eac3to - audio conversion tool

[Yraen]

Yes, that's Ratatouille.

My experience encoding it to flac differs from yours. For me, it just won't make use of the gaps file.

eac3to v2.43
command line: "D:\editing apps\eac3to\2.43\eac3to.exe" "H:\HD\RATATOUILLE\" 1) 3: "N:\test\audio.flac"
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 1:50:32
1: Chapters, 32 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 5.1 channels, 24 bits, 48khz
4: AC3, English, 5.1 channels, 640kbit/s, 48khz
5: AC3, French, 5.1 channels, 640kbit/s, 48khz
6: AC3, Spanish, 5.1 channels, 640kbit/s, 48khz
7: AC3, English, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB
[a03] Audio gap description file detected, will be used for processing...
[a03] Extracting audio track number 3...
[a03] Reading RAW/PCM...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Encoding FLAC...
[a03] Creating file "N:\test\audio.flac"...
[a03] Audio overlaps for 6ms at playtime 0:08:50.
[a03] Audio overlaps for 6ms at playtime 0:11:24.
[a03] Audio overlaps for 7ms at playtime 0:13:31.
[a03] Audio overlaps for 6ms at playtime 0:42:09.
[a03] Audio overlaps for 8ms at playtime 1:08:51.
[a03] Audio overlaps for 6ms at playtime 1:11:58.
[a03] Audio overlaps for 5ms at playtime 1:15:28.
[a03] Audio overlaps for 6ms at playtime 1:30:31.
[a03] Audio overlaps for 6ms at playtime 1:31:39.
[a03] Audio overlaps for 5ms at playtime 1:33:24.
[a03] The audio file was demuxed without making use of the gap/overlap information.
[a03] Please rerun the same eac3to command line. That will correct the gaps/overlaps.
Video track 2 contains 159042 frames.
eac3to processing took 20 minutes, 14 seconds.
Done.

[Bluestraw]

That's the same thing that happened for me when I ran it on the BD directly. Try running eac3to on the flac file directly, e.g. 'eac3to audio.flac audionew.flac'.

[rack04]

When should -stripPulldown be used?

[madshi]

When should -stripPulldown be used?
IIRC it helped making stuff play for the Xbox. But if you don't know what purpose you need it for then you probably don't need it.

[BLKMGK]

Okay, trying this on Golden Compass which appears to NOT have branching...

First I tried:
D:\Video\eac3to>eac3to x:
1) 00010.mpls, 00008.m2ts, 1:53:18
- VC-1, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48khz
- DTS, English, stereo, 48khz

2) 00013.mpls, 00011.m2ts, 1:53:18
- VC-1, 1080p24 /1.001 (16:9)
- DTS, English, multi-channel, 48khz

3) 00011.mpls, 00008.m2ts, 1:53:18
- VC-1, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48khz
- DTS, English, stereo, 48khz

4) 00012.mpls, 00011.m2ts, 1:53:18
- VC-1, 1080p24 /1.001 (16:9)
- DTS, English, multi-channel, 48khz


Then I tried:
D:\Video\eac3to>eac3to x:\BDMV\STREAM\
1) 00010.mpls, 00008.m2ts, 1:53:18
- VC-1, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48khz
- DTS, English, stereo, 48khz

2) 00013.mpls, 00011.m2ts, 1:53:18
- VC-1, 1080p24 /1.001 (16:9)
- DTS, English, multi-channel, 48khz

3) 00011.mpls, 00008.m2ts, 1:53:18
- VC-1, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48khz
- DTS, English, stereo, 48khz

4) 00012.mpls, 00011.m2ts, 1:53:18
- VC-1, 1080p24 /1.001 (16:9)
- DTS, English, multi-channel, 48khz


Now, file 10 is 666KB, file 13 is non-existant, file 11 is 21.1Gigs, file 12 is 6.5megs. Looking closer I have a file 8 that is 25Gigs. So I tried:
D:\Video\eac3to>eac3to x:\BDMV\STREAM\00008.m2ts
M2TS, 1 video track, 2 audio tracks, 1:53:18
1: VC-1, 1080p24 /1.001 (16:9)
2: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
3: DTS, English, 2t channels, 24 bits, 256kbit/s, 48khz


I cannot tell if file 8 and file 11 are supposed to be together or what. I'm ripping file 8 now ->
D:\Video\eac3to>eac3to x:\BDMV\STREAM\00008.m2ts 1: f:\gold\gold-vid.mkv 2: f:\gold\gold-aud.ac3 -640 -libav
M2TS, 1 video track, 2 audio tracks, 1:53:18
1: VC-1, 1080p24 /1.001 (16:9)
2: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
3: DTS, English, 2t channels, 24 bits, 256kbit/s, 48khz
[a02] The libav DTS decoder doesn't decode the full DTS-HD information.
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Extracting DTS core...
[a02] Remapping channels...
Loading white noise (needed for dithering)...
[a02] Encoding AC3...
[v01] Muxing video to Matroska...
[a02] Creating file "f:\gold\gold-aud.ac3"...

Fingers crossed but it seems odd that ONLY DTS is listed, that file 8 wasn't listed until I specified it, and I'm concerned that the "core" from the DTS may not give me 5.1 sound - not sure on that, anyone? Cannot watch it till it's ripped so hopefully I'll be able to see what's going on then.

Ideas? Am I screwing this up? (lol) Really want to see this movie! :p

Edit: Okay start looks good, I see credits at end, length is correct! Wonder what the 21Gig piece is....

[Yraen]

In your title listings if you'll look closely you'll see that two files are actually listed: .mpls and .m2ts. 0008.m2ts and 0011.m2ts appear to be the two movie files. Does the movie have an alternate ending or deleted scenes besides the theatrical release? It looks like they did this movie like they did I Am Legend, put the movie on there twice. It is weird that it only has DTS audio though. Does the box mention anything about any other audio formats?

[BLKMGK]

Ah yeah you're right about some MPLs there - weird. There's an option for a Director's commentary but doing that as a second movie would be silly! I have Legend, not yet watched the second ending though - I did giggle at the structure on that one..

There's also mention of Dolby Digital it looks like - disk itself says Dolby HD and DTS on it but surely it has something else too since my receiver cannot even decode either of those I don't think and I'm not alone - maybe DTS but I'd have to look. Since I have no stand-alone BD player and actually no BD player software it's kind of hard to look at some stuff lol, I just rip them to drive, compress, and then watch from HTPC with XBMC!

Here's a description -> http://www.amazon.com/Golden-Compass-Blu-ray-Nicole-Kidman/dp/B00139XZF4/ref=sr_1_4?ie=UTF8&s=dvd&qid=1209508002&sr=8-4

Might rip that second one for grins. I'm about 4 hours or so away from being able to watch this one so I've got time to prep the second part :p

P.S. file 8 never showed up in any of the eac3to listings until I pointed at it specifically - it was THE largest M2TS file on there too!

[Bluestraw]

I think you're confusing the mpls files with the m2ts ones. m2ts files 8 and 11 are the ONLY ones that show up. They happen to each be listed twice, i.e. mpls files 10,11 and 12,13 are identical pairs. I've seen that before - must be a quirk of how some BDs are authored.

Looking at the runtimes, it seems 8 and 11 are identical, except one has the single soundtrack. Definitely file 8 will be the 'normal' one, and instead of adding a soundtrack they've repeated the whole movie for director commentary on 11. Maybe they also overlaid some graphics or something on 11 - would definitely be interested to hear what you find!

[BLKMGK]

Oh! Now I see it - duh! Okay, am about to rip the smaller one, number 11 now. I have to laugh, I heard on and on about how BD was better because it had more space blah blah. However I still see some movies doing silly things like having multiple copies instead of having a single movie and multiple tracks of audio. If there's nothing visually different in the second one that will be odd indeed!

D:\Video\eac3to>eac3to x:\BDMV\STREAM\00011.m2ts
M2TS, 1 video track, 1 audio track, 1:53:18
1: VC-1, 1080p24 /1.001 (16:9)
2: DTS, English, 5.1 channels, 24 bits, 1536kbit/s, 48khz

Thanks!

P.S. Got a framecount after that first rip - woohoo!

[xxx666yyy777]

@Madshi:
Very cool tool!!!

Suggestion for additional feature (or maybe I just don't understand how to use it...)

If I have a source with chapters, video and several audio streams, it would be nice to be able to issue one single command to mux, for example, the chapters, with the 2nd video stream and the 5th audio stream into one .mkv file at once. As far as I see it now, I have to demux the chapter file to a .txt, the video stream into .mkv and the audio stream into thre separate files and then mux them into another .mkv file with video, chapters and audio...

Just a thought...

Thx

[bkman]

Hi,

just a question: Any possibility of making use of Cyberlink filters for Dolby Headphone mixing?

Because that would be great.

Thanks.

[madshi]

I think you're confusing the mpls files with the m2ts ones. m2ts files 8 and 11 are the ONLY ones that show up. They happen to each be listed twice, i.e. mpls files 10,11 and 12,13 are identical pairs. I've seen that before - must be a quirk of how some BDs are authored.

Looking at the runtimes, it seems 8 and 11 are identical, except one has the single soundtrack. Definitely file 8 will be the 'normal' one, and instead of adding a soundtrack they've repeated the whole movie for director commentary on 11. Maybe they also overlaid some graphics or something on 11 - would definitely be interested to hear what you find!
That's exactly my thinking. Probably file 11 has some PIP windows hard coded to it in some parts of the movie.

Suggestion for additional feature (or maybe I just don't understand how to use it...)

If I have a source with chapters, video and several audio streams, it would be nice to be able to issue one single command to mux, for example, the chapters, with the 2nd video stream and the 5th audio stream into one .mkv file at once. As far as I see it now, I have to demux the chapter file to a .txt, the video stream into .mkv and the audio stream into thre separate files and then mux them into another .mkv file with video, chapters and audio...
eac3to is using the Haali Matroska Muxer which doesn't support muxing chapters. So what you're suggesting is technically not even possible with the code I'm using now. I'd have to write my own Matroska Muxer to make that work which is quite a lot of work.

One thing that would be possible is for eac3to is to do 2 steps:

(1) Do what it does now.
(2) Automatically call mkvtoolnix to mux everything into MKV in a 2nd step.

That would be possible, but it would take just as long as what you can do today. It would just be slightly more comfortable. So I don't know if it's worth it. Furthermore I think the Eac3to and More GUI can already do that, anyway.

just a question: Any possibility of making use of Cyberlink filters for Dolby Headphone mixing?
Most Cyberlink filters can not be used outside of PowerDVD, as far as I know.

[madshi]

I 've been dealing with Surf's Up the last few hours... The movie consists of 3 files. Joint points are at 3:19 mins and 3:49 mins.

* FLAC track made by the TrueHD track is NOT in sync (no gaps/overlaps reported for that). I also converted this track to flac alone, in a separate command line, to be sure that no overlaps/gaps are reported.

You can see that all 3 tracks are in sync before, but the TrueHD goes off sync after 1st joint. The TrueHD is leading by 30ms the AC3 before the joints and is leading 160ms after 1st (and 2nd joint). I guess there is a gap in the audio stream.
Just checked this. I can reproduce the sync problem when using the Nero decoder. But with the libav decoder audio is in sync! Well, compared to the PCM track the TrueHD track is 4ms off after the first joint point and 8ms off after the 2nd joint point. But that's within eac3to's tolerance.

* Libav is unable to decode the TrueHD track (lossless check fail).
Actually libav decodes the TrueHD track perfectly fine. *Nero* has a problem with it. Those "lossless check fail" messages just mean that the checksum calculation isn't working correctly on the join points. But that was to be expected. It doesn't affect audio quality at all.

I'll surpress these "lossless check failed" messages at join points in the next version to avoid confusion...

[Kurtnoise]

eac3to is using the Haali Matroska Muxer which doesn't support muxing chapters.
using gdsmux (the GUI included in the Haali package which uses the filters), there is one tab dedicated to the chapters. Dunno if it's available through the directshow pins though...

[madshi]

eac3to v2.44 released

http://madshi.net/eac3to.zip

* libav is now automatically used when Nero/Sonic decoders are not working
* gap/overlap correction of RAW/PCM tracks sometimes aborted
* rerunning de/remuxing to correct gaps/overlaps ignored RAW/PCM tracks
* "lossless check failed" messages are surpressed on join points now
This should fix all problems with seamless branching Blu-Rays which were reported to me - except the one problem that audio sync gets lost when using Nero to decode TrueHD tracks. But I don't consider that a big problem since libav is default for TrueHD decoding and gets it right.

[nautilus7]

Just checked this. I can reproduce the sync problem when using the Nero decoder. But with the libav decoder audio is in sync! Well, compared to the PCM track the TrueHD track is 4ms off after the first joint point and 8ms off after the 2nd joint point. But that's within eac3to's tolerance.

Actually libav decodes the TrueHD track perfectly fine. *Nero* has a problem with it. Those "lossless check fail" messages just mean that the checksum calculation isn't working correctly on the join points. But that was to be expected. It doesn't affect audio quality at all.

I checked the logs regarding all commands i run for Surf's Up. I was mistaken. I checked sync only with nero decoder, considering that libav had trouble decoding the stream correctly. So, the screens above are with nero. I am currently checking with libav and newer version.

[shambles]

i redid spider-man 2 now with 2.44 and it does indeed stay in sync.. sorry for jumping to conclusions :o

masterful job, as always. many many thanks!

[madshi]

i redid spider-man 2 now with 2.44 and it does indeed stay in sync..
So gap/overlap correction for the TrueHD track executed and it worked ok? Just asking because I haven't seen confirmation for working TrueHD gap/overlap correction yet. Thanks...

[Greif]

Since libav is the default for TrueHD, and since I do not have Nero7 or Sonic filters installed, should I always go with the TrueHD track over EAC3?

Which will result in better sound quality using libav?

[nautilus7]

Since libav is the default for TrueHD, and since I do not have Nero7 or Sonic filters installed, should I always go with the TrueHD track over EAC3?

Which will result in better sound quality using libav?

If both TrueHD and E-AC3 is available for a movie, then TrueHD is always the way to go since it uses lossless compression (while E-AC3 is a lossy format). As a result, all TrueHD decoders have to output bit identical audio streams (that's the case with nero and libav truehd decoders). On the other hand, nero E-AC3 decoder uses Dolby's reference code, while libav not that may or may not result in somewhat lower sound quality. Anyway, the difference should be very small to hear.

[shambles]

So gap/overlap correction for the TrueHD track executed and it worked ok? Just asking because I haven't seen confirmation for working TrueHD gap/overlap correction yet. Thanks...

well, no. it doesn't seem to write any gap/overlap file. here's what i got:

eac3to v2.44
command line: eac3to "E:\Spider-Man 2 Blu-Ray" 1) 2: f:\sm2.mkv 4: f:\sm2.flac
------------------------------------------------------------------------------
M2TS, 1 video track, 7 audio tracks, 2:07:27
1: Chapters, 52 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
4: TrueHD/AC3, English, 5.1 channels, 48khz
5: AC3, Czech, 5.1 channels, 448kbit/s, 48khz, dialnorm: -30dB
6: AC3, Czech, 5.1 channels, 448kbit/s, 48khz, dialnorm: -30dB
7: AC3, Polish, 5.1 channels, 448kbit/s, 48khz, dialnorm: -28dB
8: AC3, Polish, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB
9: AC3, English, 2.0 channels, 192kbit/s, 48khz, dialnorm: -30dB
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Encoding FLAC...
[v02] Muxing video to Matroska...
[a04] This audio track contains more than 16 bit of information.
[a04] Creating file "f:\sm2.24bit.flac"...
Added fps value to MKV header.
Video track 2 contains 183372 frames.
eac3to processing took 42 minutes, 4 seconds.
Done.

still the track is in sync.

earlier in this thread, drmpeg posted that truehd pes granularity is 0.83 ms (http://forum.doom9.org/showthread.php?p=1076017#post1076017), so each overlap should be less than 1ms. does eac3to list overlaps that are less than 1ms?

[Bluestraw]

I re-tested Ratatouille and the 2 problems I had are both fixed - it DOES now pick up the GAPS file on re-running the command-line, and also if you process the flac by itself it no longer errors out.

Great job Madshi - once again :)

[nautilus7]

does eac3to list overlaps that are less than 1ms?No. Overall overlap should be more than 40ms to be displayed.

I suppose we 'll never see overlaps/gaps for TrueHD streams. :eek:

BTW, i redone the TrueHD stream from Surf's Up and it's perfect now.

[madshi]

well, no. it doesn't seem to write any gap/overlap file.

[...]

earlier in this thread, drmpeg posted that truehd pes granularity is 0.83 ms (http://forum.doom9.org/showthread.php?p=1076017#post1076017), so each overlap should be less than 1ms.
That makes sense. But there's one thing I don't understand: I've heard from different people that audio went out of sync for them when remuxing Spider-Man 2. Well, that was before eac3to supported m2ts joining, of course. But still if the TrueHD tracks have no noticable overlaps then I'm wondering why sync ever was a problem with Spider-Man 2.

I re-tested Ratatouille and the 2 problems I had are both fixed
BTW, i redone the TrueHD stream from Surf's Up and it's perfect now.
Good to hear! :)

[nautilus7]

That makes sense. But there's one thing I don't understand: I've heard from different people that audio went out of sync for them when remuxing Spider-Man 2. Well, that was before eac3to supported m2ts joining, of course. But still if the TrueHD tracks have no noticable overlaps then I'm wondering why sync ever was a problem with Spider-Man 2.Maybe they decoded with nero and run into the same problem i did with Surf's Up. Maybe nero can't deal with joint truehd streams at all. Maybe it should be blocked with seamless branching blu-rays. Time will tell i guess...

[rack04]

Here's what I recommend (and what I'm using myself):

(1) Extend the search path to include eac3to's folder. In Windows XP you can do that by:

right click on My Computer -> "extended" tab -> "environment variables" button

In the first list double click on "PATH". Then add ";c:\program files\eac3to" (replace that path with the one you're using on your PC) to the list of paths. This change will make sure that you can always directly type "eac3to" in the command line, no matter in which folder you are. Windows will then automatically find eac3to.exe for you. Very comfortable. You may need to log off/on to make the "PATH" change active.

(2) In the command line go to the folder where you want to have the files demuxed to. E.g. if you want to have the movie "c:\fresh rips\some HD DVD movie" demuxed to "c:\demuxed\some HD DVD movie", you should in the command line go to "c:\demuxed\some HD DVD movie". I mean the command line window should look like this "c:\demuxed\some HD DVD movie> _".

(3) Now type:

eac3to "c:\fresh rips\some HD DVD movie" -demux

This will demux everything to "c:\demuxed\some HD DVD movie".

MKVToolnix is already defined in the PATH. Will changing this to eac3to cause problems with MKVToolnix?

[shambles]

Maybe they decoded with nero and run into the same problem i did with Surf's Up. Maybe nero can't deal with joint truehd streams at all. Maybe it should be blocked with seamless branching blu-rays. Time will tell i guess...

i did spider-man 2 with both libav and nero before eac3to had branching support (as in 1.thd+2.thd+...) and both were massively out of sync by the end.

but sync was fine when doing the same files one by one with eac3to and then joining them after (with libav, didn't try nero this way iirc).

[madshi]

MKVToolnix is already defined in the PATH. Will changing this to eac3to cause problems with MKVToolnix?
No, not at all.

i did spider-man 2 with both libav and nero before eac3to had branching support (as in 1.thd+2.thd+...) and both were massively out of sync by the end.
Strange...

[bkman]

Most Cyberlink filters can not be used outside of PowerDVD, as far as I know.

Maybe there is a special way to talk to them..? A secret handshake? :P

It's just the Dolby Headphone sounds AMAZING, and it's very elusive to encode or convert to...

[nautilus7]

Regarding Cyberlink decoders, i 've read that new PowerDVD 9 supports full DTS-HD decoding now, so maybe it worths spending some time with that decoder in case it supports 7.1 channels. We need a 7.1 channel DTS-HD decoder.

[Rectal Prolapse]

Wow I thought PowerDVD 8 (EIGHT) was the latest. :)

[nautilus7]

Yes... 8 is the latest. :D

I messed up versions with WinDVD 9. :D

[Thunderbolt8]

madshi, could you please at least add a switch or something like that that the user is able get rid of that seamless branching delay, even in case its below those 40ms for the movie?

i know that such low values are most likely not perceivable, but for me the point is more coming as close to the source as possible. eac3to provides best = lossless results when it comes to adapting the picture and sound 1 : 1 (flac conversion), but in the respect with these little overlaps it gets a bit away from that point, though its technically possible. when playing the complete movie with a commercial programm like powerdvd or in a hardware player then those gaps are/should not be present as well and this is then the way a lossless 'copy' (=remux) to another format should look like as well.
i know that another pass is most likely needed then, but for me its not a question of time, whether eac3to needs 30 mins or 2 hours to finish, since its a singular process anyway for each movie.

so it would be nice if you give the user at least the choice to decide himself if he likes to bring it to perfection or chooses the more practical way, which is 'less closer' to the source though. its a matter of feeling perfection ;)

[Rectal Prolapse]

40 ms is too large - especially for those with displays with large input lag (ie. some LCD monitors/TVs have 40 to 60 ms of input lag - add on 40 ms and things will be very noticeably out of sync).

[saint-francis]

I would like to make a feature request/suggestion. I think it would be helpful if there were an option in eac3to to set up profiles. For example I never use a flack track. My sound system can almost do justice to a 640 Kbps ac3 and I have absolutely NO plans on upgrading it for years to come. By then people will be laughing at the poor quality of the DTS-HD tracks they used to have in BD. I always erase the flack track when I do eac3to.exe source movie.mkv but I appreciate why that automatically encoding to flack function is there. Also just about every 2.0 channel track I convert to ac3 and then ultimately to AAC. If I could just once make a profile that would mux the video to .mkv, convert all 5.1 channel and up tracks to ac3 and any 2.0 channel tracks to AAC while demuxing the subtitles and chapter file it would save me an enormous amount of time and potentially save me from carpel tunnel syndrome. I know that eac3to is under pretty heavy development as it is just trying to iron out all of the kinks and complete it's current functionality so this is just food for thought.

P.S. madshi, you rock.

[bmnot]

hey madshi, I'm joining two m2ts' into an mkv. The first one is 99% of the movie, and the 2nd one is the credits. The 2nd one comes in Japanese or English. I'm joining the Japanese, so it's 00000.m2ts+00001.m2ts

Once it gets to the credits m2ts I get the error:
"[v01] The h264 muxer received invalid h264/AVC data.
Aborted at file position 30684829696."

What should I do the successfully join these? This is with v2.44. Both m2ts' convert to mkv fine on their own.

EDIT: Also errored with the M:\Movie 1) setup

[gregt]

Hi,
Not sure if the following is by design or if it is a bug:
D:\video>eac3to "Ice Age The MeltDown" 2: "Ice Age The MeltDown.mkv" 3: "Ice Age
The MeltDown.flac"
M2TS, 1 video track, 5 audio tracks, 1:30:36
1: Chapters, 25 chapters
2: MPEG2, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
4: AC3, French, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB
5: AC3, Spanish, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB
6: AC3, English, 2.0 channels, 224kbit/s, 48khz, dialnorm: -27dB
7: AC3, English, 2.0 channels, 224kbit/s, 48khz, dialnorm: -27dB
[a03] The Sonic decoder doesn'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...
[v02] Extracting video track number 2...
[a03] The libav decoder received a non-supported data format.
[v02] Muxing video to Matroska...
Aborted at file position 65536.

D:\video>eac3to "Ice Age The MeltDown" 2: "Ice Age The MeltDown.mkv" 3: "Ice Age
The MeltDown.flac" -libav
M2TS, 1 video track, 5 audio tracks, 1:30:36
1: Chapters, 25 chapters
2: MPEG2, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
4: AC3, French, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB
5: AC3, Spanish, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB
6: AC3, English, 2.0 channels, 224kbit/s, 48khz, dialnorm: -27dB
7: AC3, English, 2.0 channels, 224kbit/s, 48khz, dialnorm: -27dB
[a03] The libav DTS decoder doesn't decode the full DTS-HD information.
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Extracting DTS core...
[v02] Muxing video to Matroska...
[a03] Remapping channels...
Loading white noise (needed for dithering)...
[a03] Encoding FLAC...
[a03] Creating file "Ice Age The MeltDown.flac"...
Added fps value to MKV header.
Video track 2 contains 130338 frames.
eac3to processing took 32 minutes, 11 seconds.
Done.

The first run it tried to use libav automatically I think, but failed for some reason. The 2nd run I specified -libav and it worked correctly. Just thought I would report it in case it was important.

[Bluestraw]

Madshi,

How does eac3to handle multiple languages within a single mpls file? For example, on the Cars BD, my understanding is that within a single mpls file there are different orders based on the language chosen. If you examine the mpls file in a text editor, you can see the languages and numbers listed. Not sure which language it would pick by default today?

I uploaded an example mpls file here:

http://rapidshare.com/files/111721488/00072.mpls.html

[madshi]

Maybe there is a special way to talk to them..? A secret handshake? :P
Regarding Cyberlink decoders, i 've read that new PowerDVD 9 supports full DTS-HD decoding now, so maybe it worths spending some time with that decoder in case it supports 7.1 channels. We need a 7.1 channel DTS-HD decoder.
That's all nice and fine. But as long as you don't find a way to make these filters work outside of PowerDVD, we don't even need to begin to think about using the filters in eac3to. Try making them work in GraphEdit. If you succeed, let me know... ;)

madshi, could you please at least add a switch or something like that that the user is able get rid of that seamless branching delay, even in case its below those 40ms for the movie?
You know what? I already expected you to complain about those 40ms... ;)

i know that another pass is most likely needed then, but for me its not a question of time, whether eac3to needs 30 mins or 2 hours to finish, since its a singular process anyway for each movie.
It's not a question of processing time at all. eac3to already uses 2 steps for gap/overlap correction.

40 ms is too large - especially for those with displays with large input lag (ie. some LCD monitors/TVs have 40 to 60 ms of input lag - add on 40 ms and things will be very noticeably out of sync).
If your display has an input lag you should adjust your receiver or media player to delay audio accordingly.

Anyway, I hear you. I can lower the threshold to something lower than 40ms. But there's one thing you should know: The timestamps are not 100% reliable. If I lower the threshold too much, we'll get false alarms. I mean then eac3to will report audio gaps/overlaps for movies which don't really have any gaps/overlaps. We don't want that, do we? So it's kind of difficult to find the right threshold. But maybe 40ms is really a bit too much. I chose this value because an AC3 frame is usually 32ms long and I wanted to allow one AC3 frame "off" without reporting a gap/overlap.

I would like to make a feature request/suggestion. I think it would be helpful if there were an option in eac3to to set up profiles. For example I never use a flack track.
I generally find this suggestion useful. Actually I thought about it myself already. However, I think it's not time for such features yet. First I need to get everything working alright, fix all bugs etc. Afterwards I may think about comfortability features like this.

hey madshi, I'm joining two m2ts' into an mkv. The first one is 99% of the movie, and the 2nd one is the credits. The 2nd one comes in Japanese or English. I'm joining the Japanese, so it's 00000.m2ts+00001.m2ts

Once it gets to the credits m2ts I get the error:
"[v01] The h264 muxer received invalid h264/AVC data.
Aborted at file position 30684829696."
Hmmmm... That's quite strange. Could you send me the last 20MB of 00000.m2ts and the first 20MB of 00001.m2ts?

Not sure if the following is by design or if it is a bug: [...] The first run it tried to use libav automatically I think, but failed for some reason. The 2nd run I specified -libav and it worked correctly. Just thought I would report it in case it was important.
Thanks, that's a useful bug report. Will fix that in the next build.

How does eac3to handle multiple languages within a single mpls file? For example, on the Cars BD, my understanding is that within a single mpls file there are different orders based on the language chosen.
No, there are no different orders for different languages in one mpls file. If there are different orders for different languages, then that is realized through using multiple mpls files. So there is no problem.

[Bluestraw]

Anyway, I hear you. I can lower the threshold to something lower than 40ms. But there's one thing you should know: The timestamps are not 100% reliable. If I lower the threshold too much, we'll get false alarms. I mean then eac3to will report audio gaps/overlaps for movies which don't really have any gaps/overlaps. We don't want that, do we? So it's kind of difficult to find the right threshold. But maybe 40ms is really a bit too much. I chose this value because an AC3 frame is usually 32ms long and I wanted to allow one AC3 frame "off" without reporting a gap/overlap.Would it be possible to make it user-configurable? Maybe you could set it at your preferred 'default' but then let certain 'perfectionists' experiment on a per-movie basis to get the desired result?

No, there are no different orders for different languages in one mpls file. If there are different orders for different languages, then that is realized through using multiple mpls files. So there is no problem.Are you absolutely sure about this? For example, in the Cars BD I was talking about, there are 3 language versions. All the playlists shown by eac3to are identical, yet different languages do result in a different required join order. The same thing happens in Open Season - there are different orders for the different languages, simply in order to show the movie title on screen in the right language, and all this in the single mpls file.

If you load that mpls file I sent above into the latest version of BDEdit, you'll see that it contains 3 'angles' - I presume these correspond to the 3 languages. There's more about mpls files here, though it doesn't actually cover the multi angles so well:

http://forum.doom9.org/showpost.php?p=998714&postcount=1543

[madshi]

Are you absolutely sure about this? For example, in the Cars BD I was talking about, there are 3 language versions. All the playlists shown by eac3to are identical, yet different languages do result in a different required join order. The same thing happens in Open Season - there are different orders for the different languages, simply in order to show the movie title on screen in the right language, and all this in the single mpls file.

If you load that mpls file I sent above into the latest version of BDEdit, you'll see that it contains 3 'angles' - I presume these correspond to the 3 languages.
Hmmmm... You're right, I forgot about angles. eac3to always uses the first angle, IIRC. Do you need access to the other ones, too? I don't think the order of the parts changes, though. Instead different languages are using different parts, I believe.

[Bluestraw]

Hmmmm... You're right, I forgot about angles. eac3to always uses the first angle, IIRC. Do you need access to the other ones, too? I don't think the order of the parts changes, though. Instead different languages are using different parts, I believe.That's right I think - e.g. you could have

0000
0001
0004
0005
0008

as the English version,

0000
0002
0004
0006
0008

as the German one

0000
0003
0004
0007
0008

as the French one.

where for example part 0001 has some on-screen English text, 0002 has German and 0003 has French. From what I've seen so far, each one has all the audio tracks, so for example you could watch the 'French version' but keep the English soundtrack.

From my pov, I always want the English one, so it depends for me whether or not which angle that is. My guess would be that I'd typically be lucky and get the first one, but others (like you!) may not find their first choice available!

[madshi]

others (like you!) may not find their first choice available!
Actually I want the English one, too, which is usually the first. But there may be other people who want their local version. So I guess angle support would make sense...

[Bluestraw]

Actually I want the English one, too, which is usually the first. But there may be other people who want their local version. So I guess angle support would make sense...:) Sorry didn't mean to offend, I thought I remember you saying you weren't English. Not that I could tell from what you write - impeccable English!

[nautilus7]

Surf's Up consists of 3 m2ts files. The 2nd is language specific. It might uses angles too.

[Thunderbolt8]

Anyway, I hear you. I can lower the threshold to something lower than 40ms. But there's one thing you should know: The timestamps are not 100% reliable. If I lower the threshold too much, we'll get false alarms. I mean then eac3to will report audio gaps/overlaps for movies which don't really have any gaps/overlaps. We don't want that, do we? So it's kind of difficult to find the right threshold. But maybe 40ms is really a bit too much. I chose this value because an AC3 frame is usually 32ms long and I wanted to allow one AC3 frame "off" without reporting a gap/overlap.

false alarms would indeed be really bad, you are right that we dont want that. hm is there maybe a way to combine the gap/overlap checking with the audio tracks used? for example if I only decide to use a pcm or truehd/dts-hd and no ac3 track for my muxing at all would it be possible then to have that checking only apply to those tracks, which have quite shorter frames as ac3?
or would this be too complicated?

[bmnot]

Hmmmm... That's quite strange. Could you send me the last 20MB of 00000.m2ts and the first 20MB of 00001.m2ts?

Ok, I will, but I don't understand how to use this HxD program.

[nautilus7]

Ok, I will, but I don't understand how to use this HxD program.You load the file. Then go to edit--> select block and set the size (20MB) in bytes. Make sure dec is chosen. To select the start of the file you put 0 to start-offset and 20971520 in end-offset. To select the end, click the mouse to the end of the file and then put in the start-offset (end-offset - 20971520). Final step is to copy-paste the selected part to a new file. Be careful not to modify the original file.

[Snowknight26]

Or use clip.exe (http://www.stfcc.org/misc/clip.exe).

usage: clip <infile> <outfile> <start offset> <length>

[Rectal Prolapse]

madshi, Ratatouille also makes use of multi-angles for different languages.

Is there a way to lower the overlap/gap threshold only for m2ts files that are to be joined from a single playlist, and only at the endpoints of the m2ts files? That may give better results overall - for multiple m2ts and single m2ts files.

Anyways - thanks for the great tool - again. :)

[jchappo]

howdy, been using your nice program here for awhile. I recently ran into a problem decoding DTS-HD to WAV using Sonic Decoder 4.3. When remuxing the WAV file back with the video in M2TS container, TsMuxer reports the WAV has no channel info.

When playing the movie back over a popcorn hour A-100, the audio plays fine for about 20-30% of the movie, then it goes all screetchy and is basically garbage. I can't determine what the problem is. How to I test if Sonic is not causing the problem? It's weird because the audio plays fine for a little while, then goes bad for some reason.

[itsancho]

hi all
strange problem with Basic Instinct, here is the log:

eac3to v2.44
command line: eac3to244\eac3to "K:\Basic Instinct (1992) BluRay AVC dts-HD Hi-Res" 1) 2: i:\video.mkv
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 2:08:15
1: Chapters, 17 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Hi-Res, English, 6.1 channels, 24 bits, 3093kbit/s, 48khz, dialnorm: -4dB
4: AC3, English, 5.1 channels, 640kbit/s, 48khz, dialnorm: -27dB
5: AC3, English, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB
6: AC3, English, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
Unfortunately the Haali Muxer cannot handle this source file.
It doesn't contain enough seek/recovery points.
Aborted at file position 142065664. i've tried to remux raw h264 stream (using 3 different programs) and then mux it with mkvtollnix, but... :-( runtime was 2:09:44
big 10x in advance 4 helping with this one!

Yraen, nope, for sorry that is not the problem, with or without any " the result is the same...

[Yraen]

Looks like you missed an " here:
i:\video.mkv"

That might be the problem. Try it and if it fails post that log.

[bmnot]

Hmmmm... That's quite strange. Could you send me the last 20MB of 00000.m2ts and the first 20MB of 00001.m2ts?
Ok here you go: http://www.mediafire.com/?g499bligrz0

end of first m2ts doesn't play, i hope it did right

BTW the end of the file is at 30684420096, while the error is at 30684829696, BUT the file converts to mkv fine when there is no joining.

[gregt]

Hi,
I seem to have run into another bug as follows:
M:\to convert\SciFi\bluray>eac3to UNDERWORLD
1) 00106.mpls, 2:13:39
[93+94+95+96+97+98+99+100].m2ts
- MPEG2, 480i30 /1.001 (4:3)
- AC3, English, stereo, 48khz

2) 00011.mpls, 00011.m2ts, 2:13:37
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, multi-channel, 48khz
- AC3, English, multi-channel, 48khz
- AC3, French, multi-channel, 48khz
- RAW/PCM, Italian, multi-channel, 48khz
- AC3, Italian, multi-channel, 48khz
- AC3, English, stereo, 48khz

3) 00095.mpls, 00093.m2ts, 0:47:18
- MPEG2, 480i30 /1.001 (4:3)
- AC3, English, stereo, 48khz

4) 00101.mpls, 00099.m2ts, 0:19:13
- MPEG2, 480i30 /1.001 (4:3)
- AC3, English, stereo, 48khz

M:\to convert\SciFi\bluray>eac3to UNDERWORLD UNDERWORLD.mkv
M2TS, 1 video track, 1 audio track, 2:13:38
1: Chapters, 13 chapters
2: MPEG2, 480i60 /1.001 (4:3)
3: AC3, English, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB
Creating file "UNDERWORLD - Chapters.txt"...
[a03] Extracting audio track number 3...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[a03] Removing dialog normalization...
[a03] Creating file "UNDERWORLD - 3 - AC3, English, 2.0 channels, 192kbps, 48khz
.ac3"...
[v02] Video overlaps for 1 frames at playtime 0:47:18.
[v02] Video overlaps for 1 frames at playtime 1:00:21.
[v02] Video overlaps for 1 frames at playtime 1:10:18.
[v02] Video overlaps for 1 frames at playtime 1:22:48.
[v02] Video overlaps for 1 frames at playtime 1:34:31.
[v02] Video overlaps for 1 frames at playtime 1:45:18.
[v02] Video overlaps for 1 frames at playtime 2:04:31.
[v02] The MKV file was created without making use of the gap/overlap information
.
[v02] Please check whether audio is in sync. If it is in sync everything is fine
.
[v02] Otherwise you can ask eac3to to repeat the muxing. It will then automatica
lly
[v02] make use of the detailed gap/overlap information.
[a03] Audio overlaps for 31ms at playtime 0:47:18.
[a03] Audio overlaps for 42ms at playtime 1:00:21.
[a03] Audio overlaps for 27ms at playtime 1:10:18.
[a03] Audio overlaps for 24ms at playtime 1:22:48.
[a03] Audio overlaps for 46ms at playtime 1:34:31.
[a03] Audio overlaps for 39ms at playtime 1:45:18.
[a03] Audio overlaps for 25ms at playtime 2:04:31.
[a03] The audio file was demuxed without making use of the gap/overlap informati
on.
[a03] Please rerun the same eac3to command line. That will correct the gaps/over
laps.
Added fps value to MKV header.
Video track 2 contains 240318 frames.
eac3to processing took 8 minutes, 15 seconds.
Done.

For some reason the main (listed under 2) movie is not converted by default. The main movie file is 38GB so may be that is the reason. When I went into the stream directory and converted the 00011.m2ts file directly it converted correctly.

[madshi]

:) Sorry didn't mean to offend, I thought I remember you saying you weren't English. Not that I could tell from what you write - impeccable English!
You didn't offend me at all, don't worry. You are right in that my native language is not English. So it makes sense that you thought I'd prefer having my native language muxed. But it's just not that way. And thanks for the compliment on my English capabilities... :) I guess reading/writing English every day helps!

false alarms would indeed be really bad, you are right that we dont want that. hm is there maybe a way to combine the gap/overlap checking with the audio tracks used? for example if I only decide to use a pcm or truehd/dts-hd and no ac3 track for my muxing at all would it be possible then to have that checking only apply to those tracks, which have quite shorter frames as ac3?
Not sure if I've understood you correctly. eac3to already checks only those tracks you are demuxing.

Is there a way to lower the overlap/gap threshold only for m2ts files that are to be joined from a single playlist, and only at the endpoints of the m2ts files? That may give better results overall - for multiple m2ts and single m2ts files.
That sounds like a good idea in theory. The problem is that I've two different thresholds: (1) For one gap/overlap it's 5ms. (2) For all gaps/overlaps combined it's 40ms. The small threshold for one gap/overlap is not problematic because even if the timestamps are unstable, they are always fluctuating around the right value. And thanks to 2 pass processing eac3to can clear out most short time fluctuations.

I guess I'll try lowering the threshold in the next build. Maybe to 20ms. And maybe I can also offer a switch to lower it even further. But I'll depend on you guys reporting your results with the thresholds, so that we can find the optimal default value (which would be the lowest possible value with which there are no false alarms with 99.9% of the movies).

I recently ran into a problem decoding DTS-HD to WAV using Sonic Decoder 4.3. When remuxing the WAV file back with the video in M2TS container, TsMuxer reports the WAV has no channel info.
There are two different types of WAV headers. Some programs prefer one type, others prefer the other type. eac3to by default writes the simpler type. It can also write the more complicated one, but currently there's no option for that. Will add that in the next build. It seems that TsMuxer wants to have the more complicated type. That most likely doesn't have anything to do with the garbage problems you're having, though...

When playing the movie back over a popcorn hour A-100, the audio plays fine for about 20-30% of the movie, then it goes all screetchy and is basically garbage. I can't determine what the problem is. How to I test if Sonic is not causing the problem?
By playing the WAV file itself (without the video/movie) on your HPTC or on your popcorn hour.

strange problem with Basic Instinct, here is the log:

Unfortunately the Haali Muxer cannot handle this source file.
It doesn't contain enough seek/recovery points.
There's a bug in the Haali Matroska Muxer which results in a total freeze when muxing some h264 movies. eac3to detects this situation and aborts processing directly before the Haali Matroska Muxer freezes. Currently there's no way to properly mux such a movie with eac3to respectively with Haali's Matroska Muxer. You can use the option "-seekToIFrames" to mux the movie in a different way. This should succeed. However, this comes at a cost: Sometimes when seeking such a movie there are very noticable image artifacts for a short time (max a few seconds). Haali knows about this problem but hasn't fixed it yet, sadly.

Ok here you go: http://www.mediafire.com/?g499bligrz0

end of first m2ts doesn't play, i hope it did right

BTW the end of the file is at 30684420096, while the error is at 30684829696, BUT the file converts to mkv fine when there is no joining.
Thanks, will check this out. If it's a bug in eac3to, it will be fixed in the next build.

I seem to have run into another bug as follows: [...] For some reason the main (listed under 2) movie is not converted by default. The main movie file is 38GB so may be that is the reason. When I went into the stream directory and converted the 00011.m2ts file directly it converted correctly.
There's no easy to find information (I know of) in the Blu-Ray disc structure which tells us which is the main movie. Consequently eac3to can only list all playlists sorted by their runtime. The playlist with the longest runtime is listed first cause that's usually the movie. In your case it's not the movie. So you have to do "eac3to UNDERWORLD 2) UNDERWORLD.mkv".

[sundansx]

Madshi,
first, thanks for a great program.
I am working on Serenity HDDVD and it fails to use libav when I specify it with other streams as shown below. It tries to use nero, which I dont have installed. When I run a pass with the audio separately followed by a separate pass to pick up the rest it works fine and uses libav. As a note, this same thing also happened on Beowulf. I dont think it is title dependent.

F:\VideoStore\DVD\Serenity>c:\utils\HDCruncher\eac3to.exe d:\HVDVD_TS 1) 2: chap
ters.txt 3: out.mkv 4: out.ac3 8: subtitle.sup -libav -down6 -640 -24
EVO, 1 video track, 4 audio tracks, 3 subtitle tracks, 1:59:01
"MainMovie"
1: Joined EVO file
2: Chapters, 20 chapters with names
3: VC-1, 1080p24 /1.001 (16:9)
4: E-AC3, English, 5.1 channels, 1536kbit/s, 48khz, dialnorm: -27dB
5: E-AC3, Spanish, 5.1 channels, 768kbit/s, 48khz, dialnorm: -27dB
6: E-AC3, French, 5.1 channels, 768kbit/s, 48khz, dialnorm: -27dB
7: AC3, English, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB
"Director's Commentary"
8: Subtitle, English, "SDH"
9: Subtitle, Spanish
10: Subtitle, French
Creating file "chapters.txt"...
[a04] The Nero decoder doesn't seem to work, will use libav instead.
[v03] Extracting video track number 3...
[v03] Muxing video to Matroska...
[a04] Extracting audio track number 4...
[a04] Removing dialog normalization...
[a04] Decoding with DirectShow (Nero Audio Decoder 2)...
[a04] Getting "Nero Audio Decoder 2" instance failed.
Aborted at file position 32768.

[Beastie Boy]

The -libav switch should be used in the audio 'group of the command line, so your command lines becomes:
F:\VideoStore\DVD\Serenity>c:\utils\HDCruncher\eac3to.exe d:\HVDVD_TS 1) 2: chap
ters.txt 3: out.mkv 4: out.ac3 -libav 8: subtitle.sup

Note that -640 is not required as that is the default.

Cheers, Beastie

[monohouse]

hi, I was wondering, is there a way to use the libav/ffmpeg TrueHD decoder in real time ? as in decoding the stream as it being played ? not for reconversion purpose ?

I have demuxed the truehd track using evodemux and now wondering how to replace the FLAC I created with the truehd track that is present, is there a program that is used to do this or evodemux can do this also ? I seemed to notice that eac3to appears to be able to mux it as well, but I was unable to find the manual for eac3to.

I have a few questions: why doesn't eac3to tell the original bit-depth of the stream ? only resolution and channels, and can this stream be directly muxed with a video without additional operations, I understand that the use of FLAC in video is new, so I am not entirely sure if any additional procedures are required, are there any guides that explain how to handle evo+truehd ?

[itsancho]

There's a bug in the Haali Matroska Muxer which results in a total freeze when muxing some h264 movies. eac3to detects this situation and aborts processing directly before the Haali Matroska Muxer freezes. Currently there's no way to properly mux such a movie with eac3to respectively with Haali's Matroska Muxer. You can use the option "-seekToIFrames" to mux the movie in a different way. This should succeed. However, this comes at a cost: Sometimes when seeking such a movie there are very noticable image artifacts for a short time (max a few seconds). Haali knows about this problem but hasn't fixed it yet, sadly. BIG, BIG 10x madshi!!! Everything is absolutely OK and btw there is no any "seeking artifacts"!

[crazydane]

Madshi, do you know if Sonic plans to fix their DTS decoder so that it can handle DTS-HD MA 7.1 titles properly?

Alternatively, do you know if there is a libav/ffmpeg DTS decoder in the works that can replace the Sonic one that does 7.1?

Granted, there aren't that many 7.1 titles yet, but I do own 6 so far and would love to be able to play them off my media server in their full 7.1 glory. Until then, I'll bitstream them from my Panny BD30 player to my Onkyo 885 pre/pro.

And thanks again for this wonderful program!

[moshmothma]

There's a bug in the Haali Matroska Muxer which results in a total freeze when muxing some h264 movies. .... .... Haali knows about this problem but hasn't fixed it yet, sadly.


And others Madshi. This is exactly why we need another dshow splitter from someone who understands the issues and is willing to respond to them. Please reconsider this. No pressure though :)

BTW, have you ever considered writing an editing app? Seems like you have the basis of all you would need at this point. Just wondering.

[Bluestraw]

I haven't seen it yet myself, but I heard promising things about this editing app due for release soon...

(sorry for going slightly OT - but the point being that I guess Madshi's skills may be better invested elsewhere!)

EDIT - oops I forgot the URL first time around!:

http://bitstreamtools.com/

[wildchild22]

I have a question I have just finished spiderman2 the ac3 track I made and dts track made with eac3to are fine but the lpcm track around 35% way though goes all static and then the volume of what should be heard is very low. ( the funny thing is the ac3 and dts track are made from the wav) I am running the eac3to wav file through delay cut and I am going to re-encode again. I have tested playback on both the ps3 and also the istar mini hd 1.3 same result.

[jchappo]

Madashi, I have more info regarding the bad DTS-HD to WAV conversion.

It seems that even DD+ tracks that I convert to WAV suffer from the same problem. Someone on the A-100 forum said it was related to the >2gb WAV header problem? They also said if they split the file up and then convert it, each slice plays perfect.

Any ideas? The popcorn hour A-100 passes the PCM audio directly to my receiver.

Edit: just played the WAV file by itself, it plays fine for the first 10% or so, then it sounds like people are talking in slow mo, and there is no background noise. Does this mean it is a decoder problem? It doesn't seem like eac3to is producing the same LPCM track that ships with some Bluray discs.

[saint-francis]

Okay, trying this on Golden Compass which appears to NOT have branching...

First I tried:
D:\Video\eac3to>eac3to x:
1) 00010.mpls, 00008.m2ts, 1:53:18
- VC-1, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48khz
- DTS, English, stereo, 48khz

2) 00013.mpls, 00011.m2ts, 1:53:18
- VC-1, 1080p24 /1.001 (16:9)
- DTS, English, multi-channel, 48khz

3) 00011.mpls, 00008.m2ts, 1:53:18
- VC-1, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48khz
- DTS, English, stereo, 48khz

4) 00012.mpls, 00011.m2ts, 1:53:18
- VC-1, 1080p24 /1.001 (16:9)
- DTS, English, multi-channel, 48khz



Fingers crossed but it seems odd that ONLY DTS is listed, that file 8 wasn't listed until I specified it, and I'm concerned that the "core" from the DTS may not give me 5.1 sound - not sure on that, anyone? Cannot watch it till it's ripped so hopefully I'll be able to see what's going on then.

Edit: Okay start looks good, I see credits at end, length is correct! Wonder what the 21Gig piece is....

Did you figure out why each is listed twice? I am tinkering with this one right now and I noticed that the first file (00008.m2ts I believe) is the main movie and the second file (00011.m2ts) is commentary with a picture in picture of extras and the like. So that is still only two movies. Why are there four listed? Also no chapters?

[madshi]

hi, I was wondering, is there a way to use the libav/ffmpeg TrueHD decoder in real time ?
There are some mplayer builds with the latest libav decoders in them, I think. But I'm not sure.

I have demuxed the truehd track using evodemux and now wondering how to replace the FLAC I created with the truehd track that is present
I don't understand what you mean. Can you explain?

I have a few questions: why doesn't eac3to tell the original bit-depth of the stream ? only resolution and channels
Of which stream? TrueHD? TrueHD streams have no fixed bit-depth. They are all encoded in 24bit, but some of those 24bit may be zeroed out, which would make the track lower than 24bit. eac3to shows the bitdepth of all tracks which have a fixed bitdepth.

and can this stream be directly muxed with a video without additional operations, I understand that the use of FLAC in video is new, so I am not entirely sure if any additional procedures are required, are there any guides that explain how to handle evo+truehd ?
You're confusing me. One time you're talking about FLAC, then you're talking about TrueHD. Then you're talking about muxing and EVO. I've no idea what you really want to do.

Madshi, do you know if Sonic plans to fix their DTS decoder so that it can handle DTS-HD MA 7.1 titles properly?
I've no idea.

Alternatively, do you know if there is a libav/ffmpeg DTS decoder in the works that can replace the Sonic one that does 7.1?
Not that I knew of.

This is exactly why we need another dshow splitter from someone who understands the issues and is willing to respond to them. Please reconsider this.
No. :)

BTW, have you ever considered writing an editing app? Seems like you have the basis of all you would need at this point. Just wondering.
I thought about it, but I don't plan to do this. Now we have Blu-Ray which is practically perfectly cut/edited. So I don't really see the need for an editing app, anymore. It might be useful for broadcasts. But for my personal needs Blu-Ray will more and more replace broadcasts...

I have a question I have just finished spiderman2 the ac3 track I made and dts track made with eac3to are fine but the lpcm track around 35% way though goes all static and then the volume of what should be heard is very low. ( the funny thing is the ac3 and dts track are made from the wav)
If the ac3 and dts tracks are made from the WAV play ok then probably the WAV is fine and the PS3 and Istar are at fault. As you may have heard, WAV "officially" doesn't support files bigger than 4GB. Some applications and media players even have problems with WAV files bigger than 2GB. eac3to supports >4GB WAVs (just like some other applications).

It seems that even DD+ tracks that I convert to WAV suffer from the same problem. Someone on the A-100 forum said it was related to the >2gb WAV header problem? They also said if they split the file up and then convert it, each slice plays perfect.

Any ideas? The popcorn hour A-100 passes the PCM audio directly to my receiver.

Edit: just played the WAV file by itself, it plays fine for the first 10% or so, then it sounds like people are talking in slow mo, and there is no background noise. Does this mean it is a decoder problem? It doesn't seem like eac3to is producing the same LPCM track that ships with some Bluray discs.
Please try to play the WAV file on your PC. Does it play fine there? To be honest, I'm not even sure if the PC WAV source filter supports >4GB (or >2GB) files, though. The best test would be to ask eac3to to convert the WAV file to AC3. If the final AC3 file plays fine from beginning to end then the WAV file is probably alright. The problem is most probably that all the playback software/hardware does not support WAV files this big. There's nothing I can do about it. You need to ask those people who don't support big WAV files to add support for it, if you actually have to use WAV. I'd suggest using FLAC instead, if possible.

Did you figure out why each is listed twice? I am tinkering with this one right now and I noticed that the first file (00008.m2ts I believe) is the main movie and the second file (00011.m2ts) is commentary with a picture in picture of extras and the like. So that is still only two movies. Why are there four listed? Also no chapters?
Most probably those 2 with the Master Audio track in them are the real movie and the other 2 are the encodings with the PIP in them. Don't know why there are 2 of each. But it doesn't matter much. eac3to only takes the m2ts part numbers from the playlist. So it doesn't matter which of the double playlists you're using for conversion. You will only have to decide between the one with the Master Audio track and the one without.

[robena]

Thanks for this great program.

I have a problem though. I am trying to demux and speedown an AC3 track from an UK Sky cap.

Here is what I get:

d:\m1\cmd\eac3to\eac3to "test.ts" 2: "test.ac3" -slowdown -384
This doesn't seem to be a valid (E-)AC3 stream.
TS, 1 video track, 0:45:05
1: h264/AVC, 1080i50 (16:9)
The source file doesn't contain a track with the number 2.


The file does have an AC3 track (stream_type = 0x81) associated with pid 0x14.

I am using version 2.44. Nero 7 is installed:

8{prob}% d:\m1\cmd\eac3to\eac3to -test
Sonic Audio Decoder (2.44.0.0) doesn't seem to be installed
Nero Audio Decoder (Nero 7 or older) works fine
Haali Media Splitter doesn't seem to be installed
Surcode DTS Encoder doesn't seem to be installed
MkvToolnix doesn't seem to be installed

Any idea?

[Beastie Boy]

First run eac3to without specifying an output, eg:
d:\m1\cmd\eac3to\eac3to "test.ts"

This will list the streams with their ID numbers.

Cheers, Beastie.

[robena]

First run eac3to without specifying an output, eg:
d:\m1\cmd\eac3to\eac3to "test.ts"

This will list the streams with their ID numbers.

Cheers, Beastie.

Thanks for the tip. I get the same error:

c8{prob}% d:\m1\cmd\eac3to\eac3to "test.ts"
This doesn't seem to be a valid (E-)AC3 stream.
TS, 1 video track, 0:45:05
1: h264/AVC, 1080i50 (16:9)

I know that there is an AC3 track. TSPE for example shows it and can play it.

[rica]

Sorry in advance guys if this has been answered already:

I follow this way while demuxing my HD/BD rips:

Demux video with TSMuxer
Demux audio with EAC3to..

The first time i pulldown 23.97 with TS muxer while demuxing.
So it is sure i have to sync the audio this time.
How can i do this?

Thanks.

[madshi]

I have a problem though. I am trying to demux and speedown an AC3 track from an UK Sky cap.

Here is what I get:

d:\m1\cmd\eac3to\eac3to "test.ts" 2: "test.ac3" -slowdown -384
This doesn't seem to be a valid (E-)AC3 stream.
The AC3 track seems to be slightly damaged. It might be 99% alright, but if there is one wrong bit anywhere in the beginning of the AC3 stream, eac3to won't accept it. You'll have to demux the AC3 track with another tool (e.g. xport or TsMuxer) and then run the demuxed AC3 track through delaycut. delaycut will probably find something to fix. Afterwards eac3to can do the slowdown for you.

At this point in time eac3to is VERY picky about the source material. The source must be in perfect state. If there's any problem with the source, eac3to will refuse to work properly. I may improve this in a future version, but that's the way it is right now.

Sorry in advance guys if this has been answered already:

I follow this way while demuxing my HD/BD rips:

Demux video with TSMuxer
Demux audio with EAC3to..

The first time i pulldown 23.97 with TS muxer while demuxing.
So it is sure i have to sync the audio this time.
How can i do this?
I'm not sure what your question is. You're describing what you're doing. And then you're asking "how can i do this"?

[rica]

I'm not sure what your question is. You're describing what you're doing. And then you're asking "how can i do this"?

I mean audio stays matching to 29.97 fps.
The question was how i could syncronize audio to downmixed video?

[madshi]

Audio is never 29.97 for a movie. Why do you think it is?

[monohouse]

-----

[Beastie Boy]

madshi, your reply to robena above seems to quote a different error to the one he posted when I replied. In the post I replied to, eac3to could not detect an audio stream, whereas your quote suggests eac3to found a damaged stream.
Not sure what happened there.
robena, if eac3to cannot detect the audio that is definately there, perhaps a sample would be useful for madshi.

Cheers, Beastie.

[rica]

Audio is never 29.97 for a movie. Why do you think it is?

OK.
I used a short clip in my trial.
When i demux video and audio as is and after remux to TS,
clip takes 2:25 minutes. Audio and video matches to each other; no syncronization issue.

When i demux video to 23.97 and leave audio as is,
this time clip takes 3:05 minutes; audio and video never overlap and while video keeps playing, audio finishes at 2:25 minutes.

EDIT: Sorry, i found my mistake, i made 23.96 by manually, this created the issue.
I tried with "remove pulldown" option selected in TSMuxer; no any problem left.

[gregt]

Hi,
I am unable to convert Xmen3 at all. It is the first 7 channel movie that I have run across. Here is the output:
eac3to v2.44
command line: eac3to xmen3 XMEN3.mkv -libav -core
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 1:44:05
1: Chapters, 31 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 6.1 channels, 24 bits, 48khz
4: AC3, Spanish, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB, -13ms
5: AC3, French, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB, -13ms
6: AC3, English, 2.0 channels, 224kbit/s, 48khz, dialnorm: -27dB, -19ms
7: AC3, English, 2.0 channels, 224kbit/s, 48khz, dialnorm: -27dB, -19ms
Creating file "XMEN3 - Chapters.txt"...
[a03] The libav DTS decoder doesn't decode the 7th channel.
[a03] The libav DTS decoder doesn't decode the full DTS-HD information.
[a03] Doubling 7th channel...
[v02] Extracting video track number 2...
[a05] Extracting audio track number 5...
[a04] Extracting audio track number 4...
[a03] Extracting audio track number 3...
[a03] Extracting DTS core...
[a06] Extracting audio track number 6...
[a07] Extracting audio track number 7...
[a03] Remapping channels...
[a03] The channel modder was started with incorrect parameters.
[v02] Muxing video to Matroska...
Aborted at file position 16384.

I tried it without -libav and without -core, but to no avail.

[nautilus7]

What do you want to do with audio and which audio is that? Change the command to make use of ID numbers for the tracks. E.g:

eac3to xmen3 1: xmen3.mkv 2: audio.dts -core

[Thunderbolt8]

Not sure if I've understood you correctly. eac3to already checks only those tracks you are demuxing.
so would it then be possible to adapt the gap value to those specific tracks? from what I understood its only 40ms, because an ac3 frame is 32ms, but when for example only decoding a truehd track then it should be possible to lower the gap detection to a value, which is only slightly higher than a truehd frame is, shouldnt it?

or did I understood it wrong, about those 40ms?

[robena]

The AC3 track seems to be slightly damaged. It might be 99% alright, but if there is one wrong bit anywhere in the beginning of the AC3 stream, eac3to won't accept it. You'll have to demux the AC3 track with another tool (e.g. xport or TsMuxer) and then run the demuxed AC3 track through delaycut. delaycut will probably find something to fix. Afterwards eac3to can do the slowdown for you.

At this point in time eac3to is VERY picky about the source material. The source must be in perfect state. If there's any problem with the source, eac3to will refuse to work properly. I may improve this in a future version, but that's the way it is right now.

Thanks for the explanation and for the workaround.

[jchappo]

Madashi, yet more info on DTS-HD to WAV problem.

I just converted a TrueHD track to WAV and it plays fine on the popcorn hour. Looks like libav works, but got this error:

C:\DVDRips\HD>eac3to.exe Ripped\legend_audio.truehd Ripped\output.wav
TrueHD/AC3, 5.1 channels, 48khz, dialnorm: -27dB
Extracting TrueHD stream...
Removing dialog normalization...
Writing WAV...
Creating/writing file "Ripped\output.24bit.wav"...
[libav] Lossless check failed - expected 0, calculated d1
Caution: The WAV file is bigger than 4GB.
Some WAV readers might not be able to handle this file correctly.
This audio track contains only 16 bit of information.
The zero bytes were successfully removed.
eac3to processing took 9 minutes, 14 seconds.
Done.

Sonice and Nero both produce garbage audio after about 20%

[Encoder888]

@madshi

I had trouble extracting the chapters from The Terminator BD (it's MPEG-2). I got this:

CHAPTER01=00:00:00.000
CHAPTER01NAME=
CHAPTER02=00:06:23.925
CHAPTER02NAME=
CHAPTER03=00:13:38.234
CHAPTER03NAME=
CHAPTER04=00:15:04.737
CHAPTER04NAME=
CHAPTER05=00:22:42.528
CHAPTER05NAME=
CHAPTER06=00:29:42.698
CHAPTER06NAME=
CHAPTER07=00:38:40.026
CHAPTER07NAME=
CHAPTER08=00:41:18.101
CHAPTER08NAME=
CHAPTER09=00:47:39.398
CHAPTER09NAME=
CHAPTER10=00:00:00.000
CHAPTER10NAME=
CHAPTER11=00:00:00.000
CHAPTER11NAME=
CHAPTER12=00:00:00.000
CHAPTER12NAME=
CHAPTER13=00:00:00.000
CHAPTER13NAME=
CHAPTER14=00:00:00.000
CHAPTER14NAME=
CHAPTER15=00:00:00.000
CHAPTER15NAME=
CHAPTER16=00:00:00.000
CHAPTER16NAME=
CHAPTER17=00:00:00.000
CHAPTER17NAME=

So, the first 8 are okay, but the last 8 are all 00:00:00.000 for some reason... Any idea what may be causing this?

[madshi]

I am unable to convert Xmen3 at all. It is the first 7 channel movie that I have run across. Here is the output:
eac3to v2.44
command line: eac3to xmen3 XMEN3.mkv -libav -core
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 1:44:05
1: Chapters, 31 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 6.1 channels, 24 bits, 48khz
4: AC3, Spanish, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB, -13ms
5: AC3, French, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB, -13ms
6: AC3, English, 2.0 channels, 224kbit/s, 48khz, dialnorm: -27dB, -19ms
7: AC3, English, 2.0 channels, 224kbit/s, 48khz, dialnorm: -27dB, -19ms
Creating file "XMEN3 - Chapters.txt"...
[a03] The libav DTS decoder doesn't decode the 7th channel.
[a03] The libav DTS decoder doesn't decode the full DTS-HD information.
[a03] Doubling 7th channel...
This looks like a bug. Should be fixed in the next build (I hope).

Madashi, yet more info on DTS-HD to WAV problem.

I just converted a TrueHD track to WAV and it plays fine on the popcorn hour. Looks like libav works, but got this error:

C:\DVDRips\HD>eac3to.exe Ripped\legend_audio.truehd Ripped\output.wav
TrueHD/AC3, 5.1 channels, 48khz, dialnorm: -27dB
Extracting TrueHD stream...
Removing dialog normalization...
Writing WAV...
Creating/writing file "Ripped\output.24bit.wav"...
[libav] Lossless check failed - expected 0, calculated d1
Caution: The WAV file is bigger than 4GB.
Some WAV readers might not be able to handle this file correctly.
This audio track contains only 16 bit of information.
The zero bytes were successfully removed.
eac3to processing took 9 minutes, 14 seconds.
Done.
Which error do you mean? The track seems to have been converted successfully?

Sonice and Nero both produce garbage audio after about 20%
With which movie/audio track? Also with the TrueHD track of "I Am Legend"? And the libav conversion of the same track works ok? That sounds extremely strange to me. I almost cannot believe that. How long are the final WAV files produced by libav, Nero and Sonic for this one specific TrueHD track?

I had trouble extracting the chapters from The Terminator BD (it's MPEG-2). [...] the first 8 are okay, but the last 8 are all 00:00:00.000 for some reason... Any idea what may be causing this?
Not right now. Can you please upload the playlist (mpls) file you used?

[Encoder888]

@madshi

It's not a seamless branching movie, it's just one m2ts... It's the first title that's behaved like this, very weird...

[madshi]

It's not a seamless branching movie, it's just one m2ts... It's the first title that's behaved like this, very weird...
Can you please upload the playlist (mpls) file you used?

P.S: Do "eac3to BluRayMovieFolder" and then upload the first mpls file that gets listed, if that's the movie playlist.

[bmnot]

madshi, did I create the sample correctly? concerning this: http://forum.doom9.org/showpost.php?p=1133416&postcount=4558

[madshi]

madshi, did I create the sample correctly? concerning this: http://forum.doom9.org/showpost.php?p=1133416&postcount=4558
Yes. The bug is already fixed in my sources.

[PHD_1976]

madshi, hi.
Is it possible to have a piece of your source code where you handle seamless branching audio?
If not could you please explain the algorithm of fixing multi file audio without reencoding?
I've tried to PM you, but your inbox limit is exceeded.
Thanx in advance for your help -)

[madshi]

it only shows the information after the process is complete, I cannot force it to create only one file (instead of 2 which it makes automatically) of the bit depth that is relevant, eac3to makes both files and then deletes the wrong one, what I wanted to ask is how is it possible to know the bit-depth of the stream before starting to convert it (and yes I mean "truehd" or should we call it "craphd", what kind of bs is that creating a 24-bit if only 16-bit is used?)
The bitdepth is simply not known before the processing. The information about the bitdepth is not contained anywhere in the TrueHD track. eac3to has no other choice than to fully decode the full track and check the bitdepth of every single audio sample (of which there are usually 48000 for every second of the runtime). If all 24bit audio samples have 8bits zeroed out, eac3to will state that it's a 16bit track only and will remove the 24bit file. There's nothing else I can do.

but I do have one question though: is it possible to demux video+demux audio+convert audio into other format+remux into mkv without saving the files, process them directly into mkv (or maybe something else ?) ?
No.

is there a way to remove dialnorm from ac3 in real time ? without re-converting ? and if reconverting is necessary, is it possible to convert ac3 to ac3 without loosing the quality (demux?) ? because I was wondering, what is dialnorm ? im guessing that dialnorm is a form of processing that is being done on the audio stream to increase the volume of the dialogs if necessary based on current level of the other data, if so it is a form of processing that involves DRC, and according to pro audio rules, any type of processing is a sure degradation of quality, however if it is not a form of processing that is being performed in real-time on a stream and instead data that exists in that form (pre-amplified) then processing it with eac3to and removing the normalization is catastrophic (reencoding+modifying)
Dialnorm is a simple number stored in every AC3 audio frame. Removing dialnorm is a simple process which just sets this number to zero and recalculates the checksum. So basically there's no reencoding done. It's a simple modification of the compressed stream. There's no quality loss whatsoever.

another question: how can you be sure that The Sonic DTS decoder actually gives you 1:1 sample values ? I can appreciate the ability of eac3to when it handles trueHD tracks to give you 1:1 sample values because it's code is in your program, however when it comes to external filters, how do you know that, for example dialog normalization hasn't been applied ? and that's just the lite case, how about bass boost that was added just for fun or DRC that suddenly decided to kick in because the decoder detected what might be clipping so it compressed by say 2 db, who knows what other thresholds there may be out there and what other code that might do some of those "other stuff" you mentioned earlier, not all of these can be detected by recording and monitoring the output of the decoder.
The TrueHD decoder is not in my sources, either. I'm using libav for decoding TrueHD. Anyway, there've tests done on the decoders which include comparing the decoded results in a WAV editor and even in a hexeditor. E.g. some movies come with both PCM and DTS-HD MA tracks. And decoding the DTS-HD MA tracks with the Sonic decoder resulted in identical data compared to the PCM track. So it's fairly safe to say that the Sonic DTS-HD decoder outputs bit perfect results.

it is very hard to determine whether it only reads the "core" or the additional data beyond the core
No, it's not hard at all. I've fed the decoder only the core (test A). And I've fed it the full DTS-HD stream (test B). And the decoder output different data for both tests. So obviously the decoder makes use of the DTS-HD information.

it could even be in reverse, that it only reads the lossless track and not the core
That's technically not possible. The result would be garbage.

if eac3to contains code to properly decode truehd, is there any chance that it might include the code to decode DTS-HDMA just as well ?
eac3to does not contain code to decode TrueHD. If the libav library will ever include DTS-HD MA decoding then eac3to will offer that functionality. But today it doesn't. And I'm not aware of any plans by the libav programmers to add DTS-HD decoding.

when source is ac3 and I asked eac3to to save that source stream also as ac3, does that mean it will demux it without reencoding ?
Yes. Except if you use one of the bitrate switches to change the bitrate. Then the stream is decoded and reencoded. eac3to logs to the command line window exactly what is does. If it decodes/reencodes the AC3 stream, you'll see that in the eac3to text output. Just try it out. You'll easily be able to tell when decoding/reencoding is done and when not.

for that matter, is it true to assume that in any event that input type=output type the operation will be demuxing and not re-encoding ?
Yes. As long as you don't use any additional parameters (like bitrate switches) which force eac3to to reencode the audio stream. Of course some manipulations are done nevertheless, e.g. dialnorm removal. But as explained above that does not harm quality at all.

I noticed that when the output format is different than the input format there is a dithering process taking place, while when input=output it isn't.
Dithering is done when bitdepth is reduced. Usually that's the case when using the libav AC3, E-AC3, MP2 and DTS decoders cause I've hacked these decoders in such a way that they outputs a very high bitdepth. eac3to then internally dithers the bitdepth down to a reasonable value.

[madshi]

madshi, your reply to robena above seems to quote a different error to the one he posted when I replied. In the post I replied to, eac3to could not detect an audio stream, whereas your quote suggests eac3to found a damaged stream.
Actually eac3to did find an audio track, it's just not listed in the track listing because eac3to found the stream to be damaged. See the line "This doesn't seem to be a valid (E-)AC3 stream" in the original post. This line indicates that an (E-)AC3 stream was found but ignored cause it's damaged.

Is it possible to have a piece of your source code where you handle seamless branching audio?
If not could you please explain the algorithm of fixing multi file audio without reencoding?
What do you need this for?

[PHD_1976]

What do you need this for?

Currently I'm trying to create a BD demuxer, which makes Scenarist compliant AVC streams while demuxing.
I already know what should be done to make AVC streams compliant, but I thought it would be great to demux all needed audio and subs as well, but I don't know how to handle audio from seamless branching movie without reencoding.
I will certainly share my tool as soon as its ready.

[Encoder888]

@mashi.

Sorry I totally wasn't paying attention, I thought you needed the actual list, not the file :P Here it is:

http://www.mediafire.com/?dmiyggjnjit

[monohouse]

-----

[tebasuna51]

I know that this is not your area but was wondering if there is a chance you might know, if I use vobblanker to blank a cell in the video then it sure does blank the audio of that area as well, does it mean that ac3 stream was modified ? or is it possible to insert data into the ac3 stream without reencoding it ?
Is possible insert silence in the middle of an ac3 stream without reencoding.

I don't know vobBlanker but the author is jsoto (http://jsoto.posunplugged.com/vobblanker.htm). also DelayCut author and the method to delay an ac3 can be used to insert ac3 silence in the middle

[monohouse]

-----

[madshi]

Currently I'm trying to create a BD demuxer, which makes Scenarist compliant AVC streams while demuxing.
I already know what should be done to make AVC streams compliant
Maybe (if it's not too much work) it would make sense to add this functionality to eac3to? Or are there more reasons why you want to create your own tool? Not that it would be a problem for me, of course!

I thought it would be great to demux all needed audio and subs as well, but I don't know how to handle audio from seamless branching movie without reencoding.
It's quite complicated. You need this:

(1) A full stream parser for all audio formats you want to support, which can also calculate the exact runtime of each audio frame.
(2) Your tool will need to read out the PTS timestamps of the m2ts container and compare them to the runtime information your audio stream parsers (see (1)) calculated. This will give you information about where in the stream are gaps/overlaps. Unfortunately the PTS values are not always 100% correct, they may vary slightly (a bit too low or a bit too high). So you'll need to find a good way to even out the variances.
(3) You have to take the gap/overlap information from step (2) and perform the necessary changes on the audio streams. For AC3, E-AC3, DTS and MP2 you'll have to insert (or delete) single audio frames to get rid of the gaps (or overlaps). For LPCM you can insert (or delete) single samples. For TrueHD things are difficult cause it's not really possible to do such corrections on the TrueHD bitstream without introducing problems. So for TrueHD you can do the correction only if you decode the audio data first.

The eac3to source code is a complicated class structure. There's not just a few lines of code which do the audio gap/overlap correction. There's code in several classes which works together. So it's nearly impossible to give you just "some code" to duplicate eac3to's audio gap/overlap correction functionality.

[jchappo]

Madashi,

This is not an error?

[libav] Lossless check failed - expected 0, calculated d1

anyway, is there anything I could provide you to help fix Sonic and Nero decoding to WAV?

[madshi]

[libav] Lossless check failed - expected 0, calculated d1
That's a bit strange. Can be a bug in the libav decoder. Or it can be a slightly damaged TrueHD stream. Don't know. Since this warning only occurred once in the whole audio track, you probably don't need to bother much. The problem can only affect max 100ms worth of audio data in the whole audio track, I believe.

anyway, is there anything I could provide you to help fix Sonic and Nero decoding to WAV?
I'm not convinced yet that there is any problem with Sonic and Nero decoding to WAV. Can you give me some more details about why you came to the conclusion that there's something wrong with Sonic and Nero decoding to WAV? Which audio track did you decode with Sonic, Nero *and* libav? How big were the WAV files created by the 3 decoders?

It's important to decode the very same track with all 3 decoders. Only if you do that and if you then run into a problem with some decoders while other decoders don't show the same problem, only then it's a hint that some decoders might have a problem. But still even in that case the problem might be elsewhere. E.g. if the TrueHD stream is really damaged (see the "lossless check failed" libav warning you posted) this might throw off Sonic&Nero but might have a less bad effect on libav.

[jchappo]

I will do more testing, but so far, the only tracks I can play on the popcorn hour are:

LPCM tracks included with the Bluray
TrueHD converted to WAV (PCM) with eac3to -libav

Both DTS-MA and DD+ (Sonic, and Nero) both produce bad WAV files.

[madshi]

I will do more testing, but so far, the only tracks I can play on the popcorn hour are:

LPCM tracks included with the Bluray
TrueHD converted to WAV (PCM) with eac3to -libav

Both DTS-MA and DD+ (Sonic, and Nero) both produce bad WAV files.
That is not a conclusive test. The bad WAV playback probably only depends on the WAV file size and on nothing else. Most probably the WAV files created from DTS-MA and DD+ are bigger than 4GB while the TrueHD created WAV file is probably smaller than 4GB. Probably that's the difference...

[madshi]

eac3to v2.45 released

http://madshi.net/eac3to.zip

* Blu-Ray angles are now reported as separate titles
* duplicate playlists are not listed in the "folder view", anymore
* reduced TrueHD and RAW/PCM gap/overlap threshold to 7ms
* reduced (E-)AC3 gap/overlap threshold to 60% of the runtime of one audio frame
* reduced MP2 gap/overlap threshold to 60% of the runtime of one MP2 frame
* reduced DTS threshold to 60% of the runtime of one DTS frame, but at least 7ms
* fixed: Blu-Ray chapter export sometimes wrote incorrect "00:00:00.000" items
* improved handling of MPEG2 streams (changes from interlaced to progressive)
* video information now shows "with pulldown flags", if applicable
* removed "-ignoreDiscon" from help; hint is shown when a discontinuity occurs
* added "-ignoreEncrypt" option; hint is shown when a source is encrypted
* new option "-extensible" creates WAV files with a slightly different header
* fixed some smaller bugs
Notes:

(1) As demanded, I've noticably decreased the audio gap/overlap thresholds. For most audio tracks the threshold is only 7ms now. If you run into any trouble with this (e.g. gaps/overlaps reported in situations where you didn't expect it), please let me know. Also if it works well for you, I'd also like to hear about it and maybe see some success logs. Thanks!

(2) Please note that MPEG2 handling of problematic streams has improved, but I still not to improve it further. As an explanation: Many MPEG2 streams are 60i. Some of them are natively interlaced, others are movies. Movies are often encoded as 24p but with pulldown flags. HD DVD uses this method. Now the problem is this: Sometimes a stream begins as 24p with pulldown flags but then suddenly switches to natively 60i interlaced. Or the other way round. Some streams even switch multiple times. Of course we want eac3to to handle 24p movies with pulldown flags as 24p. But doing this runs into problems when the stream suddenly switches to 60i interlaced content. Currently any stream with changes between 24p (with pulldown flags) and 60i is not handled in the best possible way by eac3to. I still need to improve that. But at least processing is not aborted, anymore, if there's such a change in the stream. The previous eac3to version aborted in such a situation. That means with v2.45 you can at least use eac3to for demuxing video and for handling the audio streams etc...

(3) I've heard that TsMuxer complains about eac3to's WAV files. Something like "channel mapping information missing". You can now use the new "-extensible" switch to make eac3to write slightly different WAV headers. This may satisfy TsMuxer (or not).

[jchappo]

Actually the TrueHD WAV file was 6GB, DD+ and DTS-HD both produce WAV files around that size as well.

[markrb]

[a04] Encoding AC3...
[a04] Creating file "D:\audio.ac3"...
[a04] Audio overlaps for 14ms at playtime 0:00:02.
[a04] Audio overlaps for 25ms at playtime 0:05:09.
[a04] Audio overlaps for 12ms at playtime 0:09:18.
[a04] Audio overlaps for 5ms at playtime 0:09:53.
[a04] Audio overlaps for 29ms at playtime 0:12:14.
[a04] Audio overlaps for 23ms at playtime 0:13:32.
[a04] Audio overlaps for 10ms at playtime 0:26:09.
[a04] Audio overlaps for 23ms at playtime 0:28:19.
[a04] Audio overlaps for 32ms at playtime 0:34:52.
[a04] Audio overlaps for 24ms at playtime 1:03:31.
[a04] Audio overlaps for 8ms at playtime 1:04:06.
[a04] Audio overlaps for 11ms at playtime 1:20:23.
[a04] Audio overlaps for 7ms at playtime 1:22:20.
[a04] Audio overlaps for 13ms at playtime 1:24:06.
[a04] Audio overlaps for 14ms at playtime 1:26:21.
[a04] Audio overlaps for 21ms at playtime 1:29:26.
[a04] Audio overlaps for 11ms at playtime 1:29:54.
[a04] Audio overlaps for 21ms at playtime 1:37:21.
[a04] Audio overlaps for 28ms at playtime 1:39:05.
[a04] Audio overlaps for 27ms at playtime 1:39:27.
[a04] The audio file was demuxed without making use of the gap/overlap information.
[a04] Please rerun the same eac3to command line. That will correct the gaps/overlaps.

I don't understand what it's asking me to do.
Am I suppose to just run the program again and it will somehow remember and fix the issue or am I suppose to put in some sort of command to fix it?

Thanks,
Mark

[madshi]

Actually the TrueHD WAV file was 6GB, DD+ and DTS-HD both produce WAV files around that size as well.
Ok. Then please try decoding the very same audio track with all 3 decoders and compare. I'd be surprised if it depended on the decoder which WAV plays fine for you and which doesn't. But of course anything is possible.

[Yraen]

[a04] The audio file was demuxed without making use of the gap/overlap information.
[a04] Please rerun the same eac3to command line. That will correct the gaps/overlaps.

I don't understand what it's asking me to do.
Am I suppose to just run the program again and it will somehow remember and fix the issue or am I suppose to put in some sort of command to fix it?

Thanks,
Mark

Yes, just run the same command line again.

[madshi]

I don't understand what it's asking me to do.
Am I suppose to just run the program again and it will somehow remember and fix the issue
Yes. You're supposed to simply rerun eac3to with exactly the same parameters as in your first try. The information needed for fixing the issue were stored on harddisk during the first pass (look for a *.gap file). This is then made use of when you run eac3to a 2nd time.

If you want you can save the original "d:\audio.ac3" file and compare it with the final result. You should notice that the current "d:\audio.ac3" file is not in sync with the video. The AC3 file you get from the 2nd pass should be in sync.

Edit: Yraen was faster... :)

[bmnot]

"The h264 muxer received invalid h264/AVC data" error problem fixed as expected. Thanks. :)

EDIT: I have a potentially silly question. If I have a 16-bit lossless source I want to convert to DTS, should I encode the DTS at 16-bit, because there is no need for 24, and so the size will be smaller? Or would encoding to DTS at 24-bit help the sound quality?

[nautilus7]

Surcode will encode at 24 bit. You don't have/need to do anything.

[Thunderbolt8]

thanks for the new version :)

2 questions:

"reduced DTS threshold to 60% of the runtime of one DTS frame, but at least 7ms" - does this affect dts-hd (MA) as well?

and regarding those mpeg2 sources, which sometimes switch flags or show other problems, should I still continue to make samples of these? or do you have already enough and just need more time for those improvements atm?

[Atak_Snajpera]

Is it possible to demux only chapters?

[Bluestraw]

Many thanks for the new version - I tested the angle support, and successfully converted the French version of Cars (which is at least 12 parts) to MKV with FLAC, and the resulting file is perfectly in sync. Only reported a couple of small overlaps, each 4 ms I think.

[Yraen]

Is it possible to demux only chapters?

Yes. Most chapters are track 1, but check to be sure.

eac3to H:\moviefolder\ 1) 1: chapters.txt

[jchappo]

Ok Madashi,

Converted the DTS-HD track to WAV with all decoders, all suffered from same problem, after 20-30% it was all garbage.

Same thing converting from DD+.

Only thing that works is TrueHD.

[nautilus7]

Ok Madashi,

Converted the DTS-HD track to WAV with all decoders, all suffered from same problem, after 20-30% it was all garbage.

Same thing converting from DD+.

Only thing that works is TrueHD.There must be something wrong with your installation of the sonic decoder. Which version do you have? Which movie did you try?

[gregt]

Thanks for this update (2.45)!
- Fixed the Xmen3 problem which now converts successfully.
- Fixed the DVD interlaced problem and my interlaced (480i60 and 480p24 with pulldown flags) files now process with no problems (so far)! Most of my DVDs the changed occurs in the first 2 seconds.
- the gap change seems to create a lot gap data where there was none before. I ignore it on files that I know it is not needed.
One issue that I am having with DVDs is that the language seems to be missing for audio and subtitles as follows:
eac3to v2.45
command line: eac3to vts_01_1.vob+vts_01_2.vob+vts_01_3.vob+vts_01_4.vob+vts_01_5.vob t.mkv
------------------------------------------------------------------------------
VOB, 1 video track, 2 audio tracks, 3 subtitle tracks, 13:17:28
1: Joined VOB file
2: MPEG2, 480p24 /1.001 (4:3) with pulldown flags
3: AC3, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB, -83ms
4: AC3, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB, -100ms
5: Subtitle
6: Subtitle
7: Subtitle

But, thanks for the update as that is a huge improvement for me.

[nautilus7]

To make eac3to list languages you have to load the DVD folder, like you do for HD DVDs and Blu-ray Discs. Reading DVD folders isn't supported (yet), though, so you can't do it currently.

[PHD_1976]

Maybe (if it's not too much work) it would make sense to add this functionality to eac3to? Or are there more reasons why you want to create your own tool? Not that it would be a problem for me, of course!

Madshi,
There's a lot of work fixing h264 streams for Scenarist.
Maybe we can discuss it outside this thread.
I don't mind sharing my code with you.
Where can I write you a letter with a description of what should be done to fix H264 streams?

[madshi]

"reduced DTS threshold to 60% of the runtime of one DTS frame, but at least 7ms" - does this affect dts-hd (MA) as well?
Yes.

and regarding those mpeg2 sources, which sometimes switch flags or show other problems, should I still continue to make samples of these? or do you have already enough and just need more time for those improvements atm?
Right now I don't need further samples. Once I'm done with the improvements and there are still problems, I'd be interested in further samples. But for now I think I have enough samples.

Many thanks for the new version - I tested the angle support, and successfully converted the French version of Cars (which is at least 12 parts) to MKV with FLAC, and the resulting file is perfectly in sync. Only reported a couple of small overlaps, each 4 ms I think.
Good to hear, thanks!

Converted the DTS-HD track to WAV with all decoders, all suffered from same problem, after 20-30% it was all garbage.

Same thing converting from DD+.

Only thing that works is TrueHD.
Do you have different movies with DTS-HD, DD+ and TrueHD tracks? If possible, it would be helpful if you could test other movies, too.

I still believe that the size of the WAV file is the only problem here. Let me explain: The WAV header has a "size" field which is only 4 byte long. The max size you can put in 4 bytes is about 2GB or 4GB (depending on whether you interpret that number to be able to hold negative numbers, too). Now if you have a WAV file which is longer than 2GB or 4GB, the "size" field in the WAV header is invalid. It's difficult to explain to a non-programmer what value this field has when it's invalid. But you can think of it as being "random", but the random value depends on the full WAV file size. Now if a media player is stupid enough to rely on the "size" field, it will interpret the random value in the WAV header as being the correct size. Since the value is random, a 5.9GB WAV file could eventually work much better than a 6.1GB WAV file or vice versa.

When you decode the same track with different decoders, they will likely still output the same WAV size. That's why all 3 decoders behave the same for you. I believe the only problem is the random WAV "size" field. You probably just had luck with your TrueHD track that the random size field somehow made it work more or less ok.

Fixed the DVD interlaced problem and my interlaced (480i60 and 480p24 with pulldown flags) files now process with no problems (so far)! Most of my DVDs the changed occurs in the first 2 seconds.
- the gap change seems to create a lot gap data where there was none before.
Yeah, that's still something I'm working on. It happens especially when a movie begins as 480p24 (with pulldown flags) and then changes to 480i60.

One issue that I am having with DVDs is that the language seems to be missing
As nautilus7 already explained, this is a feature which is just not implemented in eac3to. eac3to parses the special HD DVD and Blu-Ray files where the language information is contained. But eac3to doesn't parse the special DVD files where the info is contained.

There's a lot of work fixing h264 streams for Scenarist.
Maybe we can discuss it outside this thread.
I don't mind sharing my code with you.
Where can I write you a letter with a description of what should be done to fix H264 streams?
You can email to dear (at) madshi (dot) net.

[PHD_1976]

You can email to dear (at) madshi (dot) net.
Just emailed you from genetr (at) mail (dot) ru

[sshd]

The Machinist HD DVD:
3: h264/AVC, 1080p24 /1.001 (16:9) with pulldown flags

This title does not appear to have pulldown flags.
Video is destroyed when removing pulldown with h264info and plays fine at 23.976 fps without.

[tebasuna51]

Thanks for your new eac3to version.

You includes the -extensible parameter to output wav with the WAVE_FORMAT_EXTENSIBLE header (henceforth WFE).
I like this improvement but I have some comments:
- When use WFE header is mandatory a fix channel order.
- The most useful new data (maybe the unique) is the Channel Mask to specify the channels present in the audiodata.
- When is unknown the channels present in the source is recommended use a default Channel Mask based in the number of channels.
- When we know the source channels the channel mask must be set in accord.

1) ABOUT DEFAULTS

I think the default used in eac3to for 4 channels (0x000F = FL FR FC LF) must be changed for 0x0033 = FL FR BL BR (Quadro) more usual and compatible with Flac/Ogg multichannel.
A WFE wav with maskchannel 0x000F is rejected by flac.

The default used for 8 channel (0x00FF = FL FR FC LF BL BR FLC FRC) is not usual, maybe the default can be 0x063F = FL FR FC LF BL BR SL SR

When I try know the default for 7 channels using a 3/3.1 wav file:
eac3to v2.45
command line: "D:\Test\AudioN\eac3to\eac3to.exe" "E:\Test\7_61.wav" "E:\Test\z331.wav" -extensible
------------------------------------------------------------------------------
WAV, 6.1 channels, 0:00:20, 16 bits, 48khz
Doubling 7th channel...
Reading WAV...
Writing WAV...
Creating file "E:\Test\z331.wav"...
eac3to processing took 1 second.
Done.
For what "Doubling 7th channel..."?
Maybe the default for 7 channel can be: 0x013F = FL FR FC LF BL BR BC

2) ABOUT KNOW SOURCE AC3

At least with ac3 source we can obtain perfect WFE wav output:
acmod.lfe ac3 channels Mask and MS channels ordered Detect-MaskCh eac3to libav remap
------------------------- --------------------------------- --------------------- ------------
1 1/0.0 C 0x0004 FC 1.0 0x0004 ok not needed
1 1/0.1 C LFE 0x000C FC LF 1.1 0x0003 (1)(2) not needed
2 2/0.0 L R 0x0003 FL FR 2.0 0x0003 ok not needed
2 2/0.1 L R LFE 0x000B FL FR LF 2.1 0x0007 (2) not needed
4 2/1.0 L R S 0x0103 FL FR BC 2/1 0x0007 (2) not needed
4 2/1.1 L R S LFE 0x010B FL FR LF BC 2/1.1 0x000F (2)(3) -0,1,3,2,4,5
6 2/2.0 L R SL SR 0x0033 FL FR BL BR 2/2 0x000F (2) not needed
6 2/2.1 L R SL SR LFE 0x003B FL FR LF BL BR 2/2.1 0x0037 (2)(3) -0,1,4,2,3,5
3 3/0.0 L C R 0x0007 FL FR FC 3/0 0x0007 (3) -0,2,1,3,4,5
3 3/0.1 L C R LFE 0x000F FL FR FC LF 3/0.1 0x000F (3) -0,2,1,3,4,5
5 3/1.0 L C R S 0x0107 FL FR FC BC 3/1 0x000F (2)(3) -0,2,1,3,4,5
5 3/1.1 L C R S LFE 0x010F FL FR FC LF BC 3/1.1 0x0037 (2)(3) -0,2,1,4,3,5
7 3/2.0 L C R SL SR 0x0037 FL FR FC BL BR 5.0 0x0037 (3) -0,2,1,3,4,5
7 3/2.1 L C R SL SR LFE 0x003F FL FR FC LF BL BR 5.1 0x003F ok already done

(2) Default mask for channel number, the correct mask can be easyly put based in detection
(3) When decoded with libav need remapping channels.
EDIT: My previous note (1) was:
(1) Wrong detection, must be 1.1 (not important because mono + LFE is really strange).
But is my fault, eac3to detect 1.1 correctly

[calinb]

At this point in time eac3to is VERY picky about the source material.Indeed! Oftentimes, source that produces the "This track is not clean...Please clean the track with delaycut" error can be processed with eac3to ver. 1.16--after remuxing/joining the .evo(s) with EVOdemux, if necessary.

[gregt]

I seem to have run into a consistent problem with certain DVDs that are 480i60. The following is an example:
eac3to v2.45
command line: eac3to vts_01_1.vob+vts_01_2.vob+vts_01_3.vob+vts_01_4.vob t.mkv
------------------------------------------------------------------------------
VOB, 1 video track, 3 audio tracks, 3 subtitle tracks, 1:53:11
1: Joined VOB file
2: MPEG2, 480i60 /1.001 (16:9)
3: AC3, 2.0 channels, 192kbit/s, 48khz, dialnorm: -26dB
4: AC3, 2.0 channels, 192kbit/s, 48khz, dialnorm: -25dB
5: AC3, 2.0 channels, 192kbit/s, 48khz, dialnorm: -23dB
6: Subtitle
7: Subtitle
8: Subtitle
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a05] Extracting audio track number 5...
[v02] Muxing video to Matroska...
[a03] Extracting audio track number 3...
[a05] Removing dialog normalization...
[a04] Removing dialog normalization...
[a03] Removing dialog normalization...
[v02] The MPEG2 stream changed from 60000i to 48000i at runtime 0:00:02.
[s07] Extracting subtitle track number 7...
[s08] Extracting subtitle track number 8...
[s06] Extracting subtitle track number 6...
[a03] Creating file "t - 3 - AC3, 2.0 channels, 192kbps, 48khz.ac3"...
[a04] Creating file "t - 4 - AC3, 2.0 channels, 192kbps, 48khz.ac3"...
[a05] Creating file "t - 5 - AC3, 2.0 channels, 192kbps, 48khz.ac3"...
[s08] Creating file "t - 5 - Subtitle.sup"...
[s06] Creating file "t - 3 - Subtitle.sup"...
[s07] Creating file "t - 4 - Subtitle.sup"...
Added fps value to MKV header.
Video track 2 contains 162910 frames.
eac3to processing took 1 minute, 50 seconds.
Done.

The "stream changed from 60000i to 48000i " is the consistancy. The mkv file produced is jerky during playback and this happens to all that I try to convert where this exact stream change occurs. Is there an option on eac3to that will solve this or is this a bug?

[jchappo]

Madashi,

Next time I get a Bluray in with LCPM audio, and a DTS-HD track, I will do more testing.

But for now, your assumption is incorrect. All the tracks I am converting are from full length feature films. The TrueHD, DTS-HD, and DD+ tracks come out roughly the same size, between 1 and 3gb depending on movie length. Once uncompressed to WAV, all files are > 4gb, even the TrueHD.

My question is, why do the LPCM tracks that ship with the Blurays work perfectly, while ones converted using eac3to don't? What is difference between the LPCM provided by the movie studio, and the one your program is making? Is there some kind of program I can use to analyze the two LPCM files?

Bascially, I wan't to do something similar listed in this thread:
http://forum.doom9.org/showthread.php?p=968533#post968533

But I want to go from DTS-HD/TrueHD/DD+ to LPCM. Would I use the RAW output option on eac3to, and map the channels like it says in that thread?

[nautilus7]

Madashi,

Next time I get a Bluray in with LCPM audio, and a DTS-HD track, I will do more testing.

But for now, your assumption is incorrect. All the tracks I am converting are from full length feature films. The TrueHD, DTS-HD, and DD+ tracks come out roughly the same size, between 1 and 3gb depending on movie length. Once uncompressed to WAV, all files are > 4gb, even the TrueHD.

My question is, why do the LPCM tracks that ship with the Blurays work perfectly, while ones converted using eac3to don't? What is difference between the LPCM provided by the movie studio, and the one your program is making? Is there some kind of program I can use to analyze the two LPCM files?It seems that you didn't understand what madshi told you. Read again and tell what movie is that you 're converting. Maybe someone else can try it also.

[DeepBeepMeep]

Madshi,

When converting a TrueHD track to a DTS track with a multiparts seamless branching .m2ts file, are two passes really needed?

In my example I try to extract two tracks: an AC3 track and a TrueHD track (converted to DTS). Overlap is only detected on the AC3 track, is this normal? If so, could you automate the two passes to avoid calling surcode twice ? In fact, if the second pass was handled directly by EAC3TO you could save a lot of time since the demuxing has been already done.

Thanks for your great tool!

[tebasuna51]

My question is, why do the LPCM tracks that ship with the Blurays work perfectly, while ones converted using eac3to don't? What is difference between the LPCM provided by the movie studio, and the one your program is making? Is there some kind of program I can use to analyze the two LPCM files?

Like madshi say you the difference in only the header, the LPCM is raw data without header and the length is passed by other method.
The wav file >4GB have a field to inform about the length than must be wrong because have only 4 bytes and the max number is 2^32.

There are soft to manage these big wave files, you can encode to ac3 with Aften -readtoeof 1, or to mp4 with Neroaacenc -ignorelength. You can split, change bitdepth, samplerate, ... with WaveWizard. An also can be managed in AviSynth (BeHappy, SoundOut, Wavi) if you open the file with RaWavSource() method.

You can also convert (SoundOut) the header to w64 (Sonic Foundry, now acquired by Sony) format with fields than support until 2^64 filelength, but not all soft can accept this format.

[nautilus7]

When converting a TrueHD track to a DTS track with a multiparts seamless branching .m2ts file, are two passes really needed?

In my example I try to extract two tracks: an AC3 track and a TrueHD track (converted to DTS). Overlap is only detected on the AC3 track, is this normal?Yes, it is normal. Overlaps for TrueHD are really small and as a result not often met. If no overlaps are detected for a track, the 2nd pass isn't needed (there's no .gaps file created fo that track).
If so, could you automate the two passes to avoid calling surcode twice?Even if you needed 2 passes in your case, to avoid calling surcode twice (it is very slow), i suggest converting truehd (or whatever) to wav, do the 2nd pass (if needed) and then run a simple command to make the wav 2 dts encoding. This is what i always do when i want to encode to dts. Another good reason for feeding surcode direclty with wav files is to avoid the "surcode button didn't seem to work" error when you specify long filenames/paths as inputs. In that case you won't need to decode the source track twice.

[DeepBeepMeep]

Nautilus7,

Thans for your answer. I think it would be greatly appreciated anyway to automate the two passs processes since there is little reason why one would not go for the second pass when overlaps are discovered.

[tebasuna51]

Bascially, I wan't to do something similar listed in this thread:
http://forum.doom9.org/showthread.php?p=968533#post968533

But I want to go from DTS-HD/TrueHD/DD+ to LPCM. Would I use the RAW output option on eac3to, and map the channels like it says in that thread?

You don't need xport, sox, wavewizard or mediacoder, eac3to do all the job.
The 5.1 channels are automatically remapped from LPCM/DTS-HD/TrueHD/DD+
The RAW is the same than the wav if you delete the first 44 bytes.

[jchappo]

Let me fully explain the problem I have so everyone can understand it better.

The Popcorn hour A-100, is a network media player which can play uncompressed Bluray audio/video. It will accept a M2TS stream and passthrough the LPCM audio to my receiver. Some Bluray movies come with an LPCM audio stream already on them, some do not. For the ones that do not, I want to use eac3to to convert the TrueHD/DTS-HD/DD+ to LCPM so the A-100 can play it. But, I have only been successful with TrueHD so far, both DTS-HD and DD+ produce files that are garbage after the first 20-30% of the movie.

This is the problem. My question is, why can't eac3to produce LPCM files like that ones the movie studio includes on some discs? If I could produce those exact files, then that would be awesome and everything would just work. I am trying to figure out why eac3to is not producing these same files, and how to fix it.

[jchappo]

Would it help if I posted a sample of a movie studio created LPCM file, and one decoded from a DTS-HD track from eac3to which has garbage audio after 30%?

[nautilus7]

You keep saying your "problem", but you didn't listen to what madshi already said.

Do a simple test: Instead of making multi channel wav files make mono wavs. (eac3to input.thd/dtshd/eac3/etc output.wavs). Then check the center channel of all tracks. I bet every single one would be fine.

[wildchild22]

I have the same problem with lpcm as the above poster. What we really need is a program that can accept wavs not an interleaved wav file and mux that with video like scenarist. That way the audio track will be under 4 gigs. either that or maybe madshi could add an option to split the demuxed video and audio file every 30 mins. That way we maybe able to use tsmuxer and append the file and the sound may be correct.

[jchappo]

nautilis7,

I tried what you said and you are correct. If by dump luck, the TrueHD stream decoded into a working WAV, then why can't I edit the value in the WAV header to make it always work?

I'm just trying to get a working solution to producing working LPCM files.

[tebasuna51]

I have the same problem with lpcm as the above poster. What we really need is a program that can accept wavs not an interleaved wav file and mux that with video like scenarist. That way the audio track will be under 4 gigs. either that or maybe madshi could add an option to split the demuxed video and audio file every 30 mins. That way we maybe able to use tsmuxer and append the file and the sound may be correct.

Then the problem is in tsmuxer, must accept wav > 4GB or multiple monowavs or w64/RF64 wav headers.

[EPiPH0NE]

I have the same problem with lpcm as the above poster. What we really need is a program that can accept wavs not an interleaved wav file and mux that with video like scenarist. That way the audio track will be under 4 gigs. either that or maybe madshi could add an option to split the demuxed video and audio file every 30 mins. That way we maybe able to use tsmuxer and append the file and the sound may be correct.

I have already asked madshi about this and have been already trying to do this so I'll let you guys take over. Splitting the M2TS up then processing works for DTS-MA -> WAV but they can't be rejoined at this current time. We DO need to know how to make these files just like the studios are doing cause we are obviously missing something here. Maybe we should do:

eac3to.exe source.m2ts destination.wavs

then use another encoder to make the multi channel PCM/WAV? But, what encoder and what settings and how do we ensure proper LPCM channel configuration?

[Beastie Boy]

Have you tried

eac3to source.m2ts destination.raw

and then changing the file extention if necessary. Maybe this will get around any incompatabilities with the wav header.

Cheers, Beastie.

[madshi]

One issue that I am having with DVDs is that the language seems to be missing for audio and subtitles
Just as a quick follow up:

The VOB format allows to have language identifiers inside the VOB stream. If there are any, eac3to will read them and display them. I do have some DVDs where language detection works. However, there are many DVDs where the language information is not contained in the VOB itself, but in the other files (*.IFO, I believe). eac3to currently doesn't parse these files, so the language cannot be displayed in such cases.

The Machinist HD DVD:
3: h264/AVC, 1080p24 /1.001 (16:9) with pulldown flags

This title does not appear to have pulldown flags.
Video is destroyed when removing pulldown with h264info and plays fine at 23.976 fps without.
As far as I know, every single HD DVD movie (regardless of codec) does have pulldown flags. I think it's even a requirement of the HD DVD specification! If video is destroyed when using h264info then that probably indicates a bug in h264info. I'd suggest uploading a little sample and letting the h264info programmer know about the problem. He should be able to easily fix it.

Indeed! Oftentimes, source that produces the "This track is not clean...Please clean the track with delaycut" error can be processed with eac3to ver. 1.16--after remuxing/joining the .evo(s) with EVOdemux, if necessary.
v1.16? :eek: That feels like ages ago...

Can you upload a little sample which works with v1.16 but not with v2.45?

I seem to have run into a consistent problem with certain DVDs that are 480i60. [...] The "stream changed from 60000i to 48000i " is the consistancy. The mkv file produced is jerky during playback and this happens to all that I try to convert where this exact stream change occurs. Is there an option on eac3to that will solve this or is this a bug?
This is why I stated in the v2.45 release notes that I still have to further improve eac3to for MPEG2 clips with such mode changes. For now I recommend to either wait with converting such clips. Or to demux video instead of muxing it to MKV. I'm planning to complete the MPEG2 muxing improvements in the next build, which should then handle such mode changes just fine.

When converting a TrueHD track to a DTS track with a multiparts seamless branching .m2ts file, are two passes really needed?
I cannot do it in one pass. The 2 passes are necessary to even out timecode instabilities.

In my example I try to extract two tracks: an AC3 track and a TrueHD track (converted to DTS). Overlap is only detected on the AC3 track, is this normal?
Yes.

If so, could you automate the two passes
I've already thought about this. Maybe I will.

Do a simple test: Instead of making multi channel wav files make mono wavs. (eac3to input.thd/dtshd/eac3/etc output.wavs). Then check the center channel of all tracks.
Good idea... ;)

What we really need is a program that can accept wavs not an interleaved wav file and mux that with video like scenarist. That way the audio track will be under 4 gigs. either that or maybe madshi could add an option to split the demuxed video and audio file every 30 mins. That way we maybe able to use tsmuxer and append the file and the sound may be correct.
If by dump luck, the TrueHD stream decoded into a working WAV, then why can't I edit the value in the WAV header to make it always work?

I'm just trying to get a working solution to producing working LPCM files.
Then the problem is in tsmuxer, must accept wav > 4GB or multiple monowavs or w64/RF64 wav headers.
I agree with tebasuna51. This is clearly a problem with tsMuxer. So why should I spend hours and hours on adding fancy and complicated features to eac3to just to workaround a simple little "bug" in tsMuxer? Instead you guys should really pester the tsMuxer programmer about adding support for big WAV files and/or for multiple mono WAV files. That will solve all your problems at once and it will furthermore make things much easier for you, as well, cause you won't have to deal with manually splitting and rejoining files, anymore.

[madshi]

You includes the -extensible parameter to output wav with the WAVE_FORMAT_EXTENSIBLE header (henceforth WFE).
I like this improvement but I have some comments:
- When use WFE header is mandatory a fix channel order.
- The most useful new data (maybe the unique) is the Channel Mask to specify the channels present in the audiodata.
- When is unknown the channels present in the source is recommended use a default Channel Mask based in the number of channels.
- When we know the source channels the channel mask must be set in accord.

1) ABOUT DEFAULTS

I think the default used in eac3to for 4 channels (0x000F = FL FR FC LF) must be changed for 0x0033 = FL FR BL BR (Quadro) more usual and compatible with Flac/Ogg multichannel.
A WFE wav with maskchannel 0x000F is rejected by flac.

The default used for 8 channel (0x00FF = FL FR FC LF BL BR FLC FRC) is not usual, maybe the default can be 0x063F = FL FR FC LF BL BR SL SR

When I try know the default for 7 channels using a 3/3.1 wav file:
eac3to v2.45
command line: "D:\Test\AudioN\eac3to\eac3to.exe" "E:\Test\7_61.wav" "E:\Test\z331.wav" -extensible
------------------------------------------------------------------------------
WAV, 6.1 channels, 0:00:20, 16 bits, 48khz
Doubling 7th channel...
Reading WAV...
Writing WAV...
Creating file "E:\Test\z331.wav"...
eac3to processing took 1 second.
Done.
For what "Doubling 7th channel..."?
Maybe the default for 7 channel can be: 0x013F = FL FR FC LF BL BR BC

2) ABOUT KNOW SOURCE AC3

At least with ac3 source we can obtain perfect WFE wav output:
acmod.lfe ac3 channels Mask and MS channels ordered Detect-MaskCh eac3to libav remap
------------------------- --------------------------------- --------------------- ------------
1 1/0.0 C 0x0004 FC 1.0 0x0004 ok not needed
1 1/0.1 C LFE 0x000C FC LF 2.0 0x0003 (1)(2) not needed
2 2/0.0 L R 0x0003 FL FR 2.0 0x0003 ok not needed
2 2/0.1 L R LFE 0x000B FL FR LF 2.1 0x0007 (2) not needed
4 2/1.0 L R S 0x0103 FL FR BC 2/1 0x0007 (2) not needed
4 2/1.1 L R S LFE 0x010B FL FR LF BC 2/1.1 0x000F (2)(3) -0,1,3,2,4,5
6 2/2.0 L R SL SR 0x0033 FL FR BL BR 2/2 0x000F (2) not needed
6 2/2.1 L R SL SR LFE 0x003B FL FR LF BL BR 2/2.1 0x0037 (2)(3) -0,1,4,2,3,5
3 3/0.0 L C R 0x0007 FL FR FC 3/0 0x0007 (3) -0,2,1,3,4,5
3 3/0.1 L C R LFE 0x000F FL FR FC LF 3/0.1 0x000F (3) -0,2,1,3,4,5
5 3/1.0 L C R S 0x0107 FL FR FC BC 3/1 0x000F (2)(3) -0,2,1,3,4,5
5 3/1.1 L C R S LFE 0x010F FL FR FC LF BC 3/1.1 0x0037 (2)(3) -0,2,1,4,3,5
7 3/2.0 L C R SL SR 0x0037 FL FR FC BL BR 5.0 0x0037 (3) -0,2,1,3,4,5
7 3/2.1 L C R SL SR LFE 0x003F FL FR FC LF BL BR 5.1 0x003F ok already done

(1) Wrong detection, must be 1.1 (not important because mono + LFE is really strange)
(2) Default mask for channel number, the correct mask can be easyly put based in detection
(3) When decoded with libav need remapping channels.
:eek:

That's extremely detailed and helpful information - thanks much!! :)

When I try know the default for 7 channels using a 3/3.1 wav file:
For what "Doubling 7th channel..."?
This is complicated. Let me try to explain: When I tried to convert my "The Descent" Blu-Ray to MKV, I found out that the LPCM track (which was credited as 6.1) was in reality 7.1 with both back channels being identical. I'm not fully sure but I guess that multichannel Blu-Ray LPCM tracks must not be 6.1 but can only be either 5.1 or 7.1. So I thought it'd make sense to convert all 6.1 stuff to 7.1 in eac3to to make it Blu-Ray compatible. Not sure if that is/was the right thing to do. Maybe not...

At least with ac3 source we can obtain perfect WFE wav output: [...]
As your charts clearly show, eac3to currently only handles mono, stereo, 5.1 (and 7.1) correctly. For all other formats eac3to's channel mapping behaviour is more or less "random". The reason for that is that I'm not really having any true channel mapping in my whole processing chain right now. Instead I've hard coded extra code everywhere for the supported channel configurations. This is not good, and I definitely need to change it. It's just that there was always something more important to do first... ;) But I think it might be time now (or soon) to implement "proper" channel mapping support.

BTW, do you have any experience with which WAV format is the more compatible one with most applications/media players? Is it the extensible one or the simpler one? I'm wondering which WAV header format I should make default...

[madshi]

We DO need to know how to make these files just like the studios are doing cause we are obviously missing something here.
The missing thing is simply that TsMuxer needs to be changed to support either big multichannel WAV files or multiple mono WAV files.

Maybe we should do:

eac3to.exe source.m2ts destination.wavs

then use another encoder to make the multi channel PCM/WAV? But, what encoder and what settings and how do we ensure proper LPCM channel configuration?
That doesn't make any sense at all. Guys, please understand that eac3to is doing all it can. You can try 20 other programs to create the multi channel WAV file and you'll always end up with similar problems. The problem is most probably NOT in eac3to's WAV creation. The problem is in TsMuxer. And unless you get TsMuxer fixed, you can forget about making WAV -> TsMuxer working properly. It's as simple as that.

[EPiPH0NE]

That doesn't make any sense at all. Guys, please understand that eac3to is doing all it can. You can try 20 other programs to create the multi channel WAV file and you'll always end up with similar problems. The problem is most probably NOT in eac3to's WAV creation. The problem is in TsMuxer. And unless you get TsMuxer fixed, you can forget about making WAV -> TsMuxer working properly. It's as simple as that.


Well if that doesn't make sense then I guess I've hit a brick wall and my axe is dull. I'm taking off the hardhat and I'll let someone else figure it out and in the mean time just use segmented files. I could understand these files not working with tsMuxeR, which I and others have reported it but Roman is not as active here as you, but I can't even get Scenarist to take them either because of improper channel config. What is the proper channel config I need to set in eac3to to get these into Scenarist or will your new "-extensible" option fix this?

[madshi]

Yeah, the "-extensible" option might things work with Scenarist. Not fully sure though if Scenarist handles big WAV files properly!

[EPiPH0NE]

Yeah, the "-extensible" option might things work with Scenarist. Not fully sure though if Scenarist handles big WAV files properly!


I would hope a $60,000 muxing app would handle big WAV files properly. It does take WAV as LPCM input so as long as I can get around MUI Generator and actually get the files into Scenarist I think I have a 50/50 shot at actually getting DTS-MA/TrueHD -> LPCM ;) I am re-ripping Hitman for the umpteenth time as we speak and then I'll test it.

[jchappo]

Hmm, is it not possible to examine one of the studio created LPCM tracks and determine what exactly is making those work?

Edit: From my experience TsMuxer handles the LPCM files that the studio created, because I remux the original M2TS and move the LPCM track to audio track 1.

[gregt]

Just as a quick follow up:

This is why I stated in the v2.45 release notes that I still have to further improve eac3to for MPEG2 clips with such mode changes. For now I recommend to either wait with converting such clips. Or to demux video instead of muxing it to MKV. I'm planning to complete the MPEG2 muxing improvements in the next build, which should then handle such mode changes just fine.


Ok. No problem. I was just reporting the data for you to use. I have a lot of DVDs for testing.
Here is a new case:
eac3to v2.45
command line: eac3to vts_10_1.vob+vts_10_2.vob 2: sg1sXe16.mkv 3: sg1sXe16.ac3
------------------------------------------------------------------------------
VOB, 1 video track, 1 audio track, 6 subtitle tracks, 0:43:30
1: Joined VOB file
2: MPEG2, 480p30 /1.001 (16:9)
3: AC3, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB
4: Subtitle
5: Subtitle
6: Subtitle
7: Subtitle
8: Subtitle
9: Subtitle
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[a03] Extracting audio track number 3...
[a03] Removing dialog normalization...
[v02] The MPEG2 stream changed from 30000p to 24000p at runtime 0:00:01.
[a03] Creating file "sg1sXe16.ac3"...
Added fps value to MKV header.
Video track 2 contains 62598 frames.
eac3to processing took 43 seconds.
Done.

During playback each frame snaps back to the previous one before going on to the next one. Very strange. For some reason not every episode has the "changed from 30000p to 24000p" and the ones that don't play perfectly.

[EPiPH0NE]

Hmm, is it not possible to examine one of the studio created LPCM tracks and determine what exactly is making those work?

Edit: From my experience TsMuxer handles the LPCM files that the studio created, because I remux the original M2TS and move the LPCM track to audio track 1.

Yes all BluRay LPCM tracks I use are always track 1 in tsMuxeR followed by a secondary AC3/DTS track which necessary for the my target player. But they always work fine. It's just the ones I'm making myself that don't work right.

[EPiPH0NE]

Ok I just tried this:

eac3to.exe source.m2ts destination.wav -extensible


And this is what I get:

M2TS, 1 video track, 2 audio tracks, 1:34:15
1: Chapters, 25 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: AC3, Spanish, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
This audio conversion is not supported.

[madshi]

Hmm, is it not possible to examine one of the studio created LPCM tracks and determine what exactly is making those work?
How often do I need to explain this? The studio created LPCM track works because it's stored inside of an m2ts container and not inside of the WAV container. If you put the studio created LPCM track into a WAV container, TsMuxer would fail to work correctly, too.

Ok I just tried this:

eac3to.exe source.m2ts destination.wav -extensible

And this is what I get:

M2TS, 1 video track, 2 audio tracks, 1:34:15
1: Chapters, 25 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: AC3, Spanish, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
This audio conversion is not supported.
Hmmmm... That's strange. Try "eac3to source.m2ts 4: dest.wav -extensible". Does that work? Or try without "-extensible", then do another pass "eac3to dest.wav dest2.wav -extensible". Does that work?

[deathlord]

So I thought it'd make sense to convert all 6.1 stuff to 7.1

I have a 7.1 setup, so I'm happy with that. :)
Though maybe you should lower the volume a bit, because simply doubling the channel would make it too loud?

In any event, should you decide to disable the doubling, please make it an option!

[Rectal Prolapse]

Keep in mind that Sonic Scenarist 4.2 has a problem handling WAV files > 4 GB. So even commercial apps can be no better than the free stuff. :)

[Inventive Software]

Keep in mind that Sonic Scenarist 4.2 has a problem handling WAV files > 4 GB. So even commercial apps can be no better than the free stuff. :)

You could of course get around that completely and use FLAC....

[Rectal Prolapse]

Scenarist handles FLAC? I didn't know that! (I don't have scenarist btw - but I've seen posts about it)

[EPiPH0NE]

Keep in mind that Sonic Scenarist 4.2 has a problem handling WAV files > 4 GB. So even commercial apps can be no better than the free stuff. :)

Well crap to that then. Do the newer versions 4.3 or 4.5 fix any of this?

[EPiPH0NE]

You could of course get around that completely and use FLAC....

Multi channel FLAC is not supported on my player.

[EPiPH0NE]

Hmmmm... That's strange. Try "eac3to source.m2ts 4: dest.wav -extensible". Does that work? Or try without "-extensible", then do another pass "eac3to dest.wav dest2.wav -extensible". Does that work?

Yes my full command included "eac3to source.m2ts 4:' I didn't want to oput the whole CMD line for security reasons. I am trying:

eac3to dest.m2ts dest2.wav

Then will do:

eac3to.exe dest2.wav dest3.wav -extensible

Seeing as my version of Scenarist BDA has probs with >4GB WAV files, I'm guessing this is fixed in newer versions, I will still be at square one unless there is way for me to properly join my other segmented files in Scenarist but I doubt that either.

[Chouonsoku]

I've got something new going on with a seamless branching disc. Since the latest versions of eac3to support seamless branching I decided to give Nine Inch Nails: Beside You in Time another try.

F:\Video\Movies>eac3to HALO_22
1) 00000.mpls (angle 1), 1:32:30
[0+29+31+28+27].m2ts
- VC-1, 1080i30 /1.001 (16:9)
- TrueHD, English, multi-channel, 48khz
- AC3, English, stereo, 48khz

2) 00000.mpls (angle 2), 1:32:30
[0+30+31+28+27].m2ts
- VC-1, 1080i30 /1.001 (16:9)
- TrueHD, English, multi-channel, 48khz
- AC3, English, stereo, 48khz

3) 00006.mpls, 00010.m2ts, 0:19:05
- VC-1, 1080i30 /1.001 (16:9)
- TrueHD, English, multi-channel, 48khz
- AC3, English, stereo, 48khz

eac3to v2.45
command line: eac3to HALO_22 1) -demux
------------------------------------------------------------------------------
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
M2TS, 1 video track, 2 audio tracks, 1:32:30
1: Chapters, 22 chapters
2: VC-1, 1080i60 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
4: AC3, English, 2.0 channels, 256kbit/s, 48khz
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
The program channel mapping changes in the middle of the stream.
Creating file "00000 - Chapters.txt"...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Extracting audio track number 3...
[a04] Extracting audio track number 4...
[a03] Extracting AC3 stream...
[a03] Extracting TrueHD stream...
[a03] Creating file "00000 - 3 - TrueHD+AC3, English, 5.1 channels, 48khz.thd"...
[v02] Creating file "00000 - 2 - VC-1, 1080i60.vc1"...
[a03] Creating file "00000 - 3 - TrueHD+AC3, English, 5.1 channels, 48khz.ac3"...
[a04] Creating file "00000 - 4 - AC3, English, 2.0 channels, 256kbps, 48khz.ac3"...
[a03] The program channel mapping changes in the middle of the stream.
[v02] The program channel mapping changes in the middle of the stream.
[a03] The program channel mapping changes in the middle of the stream.
[a04] The program channel mapping changes in the middle of the stream.
Aborted at file position 16586463232.

This has been a pain in the ass disc since the day I bought it, and every time I try to fool around with it again, something new stops me. >.<

Edit: I've also tried using 2) instead of 1), but I get the same error.

[jchappo]

How often do I need to explain this? The studio created LPCM track works because it's stored inside of an m2ts container and not inside of the WAV container. If you put the studio created LPCM track into a WAV container, TsMuxer would fail to work correctly, too.


Excuse my ignorance madshi, I'm still learning. I'm not saying there is anything wrong with eac3to, I am just trying to find a solution to this problem. I imagine there will be many more people like myself who have this same problem in the recent future.

[EPiPH0NE]

Excuse my ignorance madshi, I'm still learning. I'm not saying there is anything wrong with eac3to, I am just trying to find a solution to this problem. I imagine there will be many more people like myself who have this same problem in the recent future.


Yeah it seems you will need to use Scenarist BDA 4.3 or above to be able to do proper DTS-MA/TrueHD -> PCM/WAV conversions as tsMuxeR needs to support multi mono WAV like Scenarist does for this to happen for the open source peeps. Madshi already said this but I'll say it again for good measure.

[wildchild22]

I have spiderman2 bluray with truehd converted to lpcm. It cannot be done without scenarist as of yet. After scenarist builds the file. You can demux the file into elementry streams and rebuild it and the audio is perfect with lpcm no 20% freezeup . However if you choose to demux and remux you need to use the append button as the audio gets split into two wav files.


Also when you import a wav file from eac3to into tsmuxer. Ts muxer says it s applying default channel information while the wav from scenarist doesnt do this so I am guessing that scenarist must add something different to the header. here is the first few lines of the scenarist file opened using XVI32 ( all you have to do is copy the garbled mess below into notepad then save it as a file then open it with xvi32 I think this may have relevant info )

RIFFüñÿWAVEfmt ( þÿ €» /
    ?
 € ª 8›qdataÀñÿ

[jchappo]

I just demuxed a studio created lpcm track from Underworld. Here is what eac3to said about it:

C:\DVDRips\HD>eac3to.exe Ripped\output.wav
The format of the source file could not be detected.

This means it's not a WAV. But, TsMuxer recognizes the file as LPCM, and I can remux it back with the video, and it still plays fine on the Popcorn Hour.

Can eac3to be made to create the decompressed audio into this format? It could be a new output option like -popcorn?

[wildchild22]

here is the hex edit from the eac3to wav file

RIFFêÿ~WAVEfmt 
 €» /
  dataÆÿ~

[nautilus7]

I just demuxed a studio created lpcm track from Underworld. Here is what eac3to said about it:

C:\DVDRips\HD>eac3to.exe Ripped\output.wav
The format of the source file could not be detected.

This means it's not a WAV. But, TsMuxer recognizes the file as LPCM, and I can remux it back with the video, and it still plays fine on the Popcorn Hour.

Can eac3to be made to create the decompressed audio into this format? It could be a new output option like -popcorn?

How did you demuxed it?

[jchappo]

using tsremux

[tebasuna51]

I just demuxed a studio created lpcm track from Underworld. Here is what eac3to said about it:

C:\DVDRips\HD>eac3to.exe Ripped\output.wav
The format of the source file could not be detected.

This means it's not a WAV. But, TsMuxer recognizes the file as LPCM, and I can remux it back with the video, and it still plays fine on the Popcorn Hour.

Please upload first 20MB from this LPCM file recognized by TsMuxer

@wildchild22, here you have two links about wav header:
http://ccrma.stanford.edu/CCRMA/Courses/422/projects/WaveFormat/
http://www.lightlink.com/tjweber/StripWav/Canon.html

The RIFF ChunkSize must be FileLength - 8 and the Data SubChunkSize can be FileLength - 44, both fields have 4 bytes.
How can you hexedit 4 bytes to put a number greater than 2^32 (4 GB)?

[jchappo]

Here is the first 20mb:
http://hd-r-us.kicks-ass.net/public/output.wav.001

I checked, and this chunk is recognized by TsMuxer.

Also, the solution may not involve the WAV container, because it obviously doesn't support file size > 4gb.

[gregt]

Hi,
I am trying to convert - using eac3to - BR Pirates of the Carribean - Deadman's Chest and the resultant mkv video height is about 30% shorter than the source m2ts file. I then demuxed using eac3to and the resultant h264 file matches the source. I then used mkvmerge to convert the h264 file and the resultant mkv video is again about 30% shorter than the source h264 video.
Is there an option that I need to use or is this a bug in the mkv code?

[gregt]

Hi,
I am trying to convert - using eac3to - BR Pirates of the Carribean - Deadman's Chest and the resultant mkv video height is about 30% shorter than the source m2ts file. I then demuxed using eac3to and the resultant h264 file matches the source. I then used mkvmerge to convert the h264 file and the resultant mkv video is again about 30% shorter than the source h264 video.
Is there an option that I need to use or is this a bug in the mkv code?

ok. I solved my own problem. It is not an mkv issue. The issue is with the h264 decoder in ffdshow. Nero 8 uses it's own internal codecs and plays the mkv video properly. If I disable libavcodec as the default codec for h264 in ffdshow then WMP will use the Nero codec and the mkv then plays back properly. I will have to check to see if there is an update to ffdshow to fix this issue.

Anyway, not sure why it did not affect other mkv videos.

[nautilus7]

I just demuxed a studio created lpcm track from Underworld. Here is what eac3to said about it:

C:\DVDRips\HD>eac3to.exe Ripped\output.wav
The format of the source file could not be detected.

This means it's not a WAV. But, TsMuxer recognizes the file as LPCM, and I can remux it back with the video, and it still plays fine on the Popcorn Hour.

Can eac3to be made to create the decompressed audio into this format? It could be a new output option like -popcorn?

Here is the first 20mb:
http://hd-r-us.kicks-ass.net/public/output.wav.001

I checked, and this chunk is recognized by TsMuxer.

Also, the solution may not involve the WAV container, because it obviously doesn't support file size > 4gb.
Will you stop filling this thread with nonsense? Will you stop wasting people's time?

1: WAV supports >4GB files and eac3to handles them fine. It's just tsremux that doesn't support them! Post a request THERE!
2: eac3to is just fine and no special switches (like -popcorn) are needed!
3: The link to your sample does not work!

[wildchild22]

All I did was slit the file into a 100 meg slice then opened the 100 meg slice with the hex editor and posted the bytes before they were all zero till later on. There is a difference in the first 2 lines so maybe eac3to needs to write the differences into the header.

[tebasuna51]

Will you stop filling this thread with nonsense? Will you stop wasting people's time?

1: WAV supports >4GB files and eac3to handles them fine. It's just tsremux that doesn't support them! Post a request THERE!
2: eac3to is just fine and no special switches (like -popcorn) are needed!
3: The link to your sample does not work!

I agree with nautilus7. Maybe you want continue with this problem, related to the the 4GB WAV limit, in this thread (http://forum.doom9.org/showthread.php?t=137541)

A working link to the lpcm sample please.

[tosehee]

Hi.

Thanks for a wonderful tool.

I have used this tool to rip my BD and HDDVDs. Two days ago, I tried to rip Chris Botti Live with friends, but for some reason, I can't seem to play this after the conversion. I have tried this on 2.44 and 2.45 version. Both of them produces the gap information. I re-ran the same command to produce the new mkv and flac file, however, after I remux them with mkvmergeGui, it doesn't play. I have tried both with and without gap and the results are same.

The symtom I have is, the mkvtoolnix remux them fine, but when I tried to play in any player (zoomplayer, mpc, kmp, media portal, and etc), none of them works. They just hangs in the beginning, and nothing happens.

I don't see any errors from the eac3to or mkxtoolnix. Oh, here is the command I used.

C:\Users\Public\Videos\Recent>eac3to 00000.m2ts 1: v.mkv 3: a.flac
M2TS, 1 video track, 3 audio tracks, 1:29:02
1: h264/AVC, 1080i60 /1.001 (16:9)
2: AC3, 2.0 channels, 640kbit/s, 48khz
3: RAW/PCM, 5.1 channels, 24 bits, 96khz
4: AC3, 5.1 channels, 640kbit/s, 48khz
[v01] Extracting video track number 1...
[a03] Extracting audio track number 3...
[a03] Reading RAW/PCM...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Encoding FLAC...
[v01] Muxing video to Matroska...

[jchappo]

Looks like my own hosting solution didn't work out, try this:
http://www.mediafire.com/?yjjvm0l1zfl

Why is everyone hung up on WAV now? I was only using WAV because I thought that was the only way to make it work.

What I want is to create the exact format of this studio LPCM, which is recognized by TsMuxer AND plays perfectly on my A/V receiver. >4gb WAV files are recognized by TsMuxer BUT do not play perfectly on my A/V receiver.

[tosehee]

The only thing that I haven't tried is demux the video as .h264. If I do that, what frame rates do I need to assign when remuxing with mkvmerge?

Is it 30 or is it 60? It's 1080i material.. So, it it 30 then? or is 60 still?

[jchappo]

usually 29.9 30000/1001

[tosehee]

usually 29.9 30000/1001

Do I type in 29.97 or 29.9 30000/1001 exactly as you typed? Sorry for dumb question. :-(

[nautilus7]

I believe it's a player/decoder issue. Change the video decoder (ffdshow/coreavc/mpc internal decoder) and report back. Try also the .mkv file that eac3to makes (without audio).

From 1st post:

-24p/30p/60i force the use of the specified framerate for h264 muxing

But, eac3to should detect fps automatically.

[jchappo]

Ok here is yet more info,

Looks like if I change the extension to RAW, then eac3to has something to say about it:

C:\DVDRips\HD>eac3to.exe Ripped\output.raw
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 big 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, 2:13:48, 16 bits, 48khz

I will try converting a DTS-HD track to RAW, and then test is TsMuxer will accept that.

[tosehee]

I believe it's a player/decoder issue. Change the video decoder (ffdshow/coreavc/mpc internal decoder) and report back. Try also the .mkv file that eac3to makes (without audio).

From 1st post:



But, eac3to should detect fps automatically.

I have done that, but when it tries to open the file, nothing happens and it hangs forever. I don't see any video or hear any sound. That's with eac3to making the video mkv. I haven't tried demuxing it manually and remux again with mkvmerge. That's what I was at. I tried the latest version of ffdshow tryout (May 04, 2008) and I tried coreavc 1.7 professional, and same result.

Mind you that I have ripped about 30+ movies so far without this type of issue now.

[jchappo]

Do I type in 29.97 or 29.9 30000/1001 exactly as you typed? Sorry for dumb question. :-(

Well 30000/1001 = 29.97, some programs list it either in 29.9 or 30000/1001. If it asks for just fps use 29.97

[tosehee]

Well 30000/1001 = 29.97, some programs list it either in 29.9 or 30000/1001. If it asks for just fps use 29.97

Thought so. Thought I confirm with you before wasting another 40 min. :-)

I will report back if I am successful with demuxing the video manually and remux them with mkvmerge.

[nautilus7]

Ok here is yet more info,

Looks like if I change the extension to RAW, then eac3to has something to say about it:

C:\DVDRips\HD>eac3to.exe Ripped\output.raw
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 big 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, 2:13:48, 16 bits, 48khz

I will try converting a DTS-HD track to RAW, and then test is TsMuxer will accept that.I just got your sample. So, it seems you demuxed a pcm track, set the extension to wav and then post here saying that eac3to can't detect your "wav" file. Wtf? In addition, i opened tsmuxer and loaded the pcm track. Its parameters were detected correctly!!!

SO MY CONCLUSION IS THAT YOU YOU CAN USE EAC3TO TO CONVERT YOUR AUDIO TO .PCM OR .RAW AND LOAD THEM IN TSMUXER. THAT WOULD WORK FINE FOR YOU.

Well 30000/1001 = 29.97, some programs list it either in 29.9 or 30000/1001. If it asks for just fps use 29.97
You can't set what ever you like. There are switches (i posted them above). 24p stand for 24/1001 or 23,976... progressive, 30p stands for 30/1001 or 29,97... progressive and 60i stands for 60/1001 or 59,94... interlaced.

[jchappo]

Actually nautilus7, I have been posting the whole time that I am converting to WAV(PCM) because I thought that was the only way to make it work. Once I hit the 4gb barrier and things stopped working, I asked around here and found out what was wrong. Now it seems I have determined that RAW is the way to go, not WAV.

Hence, this is the solution many other people were looking for...

1) Convert TrueHD/DTS-HD to RAW
2) Rename file to .wav and mux with video in TsMuxer

Thanks for your time.

[Beastie Boy]

Hence, this is the solution many other people were looking for...

1) Convert TrueHD/DTS-HD to RAW
2) Rename file to .wav and mux with video in TsMuxer

Thanks for your time.

It's a shame you didn't spot my post (http://forum.doom9.org/showthread.php?p=1134707#post1134707)earlier :)

[nautilus7]

Actually nautilus7, I have been posting the whole time that I am converting to WAV(PCM) because I thought that was the only way to make it work. Once I hit the 4gb barrier and things stopped working, I asked around here and found out what was wrong. Now it seems I have determined that RAW is the way to go, not WAV.How are we supposed to know what tsmuxer does and does not accept as input????

You posted here saying you found some weird problem regarding TrueHD & DTS-HD decoding based on the fact that the wav files you made with eac3to weren't accepted by tsmuxer. Madshi respond that everything is ok with the decoders and problem is that tsmuxer can't deal with BIG wav files, but you kept posting and posting and posting here, until i told that tsmuxer (which i don't use and i wasn't aware) can accept raw and pcm format.

So, the correct order to do things should have been:
1. Find what tsmuxer needs/accepts
2. Find a way to make what it needs.
3. Post here questions about using eac3to (since you chose it for make your audio).

Hence, this is the solution many other people were looking for...

1) Convert TrueHD/DTS-HD to RAW
2) Rename file to .wav and mux with video in TsMuxer

Thanks for your time.I don't see any...

It's a shame you didn't spot my post (http://forum.doom9.org/showthread.php?p=1134707#post1134707)earlier :)
Yeah beastie, it's really shame.

[madshi]

Don't have much time right now, but the "WAV" file posted by jchappo is neither WAV nor RAW. It seems to contain blocks of LPCM data with a 4 byte header for each block. It seems to me that either tsMuxer uses its own private data format for storing LPCM data, or there's a bug in tsMuxer's demuxing code, or there's a new "container" format I've never heard about yet.

[nautilus7]

He demuxed it with tsremux he said, not tsmuxer. Maybe that's the problem. Anyway the conclusion is that tsmuxer accepts .raw/.pcm

[jchappo]

It's a shame you didn't spot my post (http://forum.doom9.org/showthread.php?p=1134707#post1134707)earlier :)

Doh, I should have seen it, I was just skimming for madshi replys :stupid:

[jchappo]

How are we supposed to know what tsmuxer does and does not accept as input????

You posted here saying you found some weird problem regarding TrueHD & DTS-HD decoding based on the fact that the wav files you made with eac3to weren't accepted by tsmuxer.

Maybe you should re-read all my posts because I never said such a thing. I merely said the decoded DTS-HD plays garbage audio after 20-30%. I never once indicated a problem with TsMuxer

[nautilus7]

You still don't get it huh? How did you determine that dts-hd tracks decoded to garbage? Because you couldn't play them in popcornhour once muxed with tsmuxer!!! Who said that you indicated a problem with tsmuxer? Madshi did!

I won't continue this nonsense any further. PM me if you want to.

[jchappo]

Actually, I was able to copy the WAV file itself onto the popcorn hour and play it without muxing to M2TS. Try again.

[tosehee]

K.

The issue remains. But, unlike what I originally thought, it's not the video, but the audio. When I play the flac itself, the audio sounds like a tape dac playing at x10 slower. Does that make sense?

Again, this is what eac3to reports...

C:\Users\Public\Videos\Recent>eac3to 00000.m2ts
M2TS, 1 video track, 3 audio tracks, 1:29:02
1: h264/AVC, 1080i60 /1.001 (16:9)
2: AC3, 2.0 channels, 640kbit/s, 48khz
3: RAW/PCM, 5.1 channels, 24 bits, 96khz
4: AC3, 5.1 channels, 640kbit/s, 48khz

and I converts channel 3 to flac using..

eac3to 00000.m2ts 1: v.mkv 3: a.flac

[nautilus7]

The mkv file created by eac3to plays fine then? You said it didn't.

Regarding audio... Maybe pcm parameters aren't correctly detected. Is the pcm track 5.1 ch 24 bits 96 khz? Or maybe it's a decoder issue (flac decoder i mean). Try a different one, or convert to another format (e.g. pcm --> ac3) to ensure it's not the decoder.

[tosehee]

The mkv file created by eac3to plays fine then? You said it didn't.

Regarding audio... Maybe pcm parameters aren't correctly detected. Is the pcm track 5.1 ch 24 bits 96 khz? Or maybe it's a decoder issue (flac decoder i mean). Try a different one, or convert to another format (e.g. pcm --> ac3) to ensure it's not the decoder.

What decoder is it using in default? is it libav? and what should I be using to decode PCM 24bit/96khz to flac conversion?

As for video, I didn't know better it was at halt for like 2 minutes before it started to play. When it plays, the videos are fine, but the audio plays like 10x slow mode. Of course, the audio and video is not in sync and it's not watchable.. i mean inaudible as the video plays fine.

[nautilus7]

What decoder is it using in default? is it libav? and what should I be using to decode PCM 24bit/96khz to flac conversion?Flac decoder in your player (mpc). Do you use madflac or ffdshow or other? ffdshow for instance is known that can't handle 24 bit flac well.

PCM isn't compressed and doesn't need decoding. Since the pcm parameters are really those detected by eac3to, you don't need to change anything in your command line.

Play the flac track (alone) in your media player, trying different decoders. Also, convert the source pcm track to another format (wav or ac3) to narrow down the possibilities of a wrong eac3to pcm detection.

You can also make a ac3 track from the flac you have already made. If that ac3 plays ok, you definitely have a flac decoding problem in your media player.

Hope i didn't confuse you. :D

[tosehee]

Flac decoder in your player (mpc). Do you use madflac or ffdshow or other? ffdshow for instance is known that can't handle 24 bit flac well.

PCM isn't compressed and doesn't need decoding. Since the pcm parameters are really those detected by eac3to, you don't need to change anything in your command line.

Play the flac track (alone) in your media player, trying different decoders. Also, convert the source pcm track to another format (wav or ac3) to narrow down the possibilities of a wrong eac3to pcm detection.

I am using madFlac. I tried both ffdshow and madflac. They are both exhibiting same issue. I haven't converted to WAV or any other format yet, only flac. But I can give that a try next.

[tosehee]

4th channel is AC3. When demuxed to flac, it plays fine. It seems that from pcm --> flac for this particular concert is the issue.

In the case of bad ripping, I tried re-rip three times already and they all have exactly same file size and structure. I doubt it's the bad rip, but who knows.. I dont' get any errors in encoding process though.

[n0mag!c]

@Madshi
I'm needing to use your helpful tool for the first time and the questions (maybe stupid) comes to light. I'm sorry in advance.
I've decoded E-AC3 1536kbps to RAW file with GraphEdit and its Dump filter. By the way, I did it with Nero(8!) Audio Decoder 2, so how it correlates with yours "Nero 8 doesn't allow its DirectShow filters to be used from outside of Nero ShowTime"? E-AC3 track was muxed to .ts file and demuxed back with haali splitter in graph. But eac3to really can't use "Nero Audio Decoder 2". This is my first question.
And my second stupid question is - how can I get from eac3to "WAVs (multiple mono WAV files, PCM only)"? (I gave up playing with eac3to options).

[jchappo]

Hence, this is the solution many other people were looking for...

1) Convert TrueHD/DTS-HD to RAW
2) Rename file to .wav and mux with video in TsMuxer

Thanks for your time.

1) Convert TrueHD/DTS-HD to PCM, big endian is needed, where RAW is little endian

[Yraen]

And my second stupid question is - how can I get from eac3to "WAVs (multiple mono WAV files, PCM only)"? (I gave up playing with eac3to options).

You actually answered your own question there, wavs is the right answer.

eac3to input.file output.wavs

[monohouse]

-----

[jchappo]

Don't have much time right now, but the "WAV" file posted by jchappo is neither WAV nor RAW. It seems to contain blocks of LPCM data with a 4 byte header for each block. It seems to me that either tsMuxer uses its own private data format for storing LPCM data, or there's a bug in tsMuxer's demuxing code, or there's a new "container" format I've never heard about yet.

I demuxed with tsremux, and muxed with tsmuxer so I don't think it's related to a specific program?

tsmuxer won't accept the pcm output from eac3to, so that extra header is what it needs. could the header be related to the m2ts container it came from?

[jchappo]

why swapping endian is necessary for raw/pcm to flac conversion ?
is that process looses data ?

why does it remap channels ? I wasn't awear that remapping channels was required, is the process losses data ? is it possible to avoid both of the processes ?

endian swap is just a re-ordering of the bits different people use different endian, so it's needed to swap between them.

same thing with channel mapping, some people use different order.

[monohouse]

-----

[jchappo]

well you have the exact same number of bits, they are just in different order.

[Thunderbolt8]

the AVC video track of the 'things we lost in the fire' is indicated as a 16:9 track with pulldown flags. is this of any importance for me when I just want to remux the movie? so far I just remuxed it as usual and it also looks normally, but I just want to be sure.

[n0mag!c]

You actually answered your own question there, wavs is the right answer.
eac3to input.file output.wavs
Wow! Thanks a lot! I would never figure it out by myself! Well it's obvious and not obvious at the same time.

[Rectal Prolapse]

Well crap to that then. Do the newer versions (of Scenarist) 4.3 or 4.5 fix any of this?

I think it was fixed in 4.3, but I'm not 100% sure.

[EPiPH0NE]

I think it was fixed in 4.3, but I'm not 100% sure.


I just got 4.3 and I know you can do multi mono WAVs -> LPCM for sure.

[Momber]

Hey madshi!

Thanks for your continuing work on eac3to. You have developed it into one helluva tool!

This week I've been working on converting the Galactica Season 1 HD DVDs to BluRay, using eac3to for audio and EVOdemux + vc1conv for the video.
The VC-1 video processed like this is recognized by tsMuxeR as 1080p @ 23.976 fps and the remuxed BluRays play back perfectly smooth in PDVD 8 at a refresh rate of 2x 23.976 Hz.

So I thought I'd give eac3to a go on the video front, too, and had it demux and remove the pulldown flags. The resulting vc-1 file is recognized by tsMuxeR as 1080i @ 23.976 fps and what's worse, playback got really jerky about 17 minutes into the episode I tried.

Is it so that eac3to only removes the pulldown=1 flag but leaves the interlaced=1 flag in place? And if that is the case, would you please be so kind as to correct it?

TIA!
S.

[dorati]

I have a question:
Eac3To reportet this:

1) 00120.mpls (angle 1), 00152.m2ts+00153.m2ts+00154.m2ts, 1:26:15
- h264/AVC, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48khz
- AC3, German, multi-channel, 48khz
- RAW/PCM, German, multi-channel, 48khz
- AC3, English, stereo, 48khz

2) 00120.mpls (angle 2), 00152.m2ts+00155.m2ts+00154.m2ts, 1:26:15
- h264/AVC, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48khz
- AC3, German, multi-channel, 48khz
- RAW/PCM, German, multi-channel, 48khz
- AC3, English, stereo, 48khz

3) 00037.mpls, 00004.m2ts+00005.m2ts, 0:22:50
- MPEG2, 480i30 /1.001 (16:9)
- AC3, English, stereo, 48khz

4) 00004.mpls, 00004.m2ts, 0:15:14
- MPEG2, 480i30 /1.001 (16:9)
- AC3, English, stereo, 48khz

What I must use in the commandline, to use: 2) 00120.mpls (angle 2) ??

[nautilus7]

Type the command you typed to list the above plus 2) at the end.

See my example here (http://forum.doom9.org/showpost.php?p=1123939&postcount=4226)for details.

[Yraen]

So I thought I'd give eac3to a go on the video front, too, and had it demux and remove the pulldown flags. The resulting vc-1 file is recognized by tsMuxeR as 1080i @ 23.976 fps and what's worse, playback got really jerky about 17 minutes into the episode I tried.


Here (http://forum.doom9.org/showthread.php?p=1110490#post1110490) you can see where madshi says that vc1conv is better than eac3to for removing pulldown and temporarily disabled -stripPulldown. You could go eac3to > vc1conv > tsMuxer and save a step. This will get you 1080p @ 23.976.

[BlackJack1]

I can not drag'n'drop eac3to app. and hddvd/bd files to cmd window on Vista system on my laptop. Do you know how to fix it? In desktop machine with WinXP - no problem...

[dorati]

@nautilus7:
Thx !!

Somebody has succesfully converted the Movie "Open Season"?

The Movie is 00152.m2ts + 00153.m2ts + 00154.m2ts.
or in German 00152.m2ts + 00155.m2ts + 00154.m2ts.

But the change from 00152.m2ts to 00153/00154.m2ts is not clean. My standaloneplayer (popcorn) hangs at the change.
At Windows-PC with Powerdvd is not better. It stops a short Moment and than goes on with playback.

I testet all:
- copy /b 00152.m2ts + 00153.m2ts + 00154.m2ts
- demux and mux with TSMuxer
- demux with eac3to (playlist) and Mux with TSMuxer
- convert to MKV

Somebody can help?

[nautilus7]

Can you post the eac3to log? Re ripping the disc could make the trick.

[Momber]

here you can see where madshi says that vc1conv is better than eac3to for removing pulldown and temporarily disabled -stripPulldown.
Thank you. That was an older version and I thought perhaps in the meantime things were improved.
You could go eac3to > vc1conv > tsMuxer and save a step. This will get you 1080p @ 23.976.
Right - that's what I did in the end. It's just that with eac3to -removePulldown we would be able to save another step. When I'm converting episode after episode of TV series I'm grateful for every chance to streamline my workflow.
The batch creation of eac3to_more_gui is heaven-sent in that regard btw.

Ta
S.

[Sephiroth0000]

Hiya everyone,

I am currently using EAC3TO GUI (Keymakers version) and noticed that one of the options is video quality which spans from quality 0 to quality 4....can anyone tell me the differences as I am trying to make the mkvs smaller but do not want to sacrifice too much quality.

[nautilus7]

From 1st post:

-quality=4 slowdown/speedup/resampling quality (0 = low; 4 = very high)

So, nothing to do with video. Probably a bug. My advice is to use the new eac3to gui (http://forum.doom9.org/showthread.php?t=135095), since it's up-to-date.

[mic64]

Hi All

I got a problem with all FLAC Files.
its not the encoding into it or playing them back.
Its something different.
I wanted to extract some flac from my MKVs.
mkvextract gives you this
Extracting track 2 with the CodecID 'A_FLAC' to the file 'z:\xxxxx.flac'. Container format: Ogg (FLAC in Ogg)

Well ok so far I thought. But..
those flac files are not being recognized by eas3to at all, so no way to convert them into something else.

Any idea? or solution?

[Snowknight26]

mkvextract tracks source.mka --no-ogg 1:dest.flac

[EPiPH0NE]

@nautilus7:
Thx !!

Somebody has succesfully converted the Movie "Open Season"?

The Movie is 00152.m2ts + 00153.m2ts + 00154.m2ts.
or in German 00152.m2ts + 00155.m2ts + 00154.m2ts.

But the change from 00152.m2ts to 00153/00154.m2ts is not clean. My standaloneplayer (popcorn) hangs at the change.
At Windows-PC with Powerdvd is not better. It stops a short Moment and than goes on with playback.

I testet all:
- copy /b 00152.m2ts + 00153.m2ts + 00154.m2ts
- demux and mux with TSMuxer
- demux with eac3to (playlist) and Mux with TSMuxer
- convert to MKV

Somebody can help?

If you have a PCH why not read the forums? ES streams in MKV is NOT supported, you need to keep it in TS/M2TS. The answer to AVC + seamless branching is right here:

http://www.networkedmediatank.com/viewtopic.php?t=2921

Scroll down to where it says 'Seamless Branching (AVC)'. You can replace TSSplitter with tsMuxeR for doing the first join which I will do when I update the guide.

[mic64]

mkvextract tracks source.mka --no-ogg 1:dest.flac

forgot to say..the --no-ogg option didn't help to get a pure flac from the mkv.
But I've found a way.
mplayer -dumpaudio -dumpfile works.

[madshi]

I've got something new going on with a seamless branching disc. Since the latest versions of eac3to support seamless branching I decided to give Nine Inch Nails: Beside You in Time another try.
Could I have a sample, please, with which this problem occurs?

I have used this tool to rip my BD and HDDVDs. Two days ago, I tried to rip Chris Botti Live with friends, but for some reason, I can't seem to play this after the conversion. I have tried this on 2.44 and 2.45 version. Both of them produces the gap information.
For video or audio or for both?

Can I have a sample of the m2ts file, please?

I've decoded E-AC3 1536kbps to RAW file with GraphEdit and its Dump filter. By the way, I did it with Nero(8!) Audio Decoder 2, so how it correlates with yours "Nero 8 doesn't allow its DirectShow filters to be used from outside of Nero ShowTime"?
Huh!? Maybe Nero changed something in the later Nero 8 builds? Can somebody double check this?

I demuxed with tsremux, and muxed with tsmuxer so I don't think it's related to a specific program?

tsmuxer won't accept the pcm output from eac3to, so that extra header is what it needs. could the header be related to the m2ts container it came from?
It seems to me that TsRemux outputs a specific LPCM format which TsMuxer understands and likes. People always complained about that TsRemux' LPCM demuxing produces files with garbage in them. Maybe the headers in the demuxed files are kept there intentional? But I don't think this is any "official" format, at least I've never heard about that yet.

so the flac "multichannel standard" has different positions for every channel than the track in question ? so eac3to can actually tell which channel is what ?

and the endian is also because flac uses a different endian ?
Each codec has its own private channel ordering. E.g. LPCM data stored in m2ts has a different channel ordering than WAV files have. eac3to knows the channel ordering of all supported input formats/codecs and remaps the channels to the WAV channel order in the first step. Furthermore LPCM data is stored in big endian while WAV files are usually stored in little endian. So eac3to changes that, too.

But don't worry, these are all totally lossless modifications.

the AVC video track of the 'things we lost in the fire' is indicated as a 16:9 track with pulldown flags. is this of any importance for me when I just want to remux the movie? so far I just remuxed it as usual and it also looks normally, but I just want to be sure.
All HD DVD movies have pulldown flags, older eac3to versions just didn't show this information. So there's nothing you need to do differently than before.

You have developed it into one helluva tool!
Thanks... :) It gets a bit better every week...

So I thought I'd give eac3to a go on the video front, too, and had it demux and remove the pulldown flags. The resulting vc-1 file is recognized by tsMuxeR as 1080i @ 23.976 fps and what's worse, playback got really jerky about 17 minutes into the episode I tried.

Is it so that eac3to only removes the pulldown=1 flag but leaves the interlaced=1 flag in place? And if that is the case, would you please be so kind as to correct it?
eac3to's current pulldown removal is based on an outdated vc1conv version. At some time I had even removed it (because the latest vc1conv is better), but put it back in on request. I will probably include the latest vc1conv logic in a future eac3to build. But I've only limited programming time for eac3to, so I can only do so much every week. For now if you need to have the pulldown completely removed please use eac3to for demuxing and then vc1conv afterwards.

Somebody has succesfully converted the Movie "Open Season"?

The Movie is 00152.m2ts + 00153.m2ts + 00154.m2ts.
or in German 00152.m2ts + 00155.m2ts + 00154.m2ts.

But the change from 00152.m2ts to 00153/00154.m2ts is not clean. My standaloneplayer (popcorn) hangs at the change.
At Windows-PC with Powerdvd is not better. It stops a short Moment and than goes on with playback.
That's strange! Did eac3to report any *video* gaps? How long is that 00152.m2ts file? It would help if you could send me a the last 20MB of the 00152.m2ts file and the first 20MB of the 00153.m2ts file.

[Encoder888]

@madshi

eac3to 2.45 successfully extracted the correct chapters from The Terminator, but now I'm having another chapter issue with Terminator 3. eac3to reports and demuxes 97 chapters, almost one every 1-2 minutes, which I'm sure is not the correct chapter structure. Here's a link to the .mpls file:

http://www.mediafire.com/?n2cxkcm9amv

Please, let me know if you were able to fix it, so I know whether to wait for the next eac3to release before I mux my chapter file :) Thanks.

[Atak_Snajpera]

Madshi
Could you add --demuxaudioonly switch? Or something similar which will extract all audio streams without video.

[dorati]

@epiphone:
If you have a PCH why not read the forums? ES streams in MKV is NOT supported, you need to keep it in TS/M2TS. The answer to AVC + seamless branching is right here

I read - but this way don't work - Sorry.

1. Joining with TSSplitter or copy /b (i tested both - same result)
2. Demux AVC-Videostream and AC3-Audiostream with TsRemux
3. Mux with tsMuxer

Later the join-point the video ist stuttering and later 20 Sec. hangs. The stuttering you can see in the short clips, I uploaded.

@madshi:
Audiogaps are reported - but no Videogaps

http://rapidshare.de/files/39377863/00152.m2ts.html
http://rapidshare.de/files/39377791/00155.m2ts.html

[madshi]

@tebasuna51, do you happen to have a sample for acmod 1 with LFE ("1/0.1")? I don't understand why it shows as "2.0" for you. Looking at my source code it should show as "1.1".

[EPiPH0NE]

@epiphone:


I read - but this way don't work - Sorry.

1. Joining with TSSplitter or copy /b (i tested both - same result)
2. Demux AVC-Videostream and AC3-Audiostream with TsRemux
3. Mux with tsMuxer

Later the join-point the video ist stuttering and later 20 Sec. hangs. The stuttering you can see in the short clips, I uploaded.

@madshi:
Audiogaps are reported - but no Videogaps

http://rapidshare.de/files/39377863/00152.m2ts.html
http://rapidshare.de/files/39377791/00155.m2ts.html

How are you playing these files cause I've personally tested this method several times with flawless playback from internal HDD.

[dorati]

At PC (PowerDVD) it works - at Standalone not.
I use NMT (http://www.syabas.de/)

[madshi]

eac3to 2.45 successfully extracted the correct chapters from The Terminator, but now I'm having another chapter issue with Terminator 3. eac3to reports and demuxes 97 chapters, almost one every 1-2 minutes, which I'm sure is not the correct chapter structure.
BDEdit outputs the same chapters. Why are you sure that this is not correct?

The problem is this: There are 2 different types of marks in the playlist files: "entry points" and "link points". Originally I only used "entry points" for the chapter export. This resulted in the problem you experienced with "The Terminator". The first half of the chapters in "The Terminator" were "entry points" and the 2nd half were "link points". Now with Terminator 3 there are about 3 "link points" for every "entry point". If I'd ignore the "link points", the chapter list would be more reasonable for Terminator 3. However, if I ignore the "link points" then The Terminator will only have chapters in the first half of the movie. BDEdit exports both "entry points" and "link points".

[madshi]

I read - but this way don't work - Sorry.

1. Joining with TSSplitter or copy /b (i tested both - same result)
2. Demux AVC-Videostream and AC3-Audiostream with TsRemux
3. Mux with tsMuxer

Later the join-point the video ist stuttering and later 20 Sec. hangs. The stuttering you can see in the short clips, I uploaded.
Thanks for the upload. Next time if possible please use a different server, though. rapidshare is the worst of all. I had to wait a minute to download the first sample and than over an hour to download the 2nd one. Anyway...

I've checked the samples you uploaded and they work beautifully with eac3to v2.45 for me. I've simply done this:

"eac3to 00152.m2ts+00155.m2ts movie.mkv" and the resulting MKV played perfectly fine with no stuttering at all on my HTPC with Media Player Classic (Haali Media Splitter + Cyberlink h264 decoder). The join point between the two m2ts parts was not visible at all.

Have you tried this command line? Are you using eac3to v2.45?

[madshi]

Could you add --demuxaudioonly switch? Or something similar which will extract all audio streams without video.
Of course I could. But I don't really like adding and documenting a lot of options because every option makes the help text more complicated. Maybe some day I'll split the help into "main options" and "expert options" or something like that. Then it wouldn't hurt as much if I add some more options. But right now I prefer to keep the option list as short as possible. Of course you can do the same as "-demuxaudioonly" would do by typing each audio track into the command line. So such an option would just save you a few key presses. Because of that I don't consider it very important at this point in time. I'm more worried about making everything work well right now...

[jchappo]

madshi, any chance of getting a switch -tsmuxer to output PCM with the extra data so I can remux these PCM tracks with tsMuxer?

or, do you know of a program that could create this extra data?

[tebasuna51]

@tebasuna51, do you happen to have a sample for acmod 1 with LFE ("1/0.1")? I don't understand why it shows as "2.0" for you. Looking at my source code it should show as "1.1".
You are right, is my fault, sorry. :confused:
I edited my post (http://forum.doom9.org/showthread.php?p=1134394#post1134394)

[Snowknight26]

[a03] This TS/M2TS file seems to be damaged (sync byte missing).
Aborted at file position 11482513408.
Blu-ray.. need a sample?

[madshi]

madshi, any chance of getting a switch -tsmuxer to output PCM with the extra data so I can remux these PCM tracks with tsMuxer?
I don't know exactly what tsMuxer expects. E.g. does it want to have a different block size for 2.0 tracks compared to 5.1 tracks? I don't feel like experimenting. If there's an official documentation/explanation somewhere which format tsMuxer exactly wants to have then I can add support for that. But actually I think Roman should really add support for big WAV files and/or multi mono WAV files. Because that's a standard format. That "LPCM with headers" format is totally non-standard, as far as I can say...

or, do you know of a program that could create this extra data?
No.

[a03] This TS/M2TS file seems to be damaged (sync byte missing).
Aborted at file position 11482513408.
Blu-ray.. need a sample?
I don't think a sample will help. This looks like a real damage in the source file. A missing sync byte is a very simple check. There's not really a big chance that eac3to has a bug in this situation.

Is this a one part movie? Or does the movie consist of multiple m2ts files? In the latter case check the file size of all m2ts parts. Are they all divisible by 192?

[lexor]

Hi guys, I have a couple of question on the tool use.

1) How do we use the following command
-seekToIFrames make all h264/AVC "I" frames seekable
It says that audio formats are supported only in demuxed form, so how would it know anything about avc i-frames?

2) What does this line (from the help file) do?
eac3to source.evo 1: chapters.txt 2: video.mkv 3: audio.flac 5: subtitle.sup

[madshi]

eac3to v2.46 released

http://madshi.net/eac3to.zip

* MPEG2 muxing now fully supports streams with mixed 23.976 and 29.970 content
* mixed video/movie MPEG2 streams are now always muxed with 29.970 timestamps
* if a movie MPEG2 stream goes 29video, processing is automatically restarted
* MPEG2 pulldown is now automatically removed whenever an MPEG2 stream is read
* new option "-keepPulldown" can be used to disable MPEG2 pulldown removal
* corrected default WAV channel masks for 4.0, 6.1 and 7.1
* added proper channel remaps for libav AC3 decoding of "funny" channel formats
* added general channel mask support
* WAV parser reads channel mask from extensible header
* (E-)AC3 parser sets correct channel mask
There are 2 bigger changes in this release:

(1) MPEG2 handling is noticably improved. Especially MPEG2 streams which contain a mixture of movie content (24p or 48i with pulldown flags) and video content (60i) should be handled perfectly now. Also the pulldown is now automatically removed for movie content which has pulldown flags.

(2) The audio processing chain got full support for custom channel masks. This is necessary to handle funny channel combinations like e.g. "5.0" correctly. However, the channel mask support is not complete yet. Especially the DTS, TrueHD, MLP and MP2 parsers do not always properly set the correct channel masks yet. So if you guys have any DTS/TrueHD/MP2 samples with funny channel masks, please upload samples for me. Thanks!

@tebasuna51, I think I implemented all the changes from your detailed AC3 testing post. But I couldn't really test it because I don't have AC3 samples other than 1.0, 2.0 and 5.1. If you have time, maybe you could check whether my fixed work correctly? Thanks!

[madshi]

1) How do we use the following command
-seekToIFrames make all h264/AVC "I" frames seekable
It says that audio formats are supported only in demuxed form, so how would it know anything about avc i-frames?
I'm confused. First of all AVC is video and not audio. Furthermore neither audio nor video are supported only in demuxed form. EVO/VOB/TS/M2TS containers are fully supported with most video and audio codecs.

P.S: I just noticed that the help text still stated "only in demuxed form" for the audio formats. This is outdated. I just removed it from the first page and from my source code. However, it didn't apply to AVC, cause AVC is video and not audio.

2) What does this line (from the help file) do?
eac3to source.evo 1: chapters.txt 2: video.mkv 3: audio.flac 5: subtitle.sup
It reads the source EVO file and creates all the other files listed in the command line. Of course the track numbers ("1:" etc) vary, depending on the source file. So you first have to do "eac3to sourcefile" to get a proper track listing. Then you can build the correct command line from there by using the correct track numbers.

[lexor]

P.S: I just noticed that the help text still stated "only in demuxed form" for the audio formats. This is outdated. I just removed it from the first page and from my source code. However, it didn't apply to AVC, cause AVC is video and not audio.


Yeah that's the part that got me, since if audio is in the demuxed form it wouldn't know anything about the video. With the update about container support, it makes sense now, thank you.

Also while there are recommended DTS en/decoder, there is only recommended AC3 decoder, what about encoder? Aften as good as it gets?

[Thunderbolt8]

thanks for the improvements again! :)
will test my tricky videos with the new version again, but I wont be home before 1 week, so I guess it will take 1-2 weeks until then.

[tebasuna51]

eac3to v2.46 released

@tebasuna51, I think I implemented all the changes from your detailed AC3 testing post. But I couldn't really test it because I don't have AC3 samples other than 1.0, 2.0 and 5.1. If you have time, maybe you could check whether my fixed work correctly?

Thanks madshi.
All work fine, maskchannels and channel map, with ac3 sources and libav decoder.

Using a 3/3.1 wav still:
eac3to v2.46
command line: "D:\Test\AudioN\eac3to\eac3to.exe" "E:\Test\7_61.wav" "E:\Test\z331.wav" -extensible
------------------------------------------------------------------------------
WAV, 6.1 channels, 0:00:20, 16 bits, 48khz
Doubling 7th channel...
Reading WAV...
Writing WAV...
Creating file "E:\Test\z331.wav"...
eac3to processing took 1 second.
Done.
And the final wav is 8 channels with the default channel mask from 7 channels.

[Snowknight26]

I don't think a sample will help. This looks like a real damage in the source file. A missing sync byte is a very simple check. There's not really a big chance that eac3to has a bug in this situation.

Is this a one part movie? Or does the movie consist of multiple m2ts files? In the latter case check the file size of all m2ts parts. Are they all divisible by 192?

The whole film is one m2ts. Wouldn't ignoring the missing sync byte help? I know xport can demux it though, so I guess it must be ignoring it.

[STaRGaZeR]

Ey madshi, thanks for this program :)

I'm having a weird problem I also posted here: http://forum.doom9.org/showpost.php?p=1136225&postcount=1461

The sample in that post has 4 audio tracks: 1-DTS 2-LPCM 3-AC3 4-AC3

tsMuxer only detects 1, 3 and 4, and eac3to only 3 and 4. Can you help me with this? I just want to demux that LPCM track, it plays fine but I have this problem.

[MichalHabart]

eac3to v2.46 released

http://madshi.net/eac3to.zip

* MPEG2 muxing now fully supports streams with mixed 23.976 and 29.970 content
* mixed video/movie MPEG2 streams are now always muxed with 29.970 timestamps
* if a movie MPEG2 stream goes 29video, processing is automatically restarted
* MPEG2 pulldown is now automatically removed whenever an MPEG2 stream is read
* new option "-keepPulldown" can be used to disable MPEG2 pulldown removal
* corrected default WAV channel masks for 4.0, 6.1 and 7.1
* added proper channel remaps for libav AC3 decoding of "funny" channel formats
* added general channel mask support
* WAV parser reads channel mask from extensible header
* (E-)AC3 parser sets correct channel mask
There are 2 bigger changes in this release:

(1) MPEG2 handling is noticably improved. Especially MPEG2 streams which contain a mixture of movie content (24p or 48i with pulldown flags) and video content (60i) should be handled perfectly now. Also the pulldown is now automatically removed for movie content which has pulldown flags.

(2) The audio processing chain got full support for custom channel masks. This is necessary to handle funny channel combinations like e.g. "5.0" correctly. However, the channel mask support is not complete yet. Especially the DTS, TrueHD, MLP and MP2 parsers do not always properly set the correct channel masks yet. So if you guys have any DTS/TrueHD/MP2 samples with funny channel masks, please upload samples for me. Thanks!

@tebasuna51, I think I implemented all the changes from your detailed AC3 testing post. But I couldn't really test it because I don't have AC3 samples other than 1.0, 2.0 and 5.1. If you have time, maybe you could check whether my fixed work correctly? Thanks!

Thanks Madshi for excellent work. I have just one question. Will extraction of subtitles from bluray be supported in any future release?

[Yraen]

Thanks Madshi for excellent work. I have just one question. Will extraction of subtitles from bluray be supported in any future release?

It already is. I just did one about an hour ago.

[MichalHabart]

It already is. I just did one about an hour ago.

And how did you use eac3to to export subtitles from m2ts file? What command did you use?

[bmnot]

TS, 1 video track, 1:55:47
1: MPEG2, 1080p24 /1.001 (16:9) with pulldown flags
[v01] Extracting video track number 1...
[v01] Removing MPEG2 pulldown...
[v01] Muxing video to Matroska...
[v01] The MPEG2 stream is a mixture of video and movie content.
[v01] This type of MPEG2 stream cannot be muxed with 24p timestamps.
[v01] Will have to abort processing and redo everything.
[v01] Extracting video track number 1...
[v01] Muxing video to Matroska...
[v01] The MPEG2 stream is a mixture of video and movie content.

So it includes the pulldown flags in the 2nd try? Should I hold onto the original ts in case 24p will work on it in future eac3to revisions?