

[73ChargerFan]

Perhaps this has been asked before but with so much posts an answer is hard to find and also asking again wouldn't hurt :)

Is there any chance to share the source code for eac3to? Ideally into an open git repository such that updates could be offered by anybody interested in adding features or fixing issues.
Eac3to is closed source, free software. Madshi has responded to these questions (many from me), so search this thread for answers.

I'd like to be able to add muxing into mkv with unprocessed streams for everything i.e. audio+video+subtitles for dvd, hd-dvd and blu-ray. Even though wine deals with eac3to.exe nicely, a native linux port would be also great :-)

Mkv muxing/demuxing is not the primary function of eac3to, and is fairly complex. I suggest looking at mkvtoolnix (http://forum.doom9.org/showthread.php?t=120648) for further functionality.

[sehgal.v7]

Anybody got problem playing 7.1 flac made by eac3to?

I converted "DTS Master Audio, English, 7.1 channels, 24 bits, 48khz" to flac with ArcSoft DTS Decoder v1.1.0.7

FLACTester throws this error "FLAC__STREAM_DECODER_UNPARSEABLE_STREAM"

[Inspector.Gadget]

Anybody got problem playing 7.1 flac made by eac3to?

I converted "DTS Master Audio, English, 7.1 channels, 24 bits, 48khz" to flac with ArcSoft DTS Decoder v1.1.0.7

FLACTester throws this error "FLAC__STREAM_DECODER_UNPARSEABLE_STREAM"

FLACTEster is known to be broken for multichannel 24-bit files created with the recent version(s) of FLAC. This has been happening to me for the last year, at the least. Check it with foobar2000 instead.

[Vincent Vega]

i'm new to hd audio transcoding (from blu ray, hd dvd).

can someone please provide an explanation or a link where i can read about this whole "different channel order" thing on hd discs, and the need to remap channels after decoding. does it apply to both blu ray and hd dvd? and does it apply to all hd audio formats or just some of them?

eac3to remaps channels after all decoders - libav, arcsoft, nero.. what, none of them are yet able to decode hd audio properly? also we use the same decoders in directshow players without eac3to to do the remapping, then what, my system outputs wrong channel order each time i see an hd movie?

i'm sure these are all stupid questions, but i just couldn't find a good answer, although i searched all evening -(

thanks

[Inspector.Gadget]

eac3to remaps channels only when transcoding to a format that doesn't obey the Blu-ray convention but instead obeys another (I think something do with Waveformat extensible). Trust eac3to's remapping, but don't worry about playback: Arcsoft TMT, PowerDVD, etc. map channels correctly on playback of the original audio types because they know to expect the Blu-ray order.

[popr]

Eac3to is closed source, free software. Madshi has responded to these questions (many from me), so search this thread for answers.

A pitty about the closed source issue. Free software is not really free unless it lives as open source forever :rolleyes:


Mkv muxing/demuxing is not the primary function of eac3to, and is fairly complex. I suggest looking at mkvtoolnix (http://forum.doom9.org/showthread.php?t=120648) for further functionality.

Indeed, this is why community effort would help. The current option of demuxing streams and then using other tools to remux is a bit of a pain ...

Anyway, thanks for the reply! :thanks:

[deathlord]

Anybody got problem playing 7.1 flac made by eac3to?

The newest Foobar2000 plays eac3to-7.1 flac perfectly.

[Vincent Vega]

eac3to remaps channels only when transcoding to a format that doesn't obey the Blu-ray convention but instead obeys another (I think something do with Waveformat extensible). Trust eac3to's remapping, but don't worry about playback: Arcsoft TMT, PowerDVD, etc. map channels correctly on playback of the original audio types because they know to expect the Blu-ray order.

thank you Inspector.Gadget

so one practical question:
- if i convert DTS HD MA to say AC3 or AAC to mux into an mkv HDrip, i should let eac3to remap the channels, right?
- now if i convert DTS HD MA to LPCM to use in blu ray or avchd authoring, should i still let eac3to remap the channels?

i'm sorry i just don't get the clear picture here, need a little more information..

[tebasuna51]

so one practical question:
- if i convert DTS HD MA to say AC3 or AAC to mux into an mkv HDrip, i should let eac3to remap the channels, right?
- now if i convert DTS HD MA to LPCM to use in blu ray or avchd authoring, should i still let eac3to remap the channels?

Just let eac3to do the job.

Maybe your authoring software (at least the based in TsMuxer) need the input in WAV format (file < 4GB) or W64 format (file > 4GB) instead LPCM format (without headers).

[DigitalLF]

i only got a 2.0 setup and i would like to downmix a DTS-MA file to 2.0 flac but when i try to run:

C:\eac3to>eac3to.exe "d:\audio1.dtshd" e:\audio1.flac -mixlfe -down2

i get something that i do not understand fully...

"Reducing depth from 64 to 24 bits"

why is does eac3to change the depth?

oh and is this the correct way to downmix a DTS-MA to 2.0? i am totally new to this

[tebasuna51]

@DigitalLF
Please put the full log to know the decoder used.

The downmix routines work at 64 bits float to be the more exact possible, to encode to flac we need recover a proper integer bitdepth.

The downmix process isn't lossless, I recommend you use aac output instead flac:
C:\eac3to>eac3to.exe "d:\audio1.dtshd" e:\audio1.m4a -mixlfe -down2

[DigitalLF]

@tebasuna51

eac3to v3.18
command line: eac3to.exe "d:\audio1.dtshd" f:\audio1.flac -mixlfe -down2
------------------------------------------------------------------------------
DTS Master Audio, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
Decoding with ArcSoft DTS Decoder...
Downmixing multi channel audio to stereo...
Reducing depth from 64 to 24 bits...
Encoding FLAC with libFlac...
Creating file "f:\audio1.flac"...
Clipping detected, a 2nd pass will be necessary. <WARNING>
The original audio track has a constant bit depth of 24 bits.
The processed audio track has a constant bit depth of 24 bits.
Starting 2nd pass...
Decoding with ArcSoft DTS Decoder...
Downmixing multi channel audio to stereo...
Reducing depth from 64 to 24 bits...
Encoding FLAC with libFlac...
Creating file "f:\audio1.flac"...
The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 43 minutes, 6 seconds.
Done.


--------------------- and ---------------------

eac3to v3.18
command line: eac3to.exe "d:\audio2.dtshd" f:\audio2_nolfe.flac -down2
------------------------------------------------------------------------------
DTS Master Audio, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
Decoding with ArcSoft DTS Decoder...
Downmixing multi channel audio to stereo...
Reducing depth from 64 to 24 bits...
Encoding FLAC with libFlac...
Creating file "f:\audio_nolfe.flac"...
The original audio track has a constant bit depth of 24 bits.
The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 20 minutes, 39 seconds.
Done.


the outputfile differ with around 50mb so i don't think there wore any LFE data. (as i said before i am a complete n00b when it comes to this..)

i will also try to output a m4a file...

EDIT:
@tebasuna51 i have now made a test with m4a.... from 6gb DTS-MA i got a 200mb m4a... somethings wrong here.. :P

[Blue_MiSfit]

No, there's nothing wrong. AAC is not lossless, so 200mb output is not unreasonable.

~MiSfit

[shambles]

Original audio track, L+R+C+SL+SR: max 24 bits, average 16 bits.
Original audio track, LFE: constant bit depth of 16 bits.for these variable bit depth tracks, would it be possible to add a switch that dithers only >16bit samples, and truncates 16bit ones when reducing bitdepth to 16bit? or is there some software already that can do such a thing?

edit: or making it default behaviour might make more sense :)

[nurbs]

There is the -down16 switch. That won't do anything if the content is already 16bit and will dither down higher bitdepth content.

[tebasuna51]

...
the outputfile differ with around 50mb so i don't think there wore any LFE data
The LFE is low frequency (~120 Hz) and need much less bitrate than the whole audio (~24 KHz)
...i have now made a test with m4a.... from 6gb DTS-MA i got a 200mb m4a... somethings wrong here..
Is OK, listen the flac track and the m4a track, can you hear the difference?
Like I say you the downmix is a lossy process (you can't recover the original signal) then is not sense use a lossless encoder. Store a copy of the full 5.1 flac to be played in a surround system in the future.

[shambles]

There is the -down16 switch. That won't do anything if the content is already 16bit and will dither down higher bitdepth content.sure, that applies if the source has a constant bit depth. but if the source has a variable bit depth then it dithers the whole track..

[DigitalLF]

@tebasuna51 well... i got a very nice setup (high-end) right now that i am very happy with and i would like the best sound possible with my setup... i have non to little knowledge about downmix and flac vs. m4a but it does not sound right that a aac would be a good replacement for flac? and if we would just talk about harddrive space it is not interesting to save space i prefer good sound over high compression. is there a better lossless way to downmix?

[Inspector.Gadget]

Downmixing is never lossless.

[tebasuna51]

@DigitalLF,Is your choice, my choice is don't downmix and preserve the 5.1 original audio (you can downmix at play time in a stereo audio equipment)

[DigitalLF]

@Inspector.Gadget my bad i found some info about this after that i posted here.

@tebasuna51 well.. i got a 2.0 amp and there is no mixdown and even if there wore there would not be enough bandwidth with a optical cable. so the processing most happen in the computer.

learning how to downmix is among other things just for learning and testing to see how much cpu/mem the ffdshow mixdown requires.

[Vincent Vega]

Just let eac3to do the job.

Maybe your authoring software (at least the based in TsMuxer) need the input in WAV format (file < 4GB) or W64 format (file > 4GB) instead LPCM format (without headers).

hi tebasuna51!

correct me if i'm wrong:
-when eac3to decodes .dtshd to .lpcm it remaps the channels from correct wav order output by arcsoft decoder to blu ray order because the resultant .lpcm file is supposed to be used for blu ray authoring.
-when eac3to decodes .dtshd to .wav it does not remap the channels because the resultant .wav file is not supposed to be used for blu ray authoring but should be used as lossless source for further encoding to ac3, aac, etc.

now, if above is true, i have the following problem: i cannot import the .lpcm output of eac3to into multiavchd (it shows no tracks imported after clicking on such file), and if i use the same .lpcm after fixing it with pcm2tsmu multiavchd for some reason detects it as aac(!). only importing in .wav container works fine.

so is there a switch to get eac3to to output .wav with blu ray style channel order suitable for importing into a blu ray compilation?

[b66pak]

when importing .wav (http://en.wikipedia.org/wiki/WAV) to bluray it will be muxed as .lpcm (.lpcm (http://en.wikipedia.org/wiki/LPCM) is a type of .pcm...)

if the .wav is bigger than 4gb (in 99% of the cases) then decode to .w64...
_

[tebasuna51]

... i cannot import the .lpcm output of eac3to into multiavchd (it shows no tracks imported after clicking on such file), and if i use the same .lpcm after fixing it with pcm2tsmu multiavchd for some reason detects it as aac(!). only importing in .wav container works fine.

Forget pcm2tsmu and don't worry with the channel mapping, use wav/w64 container. Like b66pak say you all is managed correctly.

[Midzuki]

eac3to does not identify .dts streams
(read: .cpt / .dca padded-with-zeroes for "SPDIFing")
if the bitrate is different than 754 / 1509 kbps and/or
the channel count is less than 5.1.

Example:

-- create a stereo .cpt @ 384kbps

-- use spdifer.exe for generating a "little-endian.raw"

-- use swab.exe for generating a "big-endian.raw"

MediaInfo (0.7.23BETA) recognizes only the little-endian file as DTS, and mkvmerge (3.2.0) recognizes both the big-endian and the little-endian files as DTS, — whereas eac3to fails in both cases. :(

<REMINDER>
Valid bitrates and sampling frequencies for "Coherent Acoustics" compression:

Bitrates (kbps) :

32 56 64 96

112 128 192 224 256 320

384 448 512 576 640 754 960

1024 1152 1234 1280 1344 1408

1411.2 1472 1509 1920 2048 3072 3840

Frequencies (kHz) :

8 11.025 12 16 22.050 24 32 44.1 48
<REMINDER>

[oddwunn]

Newbie question (and probably already answered somewhere that I can't find): On a few AnyDVD Blu-ray rips, when I try to analyze the structure, I sometimes get the message "HD DVD / Blu-Ray disc structure not found" reported by Eac3to v3.18. When I examine the structure it looks absolutely fine to me, so I don't know why Eac3to is balking at it. What tests does Eac3to run to determine proper structure? How can I "convince" Eac3to that the structure is valid? Any other ideas or suggestions?

[setarip_old]

@oddwunn

Hi!When I examine the structure it looks absolutely fine to meWhy don't you post the structure of one of your problematic BluRay discs?

[oddwunn]

Why don't you post the structure of one of your problematic BluRay discs?
Hmmm...I'm not sure exactly how to do that, but I will try:
Master directory created by AnyDVD is "E:\BDROM".
Under that I have 2 directories - "BDMV" and "CERTIFICATE".

1. Under BDMV I have several directories:
AUXDATA
BACKUP
BDJO
CLIPINF
JAR
META
PLAYLIST
STREAM

and 2 files:
index.bdmv
MovieObject.bdmv

The directories "AUXDATA", "BDJO", "JAR", and "META" are all empty. The directory "BACKUP" seems to have backups of other parts of the disc. "CLIPINF" has 15 files sequentially numbered "00000.clpi" to "00014.clpi". "PLAYLIST" has 14 files numbered sequentially from "00000.mpls" to "00013.mpls". "STREAM" has 15 files numbered sequentially from "00000.m2ts" to "00014.m2ts".


2. Under CERTIFICATE I have one directory ("BACKUP") and one file ("id.bdmv"). Under that directory I have what appears to be an identical copy of "id.bdmv".

From what I know this structure looks sound to me, but apparently there is some test that Eac3to is using to determine if a valid Blu-ray structure is found and this particular rip does not pass that test.

I have done literally hundreds of Blu-ray rips and this is now the 4th time I have run into this error message from Eac3to, and all 4 of the rips looked perfectly good to me. I have tried PM'ing madshi over at AVS, but I guess he hasn't been around lately or is too busy or just not interested (not that I blame him...:) ). I would appreciate anyone who could help explain why Eac3to is balking at certain rips and what I can do to "convince" the program to accept the structure.

[TinTime]

What happens if you run eac3to against one (or all) of the playlists?

eac3to.exe 00000.mpls

[tebasuna51]

...On a few AnyDVD Blu-ray rips, when I try to analyze the structure, I sometimes get the message "HD DVD / Blu-Ray disc structure not found" reported by Eac3to v3.18....
I get the same message with the free BluRay sample from here:
http://x264dev.multimedia.cx/?p=328%22

[oddwunn]

What happens if you run eac3to against one (or all) of the playlists?
It analyzes the playlists just fine.

[setarip_old]

@oddwunn

1) Have you tried playing this BluRay "package" from your hard drive (AS a BluRay, NOT as individual .M2TS files)? Does it play properly?

2) I do not use AnyDVDHD, but I recall seeing that it creates another folder named (IIRC) "Sly!", that contains the remnants of the original "AACS" folder/directory.

3) What is the title and region of this particular original, commercial BluRay disc?

[Snowknight26]

Sounds like you're having eac3to analyze E:\ instead of E:\BDROM.

[oddwunn]

1) Have you tried playing this BluRay "package" from your hard drive (AS a BluRay, NOT as individual .M2TS files)? Does it play properly?
No, I don't have a software Blu-ray player installed. I play the actual discs using a hardware player (Oppo BDP-83). I used to have a trial version of PowerDVD 9 installed, but it only played about 2 out of 10 discs that I tried, so I gave up on software players. Remuxing to .MKV has been a MUCH better solution...:)
2) I do not use AnyDVDHD, but I recall seeing that it creates another folder named (IIRC) "Sly!", that contains the remnants of the original "AACS" folder/directory.
I have seen that "Sly!" directory on other people's rips, but I have never seen it with my own rips, and like I said, I have done a few hundred by now.
3) What is the title and region of this particular original, commercial BluRay disc?
Region 1 - Stargaze HD - Universal Beauty
Sounds like you're having eac3to analyze E:\ instead of E:\BDROM.
I made that mistake many moons ago and have long since learned where to point Eac3to when I analyze rips. I even went as far as to try renaming the parent directory in case Eac3to saw it as an illegal directory or something. But you are right, Eac3to will give you the same error message if you point it to the wrong directory.

[Steel]

I recently reinstalled my OS (and of course EAC3TO!!) and I am having some trouble. I have eac3to v3.18, Nero v7.9.6.0, and I also have the bluray/hd plugin installed. I keep getting the message:

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.

I have all of that, any ideas why eac3to is not recognizing it?

Also, using win7 x64. BUt that makes no difference as that is what I was using before and it all worked.

Thanks

[setarip_old]

@oddwunn Remuxing to .MKV has been a MUCH better solution...That being said, for this problematic disc (and the other three that you alluded to), may I suggest that you try using only MakeMKV (No need for either AnyDVD or EAC3to) to, with one click, both decrypt the disc and convert to an .MKV file of the "movie only"?

[oddwunn]

That being said, for this problematic disc (and the other three that you alluded to), may I suggest that you try using only MakeMKV (No need for either AnyDVD or EAC3to) to, with one click, both decrypt the disc and convert to an .MKV file of the "movie only"?
Thanks for the tip! Unfortunately MakeMKV can't quite handle this disc, but at least I now know why. The first two .m2ts files are intro screens. The 3rd .m2ts is some sort of a menu that allows the user to choose any of the 12 Starscapes, and they seem to play on a one at a time basis. This probably explains why Eac3to balked at the structure, as it does not have a playlist that is "movie only" anywhere. There is no way to create a single .MKV from this disc that will function in the same manner as the original disc!

As far as MakeMKV goes, it looks like a very promising program, but it still needs work (thus the "beta"). None of the .MKVs it created would play back on Zoomplayer or MPC, my players of choice. I am going to try it out with a number of other more mainstream titles, as I have a feeling it is just a quirk with this one particular title.

Thanks again!

[setarip_old]

@oddwunn None of the .MKVs it created would play back on Zoomplayer or MPC, my players of choice.Please correct me if I'm mistaken but, I don't believe either of these players is capable of playing back .MKVs containing High Def video and or audio...

[oddwunn]

Please correct me if I'm mistaken but, I don't believe either of these players is capable of playing back .MKVs containing High Def video and or audio...
I play hi def audio and video all the time with both players. The video codecs are installed from the getgo (and others can be added if you prefer to use a different decoder) and you just need to install madFlac in order to decode hi def audio (of course the original DTS-MA or TrueHD must be decoded and recoded as FLAC first...:) ). But in my experiment with MakeMKV I unchecked the hi-def audio and just ripped the plain vanilla AC3 tracks, as I wasn't expecting MakeMKV to do the hi def audio conversion for me. Don't get me wrong, MakeMKV looks like a very promising tool, but it certainly can not replace a combination of Eac3to and MKVmerge in terms of flexibility and ability to manipulate audio and video streams.

Since you mentioned it, what software player WILL play back TrueHD and/or DTS-MA tracks when they are imbedded in an .MKV? I didn't know such a player existed.

At any rate, I think I now understand why Eac3to saw this disc as having an improper structure, as there is no playlist that combines the various .m2ts files into a single "movie" and instead has a menu which plays back the .m2ts files separately - mystery solved...:)

Edit: The more I play with MakeMKV, the more I like it. I will go find a dedicated thread elsewhere, as I don't want to continue off-topic here.

[Inspector.Gadget]

@oddwunn Please correct me if I'm mistaken but, I don't believe either of these players is capable of playing back .MKVs containing High Def video and or audio...

This is completely incorrect.

[setarip_old]

@oddwunn

I stand corrected.

BTW - My only purpose in suggesting you experiment with MakeMKV was so you could see if it generated any negative information similar to what you stated regarding EAC3to...

[madshi]

maybe you can add support for Lame.exe encoder (like NeroAacEnc or with dll).
Many people want mp3 output for compatibility issues (device or container avi related).
Already have that on my to do list. If only I had a bit more free time... ;)

Using the custom mix only for scanning the file and then output the peak value in dB.
Would it help if eac3to would output the peak volume of any raw/decoded audio track?

Example:

--- create a stereo DTS file @ 48kHz @ 384kbps;

--- run "eac3to input.dts output.wav";

--- play "output.wav" in your favorite player;

*giggles*
Can I have a sample of such a stereo DTS file with which I can reproduce the problem?

However, it only does this when the "resampleto48000" commandline flag is present.

3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
Why do you want to resample the audio from 48khz to 48khz? That doesn't make any sense to me. Ok, eac3to might be stupid enough to stumble over this, but still, it doesn't make much sense to resample audio, if it's already at the desired rate to start with!

I can't get eac3to to recognize the Monsters Inc. Bonus disc on Blu Ray. I get "HD DVD / Blu-Ray disc structure not found." from either a rip or actual disc in drive.
Newbie question (and probably already answered somewhere that I can't find): On a few AnyDVD Blu-ray rips, when I try to analyze the structure, I sometimes get the message "HD DVD / Blu-Ray disc structure not found" reported by Eac3to v3.18. When I examine the structure it looks absolutely fine to me, so I don't know why Eac3to is balking at it. What tests does Eac3to run to determine proper structure? How can I "convince" Eac3to that the structure is valid? Any other ideas or suggestions?
eac3to only accepts Blu-Ray playlists with a runtime of 15 minutes or longer. I've implemented it this way because otherwise with many Blu-Rays you'd get literally hundreds of playlists listed by eac3to, which would make things pretty ugly. My best guess is that the Monters Inc. Bonus disc only has shorter playlists than 15 minutes. Is that possible? Of course the eac3to complaint is bad/wrong, if that's the cause of the problem.

I'm wondering if would be posible support user defined FPS conversion.
Not possible, because the resampling lib I'm using is limited to do only specific conversions.

When a gain is applied by request (+3dB) eac3to send the message:
Applying +3dB gain...

Is possible send a equivalent message when a 2nd pass for 'Clipping detected', or -normalize parameter, apply a gain?
The user then know the amount of gain needed.
The value can be in dB, % or coefficient, the more easy way for you.
Sure, can do that.

When using a command line like this:

eac3to.exe source dest -slowdown -295ms

is the delay applied before or after the framerate conversion?
Uhm, (checking source code), delay first.

Actually I have some problems with "There's Something About Mary". It's a Seamless-Branching Blu-Ray, so I try to demux all streams, because I want to play it on Popcorn Hour.
But I'm always getting an error:

[v02] Writing the destination file failed. <ERROR>
Aborted at file position 18752526336. <ERROR>
If that's not a full harddisk then I don't really know what it is. It's an error reported from the win32 file writing API.

for these variable bit depth tracks, would it be possible to add a switch that dithers only >16bit samples, and truncates 16bit ones when reducing bitdepth to 16bit?
This is not so easy because the exact bitdepth of the tracks is only known after processing is fully completed, so basically the whole processing would have to be repeated. I'm not sure if it's worth it adding extra code for that and spending extra processing time on that.

i got a 2.0 amp and there is no mixdown and even if there wore there would not be enough bandwidth with a optical cable. so the processing most happen in the computer.

learning how to downmix is among other things just for learning and testing to see how much cpu/mem the ffdshow mixdown requires.
I'd suggest to do downmixing during playback, e.g. by using ffdshow.

is there a switch to get eac3to to output .wav with blu ray style channel order
There is no such thing as a "wav with blu ray style channel order". Such a wav file would be "broken". Wav files have a strictly defined channel order and eac3to is using that. Any software which reads a wav file and create a Blu-Ray from that has to internally make sure that the channels are remapped correctly. If a software isn't doing that, that would be a bug in that software.

eac3to does not identify .dts streams
(read: .cpt / .dca padded-with-zeroes for "SPDIFing")
if the bitrate is different than 754 / 1509 kbps and/or
the channel count is less than 5.1.
Can I have a sample, please?

I recently reinstalled my OS (and of course EAC3TO!!) and I am having some trouble. I have eac3to v3.18, Nero v7.9.6.0, and I also have the bluray/hd plugin installed. I keep getting the message:

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.

I have all of that, any ideas why eac3to is not recognizing it?
I don't know. Have you registered the HD plugin properly? I do know that there were a few cracked plugin serial numbers floating around, which did NOT work. Make sure you use a properly bought serial number.

[yesgrey]

Would it help if eac3to would output the peak volume of any raw/decoded audio track?
No, because we need to know the peak for a certain mix of the channels, and a peak in one mix does not necessarily means also a peak in a different mix. It would only work if we could specify the mix configuration.
For example. I have a 4.0 setup, and I send the center and LFE channels to my front left and right speakers. The surround channels remain untouched.

[QuadraX]

Could anybody help me to integrate Surcode into Eac3to?

When i do the Test (eac3to -test) then it says nothing about Surcode, but it is installed!

[Midzuki]

@ madshi:

http://cid-5acf098e0ebae8d5.skydrive.live.com/self.aspx/.Public/mb234dts.7z

[ ^ file not available anymore ]

Use spdifer.exe for generating a padded .DTS,
and swab.exe for inverting the byte-order.

P.S.: Have fun! :)

[madshi]

No, because we need to know the peak for a certain mix of the channels
Ok.

Could anybody help me to integrate Surcode into Eac3to?

When i do the Test (eac3to -test) then it says nothing about Surcode, but it is installed!
eac3to looks for the path to the Surcode exe in the registry. It should be one of these (depends on Surcode version):

- HKLM\Software\Minnetonka Audio Software\SurCode DVD-DTS
- HKLM\Software\Minnetonka Audio Software\SurCode DVD DTS

http://cid-5acf098e0ebae8d5.skydrive.live.com/self.aspx/.Public/mb234dts.7z
Works just fine for me with the ArcSoft DTS decoder. The libav decoder produces garbage, but that's not eac3to's fault. Maybe the bug would be fixed if I updated to the latest libav build. Would be about time, anyway...

[Midzuki]

madshi wrote:

Works just fine for me with the ArcSoft DTS decoder. The libav decoder produces garbage, but that's not eac3to's fault. Maybe the bug would be fixed if I updated to the latest libav build. Would be about time, anyway...

That explains it all, then. :)

Notwithstanding, what do you have to say about the "zero-padding" issue? :confused:

Also: will eac3to v3.19 recognize "pure" .DTSHD files? :)

[heerschop]

I recently reinstalled my OS (and of course EAC3TO!!) and I am having some trouble. I have eac3to v3.18, Nero v7.9.6.0, and I also have the bluray/hd plugin installed. I keep getting the message:

Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed


I use Eac3to with nero without installing nero.
Eac3to only needs a couple files and registry settings
Nero Files:
- AdvrCntr2.dll
- NeAudio2.ax
- NeEacDec.dll

Copy these files in a folder of your choice. I have copied them to my eac3to folder.
Use the following command lines to register the dll's.
regsvr32.exe C:\tools\eac3to\Nero\NeAudio2.ax
regsvr32.exe C:\tools\eac3to\Nero\AdvrCntr2.dll

The following registry keys have to be added to the registry to register the nero plugin.

[HKEY_LOCAL_MACHINE\SOFTWARE\Ahead\Installation\Families\Nero 7\Info]
etc.

[HKEY_LOCAL_MACHINE\SOFTWARE\Ahead\Installation\Families\Plugins\Info]
etc.

That all you need to use the nero plugin with eac3to

[heerschop]

Could anybody help me to integrate Surcode into Eac3to?

When i do the Test (eac3to -test) then it says nothing about Surcode, but it is installed!

Copy surcodedvd.exe and the Tables directory in a directory of your choice.
I have copied mine to C:\tools\eac3to\Surcode
check or add the following keys to the registry

[HKEY_LOCAL_MACHINE\SOFTWARE\Minnetonka Audio Software]

[HKEY_LOCAL_MACHINE\SOFTWARE\Minnetonka Audio Software\SurCode DVD DTS]
"Home"="C:\\tools\\eac3to\\Surcode"

[HKEY_LOCAL_MACHINE\SOFTWARE\Minnetonka Audio Software\SurCode DVD DTS\1.0.10]

It work's for me

[ACrowley]

madshi wrote:



That explains it all, then. :)

Notwithstanding, what do you have to say about the "zero-padding" issue? :confused:

Also: will eac3to v3.19 recognize "pure" .DTSHD files? :)

A simple Header switch Function would be perfect.
(pure DTSHD Header to stripped DTSHD Header and reversed)
There are only a few Bytes different so it should be easy for Madshi :)

[Midzuki]

A simple Header switch Function would be perfect.
(pure DTSHD Header to stripped DTSHD Header and reversed)
There are only a few Bytes different so it should be easy for Madshi :)

Just as an "irrelevant" detail:
first 140 bytes for Master Audio,
but just 116 bytes for "non-MA" content.

As long as the new eac3to removes the DTSHD footer as well, of course. :D

[ACrowley]

Just as an "irrelevant" detail:
first 140 bytes for Master Audio,
but just 116 bytes for "non-MA" content.

As long as the new eac3to removes the DTSHD footer as well, of course. :D

so far i can remember there are some bytes an the End of the Header too which contains the Timecode Value.

[Midzuki]

so far i can remember there are some bytes an the End of the Header too which contains the Timecode Value.

Besides, the footer also contains:

BUILDVER

LBUILD DATE

BUILD TIME

VERSION

REVISION

FILEINFO

BLACKOUT ( <-- :confused: :confused: :confused: )

[aleg]

I use Eac3to with nero without installing nero.
Eac3to only needs a couple files and registry settings
Nero Files:
- AdvrCntr2.dll
- NeAudio2.ax
- NeEacDec.dll

Copy these files in a folder of your choice. I have copied them to my eac3to folder.
Use the following command lines to register the dll's.
regsvr32.exe C:\tools\eac3to\Nero\NeAudio2.ax
regsvr32.exe C:\tools\eac3to\Nero\AdvrCntr2.dll

The following registry keys have to be added to the registry to register the nero plugin.

[HKEY_LOCAL_MACHINE\SOFTWARE\Ahead\Installation\Families\Nero 7\Info]
etc.

[HKEY_LOCAL_MACHINE\SOFTWARE\Ahead\Installation\Families\Plugins\Info]
etc.

That all you need to use the nero plugin with eac3to

heerschop

Could you explain a little bit further would you mean with "etc." after the registry keys?

-
aleg

[heerschop]

heerschop

Could you explain a little bit further would you mean with "etc." after the registry keys?

-
aleg

etc. means that you have to add the complete registry keys. The XX's are your personal nero serial keys.

[HKEY_LOCAL_MACHINE\SOFTWARE\Ahead\Installation\Families\Nero 7\Info]
"EulaAccepted"="X"
"Serial7_XXXXXXXXXX"="XXXX-XXXX-XXXX-XXXX-XXXX-XXXX-XXXX"
"Version"="7.11.10.0"
"User"="XXXX"
"Company"=""
"CDNR"="XXXXX"
"MissingFilesState"="0"

[HKEY_LOCAL_MACHINE\SOFTWARE\Ahead\Installation\Families\Plugins\Info]
"Serial7_XXXXXXXXXX"="*XXXX-XXXX-XXXX-XXXX-XXXX-XXXX-XXXX"
"Serial7_XXXXXXXXXX"="*XXXX-XXXX-XXXX-XXXX-XXXX-XXXX-XXXX"
"Serial7_XXXXXXXXXX"="*XXXX-XXXX-XXXX-XXXX-XXXX-XXXX-XXXX"
"Serial7_XXXXXXXXXX"="*XXXX-XXXX-XXXX-XXXX-XXXX-XXXX-XXXX"

Where do you get the serial keys? Buy nero.

[aleg]

etc. means that you have to add the complete registry keys. The XX's are your personal nero serial keys.

[HKEY_LOCAL_MACHINE\SOFTWARE\Ahead\Installation\Families\Nero 7\Info]
"EulaAccepted"="X"
"Serial7_XXXXXXXXXX"="XXXX-XXXX-XXXX-XXXX-XXXX-XXXX-XXXX"
"Version"="7.11.10.0"
"User"="XXXX"
"Company"=""
"CDNR"="XXXXX"
"MissingFilesState"="0"

[HKEY_LOCAL_MACHINE\SOFTWARE\Ahead\Installation\Families\Plugins\Info]
"Serial7_XXXXXXXXXX"="*XXXX-XXXX-XXXX-XXXX-XXXX-XXXX-XXXX"
"Serial7_XXXXXXXXXX"="*XXXX-XXXX-XXXX-XXXX-XXXX-XXXX-XXXX"
"Serial7_XXXXXXXXXX"="*XXXX-XXXX-XXXX-XXXX-XXXX-XXXX-XXXX"
"Serial7_XXXXXXXXXX"="*XXXX-XXXX-XXXX-XXXX-XXXX-XXXX-XXXX"

Where do you get the serial keys? Buy nero.

heerschop

Thank you for your additional information.

I have my Nero 7 personal registration. I have in the meantime progressed to Nero 9 and would like to try to install these old Nero7 files alongside my Nero 9 for EAC3to purpose only.

-
aleg

[heerschop]

heerschop

Thank you for your additional information.

-
aleg

Let me know if you get it to work.
I am using arcsoft, nero, sonic and surcode with eac3to, all without installing them completely on my system.

[Jynx980]

eac3to only accepts Blu-Ray playlists with a runtime of 15 minutes or longer. I've implemented it this way because otherwise with many Blu-Rays you'd get literally hundreds of playlists listed by eac3to, which would make things pretty ugly. My best guess is that the Monters Inc. Bonus disc only has shorter playlists than 15 minutes. Is that possible? Of course the eac3to complaint is bad/wrong, if that's the cause of the problem.


Your right, none of the playlists or videos are longer than 15 minutes. Thank you for clearing this up.

[Steel]

I don't know. Have you registered the HD plugin properly? I do know that there were a few cracked plugin serial numbers floating around, which did NOT work. Make sure you use a properly bought serial number.


NO, I bought and paid for it. See image....

http://i44.tinypic.com/28c0zk6.jpg

[Steel]

I use Eac3to with nero without installing nero.
Eac3to only needs a couple files and registry settings
Nero Files:
- AdvrCntr2.dll
- NeAudio2.ax
- NeEacDec.dll

Copy these files in a folder of your choice. I have copied them to my eac3to folder.
Use the following command lines to register the dll's.
regsvr32.exe C:\tools\eac3to\Nero\NeAudio2.ax
regsvr32.exe C:\tools\eac3to\Nero\AdvrCntr2.dll

The following registry keys have to be added to the registry to register the nero plugin.

[HKEY_LOCAL_MACHINE\SOFTWARE\Ahead\Installation\Families\Nero 7\Info]
etc.

[HKEY_LOCAL_MACHINE\SOFTWARE\Ahead\Installation\Families\Plugins\Info]
etc.

That all you need to use the nero plugin with eac3to

NeAudio2.ax? I have the other files, no idea why i cant find this one..

[heerschop]

Any chance you could point me to somewhere to download NeAudio2.ax? I have the other files, no idea why i cant find this one..

I have Nero installed on one windows xp system.
If you have installed Nero and activated the plugin correctly. The files should be installed in:
AdvrCntr2.dll -> C:\Program Files\Common Files\Ahead\Lib
NeAudio2.ax -> C:\Program Files\Common Files\Ahead\DSfilter
NeEacDec.dll -> C:\Program Files\Common Files\Ahead\DSfilter

[Steel]

I have Nero installed on one windows xp system.
If you have installed Nero and activated the plugin correctly. The files should be installed in:
AdvrCntr2.dll -> C:\Program Files\Common Files\Ahead\Lib
NeAudio2.ax -> C:\Program Files\Common Files\Ahead\DSfilter
NeEacDec.dll -> C:\Program Files\Common Files\Ahead\DSfilter

As I previously stated, I have no idea why I can't find the one file. The screen shot proves I have Nero installed and the plugin activated. Was just asking for a little help with one file..

[laserfan]

My W7 x64 has NeAudio2.ax in C:\Program Files (x86)\Common Files\Ahead\DSFilter but for my installation I used a "micro" installer (search on that word in this thread). Don't recall why I used it except I didn't want to pollute my new Windows 7 with Nero junk--I don't even have the Registry entries that others have posted.

Dunno if your Premium codes will work w/a micro install--my Ultra codes worked again just fine (previously was installed to an XP x64 partition).

[XhmikosR]

Nero Lite/Micro (http://updatepack.nl/) works just fine. I never used the full Nero with Nero Lite/Micro so I suggest that you uninstall it first. Even run Nero Clean Tool (http://www.nero.com/enu/tools-utilities.html#tab4) after you uninstall it.

[Steel]

Nero Lite/Micro (http://updatepack.nl/) works just fine. I never used the full Nero with Nero Lite/Micro so I suggest that you uninstall it first. Even run Nero Clean Tool (http://www.nero.com/enu/tools-utilities.html#tab4) after you uninstall it.

Thanks guys, got it working!

[dansrfe]

If anyone is interested in getting ArcSoft DTS Decoder 1.1.0.7 to work with eac3to very easily then pm me.

[Steel]

If anyone is interested in getting ArcSoft DTS Decoder 1.1.0.7 to work with eac3to very easily then pm me.

dansfre: Thanks for the offer. That information was posted in this thread a long time ago. However, I'll repost it here for others to see.

1. Copy the following files to your eac3to directory.

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

2. Run regsvr32.exe ASAudioHD.ax from a command line in your eac3to folder.

This definitely works with dtsdecoderdll.dll 1.1.0.0 and dtsdec.dll 1.0.2.2; there are so many Arcsoft updates issued that I'm not sure if every version uses the same one.

If anyone needs help with the files drop me a PM.

[Midzuki]

Yet Another Shameless Feature Request :o :)

— add switch -dtsfreq , for modifying the playback frequency of the lossy DTS streams ( most useful if someone intends to create a DTS-CD @ 1387.0828125kbps but is too lazy to deal with a hex editor :p )

— add switch -dtsrate, for modifying the apparent bitrate of .cpt files

[dansrfe]

dansfre: Thanks for the offer. That information was posted in this thread a long time ago. However, I'll repost it here for others to see.

1. Copy the following files to your eac3to directory.

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

2. Run regsvr32.exe ASAudioHD.ax from a command line in your eac3to folder.

This definitely works with dtsdecoderdll.dll 1.1.0.0 and dtsdec.dll 1.0.2.2; there are so many Arcsoft updates issued that I'm not sure if every version uses the same one.

If anyone needs help with the files drop me a PM.

Yes but in order to attain those files, you have to download the trial then get the files from it. In order to save everyone time I can give you a zip file with all the latest versions of those dlls and asaudiohd.ax however I am not quite sure if it is allowed to package those dlls even though they do exist in the trail version.

[Steel]

Yes but in order to attain those files, you have to download the trial then get the files from it. In order to save everyone time I can give you a zip file with all the latest versions of those dlls and asaudiohd.ax however I am not quite sure if it is allowed to package those dlls even though they do exist in the trail version.

If it's trialware there is nothing illegal or wrong with it. Trialware is freely available to anyone.

[dansrfe]

Simple method to install ArcSoft DTS Decoder (1.1.0.7) in eac3to:

(From Steel's instructions):

1. Copy the following files to your eac3to directory.

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

2. Run regsvr32.exe ASAudioHD.ax from a command line in your eac3to folder.

Zip File with all the above included: http://www.sendspace.com/file/6rtk1l

[dansrfe]

I actually have a question regarding ArcSoft decoding of DTS-HD MA streams in eac3to. Do I have to set any parameters in ASAudioHD.ax such as channel outputs or anything or does eac3to just use the engine of the dlls from ArcSoft DTS Decoder to decode the dts-hd ma streams in its original state?

[~Revolution~]

When I try to convert mp3 to m4a or an mp2 to ac3 I hear lots of crackling (especially in dialog) that wasn't in the original audio source.

My commands are this:

eac3to source.mp2 source.ac3 -resampleto48000 -160

and

eac3to source.mp3 source.m4a -180

Am I doing something wrong or is there some other setting I should use?

[heerschop]

When I try to convert mp3 to m4a or an mp2 to ac3 I hear lots of crackling (especially in dialog) that wasn't in the original audio source.



I had the same problem. It has something to do with a bug in the Libav / ffmpeg decoder (the default eac3to decoder).
Use nero to decode the mp3 with eac3to and the crackling should be gone.

for example,

eac3to source.mp2 source.ac3 -resampleto48000 -160 -nero

[~Revolution~]

Is there an updated version of the Libav / ffmpeg decoder that doesn't cause this problem?

[Moondust]

I had the same problem. It has something to do with a bug in the Libav / ffmpeg decoder (the default eac3to decoder).
Use nero to decode the mp3 with eac3to and the crackling should be gone.

for example,

eac3to source.mp2 source.ac3 -resampleto48000 -160 -nero

I have the same problem with crackling sound when converting a DTS-HD 7.1 file into a FLAC 7.1 file using the Arcsoft decoder. Only 7.1 goes wrong, 5.1 works like a charm. Does the same solution work here too? If not, is there another remedy?

[Midzuki]

I have the same problem with crackling sound when converting a DTS-HD 7.1 file into a FLAC 7.1 file using the Arcsoft decoder. Only 7.1 goes wrong, 5.1 works like a charm.

To me, it seems a problem in the ArcSoft DTS decoder.
I have tested DtsDec.dll 1.0.2.2 + DTSdecoderDLL.dll 1.1.0.0, and these do NOT decode stereo DTSHD Master Audio correctly (not on my computer, at least).

[Abradoks]

Is there an updated version of the Libav / ffmpeg decoder that doesn't cause this problem?
You can get latest ffmpeg here (http://ffmpeg.arrozcru.org/autobuilds/).
Eac3to has too many unresolved issues to use it as decoder:
- Nero DRC bug (http://forum.doom9.org/showthread.php?p=1353768#post1353768)
- DTS-HD 7.1 bug (http://forum.doom9.org/showthread.php?p=1363005#post1363005)
- bug, when output is different for *.wav and *.wavs (at least with libav and ac3/dts)
- libav attenuates high frequencies of non-mono dolby-encoded ac3. Seems like a libavcodec bug because for some streams (5.1 encoded using dolby encoder) ffmpeg produces two times worse PSNR than eac3to.
- ...

[~Revolution~]

So anyone have a suggested solution to fix the two bugs?

[rapscallion]

Hey guys, I'm a little confused re DTS HD master 7.1 Blu-ray tracks.

If the player or receiver is only capable of 5.1, will it revert to the 5.1 core and you lose the advantage of lossless, or will it/they downmix the 7.1 to 5.1 and you'll still have the lossless track.

Using eac3to to decode the MA 7.1 track to wavs, will those wavs be lossless, or will they be decoded from the core ?

[b66pak]

If the player or receiver is only capable of 5.1, will it revert to the 5.1 core

yes it will play the core...all the ancient 5.1 receivers have to play something!...clever design to have backwards compatibility!

Using eac3to to decode the MA 7.1 track to wavs, will those wavs be lossless, or will they be decoded from the core ?

if you have ArcSoft DTS decoder installed you will get 8 lossless wavs...if not then libav will be used and you get 6 core wavs...
_

[rapscallion]

yes it will play the core...all the ancient 5.1 receivers have to play something!...clever design to have backwards compatibility! Thanks for the response, however are you saying that a receiver capable of playing/decoding only 5.1 MA (not 7.1) will also revert to the core, if it's presented with 7.1 MA ?


if you have ArcSoft DTS decoder installed you will get 8 lossless wavs...if not then libav will be used and you get 6 core wavs...

Well, that's good news, if the above is true. Because (yes, I have Arcsoft) I can decode to the lossless wavs and then re encode to DTS MA 5.1, using DTS MA Suite. Just ignore the 2 side channels, no need to downmix, correct ?

[dansrfe]

Just ignore the 2 side channels, no need to downmix, correct ?

Correct

[Snowknight26]

Any plans to add -check capabilities to MKVs?

[Laurent]

For a particular HD-DVD, eac3to detects no subtitle track. Same thing when I open the main EVO files in MediaInfo.
But when I play the disc in my HD-DVD player, I have the possibility to switch between french audio / no subtitle and english audio / french subtitle.
eac3to detects only one video track, so I don't think there are two video tracks, one with subtitles and one without subtitles.
Can you explain where are (hidden) the subtitles please ?

[Thunderbolt8]

got a 5.1 dtsma track, which only has data for the 2.0 channels, the rest is empty. so instead of keeping it as 5.1 with 4 empty channels, I'd rather turn in into a 2.0 dtsma & flac track. how do I do this? 5.1 -> wavs? and then afterwards?

[Laurent]

Do you know what is the current status of the libav decoder and support of spectral extension for E-AC3 audio tracks ?
If eac3to could support audio tracks from our french HD DVB-T standard, that would be really cool.
It looks like ffdshow is able to decode these tracks, and ffdshow is using libav decoder, isn't it ? So why not using this version of the libav decoder in eac3to ?

[Kurtnoise]

Do you know what is the current status of the libav decoder and support of spectral extension for E-AC3 audio tracks ?
this has been officially introduced few weeks ago in the FFmpeg trunk...

[Laurent]

this has been officially introduced few weeks ago in the FFmpeg trunk...

That's a good news.
I hope it means that lots of tools, and especially eac3to, could very soon support this E-AC3 extension ?

[jruggle]

- libav attenuates high frequencies of non-mono dolby-encoded ac3. Seems like a libavcodec bug because for some streams (5.1 encoded using dolby encoder) ffmpeg produces two times worse PSNR than eac3to.

I haven't ever seen anything like this reported before. Do you have a sample? And I'm assuming you mean using eac3to with Nero rather than libavcodec for AC-3 decoding...

If you do have a sample, it would be great to have the original AC-3 snippet, along with the eac3to wav or flac output.

[Abradoks]

I haven't ever seen anything like this reported before. Do you have a sample? And I'm assuming you mean using eac3to with Nero rather than libavcodec for AC-3 decoding...

If you do have a sample, it would be great to have the original AC-3 snippet, along with the eac3to wav or flac output.
Hi, jruggle. It's nice to see you here. I thought I should better investigate the problem before reporting it to ffmpeg, but was too lazy recently.
Here are some samples:
1) 5amctest.rar (http://rghost.net/2070394) (mirror (http://www.mediafire.com/?mgy1zg2nlmi)) : test.wav — mono source; test.ac3 — 2.0 ac3 produced from source (both channels are same) by Dolby encoder. Man who originally reported the problem has made this (http://i072.radikal.ru/1003/28/d2b2dceaf900.jpg) graph, where you can see attenuation in the high frequencies when using eac3to with libav. ffmpeg (SVN-r23056) has the same problem but attenuation starts on higher frequency. The funny thing is that a52dec hasn't such issue.
2) qs.7z (http://rghost.net/2070388) (mirror (http://www.mediafire.com/?qkn2jnd3eud)) (14.5 MB) — two another samples and source file. 5.1, Dolby encoder. For 640.ac3 eac3to (with libav) shows 27.809 dB SNR, while ffmpeg 11.547 dB. liba52 (through foobar2000 plugin) gives 27.537 dB. When comparing average sound level graphs, it looks like eac3to and ffmpeg start to attenuate at ~16.8 kHZ. And ffmpeg is always ~2 dB quieter than eac3to. For 224.ac3 attenuation starts near 4.5 kHZ but only for eac3to. ffmpeg is just always ~2 dB quieter than source.
Also in all 3 samples ffmpeg has average sound level about -140 dB after frequency cutoff. Is it ok?
Graphs:
http://img20.imageshack.us/img20/3152/92026076.th.png (http://img20.imageshack.us/img20/3152/92026076.png)http://img522.imageshack.us/img522/549/224u.th.png (http://img522.imageshack.us/img522/549/224u.png)http://img17.imageshack.us/img17/3793/640w.th.png (http://img17.imageshack.us/img17/3793/640w.png)

[jruggle]

Hi, jruggle. It's nice to see you here. I thought I should better investigate the problem before reporting it to ffmpeg, but was too lazy recently.
Here are some samples:

There could be a few things going on. First, with FFmpeg did you use the -drc_scale 0 option? IIRC eac3to always disables DRC. Also, eac3to uses a patch to get floating-point output from the libavcodec AC-3 decoder, while FFmpeg always converts to 16-bit.

[Abradoks]

First, with FFmpeg did you use the -drc_scale 0 option?
No, I didn't. But those tracks haven't DRC, so with -drc_scale 0 ffmpeg produces identical files.

[jruggle]

No, I didn't. But those tracks haven't DRC, so with -drc_scale 0 ffmpeg produces identical files.
Ok, then there are a few things left to investigate.

1) If FFmpeg is modified to give floating-point output will the results be closer to eac3to. I think this is likely. If not, what other libavcodec patches is Madshi using?

2) The first drop is probably at the rematrixing start point for the stereo sample and at the coupling start point for the 5.1 samples. First step is to verify this. If so, that gives a good starting point for investigation.

3) See what the Nero decoder does compared to liba52 and libavcodec.

What program did you use to make the graphs? I want to use it in my comparisons as well.

[0x0]

Hi everyone,
it's been several days I went on with this issue, so I decided to ask for help here.
I've recently bought Spiderman trilogy in bluray, and I successfully extracted all the audio tracks I needed from the 3 discs (in truehd format, .thd).
When I tried to convert the .thd files to mono wavs (or flac, or single wav) with eac3to & libav, no issue at all with audio from movie 1 and 3, but I experienced some problems, with the second movie's track decoding... Libav decoding in eac3to gave the repeated "Lossless check failed" warnings, even if the encoded wav files seemed to be good (seemed because I couldn't stay there on listening the entire two hours' track...).
Unsatisfied, I tried to use Nero Audio Decoder plugin in eac3to, but, even if the plugin correctly works for ac3, e-ac3 tracks, with TrueHD streams it stops immediately, giving the error "Nero Directshow decoder didn't accept the input stream" or something similar.
Is there any way to make the Nero plugin work for truehd tracks, or should I wait for madshi to release support for Arcsoft hd audio plugin?

[Snowknight26]

Unless you're getting thousands of those warnings, ignore them.