eac3to - audio conversion tool [Archive] - Page 7

deacon crusher

2nd September 2008, 00:56

already know that 25, 28 and 29 are fine, they are the movie and I've run them through EAC3to and then joined the parts and works fine.

So, has to be the first or the last or both, or both versus the others. I know that 30 is one of those super short ones you've been trying to program around.

So, here's the output for first, one in the middle and last.

D:\kid movies\cj7>C:\eac3to\eac3to.exe E:\BDMV\STREAM\00011.m2ts
M2TS, 1 video track, 6 audio tracks, 10 subtitle tracks, 0:00:23
1: h264/AVC, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, Chinese, 5.1 channels, 48khz
3: TrueHD/AC3, English, 5.1 channels, 48khz
4: AC3, Thai, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
5: AC3, Chinese, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
6: TrueHD/AC3, French, 5.1 channels, 48khz
7: AC3, Chinese, 5.1 channels, 640kbps, 48khz
8: Subtitle (PGS), English
9: Subtitle (PGS), English
10: Subtitle (PGS), French
11: Subtitle (PGS), Chinese
12: Subtitle (PGS), Korean
13: Subtitle (PGS), Thai
14: Subtitle (PGS), Arabic
15: Subtitle (PGS), Indonesian
16: Subtitle (PGS), English
17: Subtitle (PGS), French

D:\kid movies\cj7>C:\eac3to\eac3to.exe E:\BDMV\STREAM\00125.m2ts
M2TS, 1 video track, 6 audio tracks, 10 subtitle tracks, 0:06:59
1: h264/AVC, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, Chinese, 5.1 channels, 48khz
3: TrueHD/AC3, English, 5.1 channels, 48khz
4: AC3, Thai, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
5: AC3, Chinese, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
6: AC3, Chinese, 5.1 channels, 640kbps, 48khz
7: TrueHD/AC3, French, 5.1 channels, 48khz
8: Subtitle (PGS), English
9: Subtitle (PGS), English
10: Subtitle (PGS), French
11: Subtitle (PGS), Chinese
12: Subtitle (PGS), Korean
13: Subtitle (PGS), Thai
14: Subtitle (PGS), Arabic
15: Subtitle (PGS), Indonesian
16: Subtitle (PGS), English
17: Subtitle (PGS), French

D:\kid movies\cj7>C:\eac3to\eac3to.exe E:\BDMV\STREAM\00130.m2ts
M2TS, 1 video track, 6 audio tracks, 10 subtitle tracks, 0:00:22
1: h264/AVC, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, Chinese, 5.1 channels, 48khz
3: TrueHD/AC3, English, 5.1 channels, 48khz
4: AC3, Thai, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
5: AC3, Chinese, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
6: AC3, Chinese, 5.1 channels, 640kbps, 48khz
7: TrueHD/AC3, French, 5.1 channels, 48khz
8: Subtitle (PGS), English
9: Subtitle (PGS), English
10: Subtitle (PGS), French
11: Subtitle (PGS), Chinese
12: Subtitle (PGS), Korean
13: Subtitle (PGS), Thai
14: Subtitle (PGS), Arabic
15: Subtitle (PGS), Indonesian
16: Subtitle (PGS), English
17: Subtitle (PGS), French

So, the first one changes the order of the audio streams.

Interesting, but unhelpful for me <G> hopefully helpful for you.

Thanks for all your amazing work.

Set eac3to lose on the separate m2ts parts (listed in "[]", e.g. for playlist 1 it's 00011.m2ts, 00125.m2ts, 00128.m2ts, 00129.m2ts and 00130.m2ts). I'm sure that eac3to has a good reason to complain. Some of these m2ts parts must have a different video or audio track listing compared to the other parts. Joining m2ts parts with different track lists can't work properly. So you need to find out which parts are different. Then we can go from there...

rica

2nd September 2008, 01:32

Hi guys, wiki how to guide:

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

sehgal.v7

2nd September 2008, 05:46

Jan Marijniszoon

2nd September 2008, 06:05

* extension ".thd+ac3" is supported now to define destination format
* TrueHD tracks without AC3 core can be converted to TrueHD/AC3 now
* demuxing a single-part Blu-Ray title keeps the original "TrueHD/AC3" data
* demuxing a multi-part Blu-Ray title automatically redoes the AC3 substream
* added workaround for Blu-Ray playlists with multiple last "invalid" parts
* fixed: "-check" didn't work for LPCM tracks

I must be dreaming man! Thank you very much for these super fine new abilities!

madshi

2nd September 2008, 07:03

So, the first one changes the order of the audio streams.

Interesting, but unhelpful for me <G> hopefully helpful for you.
Could you please upload 00011.m2ts and 00130.m2ts for me? Those two seem to be really small. Thanks!

sehgal.v7

2nd September 2008, 09:34

Jus saw today that this thread is the largest thread in Doom9 Forum, containing 6,021 replies!!
Cheers.

G_M_C

2nd September 2008, 11:10

Hi guys, wiki how to guide:

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

-core, "extract the DTS core of a DTS-HD track"

I remember using this switch on a TrueHD track (HDScape HDWindow), to extract the core AC3; That worked fine afaik.

rica

2nd September 2008, 12:31

-core, "extract the DTS core of a DTS-HD track"

I remember using this switch on a TrueHD track (HDScape HDWindow), to extract the core AC3; That worked fine afaik.

I got the core dts from a dts-hd source. (-core) option is working for dts as well..

G_M_C

2nd September 2008, 14:04

I got the core dts from a dts-hd source. (-core) option is working for dts as well..

;)

I mentioned this so that this info can be added to the wiki (where only the DTS-extraction is mentioned).

yonta

2nd September 2008, 14:11

I mentioned this so that this info can be added to the wiki (where only the DTS-extraction is mentioned).

-core option is only for DTS core extraction.
AC3 track from a Blu-ray TrueHD/AC3 stream can be extracted by a simple command like below.

eac3to xxxxx.m2ts 3: target.ac3

G_M_C

2nd September 2008, 14:26

-core option is only for DTS core extraction.
AC3 track from a Blu-ray TrueHD/AC3 stream can be extracted by a simple command like below.

eac3to xxxxx.m2ts 3: target.ac3

Ah, i see. That explains why my use of -core worked, because I used target.ac3 offcourse ;)

rack04

2nd September 2008, 14:28

How does eac3to retreive the version of mkvtoolnix that is installed? I have updated to the latest beta and it still complains that a new version exists. Thanks.

yonta

2nd September 2008, 14:41

How does eac3to retreive the version of mkvtoolnix that is installed? I have updated to the latest beta and it still complains that a new version exists. Thanks.

This was answered a few pages back.

I also have the latest beta(Aug 21).

f:\Movie.Utilz\MKVtoolnix>mkvmerge.exe --version
mkvmerge v2.2.0 ('Turn It On Again') built on Aug 19 2008 15:59:25

rica

2nd September 2008, 15:35

Hi madshi, how would you edit wiki, i really wonder?

Especially after the latest versions?

nautilus7

2nd September 2008, 20:19

Hi madshi, how would you edit wiki, i really wonder?

Especially after the latest versions?The wiki is basically developed by me, not madshi. Everyone is welcome to contribute, of course. That's why it is in the wiki in the first place. I would appreciate if you contact me (by pm) if you are willing to contribute. :)

rica

2nd September 2008, 20:25

The wiki is basically developed by me, but of course everyone is welcome to contribute. That's why it is in the wiki in the first place. I would appreciate if you contact me (by pm) if you are willing to contribute. :)

Cograts, it is really a very good job :thanks:

It would be great plasure to me to contribute but i have to do my homework first; i'll contact you.

Thanks again :)

nautilus7

2nd September 2008, 20:31

Cograts, it is really a very good job :thanks:

It would be great pleasure to me to contribute but i have to do my homework first; i'll contact you.

Thanks again :)
Thank you. I 've been busy also, that's why it's not so up-to date. :rolleyes:

odin24

3rd September 2008, 02:35

rica

3rd September 2008, 03:00

Are there any benefits to demuxing/remuxing with eac3to vs. tsMuxeR? I've been using tsMuxeR lately and demuxing through the MPLS file, this way I am ensured all multiple files (if any) are appended properly. I also have no interest muxing to mkv, only m2ts or Blu-ray.

I have also found eac3to an more GUI pretty useful... which uses tsMuxeR.

Thanks, O.

The answer will be:
TSMuxer needs eac3to's help always but eac3to doesn't need any help of TSMuxer. :)
More job in one step...

To me, eac3to is a video demuxer, audio demuxer-converter while TSMuxer is just a muxer anymore... (it is my opinion after the latest updates of eac3to)

_ _ _ _ _

odin24

3rd September 2008, 04:55

The answer will be:
TSMuxer needs eac3to's help always but eac3to doesn't need any help of TSMuxer. :)
More job in one step...

To me, eac3to is a video demuxer, audio demuxer-converter while TSMuxer is just a muxer anymore... (it is my opinion after the latest updates of eac3to)

_ _ _ _ _

Great, a few more questions.
1. How does tsMuxeR need eac3to's help?
2. does decoding take place during demuxing or just converting?
3. Is quality comprimised when using tsMuxeR over eac3to, or is it just the same since tsMuxeR uses the help of eac3to?

Thanks again, O.

nautilus7

3rd September 2008, 07:58

Main advantage of using eac3to over tsmuxer is with seamless branching Blu-rays. Those movies are split among multiple .m2ts files from which audio can overlap a little bit. This means that just joining them together (what tsmuxer does) will result in audio-video sync problems. On the contrary, eac3to is the only program available that can remove overlapping parts of the audio streams. So, I suggest treating your seamless Blu-rays with eac3to as first step.

Regarding quality between tsmuxer and eac3to (in removing pulldown, for example - not sure if tsmuxer can do that), don't know where tsmuxer stands, but I can tell for sure that eac3to is perfect. So, to put it simple... using eac3to you don't loose anything.

odin24

3rd September 2008, 08:27

odin24

3rd September 2008, 08:33

Hey nautilus7, have a look.

http://en.wikipedia.org/wiki/User_talk:Nautilus7

Nothing comes up on a search either.

madshi

3rd September 2008, 08:35

Demuxing with eac3to vs. tsMuxeR:

(1) As nautilus7 said, eac3to fixes audio overlaps in seamless branching titles. tsMuxeR does not (AFAIK).

(2) eac3to automatically detects and fixes audio delays (when the audio stream begins earlier/later than the video stream). I don't know if tsMuxeR does that.

(3) eac3to automatically removes MPEG2, VC-1 and h264 pulldown from HD DVD video streams (29.97 -> 23.976). I'm not sure whether tsMuxeR does that, too. I've seen people reporting that they get stuttering problems with VC-1 movies when using tsMuxeR, only. Those people used vc1conv to remove the VC-1 pulldown and that fixed the problem for them. Now eac3to does the same thing as vc1conv for VC-1 streams. Also eac3to does the same thing as h264info for h264 streams.

(4) eac3to can transcode/modify HD DVD audio streams (e.g. E-AC3, TrueHD) to make them Blu-Ray/m2ts compatible. tsMuxeR can't do that. At least that's where tsMuxeR needs the help of eac3to.

nautilus7

3rd September 2008, 08:41

Hey nautilus7, have a look.

http://en.wikipedia.org/wiki/User_talk:Nautilus7

Nothing comes up on a search either.http://en.wikibooks.org/wiki/eac3to (it's not finished... yet)

It was originally located in Wikipedia, but I received a warning/notification that the page didn't fit in Wikipedia. That's why I migrated it to Wikibooks.

odin24

3rd September 2008, 08:46

Yeah, that was a warning/notification that the eac3to page didn't fit in Wikipedia. That's why I migrated to Wikibooks (a long time ago).

Oops, kinda new here. Thanks anyway.

Demuxing with eac3to vs. tsMuxeR:

(1) As nautilus7 said, eac3to fixes audio overlaps in seamless branching titles. tsMuxeR does not (AFAIK).

(2) eac3to automatically detects and fixes audio delays (when the audio stream begins earlier/later than the video stream). I don't know if tsMuxeR does that.

(3) eac3to automatically removes MPEG2, VC-1 and h264 pulldown from HD DVD video streams (29.97 -> 23.976). I'm not sure whether tsMuxeR does that, too. I've seen people reporting that they get stuttering problems with VC-1 movies when using tsMuxeR, only. Those people used vc1conv to remove the VC-1 pulldown and that fixed the problem for them. Now eac3to does the same thing as vc1conv for VC-1 streams. Also eac3to does the same thing as h264info for h264 streams.

(4) eac3to can transcode/modify HD DVD audio streams (e.g. E-AC3, TrueHD) to make them Blu-Ray/m2ts compatible. tsMuxeR can't do that. At least that's where tsMuxeR needs the help of eac3to.

Great information, I'm liking this thread more and more. Time to start reading the novella that is this thread.

baudi

3rd September 2008, 16:34

Demuxing with eac3to vs. tsMuxeR:

...

(4) eac3to can transcode/modify HD DVD audio streams (e.g. E-AC3, TrueHD) to make them Blu-Ray/m2ts compatible. tsMuxeR can't do that. At least that's where tsMuxeR needs the help of eac3to.

How can I make a compatible audio stream from an E-AC3 track to a DD+ Blu-Ray compatible? (i.e. command line)

Thank you very much Madshi.

madshi

3rd September 2008, 17:55

How can I make a compatible audio stream from an E-AC3 track to a DD+ Blu-Ray compatible? (i.e. command line)
eac3to source.eac3 dest.ac3

rica

3rd September 2008, 18:18

rebkell

3rd September 2008, 18:27

I have a stupid question, how does eac3to fix gaps? Overlaps I sorta can see, but wonder what happens to/in the overlapped portions but I'm mostly wondering how exactly it works on gaps?

nautilus7

3rd September 2008, 18:29

I have a stupid question, how does eac3to fix gaps? Overlaps I sorta can see, but wonder what happens to/in the overlapped portions but I'm mostly wondering how exactly it works on gaps?It repeats last part of the audio or fills with silence depending on how long the gap is. Not sure though (madshi correct me if i'm wrong). Good thing is that there's no movie with gaps till now..

baudi

3rd September 2008, 18:30

eac3to source.eac3 dest.ac3

Making so I only got a standar ac3 file, not a DD+ Blu-Ray compliant.

Thank you again.

nautilus7

3rd September 2008, 18:36

Making so I only got a standar ac3 file, not a DD+ Blu-Ray compliant.

Thank you again.Blu-ray allows E-AC3 only for tracks with more than 5.1 channels (7.1). And these tracks are 5.1 AC3 core with E-AC3 extensions for the additional channels.

tebasuna51

3rd September 2008, 19:05

I found it. It's from Edward Scissorhands Blu-ray Disc. The movie menu states it is a 4.0 and eac3to a 3/1. That seems correct, except that the surround channel is empty (I checked a big part of the track to be sure about this).

Here (http://www.sendspace.com/file/e073sw) is a sample.

@nautilus7

This is a old post but now I understand your comment about the surround channel. Now I have ArcSoft DTS decoder and don't work fine with these atypical channel configuration.

My question is:
to decode this dts ma 3/1 sample do you use Sonic or ArcSoft decoder?

With -libav the surround channel is present, try:
eac3to yoursample31.dtshd output.wav -libav -1,2,0,3,4,5

With ArcSoft:
eac3to yoursample31.dtshd wrong_output.wav

the surround channel is empty, and is empty also with dts core:

eac3to yoursample31.dtshd core31.dts -core
eac3to core31.dts wrong_output.wav

rica

3rd September 2008, 20:50

The file is not available anymore.

tebasuna51

3rd September 2008, 20:51

nautilus7

3rd September 2008, 20:54

Well, tebashuna51 you're right. There's something really strange with this track.

My findings:

* ArcSoft and Sonic decode the DTS-HD MA track bit perfect
* ArcSoft decode the DTS core "similar" to the above
* Sonic decode the DTS core with lower volume
* In the above cases the surround channel is empty
* With Libav the surround channel is not empty
* With Libav channels needs remapping
* With Libav L,R,C channels are similar to ArcSoft core decoding

http://img214.imageshack.us/img214/4475/40dtshdgo2.th.png (http://img214.imageshack.us/my.php?image=40dtshdgo2.png)

I re-uploaded a sample (http://www.sendspace.com/file/wgdtbn) of this track.

stottle

4th September 2008, 01:08

rica

4th September 2008, 01:25

The only part of converting an HD-DVD to Bluray that I can't figure out is how to convert the subtitle data (.sup) that eac3to produces to something that tsmuxer will take. Can anyone help me on this one?

Thanks

Even it is out of the thread:

http://forum.doom9.org/showthread.php?p=1177618#post1177618

_ _ _ _ __

stottle

4th September 2008, 01:44

Even it is out of the thread:

http://forum.doom9.org/showthread.php?p=1177618#post1177618
_ _ _ _ __
I saw that, but I thought .srt was an OCR'd version of the .sup bitmaps. If so, then converting to .srt and then having tsmuxer convert to pgs means converting to text via ocr and then converting the text back to bitmap for pgs, in some other font. I was hoping there was a way to convert the .sup bitmaps to pgs directly, so that the intended font wouldn't change. This also mitigates the need to double check the OCR results for errors.

BTW, I tried the export function in supread (png format), which tsmuxer accepted, but subtitles didn't show up in the output (TMT said they were there however).

rica

4th September 2008, 01:49

I think you didn't read well what i said.
You will never read the subtitles without MPC-HC for the time being...(Zoom player?: just forget it.)
Edit: sorry; except image + daemontools method....or burning to disc method...

moshmothma

4th September 2008, 02:27

Madshi, thanks for the recent update. I was able to complete operations on the NIN BD. However, the video pixelates heavily. I am not sure what is going on. Do you want another sample? Thanks

Yes.

I've checked these. The first two look like this:

1: VC-1, 1080i60 /1.001 (16:9)
2: TrueHD/AC3, 5.1 channels, 48khz
3: AC3, 2.0 channels, 256kbps, 48khz
4: Subtitle (PGS)

The last small two look like this:

1: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
2: AC3, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
3: AC3, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
4: Subtitle (PGS)
As you can see, the last two m2ts parts have totally different video and audio track properties compared to the rest of the movie. Now a media player could easily notice the change and adjust its output accordingly. But eac3to is not a media player but just exports the raw video and audio streams. And it just doesn't make any sense to join e.g. a "TrueHD+AC3" track together with a simple "AC3" track. The end result wouldn't play correctly. That's why eac3to cannot handle this situation correctly.

I'll "fix" it in the next build by ignoring the last few m2ts parts, if they are very small and have different audio or video track parameters.

stottle

4th September 2008, 02:30

@rica,

Sorry, I'm not sure I follow. I am trying to convert HD-DVD to Bluray, iso if need be. TMT can play both HD-DVD and Bluray from hdd or from .iso files. I'm not sure that TMT will always continue to play HD-DVD however, so I want to convert my HD-DVD rips to Bluray, taking advantage of eac3to's new truehd conversion functionality.

I thought that if you demux'd a bluray and hddvd with eac, the subtitles both end up with a .sup extension. I'm pretty sure I tried this, but it was pretty late. Won't tsmuxer accept a bluray .sup, but not a hddvd .sup? If so, I am hoping there is a way to change the from an hddvd formatted .sup to a bluray one.

Does that make more sense?

Thanks for the help!

tebasuna51

4th September 2008, 02:31

There's something really strange with this track.
...
* In the above cases the surround channel is empty
...

The track is ok, the problem is the decoder, seems ArcSoft work with:
2.0, 3.0, 3.1, 3/2 and 3/2.1

But don't work with:
2/1, 2/1.1, 2/2, 2/2.1, 3/1 and 3/1.1

Please, can you test Sonic with my samples_dts (http://www.sendspace.com/file/0d5dbt)?

Thanks.

saint-francis

4th September 2008, 05:21

Does that make more sense?

Thanks for the help!

If you want your HD DVD .sup's to be BD .sup's just convert them to BD with supread. Load it up and tell it to output a BD file. There is plenty about this in the subtitle forum in threads with indicative names like "converting Blueray disk subtitles (http://forum.doom9.org/showthread.php?t=124105)" and "HD subtitle ripping (http://forum.doom9.org/showthread.php?t=121535)".
At any rate this discussion should continue down there.

xkodi

4th September 2008, 05:41

The track is ok, the problem is the decoder, seems ArcSoft work with:
2.0, 3.0, 3.1, 3/2 and 3/2.1

But don't work with:
2/1, 2/1.1, 2/2, 2/2.1, 3/1 and 3/1.1

back in the early Arcsoft days i came across the same problem too, i.e. the problem with DTS-HD MA files with strange channel order:

- the sample (from Q-tec Demo Disc):

http://forum.doom9.org/showthread.php?p=1153102#post1153102

- the problem discussion with madshi:

http://forum.doom9.org/showthread.php?p=1153303#post1153303

http://forum.doom9.org/showthread.php?p=1153571#post1153571

http://forum.doom9.org/showthread.php?p=1153688#post1153688

my idea then was to patch the DTS-HD MA header and change the channel order, so Arcsoft can decode the track correctly and then rearrange the right order in the WAV:

http://forum.doom9.org/showthread.php?p=1153692#post1153692

but then we both with madshi agreed that it wasn't worth the effort for a single sample from a demo disc:

http://forum.doom9.org/showthread.php?p=1153700#post1153700

http://forum.doom9.org/showthread.php?p=1153711#post1153711

now, we have sample from a real movie disc ("Edward Scissorhands" BD), not a demo disc. so, maybe we should look again at the problem.

it should be very easy to patch the channel order byte in the header, but then there is CRC involved and i personally know nothing about how DTS(-HD MA) CRC is calculated and where is stored in the DTS(-HD MA) header.

deacon crusher

4th September 2008, 06:33

Here you go madshi

Those 2 are uploading to mediafire right now.

When done (around midnight pacific time Wednesday night) they'll be here.

Done, now and here are the correct links

http://www.mediafire.com/file/wvj4shszfkv/00011.m2ts

http://www.mediafire.com/file/wxufltmfgfa/00130.m2ts

thanks

Could you please upload 00011.m2ts and 00130.m2ts for me? Those two seem to be really small. Thanks!

Duppie

4th September 2008, 08:04

Madshi, excellent application. Many Thanks.

I am using your tool to convert TrueHD to PCM and then mux audio and video together using tsMuxeR from ripped Blu-Rays. This works perfectly. I have read so much in this thread and just to make certain that the quality and command is OK. Am i doing it correct, if for instance the truehd audio track is 2 in the 0001.m2ts file:

Convertion from TrueHD to PCM with no loss: C:\eac3to 0001.m2ts 2: out.pcm

Do i need to use the -blu-ray switch? What is it exactly for? I have search this thread and could not find answers to these questions. Is this switch removed for a reason when only executing eac3to to obtain some help?

Thanks again.

nautilus7

4th September 2008, 09:14

I tested tebasuna51's samples with Sonic. The updated list is below:

With the help of Elektra999 I have dts samples with atypical channel map.
The samples was encoded with DTS PRO SERIES SURROUND ENCODER.
Only 3/2.0 and 3/2.1 was encoded with Surcode.

AMODE Ch Arrangement+lfe Mask and MS channels order Detect-MaskCh eac3to libav remap ArcSoft problems Sonic Problems
-------- - --------------- -------------------------- -------------------- ----------- ---------------- --------------
0b000010 2 L R 0x0003 FL FR 2.0 0x000B (1) not needed ok ok
0b000110 3 L R S 0x0103 FL FR BC 2/1 0x0103 ok not needed BC ch empty FL & FR present in BC, BC only present in FL & FR
4 L R S LFE 0x010B FL FR LF BC 2/1.1 0x010B (2) -0,1,3,2,4,5 BC ch empty (3) FL & FR present in BC, BC only present in FL & FR, LFE empty, maybe remap needed
0b001000 4 L R Ls Rs 0x0033 FL FR BL BR 2/2 0x0033 ok not needed BL+BR ch's empty ok
5 L R Ls Rs LFE 0x003B FL FR LF BL BR 2/2.1 0x003B (2) -0,1,4,2,3,5 LF ch empty ok
0b000101 3 C L R 0x0007 FL FR FC 3/0 0x000F (1)(2) -1,2,0,3,4,5 ok ok
4 C L R LFE 0x000F FL FR FC LF 3.1 0x000F (2) -1,2,0,3,4,5 ok ok
0b000111 4 C L R S 0x0107 FL FR FC BC 3/1 0x0107 (2) -1,2,0,3,4,5 BC ch empty BC empty, BC only presesnt in FL & FR
5 C L R S LFE 0x010F FL FR FC LF BC 3/1.1 0x010F (2) -1,2,0,4,3,5 LF same than BC eac3to crashes
0b001001 5 C L R Ls Rs 0x0037 FL FR FC BL BR 5.0 0x003F (1)(2) -1,2,0,3,4,5 ok ok
6 C L R Ls Rs LFE 0x003F FL FR FC LF BL BR 5.1 0x003F ok already done ok ok

(1) Wrong mask, the correct mask can be easyly put based in detection
(2) When decoded with libav need remapping channels.
(3) Same remap than libav needed.

btw, tebasuna51, is that you on the samples? :D

nautilus7

4th September 2008, 09:27

Duppie

4th September 2008, 09:41

Thanks, nautilus7

nautilus7

4th September 2008, 10:00

Thanks, nautilus7
Welcome. One thing I can suggest also, is to add -down16 to the command line if you want to save some space and you don't mind (or you can't hear) the quality loss of the 24 bit to 16 bit conversion. That's of course in case the source TrueHD track is 24 bit and not already 16 bit.

The only part of converting an HD-DVD to Bluray that I can't figure out is how to convert the subtitle data (.sup) that eac3to produces to something that tsmuxer will take. Can anyone help me on this one?

ThanksAs already saint-francis pointed out, you need SUPread to convert HD DVD .sup to Blu-ray .sup, without any OCR. That's because subtitles in these two media are little different.

nautilus7

4th September 2008, 10:17

Thanks, nautilus7Things are coming slowly to my mind...

Since version 2.59 eac3to can also convert HD DVD TrueHD tracks to Blu-ray style TrueHD tracks with AC3 track interleaved. So there's no need to convert to pcm anymore. Except you don't have a TrueHD decoder.

Duppie

4th September 2008, 11:36

nautilus7

In my case I also have to convert TrueHD to PCM for tsMuxeR to create a correct Blu-Ray structure split into 4gig chunks to be played in AVCHD format on HDD on PS3. As you know tsMuxeR does not do proper TrueHD muxing in blu-ray structure and PCM is a good workaround.

Thanks.

G_M_C

4th September 2008, 12:01

Duppie

4th September 2008, 12:15

G_M_C

4th September 2008, 12:22

G_M_C

Thanks for that. This is actually me who posted that and made those tests. Currently there is no solution to mux TrueHD in Blu-Ray structure split in 4 gig chunks due to tsMuxeR problems. TsRemux does work with TrueHD but cannot split the file in 4 gig chunks which is necessary for fat32 disks on PS3 in AVCHD format.

:p Didnt realize it was you that wrote that :p

-----

Can you not burn to a DVD-DL in UDF 2.5 ?

Or split before muxing i.e. "split manually" ?

Duppie

4th September 2008, 12:36

EPiPH0NE

4th September 2008, 18:34

G_M_C

Thanks for that. This is actually me who posted that and made those tests. Currently there is no solution to mux TrueHD in Blu-Ray structure split in 4 gig chunks due to tsMuxeR problems. TsRemux does work with TrueHD but cannot split the file in 4 gig chunks which is necessary for fat32 disks on PS3 in AVCHD format.

Wow file playback on the PS3 seems like too much of a hassle. I also do TrueHD/DTS-MA -> 5.1 PCM for my NMT but the only time I have to split the files up is when I'm archiving them to DVD-R :) Lucky me. I guess if you already have a PS3 a BluRay burner is the way to go. A lot less work.

odin24

4th September 2008, 22:27

G_M_C

Yes you can burn to a DVD-DL in UDF 2.5 and it works. But the idea is to take one big (20-30gig) m2ts file open it with a muxer select the required tracks and generate a Blu-Ray structure with one playlist on fat32 (4gig chunks) convert the Blu-Ray structure to AVCHD and play the complete film from HDD on PS3 without writing a couple of DVD's. This requires alot of disk space but it is better than having alot of DVD's and changing the DVD's a couple of times during playpack of one movie.

I use this method as well, it's perfect for rentals that are due back the next day, and you have no time to watch immediately. Also for BDs you have no intention of archiving to DVD9.

The only drawback is you need your own BD burner.

Duppie, do you use AVCHD-ME, or is there something else out there that can produce the same media type... possibly that doesn't cause the slight pause between the 4GB chunks?

nautilus7

4th September 2008, 22:39

Guys, I don't want to be a geek, but your discussion is out of topic. :thanks:

Jaja1

5th September 2008, 20:27

Remuxing Welcome to Dongmakgol Blu-ray to mkv results in async audio.

eac3to v2.60
command line: eac3to 1) 2:i:\Wel.mkv 5:i:\Wel.dts -core
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 5 subtitle tracks, 2:12:52
1: Chapters, 37 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, Korean, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
4: AC3, Korean, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
5: DTS Master Audio, Korean, 5.1 channels, 16 bits, 48khz, dialnorm: -4dB
6: AC3, Korean, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
7: AC3, Korean, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
8: Subtitle (PGS), Korean
9: Subtitle (PGS), English
10: Subtitle (PGS), Japanese
11: Subtitle (PGS), Korean
12: Subtitle (PGS), Korean
[a05] Extracting audio track number 5...
[v02] Extracting video track number 2...
[a05] Extracting DTS core...
[a05] Removing DTS dialog normalization...
[a05] Patching bitdepth to 24 bits...
[v02] Muxing video to Matroska...
[a05] Creating file "i:\Wel.dts"...
Added fps value to MKV header.
Video track 2 contains 191141 frames.
eac3to processing took 12 minutes, 32 seconds.
Done.
The resulting DTS track is not in sync with the video. The original stream is in sync.

Remarkably, the WMV decoder is not capable of proper playback of the m2ts, artefacts all over. Ffdshow however, has no problems with the video.

A sample can be found at:
http://www.sendspace.com/file/71uipv

rica

5th September 2008, 21:00

Remuxing Welcome to Dongmakgol Blu-ray to mkv results in async audio.

The resulting DTS track is not in sync with the video. The original stream is in sync.

Remarkably, the WMV decoder is not capable of proper playback of the m2ts, artefacts all over. Ffdshow however, has no problems with the video.

A sample can be found at:
http://www.sendspace.com/file/71uipv

This is a VC1mkv which has always issues with TSMuxer.
I think you remuxed that sample with TSMuxer, did you?

_ _ _ _ _

Snowknight26

5th September 2008, 21:06

Ehhem.
[v02] Muxing video to Matroska...

asarian

5th September 2008, 21:13

Hmm, getting the weirdest error (see below for full output): "The channel modder was started with incorrect parameters." What is up with that? And how do I fix it? :)

E:\migrate\devils_rejects\BDMV>eac3to 1) 3: c:\video\devils.ac3
M2TS, 1 video track, 4 audio tracks, 2 subtitle tracks, 1:50:54
1: Chapters, 15 chapters
2: MPEG2, 1080p24 /1.001 (16:9)
3: DTS Hi-Res, English, 6.1 channels, 24 bits, 3093kbps, 48khz, dialnorm: -4dB
4: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
7: Subtitle (PGS), English
8: Subtitle (PGS), Spanish
[a03] The ArcSoft decoder doesn't seem to work, will use Sonic instead.
[a03] AC3 encoding only supports 6 channels. Will mix the surround channels.
[a03] Extracting audio track number 3...
[a03] Removing DTS dialog normalization...
[a03] Decoding with DirectShow (Sonic Audio Decoder)...
[a03] DirectShow reports 5.1 channels, 24 bits, 48khz
[a03] Mixing surround channels...
[a03] The channel modder was started with incorrect parameters.
Aborted at file position 21967306752.

E:\migrate\devils_rejects\BDMV>

rica

5th September 2008, 21:31

asarian

5th September 2008, 22:27

eac3to "E:\migrate\devils_rejects\BDMV\*.m2ts" 3: "c:\video\devils.ac3"

Check that.

EDIT: Sorry i've misssed the source is DTS-HD and libav doesn't work in this case-corrected. Just install TMT trial and give it a go and compare.

Thanks. That worked a lot better. :) No errors, at least.

eac3to v2.60
command line: eac3to 1) 3: c:\video\devils.ac3
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 2 subtitle tracks, 1:50:54
1: Chapters, 15 chapters
2: MPEG2, 1080p24 /1.001 (16:9)
3: DTS Hi-Res, English, 6.1 channels, 24 bits, 3093kbps, 48khz, dialnorm: -4dB
4: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
7: Subtitle (PGS), English
8: Subtitle (PGS), Spanish
[a03] AC3 encoding only supports 6 channels. Will mix the surround channels.
[a03] Extracting audio track number 3...
[a03] Mixing surround channels...
Loading white noise (needed for dithering)...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Removing DTS dialog normalization...
[a03] Encoding AC3 <640kbps> with libAften...
[a03] Creating file "c:\video\devils.ac3"...
Video track 2 contains 159528 frames.
eac3to processing took 21 minutes, 28 seconds.
Done.

rica

5th September 2008, 23:01

:thanks: for feedback.

K1ngp1ng

6th September 2008, 09:47

and what exactly is the point of this? "4:" already is an ac3 track?

asarian

6th September 2008, 09:53

and what exactly is the point of this? "4:" already is an ac3 track?

The point is to use the DTS stream to get another AC3 stream, sans the dialogue normalization.

madshi

6th September 2008, 10:00

With the help of Elektra999 I have dts samples (http://www.sendspace.com/file/0d5dbt)with atypical channel map.
The samples was encoded with DTS PRO SERIES SURROUND ENCODER.
Only 3/2.0 and 3/2.1 was encoded with Surcode.

AMODE Ch Arrangement+lfe Mask and MS channels order Detect-MaskCh eac3to libav remap ArcSoft problems
-------- - --------------- -------------------------- -------------------- ----------- ----------------
0b000010 2 L R 0x0003 FL FR 2.0 0x000B (1) not needed ok
0b000110 3 L R S 0x0103 FL FR BC 2/1 0x0103 ok not needed BC ch empty
4 L R S LFE 0x010B FL FR LF BC 2/1.1 0x010B (2) -0,1,3,2,4,5 BC ch empty (3)
0b001000 4 L R Ls Rs 0x0033 FL FR BL BR 2/2 0x0033 ok not needed BL+BR ch's empty
5 L R Ls Rs LFE 0x003B FL FR LF BL BR 2/2.1 0x003B (2) -0,1,4,2,3,5 LF ch empty
0b000101 3 C L R 0x0007 FL FR FC 3/0 0x000F (1)(2) -1,2,0,3,4,5 ok
4 C L R LFE 0x000F FL FR FC LF 3.1 0x000F (2) -1,2,0,3,4,5 ok
0b000111 4 C L R S 0x0107 FL FR FC BC 3/1 0x0107 (2) -1,2,0,3,4,5 BC ch empty
5 C L R S LFE 0x010F FL FR FC LF BC 3/1.1 0x010F (2) -1,2,0,4,3,5 LF same than BC
0b001001 5 C L R Ls Rs 0x0037 FL FR FC BL BR 5.0 0x003F (1)(2) -1,2,0,3,4,5 ok
6 C L R Ls Rs LFE 0x003F FL FR FC LF BL BR 5.1 0x003F ok already done ok

(1) Wrong mask, the correct mask can be easyly put based in detection
(2) When decoded with libav need remapping channels.
(3) Same remap than libav needed.
Thank you!!

The next eac3to version will fix the incorrect channel masks and implement channel remapping for libav and ArcSoft. Here's how the different DTS decoders work with these samples:

- libav works fine
- ArcSoft works fine, but splits mono surround channel (x1x.dts) into Ls/Rs pair with half volume each
- Sonic heavily dislikes mono surround channel (crash or funny results), but works fine for the other tracks
- Nero works fine

Not sure yet how I should handle ArcSoft's mono surround channel splitting. Should eac3to keep ArcSoft's Ls/Rs pair? Or should I remove one of those channels and double the volume of the other to reconstruct the original mono surround channel?

Madshi, thanks for the recent update. I was able to complete operations on the NIN BD. However, the video pixelates heavily. I am not sure what is going on. Do you want another sample? Thanks
IIRC I had no pixelation. Decoding worked just fine for me with the MS VC-1 decoder. Which decoder are you using? NIN is interlaced and the ffmpeg/libav decoder (and also the MPC HC built in decoder) don't handle interlaced decoding correctly yet.

Those 2 are uploading to mediafire right now.
Thanks! This very much looks like an authoring fault. One and the same audio track suddenly changes from TrueHD to AC3 and another audio track suddenly changes from AC3 to TrueHD. I'm not sure how to handle that...

Remuxing Welcome to Dongmakgol Blu-ray to mkv results in async audio.

The resulting DTS track is not in sync with the video. The original stream is in sync.

Remarkably, the WMV decoder is not capable of proper playback of the m2ts, artefacts all over. Ffdshow however, has no problems with the video.

A sample can be found at:
http://www.sendspace.com/file/71uipv
I don't see any artifacting here with the sample you uploaded. WMV decoder? What's that? You may want to install the latest Windows Media Player to get the latest MS VC-1 decoder. This in combination with the latest Haali version results in perfect video playback on my PC.

About the DTS track: The sample is rather short and doesn't really help much with checking the correct audio delay. I don't see an obvious "desync", but it's hard to say from that sample. How did you test audio sync? Have you transcoded the DTS track to something else? Have you muxed it into some container? The audio sync mismatch could be introduced by the filters you're using for playback or by the way you muxed things. Of course it could also be introduced by eac3to. Can't say for certain...

Hmm, getting the weirdest error (see below for full output): "The channel modder was started with incorrect parameters." What is up with that? And how do I fix it? :)
Can you please run "eac3to 1) 3: c:\video\devils.ac3 -logdts"? It seems that the DTS track is decoded by Sonic only as 5.1 although it's a 6.1 track (which results in that strange eac3to error you've seen). I'd like to know why it's only decoded as 5.1. My guess is that it's one of those rare DTS tracks with a slightly unusual format. The "-logdts" parameter should tell me that. Thanks!

madshi

6th September 2008, 10:01

The point is to use the DTS stream to get another AC3 stream, sans the dialogue normalization.
You can also let eac3to just demux the original AC3 stream. Dialog normalization will then automatically be removed. The original AC3 stream will very likely have a better audio quality compared to transcoding the DTS-Hi-Res track to AC3.

nautilus7

6th September 2008, 10:25

- ArcSoft works fine, but splits mono surround channel (x1x.dts) into Ls/Rs pair with half volume each

Not sure yet how I should handle ArcSoft's mono surround channel splitting. Should eac3to keep ArcSoft's Ls/Rs pair? Or should I remove one of those channels and double the volume of the other to reconstruct the original mono surround channel?
I believe reconstructing the original DTS format (mono surround) will be better.

xkodi

6th September 2008, 10:42

should I remove one of those channels and double the volume of the other to reconstruct the original mono surround channel?

sorry, but i will ask again about DTS header patching - these tracks are plain DTS, AFAIK channel order bits and how CRC is calculated are well known for DTS header.

so, why not patch for example "L R S" as "C L R" in the DTS header, then it should decode correctly and it shouldn't need reconstruction at all or am i wrong about such possibility?

reconstruction is probably easier to do than header patching, but less precise.

Jaja1

6th September 2008, 11:08

I don't see any artifacting here with the sample you uploaded.I figured out what is wrong. Selecting Gabest's DTS source filter in ZP caused artefacting in combination with the WMV decoder. Deselecting the filter solved this. And now audio is in sync as well :confused:

WMV decoder? What's that?Well, MS still call there VC-1 decoder WMV decoder. To be precise WMVds32.ax and WMV8ds32.ax. And it shows up as WMVideo decoder in ZP and Windows Media Video Decoder in Radlight Filter Manager. Can't help that.

madshi

6th September 2008, 11:55

sorry, but i will ask again about DTS header patching - these tracks are plain DTS, AFAIK channel order bits and how CRC is calculated are well known for DTS header.

so, why not patch for example "L R S" as "C L R" in the DTS header, then it should decode correctly and it shouldn't need reconstruction at all or am i wrong about such possibility?
You're right. I've implemented that now and it fixes both ArcSoft and Sonic. So now all DTS decoders decode all of tebasuna's samples correctly.

nautilus7

6th September 2008, 12:11

Is the patching only for plain DTS or for DTS-HD also?

xkodi

6th September 2008, 14:45

You're right. I've implemented that now and it fixes both ArcSoft and Sonic. So now all DTS decoders decode all of tebasuna's samples correctly.

great! thank you, madshi!

can you try the same patching approach with the DTS-HD MA header of the sample from Q-tec Hi-Definition Reference Demo Disc:

http://forum.doom9.org/showthread.php?p=1153102#post1153102

and change the order from "C L R Ls Rs LFE Lsr Rsr", which doesn't decode correctly to the "C L R LFE Lsr Rsr Lss Rss" which is known to be good order.

if it doesn't work it's OK, i just want to try that in order to know is there any different in patching header channel order between plain DTS and DTS-HD MA.

also good sample for DTS-HD MA header patching is the 3/1 DTS-HD MA track that nautilus7 found:

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

madshi

6th September 2008, 17:02

Is the patching only for plain DTS or for DTS-HD also?
can you try the same patching approach with the DTS-HD MA header
Unfortunately patching DTS-HD channel/speaker settings is much more difficult than patching core channel settings. In the DTS-HD data blocks many fields have a variable size which depends on other fields. The changes I'd need to do would result in a change of the DTS-HD data block size. Which again is a very big change to the DTS-HD bitstream which I'm not confident to do correctly. As a result patching speaker settings in DTS-HD data blocks is no go.

rebkell

6th September 2008, 17:05

I asked earlier, but it probably got forgotten in all the posts, but was wondering how the overlaps and gaps get fixed, as in if two segments overlap, do you chop off the end of segment 1 or the start of segment 2 and also wonder how gaps are handled, filled in with silence?

xkodi

6th September 2008, 17:41

As a result patching speaker settings in DTS-HD data blocks is no go.

if so complicated then leave it, it doesn't worth wasting time for that for the time being, because i know for only two problematic samples:

- Edward Scissorhands Blu-ray DTS-HD MA 3/1 track
- Q-tec Hi-Definition Reference Demo Disc DTS-HD MA track

i hope that the studios will never use such weird channel orders in their future releases.

rica

6th September 2008, 17:53

Hi guys,
Sorry for my question which is out of the subject:

eac3to can make 23.976p video and audio to 25p PAL with "-speedup" option. (or reverse with -slowdown)

How can i convert subtitles to 25 fps, would speedup option work for subtitles as well? (or shall i use Supread's 25fps option?)

Thanks.

nautilus7

6th September 2008, 17:56

eac3to can only demux the subtitles. Editing should be done with other programs. SUPread can change framerate while keeping the subs in their original .sup format.

rica

6th September 2008, 18:00

Thx but last question:

Do you mean i have to export as 25fps on Supread ?(and later to convert to srt)

nautilus7

6th September 2008, 18:06

Thx but last question:

Do you mean i have to export as 25fps on Subread ?(and later to convert to srt)No, I though you wanted to keep them as .sup. You can do it in one step.

But since you want to convert to .srt (OCR), I suggest SupRip against SUPread, but it doesn't allow fps changing, so you'll have to use another tool like Subtitle creator.

rica

6th September 2008, 18:08

Thanks, i wanted to keep it as 25fps sup.

Edit: I think it would be a good combination:

Export to 25p sup with SUPRead
Convert to srt with SupRip.

Beastie Boy

7th September 2008, 08:40

but don't want to actually *ever* make use of the embedded AC3 track, you can ask eac3to to use a really low bitrate to save space, e.g. "eac3to source dest.thd+ac3 -192".

This doesn't work for me. If I specify a bitrate, eac3to errors with 'This audio conversion is not supported'.
I have tried different bitrates, 384, 448, but none work.

eac3to input.thd output.thd+ac3 works fine. Also, the output from this is accepted by tsMuxeR as a TrueHD file and muxes successfully :)

Cheers, Beastie.

asarian

7th September 2008, 10:47

Thank you!!

Can you please run "eac3to 1) 3: c:\video\devils.ac3 -logdts"? It seems that the DTS track is decoded by Sonic only as 5.1 although it's a 6.1 track (which results in that strange eac3to error you've seen). I'd like to know why it's only decoded as 5.1. My guess is that it's one of those rare DTS tracks with a slightly unusual format. The "-logdts" parameter should tell me that. Thanks!

Okay, I reran the thing again (adding -sonic, too). And this was the result:

+ DTS-Core
- frameSize 2012
- DTS-ES +
- channelNo 6
- lfe 1
- channelDescr 6.1
- samplingRate 48000
- bitDepth 24
- bitrate 1536000
- dialNorm 4
- extAudio XXCh
- samplesPerFrame 512
- copyHistory 1
+ DTS-HD
- fullSize 2040
- headerSize 28
- refClockCode 1/48000
- frameDurationCode 1
- activeMasks [1], [[1]]
+ Asset [0]
- fullSize 2012
- headerSize 10
- corePackets Core+XXCh
- extSubStrPackets XBR
- bitResolution 24
- maxSampleRate 48000
- totalNumChannels 7
- activeSpeakers C L R Ls Rs LFE Cs ($1f)
-------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 2 subtitle tracks, 1:50:54
1: Chapters, 15 chapters
2: MPEG2, 1080p24 /1.001 (16:9)
3: DTS Hi-Res, English, 6.1 channels, 24 bits, 3093kbps, 48khz, dialnorm: -4dB
4: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
7: Subtitle (PGS), English
8: Subtitle (PGS), Spanish
[a03] AC3 encoding only supports 6 channels. Will mix the surround channels.
[a03] Extracting audio track number 3...
[a03] Removing DTS dialog normalization...
[a03] Decoding with DirectShow (Sonic Audio Decoder)...
[a03] DirectShow reports 5.1 channels, 24 bits, 48khz
[a03] Mixing surround channels...
[a03] The channel modder was started with incorrect parameters.
-------------------------------------------------------------------------------
Aborted at file position 21967306752.
Find sync word: 7ffe8001
Find sync extension: 3f
Frame #1: Core: 1006 words unpacked
Frame #2: Core: 1006 words unpacked
Frame #3: Core: 1006 words unpacked
Frame #4: Core: 1006 words unpacked
Find sync word: 7ffe8001
Find sync extension: 3f
Frame #1: Core: 903 , XXCh: 99 , XBR: 63 words unpacked

tebasuna51

7th September 2008, 12:21

...
+ DTS-Core
- DTS-ES +
- channelNo 6
- lfe 1
- channelDescr 6.1
...

@madshi
Standards DTS-ES (encoded with DTS PRO SERIES SURROUND ENCODER) are decoded by libav and Arcsoft like 5.1 (with Cs mixed in Ls-Rs). Maybe if you expect by 6.1 and only have a 5.1 ...

tebasuna51

7th September 2008, 12:26

I tested tebasuna51's samples with Sonic. The updated list is below:

Thanks Nautilus7. I don't like install Sonic again.

btw, tebasuna51, is that you on the samples? :D
Yep, I make myself the samples.

rica

7th September 2008, 12:39

@madshi
Standards DTS-ES (encoded with DTS PRO SERIES SURROUND ENCODER) are decoded by libav and Arcsoft like 5.1 (with Cs mixed in Ls-Rs). Maybe if you expect by 6.1 and only have a 5.1 ...

Do you mean Libav decodes DTS-HD(HR or MA) as well and both Arcsoft and Libav creates just 5.1? (even the source 6.1 or 7.1)

odin24

7th September 2008, 13:58

Using the command eac3to input.thd output.thd+ac3 make a BD compatible TrueHD stream, would a similar command work for a E-AC3 stream?

eac3to input.eac3 output eac3+ac3?

Thanks, O

nautilus7

7th September 2008, 15:43

No, it won't.

To use E-AC3 on BD you must have a 7.1 stream. And to make it BD compatible, you need an E-AC3 encoder. It 's not like TrueHD + AC3 (individual tracks), but 5.1 AC3 core with E-AC3 extensions for the additional channels. 5.1 E-AC3 tracks in HD DVD are replaced by 5.1 AC3 in BD.

tebasuna51

7th September 2008, 17:29

Do you mean Libav decodes DTS-HD(HR or MA) as well and both Arcsoft and Libav creates just 5.1? (even the source 6.1 or 7.1)

I say DTS-ES, not HD (HR or MA), 6 physical channels (5.1) but 7 logical channels (6.1) with Cs mixed in Ls-Rs

madshi

7th September 2008, 17:32

I asked earlier, but it probably got forgotten in all the posts
You got a reply to your question *directly* under your question. Put your glasses back on!

i know for only two problematic samples:

- Edward Scissorhands Blu-ray DTS-HD MA 3/1 track
- Q-tec Hi-Definition Reference Demo Disc DTS-HD MA track
Edward will work with the next eac3to version and all channels will have the correct volume, but the surround channel will not be bit perfect (the other 3 channels will be bit perfect) when decoding with ArcSoft. Decoding with Sonic will result in all channels being bit perfect with the next eac3to version.

Unfortunately there's nothing I can do about the Hi-Def demo DTS track.

This doesn't work for me. If I specify a bitrate, eac3to errors with 'This audio conversion is not supported'.
Strange, it works for me. Please post your command line and the full eac3to log.

Standards DTS-ES (encoded with DTS PRO SERIES SURROUND ENCODER) are decoded by libav and Arcsoft like 5.1 (with Cs mixed in Ls-Rs).
I don't think that's right. DTS-ES Matrix is decoded as 5.1 by all decoders. DTS-ES Discrete 6.1 is correctly decoded to 6.1 by the ArcSoft decoder, I think (if you're sure that I'm wrong here I'd be happy about a sample). The Sonic decoder decodes DTS-ES Discrete 6.1 to either 5.1 or 6.1, depending on which channel extension is used. Sonic supports 6.1 XCh, but not 6.1 XXCh. libav DTS decoding is generally limited to 5.1.

Okay, I reran the thing again (adding -sonic, too). And this was the result:

+ DTS-Core
- frameSize 2012
- DTS-ES +
- channelNo 6
- lfe 1
- channelDescr 6.1
- samplingRate 48000
- bitDepth 24
- bitrate 1536000
- dialNorm 4
- extAudio XXCh
- samplesPerFrame 512
- copyHistory 1
Yeah, that explains the problem. The Sonic DTS-Decoder doesn't support XXCh. That's why this DTS track is decoded as 5.1, only, while eac3to thought it would be decoded as 6.1. I guess I need to make eac3to aware of that Sonic sometimes decodes 6.1 as 5.1 and sometimes as 6.1... <sigh>

If possible, a small sample of the DTS track would be a nice addition for my sample collection.

madshi

7th September 2008, 17:34

I say DTS-ES, not HD (HR or MA), 6 physical channels (5.1) but 7 logical channels (6.1) with Cs mixed in Ls-Rs
There is DTS-ES 5.1 (also called "DTS-ES Matrix") and DTS-ES 6.1 (also called "DTS-ES Discrete"). These are two very different things. For a decoder DTS-ES Matrix looks like normal 5.1 with just the "ES" flag set. But DTS-ES Discrete has an XCh or XXCh extension in the DTS stream which contains a full separate channel. You can use "eac3to some.dts -logdts" to get exact information about what extensions are contained in your DTS files.

rebkell

7th September 2008, 18:02

You got a reply to your question *directly* under your question. Put your glasses back on!

Nautilus cheated, he edited his answer after I had already read it ;)

nautilus7

7th September 2008, 18:19

Yes, that's true, but I hoped you would have a look in my post again.

rebkell

7th September 2008, 18:31

Yes, that's true, but I hoped you would have a look in my post again.

No problem, I sub to the thread and it goes to the newest post and it works great, but if someone edits a post it doesn't show up as a new post, even though it has a later time/datestamp than the newest post.

I have a different scenario than the majority in this thread(I suppose). I record with the HD-PVR, then reencode with x264(to get some better IDR frames for scenecut purposes) and then mux it back into a ts with tsmuxer, then I cut out commercials with h264ts_cutter and I used to occasionally get some a/v sync problems. So there is no telling what kind of audio overlaps and gaps end up in the cut file.

rica

7th September 2008, 19:49

Why do you use another cutter while TSMuxer exists?
Cut while remuxing and merge the files with eac3to. (another tip, btw there is no VC1 cutter except TSMuxer afaik.)

tebasuna51

7th September 2008, 19:54

@madshi, you are right, I was talking about DTS-ES "Matrix" (with Cs mixed in Ls-Rs) encoded with DTS PRO SERIES SURROUND ENCODER, well detected by eac3to -logdts like:
+ DTS-Core
- frameSize 2013
- DTS-ES +
- channelNo 5
- lfe 1
- channelDescr 5.1
...

I don't have samples "DTS-ES Discrete"

madshi

7th September 2008, 19:55

I don't have samples "DTS-ES Discrete"
You can create them with that Pro encoder you mentioned, I think.

madshi

7th September 2008, 19:58

eac3to v2.61 released

http://madshi.net/eac3to.zip

* added read/write support for Sony wave64 (*.w64) format
* added read/write support for RF64 wave64 (*.rf64) format
* added write support for AGM format
* true bitdepth (e.g. 18 bits) is written to extensible wav header now
* when reading 16/24 (true/storage) WAV files, zero bytes are stripped now
* added HDCD detection for WAV and FLAC files
* added HDCD detection for PCM tracks in VOB/EVO/m2ts containers
* added HDCD decoder written by Christopher Key
* added new option "-decodeHdcd" to decode HDCD information
* HDCD track -> lossy format: HDCD decoding is automatically activated
* when DTS-MA and TrueHD tracks are decoded, a check for HDCD is done
* fixed some incorrect DTS channel masks
* added automatic libav DTS channel remapping
* added automatic ArcSoft DTS channel remapping
* added channel map manipulation to make funny DTS tracks decode with Sonic
* added channel map manipulation to make funny DTS tracks decode with ArcSoft
* added channel volume modification to undo ArcSoft mono surround splitting
* for TrueHD+AC3 creation AC3 delay and gap correction are disabled now
* fixed: DTSWAV and DTSAC3 readers reported too long runtime
* fixed: sometimes processing aborted with a "bitdepth reducer" complaint
Many thanks to Christopher Key for providing the HDCD decoder!

Thunderbolt8

7th September 2008, 20:04

nautilus7

7th September 2008, 20:07

eac3to v2.61 released
Many thanks to Christopher Key for providing the HDCD decoder!And many thanks to you for all the new features and excellent support of this program. :thanks:

* added write support for AGM formatOne question though... Is this the 1st step for adding support for the official DTS encoder?

madshi

7th September 2008, 20:13

guess I might have missed, but what exactly is HDCD (good for)? according to wikipedia it seems to be some kind of technology for accounstic improvement for CDs and DVDs? does someone who just wants to convert the lossless tracks of movies to flac need this somehow?
thanks again for new!
Some audio CDs are encoded with HDCD. On a normal CD player they play just like a normal CD in 16 bit. However, there are CD players which can detect and make use of the HDCD coding. Which results in the audio tracks being enhanced to something more than 16 bit (something between 16 bit and 20 bit). This technology is usually only used for audio CDs and not for movies. Now eac3to can detect HDCD tracks and decode them which basically means increasing the bitdepth from 16 bit to something a little bit higher than 16 bit. If you have hardware which supports HDCD you will not want eac3to to decode HDCD, but let the hardware decode it instead (that saves space). But if your hardware can't do HDCD and you do have some HDCD audio tracks, eac3to can now decode them to full bitdepth.

madshi

7th September 2008, 20:14

* added write support for AGM formatOne question though... Is this the 1st step for adding support for the official DTS encoder?
Maybe. Or maybe not.

nautilus7

7th September 2008, 20:18

Maybe. Or maybe not.I see. ;)

btw, that encoder can do DTS-ES Discrete 6.1 (and Matrix 6.1).

tman24

7th September 2008, 20:29

I'm sure this has been asked a lot, but I would appreciate a quick answer.

Is is possible to extract just 'forced' subtitles from HD rips using eac3to? Primarily this will be for blu-ray. I can extract the whole subtitle stream and process it using SubtitleCreator, but forced subtitles seem to be harder (or maybe not, if anyone knows the best way!).

markrb

7th September 2008, 22:07

I just updated to 2.61 and now get this error.

eac3to v2.61
command line: "D:\HD\ea\eac3to.exe" "H:\BDMV\STREAM\" 1) 2: "E:\video.mkv" -seekToIFrames 3: "E:\audio.ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 3 subtitle tracks, 1:52:02
1: Chapters, 17 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 channels, 16 bits, 48khz
4: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
6: Subtitle (PGS), English
7: Subtitle (PGS), English
8: Subtitle (PGS), Spanish
[a03] AC3 encoding only supports 6 channels. Will mix the surround channels.
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[v02] Muxing video to Matroska...
[a03] Mixing surround channels...
[a03] The channel mixer received channels with different attributes.
[a03] Decoding with ArcSoft DTS Decoder...
[a03] The channel modder didn't receive the format information.
Aborted at file position 32768.

It all worked fine with Live Free and Die Hard DTS-HD
Now I am trying Bank Job DTS-HD and get this error.

Any thoughts?

BTW 2.60 worked fine on this movie.

rebkell

7th September 2008, 23:37

Why do you use another cutter while TSMuxer exists?
Cut while remuxing and merge the files with eac3to. (another tip, btw there is no VC1 cutter except TSMuxer afaik.)

This conversation is totally off topic for this thread, but I have no idea what you are talking about, unless you're talking about tsremuxer and if so, it is too hard to determine what I'm doing and also I've never been able to get it to even mux the aac audio of the recordings I make.

rica

8th September 2008, 00:16

This conversation is totally off topic for this thread, but I have no idea what you are talking about, unless you're talking about tsremuxer and if so, it is too hard to determine what I'm doing and also I've never been able to get it to even mux the aac audio of the recordings I make.

I was talking about TSMuxer as i said. You didn't tell you are re-encoding to aac btw.
Anyway forget about it.

I have a different scenario than the majority in this thread(I suppose). I record with the HD-PVR, then reencode with x264(to get some better IDR frames for scenecut purposes) and then mux it back into a ts with tsmuxer, then I cut out commercials with h264ts_cutter and I used to occasionally get some a/v sync problems. So there is no telling what kind of audio overlaps and gaps end up in the cut file.

Beastie Boy

8th September 2008, 07:36

This doesn't work for me. If I specify a bitrate, eac3to errors with 'This audio conversion is not supported'.

Please ignore this post by me regarding thd+ac3 output.
I was editing the command line produced by eac3to_and_more_gui, and put the bitrate inside the quotes. I feel so embarrassed. :rolleyes:

madshi

8th September 2008, 07:36

Is is possible to extract just 'forced' subtitles from HD rips using eac3to?
I think forced subtitles are usually contained in an extra subtitle track, or am I wrong? If I'm not wrong, you just need to find the right subtitle track and demux it. Or just demux all subtitle tracks and check which one is the smallest. That should be the "forced" one. But I guess not all movies have forced subtitles...

I just updated to 2.61 and now get this error.

eac3to v2.61
command line: "D:\HD\ea\eac3to.exe" "H:\BDMV\STREAM\" 1) 2: "E:\video.mkv" -seekToIFrames 3: "E:\audio.ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 3 subtitle tracks, 1:52:02
1: Chapters, 17 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 channels, 16 bits, 48khz
4: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
6: Subtitle (PGS), English
7: Subtitle (PGS), English
8: Subtitle (PGS), Spanish
[a03] AC3 encoding only supports 6 channels. Will mix the surround channels.
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[v02] Muxing video to Matroska...
[a03] Mixing surround channels...
[a03] The channel mixer received channels with different attributes.
[a03] Decoding with ArcSoft DTS Decoder...
[a03] The channel modder didn't receive the format information.
Aborted at file position 32768.

It all worked fine with Live Free and Die Hard DTS-HD
Now I am trying Bank Job DTS-HD and get this error.

Any thoughts?

BTW 2.60 worked fine on this movie.
Will be fixed in the next build.

madshi

8th September 2008, 07:37

eac3to v2.62 released

http://madshi.net/eac3to.zip

* fixed: downmixing 16 bit 7.1 DTS tracks to 5.1 stopped working in v2.61

xkodi

8th September 2008, 09:28

madshi, how i can use the AGM write support, doing:

eac3to input_file output.agm

gives me

This audio conversion is not supported.

3ngel

8th September 2008, 09:43

Madshi, is there a possibility you can develop the DPLIIx matrix for 7.1 channels downmixing? It's quite different from a standard DPLII

That's the config

http://img169.imageshack.us/img169/2431/129025gc2.jpg

The DPLIIx is the evolution of DPLII and can be used obviously on 5.1 too.

Inventive Software

8th September 2008, 12:04

Thank you madshi for adding HDCD decoding support to your program. I have been looking for a solution to play my HDCD back as intended, and you've provided it! :D

sehgal.v7

8th September 2008, 13:20

@Madshi
Why doesn't TSMuxer accept RAW, LPCM, PCM converted from TrueHD??
eac3to Audio.thd Audio.raw
eac3to Audio.thd Audio.lpcm
eac3to Audio.thd Audio.pcm

It only accepts WAV File// is that which is specified in Blu Ray specifications??
eac3to Audio.thd Audio.wav

I used to thought blu ray uses LPCM/PCM// but in that it swaps channels.

sehgal.v7

8th September 2008, 13:30

@Madshi
eac3to v2.62
command line: eac3to audio.thd audio.wav
------------------------------------------------------------------------------
TrueHD, 5.1 channels, 48khz
Decoding with libav/ffmpeg...
Writing WAV...
Creating file "audio.wav"...
This audio track has a constant bit depth of 16 bits.
eac3to processing took 16 seconds.
Done.

It again gives 24bit file :-(

madshi

8th September 2008, 13:32

madshi, how i can use the AGM write support
I've tested it with "eac3to source.wav dest.agm". Which was your exact command line?

Madshi, is there a possibility you can develop the DPLIIx matrix for 7.1 channels downmixing?
Downmixing to 5.1 or to 2.0? I'd need the exact matrix to implement that.

Why doesn't TSMuxer accept [...]
Why do you ask me? I have not written tsMuxer.

It again gives 24bit file :-(
Will check that... You can probably fix it by doing "eac3to audio.wav audio2.wav".

sehgal.v7

8th September 2008, 13:43

I just wanted to confirm whether acc to Blu Ray specifications, LPCM is correct or PCM or WAV??

madshi

8th September 2008, 13:52

Ever heard of wikipedia?

xkodi

8th September 2008, 13:57

I've tested it with "eac3to source.wav dest.agm". Which was your exact command line?

eac3to.exe 51.wav 51.agm
WAV, 5.1 channels, 0:00:14, 24 bits, 6912kbps, 48khz
This audio conversion is not supported.

[edit]the problem is solved, it turns out that i've messed my eac3to version and was using 2.60. thanks to nautilus7 for giving me the idea to double check that.

nautilus7

8th September 2008, 14:03

I just wanted to confirm whether acc to Blu Ray specifications, LPCM is correct or PCM or WAV??
Look sehgal.v7,

tsmuxer is a buggy and incomplete program. It's developer hasn't shown up since months...

This "problem" with pcm input in tsmuxer has been covered (in this thread) a while ago. In short, it doesn't accept "normal" pcm files, but pcm files in a very weird form. Tebasuna51 was kind enough to make a program (pcm2tsmu) that makes tsmuxer compatible pcm files.

Anyway, I 've seen a lot tsmuxer users asking here why this or that doesn't work with tsmuxer. I understand that they don't know what else to do, except than asking eacto's developer to help, since tsmuxer's developer has forgot them... but that's not proper for madshi.

@ xkodi

Are you sure your version is 2.61/2.62?

sehgal.v7

8th September 2008, 14:26

Why you guys heating up so much??
I jus asked// if you knew, you shared.. thn wht's being arrogant of this??

xkodi

8th September 2008, 14:28

Are you sure your version is 2.61/2.62?

you are right i messed my versions and was using 2.60. the problem is solved. thanks!

nautilus7

8th September 2008, 14:31

Why you guys heating up so much??Oh, really? Too bad i tried to explain the "situation".
I jus asked// if you knew, you shared.. thn wht's being arrogant of this??You should have used search before asking then... (I guess you like this answer better, huh?)

sehgal.v7

8th September 2008, 14:34

Again, you are sounding tough// c'mon bro, if were in ma city, would have take you to for the drink.. Chill!!
Everything thing has a way of explaining..

McFlipp

8th September 2008, 14:38

madshi, how i can use the AGM write support, doing:

eac3to input_file output.agm

gives meWorks fine here:
eac3to v2.62
command line: eac3to E:\truehd.flac D:\audio.agm
------------------------------------------------------------------------------
FLAC, 5.1 channels, 2:32:39, 24 bits, 7620kbps, 96khz
Decoding FLAC...
Writing Agm...
Creating file "audio.agm"...
This audio track has a constant bit depth of 24 bits.
eac3to processing took 9 minutes, 36 seconds.
Done.

edit: Disregard this post... I was being a blind fool....

xkodi

8th September 2008, 14:41

thank you, McFlipp!

works fine here too:

http://forum.doom9.org/showthread.php?p=1180993#post1180993

madshi

8th September 2008, 16:00

Why you guys heating up so much??
I jus asked// if you knew, you shared.. thn wht's being arrogant of this??
We're not heating up. We're just pointing out that some of the questions you're asking don't belong here while others were not really necessary at all (a simple google/wikipedia check would have answered them). You seem to take it for granted that you get good replies to all funny questions for free. Ever thought of that maybe I have a job? And that every minute I spend on answering questions here means that I give up money which I could have earned otherwise in my job? Please limit your questions to those which belong here and which haven't been asked before and where wikipedia and google don't help you. That would show respect to the worth of my time. Thanks.

sehgal.v7

8th September 2008, 16:03

@Madshi
I'm sowwie for my neglection bro// I apologise!!

Regards,
Sehgal.

Encoder888

8th September 2008, 16:10

@madshi

I have trouble demuxing the MPEG-2 stream from VOB files. Here's what I get:

eac3to v2.62
command line: D:\eac3to\eac3to.exe vts_02_1.vob+vts_02_2.vob 2: Video.mkv
------------------------------------------------------------------------------
VOB, 1 video track, 4 audio tracks, 5 subtitle tracks, 0:43:15
1: Joined VOB file
2: MPEG2, 480p24 /1.001 (16:9) with pulldown flags
3: AC3, 5.1 channels, 384kbps, 48khz, dialnorm: -25dB
4: AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -25dB
5: AC3, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB
6: AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
7: Subtitle
8: Subtitle
9: Subtitle
10: Subtitle
11: Subtitle
[v02] This doesn't seem to be a valid MPEG2 stream.
Aborted at file position 1713938432.

I tried it with an earlier version of eac3to (I think it was 2.51) and it worked fine. Please, fix it if possible :)

piratburner

8th September 2008, 16:33

It's should be a DREAM if this fantastic SW could make a m2ts output :-) :-) :-)
Then it would be the ultimate SW, Just think ..... take the movie stream, and a sound, subtitle you like from the BD source and output a new m2ts file ..... but I'm perhaps only dreaming ;-) but it's nice to dream :-)

Inventive Software

8th September 2008, 17:20

Keep dreaming. That's the job of the authoring tools, like TSRemux and TSmuxer. ;)

@madshi: HDCD decoding is PERFECT! The 2 HDCDs I have ripped to FLAC and re-encoded to FLAC via eac3to sound different, so I'm assuming the decoder's more or less perfect. Thank you. :)

madshi

8th September 2008, 18:08

I have trouble demuxing the MPEG-2 stream from VOB files. Here's what I get:

eac3to v2.62
command line: D:\eac3to\eac3to.exe vts_02_1.vob+vts_02_2.vob 2: Video.mkv
------------------------------------------------------------------------------
VOB, 1 video track, 4 audio tracks, 5 subtitle tracks, 0:43:15
1: Joined VOB file
2: MPEG2, 480p24 /1.001 (16:9) with pulldown flags
3: AC3, 5.1 channels, 384kbps, 48khz, dialnorm: -25dB
4: AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -25dB
5: AC3, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB
6: AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
7: Subtitle
8: Subtitle
9: Subtitle
10: Subtitle
11: Subtitle
[v02] This doesn't seem to be a valid MPEG2 stream.
Aborted at file position 1713938432.

I tried it with an earlier version of eac3to (I think it was 2.51) and it worked fine. Please, fix it if possible :)
Hmmmm... That is extremely strange. The code which outputs this MPEG2 error message is meant to run at the beginning of a file, but I see that the processing aborted at 1.7GB, which is WAY after the beginning of the file. How big are the two vob files?

@madshi: HDCD decoding is PERFECT! The 2 HDCDs I have ripped to FLAC and re-encoded to FLAC via eac3to sound different, so I'm assuming the decoder's more or less perfect. Thank you. :)
I hope the decoded files sound better, not just different? :) Unfortunately I don't have any way to check whether the decoder is working correctly or not. But I assume it does.

xkodi

8th September 2008, 18:40

I hope the decoded files sound better, not just different? :) Unfortunately I don't have any way to check whether the decoder is working correctly or not. But I assume it does.

i was thinking about the same - Windows Media Player has build-in HDCD decoder, because Microsoft acquired HDCD company in 2000. so, WMP decoder can be used for reference. the problem is how to make bit-perfect capture of the WMP decoder output (AFAIK, it is not DirectShow filter)?

http://thumbnails4.imagebam.com/1288/41d52e12870874.gif (http://www.imagebam.com/image/41d52e12870874)

Encoder888

8th September 2008, 18:54

@madshi

It's a LOST episode, the two files are 1GB+0.7GB so it must have aborted at the end. It didn't even start demuxing, the line just ran across very fast and I got the error message. I've gotten the same error with about 3/4ths of whatever other movies I've tried, so I don't think it depends on the number or size of VOBs. I just use an older version of eac3to for those, I cant tell you for sure in which version the problem started, but 2.51 works fine.

odin24

8th September 2008, 20:34

I just made an attempt at using a TrueHD track on a HDDVD-->Blu-ray conversion using eac3to input.thd output.thd+ac3, and the end result was a choppy AC3 track, however I did use a DTS track in there as well and it played fine, as did the video. I used vc1conv for the video and tsMuxeR for BD muxing.

Does this command work for HDDVD-->BD conversions?

yonta

8th September 2008, 21:29

eac3to v2.62 crashes on this dtswav file.
sample (http://www.sendspace.com/file/b5utf9)

Tried v2.52 and it works just fine.

nautilus7

8th September 2008, 22:01

2.60 also works. Crash report says something about HD CD decoder. I believe DTS/AC3-WAV support is broken due to HD CD support.

JustABDFan

9th September 2008, 00:37

Hi,

1) I noticed that the libFLAC.dll that comes with Eac3to is 151,552 bytes and is dated 7/22/2007 while the one that comes in MadFLAC is 258,048 bytes and is dated 10/7/2007. I wanted to put Eac3to and Madflac into one "tools" folder so which I should use OR do I just make a folder for each and keep em separate?

2) Any chance of Eac3to using arcsoft decoders for (E)AC3 instead of Nero 7? Nero 7 Ultra isn't being sold by Ahead anymore and I don't want to risk buying the Blu-ray plugin to use with an OEM version then it not work properly.

Thanks.

hdpete

9th September 2008, 07:24

Is there anyway I can remux a blu-ray keeping the TrueHD track to fit on a BD25 for playback via a surround amp connected using HDMI. I know from a couple of costly BD-R's that tsMuxer won't allow the remux of TrueHD. I see that the latest version of eac3to can extract TrueHD as is, but how do I get it onto a blu-ray as I can't use tsMuxer?
I want to keep lossless audio, is the only way to use eac3to to convert the TrueHD to PCM 5.1?
Thanks

Beastie Boy

9th September 2008, 07:51

@hdpete, I have written a post in the tsMuxeR thread (http://forum.doom9.org/showthread.php?p=1181301#post1181301)that may help you. Please keep tsMuxeR questions there.

Cheers, Beastie.

tebasuna51

9th September 2008, 13:26

Thanks madshi for your new version, for me the standard dts is all ok decoded but:

2/2.1 - ArcSoft output is FL FR FC Empty LF BL, the BR channel is missing and need remap -0,1,3,4,2,5

And using -dtslog with Samples_dts-es (http://www.sendspace.com/file/cohrtp) encoded with DTS PRO SERIES SURROUND ENCODER (thanks Elektra999) seems there are a detect problem:

3/3.0 m 3/3.1 m 3/3.0 d 3/3.1 d
+ DTS-Core ------- ------- ------- -------
- frameSize 2013 2013 2013 2013
- DTS-ES + + + +
- channelNo 5 5 5 (?) 6
- lfe 0 1 0 1
- channelDescr 5.0 5.1 5.0 (?) 6.1
- samplingRate 48000 48000 48000 48000
- bitDepth 16 16 16 16
- bitrate 1536000 1536000 1536000 1536000
- extAudio - - XCh XCh
- samplesPerFrame 512 512 512 512
- copyHistory 1 1 1 1

With libav
3/3.1 DTS-ES 6.1 Discrete, output 5.1 with BC mixed in BL-BR, ok but need remap -1,2,0,5,3,4

And ArcSoft problems:
3/3.0 DTS-ES 6.0 Discrete, detected like 5.0 and decoded 5.0 with BC channel mixed in BL-BR like 6.0 Matrix.
3/3.1 DTS-ES 6.1 Discrete, all ok but FL and FR channels broken (missing 99% of samples)

piratburner

9th September 2008, 19:04

Just a Q

What exactly can I expect when doing this command

eac3to BDMV k:\output.mkv ?
a complete mkv with all sound,subs ? or ?

hdpete

9th September 2008, 21:00

If use eac3to "D:\input.thd+ac3" "D:\output.wav" -nero -24, will the resulting wav file have all the lossless date of the trueHD source?
Thanks for your advice

nautilus7

9th September 2008, 21:13

hdpete

9th September 2008, 21:21

First, .thd+ac3 is only accepted for input. Just use .thd. Furthermore, why use nero? Libav is the default TrueHD decoder because it works better. Finally, you don't need any -24 or whatever switch. All source data will be present in the output wav file. If source is 24bit, the wav will be 24bit, if it is 16bit, the wav will be 16bit. You will get a report about the actual bitdepth at the end of the procedure.

So your command should be:

eac3to input.thd output.wav

Althogh my source thd is 24 bit and the wav says it's 24bit, tsmuxer only recognises it as a 16bit, either way the -24 bit didn't seem to make a difference.
I'm usimg nero with blu-ray plugin as I thought it was a reference decoder as opposed to libav.
I just want to be able to convert TrueHD to LPCM to mux with tsmuxer to get a BD25 (tsmuxer doesn't like TrueHD). Previous converts using eac3to have resulted in screaching audio when muxed onto a BD25

nautilus7

9th September 2008, 21:28

Althogh my source thd is 24 bit and the wav says it's 24bit, tsmuxer only recognises it as a 16bit, either way the -24 bit didn't seem to make a difference.That's tsmuxer's bug, I'm afraid.
I'm usimg nero with blu-ray plugin as I thought it was a reference decoder as opposed to libav.What you say is true for the lossy decoders (AC3, E-AC3). All lossless fortmat decoders have to output the same data. That's the case with nero and libav TrueHD decoders. In addition libav supports up to 7.1ch when nero doesnt.
I just want to be able to convert TrueHD to LPCM to mux with tsmuxer to get a BD25 (tsmuxer doesn't like TrueHD). Previous converts using eac3to have resulted in screaching audio when muxed onto a BD25Again tsmuxer's bug. It supports pcm in a weird format. Use pcm2tsmu to convert eac3to's pcm files to tsmuxer's compatible ones.

hdpete

9th September 2008, 21:38

Again tsmuxer's bug. It supports pcm in a weird format. Use pcm2tsmu to convert eac3to's pcm files to tsmuxer's compatible ones.

Thanks a lot for this, I can start backing up my TrueHD blu-rays/HD-DVD's now and keep lossless audio for my Onkyo amp.

hdpete

10th September 2008, 16:14

rack04

10th September 2008, 17:41

What file extenstion should I use to extract the 1080p blu-ray video using eac3to which will allow tsRemux to recognise and import it? Will it be a different file extention depending on the codec, i.e. H264 or VC-1?
Thanks for your help

I always change the container format to mkv and convert the audio to whatever format I fancy. So in my case the extension for the container wouldn't vary by video codec but the audio would. Clear?

nautilus7

10th September 2008, 17:56

rica

11th September 2008, 00:51

Hi,
BD which i have issues is "Welcome Jungle".

Here you will see it has an english thd:

http://img225.imageshack.us/img225/8233/48098432bo2.th.png (http://img225.imageshack.us/my.php?image=48098432bo2.png)

And i can play it with TMT with my analog outputs to my A/V receiver and no problem:

http://img225.imageshack.us/img225/8416/72750525hb8.th.jpg (http://img225.imageshack.us/my.php?image=72750525hb8.jpg)

But i can only see french thd when i open it with eac3to, i extract it to thd+ac3, remuxed to m2ts with TSMuxer and finally remuxed to m2ts with TSRemux using the file created by TSMuxer; it works but it is french:

eac3to v2.60
command line: eac3to\eac3to.exe "F:\RUNDOWN_P1_SPITASLA\BDMV\STREAM\00002.m2ts" 3: "C:\output.thd+ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 13 subtitle tracks, 1:44:15
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, French, 5.1 channels, 48khz, dialnorm: -30dB
4: TrueHD/AC3, German, 5.1 channels, 48khz
5: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
6: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -29dB
7: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
8: Subtitle (PGS), English
9: Subtitle (PGS), French
10: Subtitle (PGS), German
11: Subtitle (PGS), Dutch
12: Subtitle (PGS), Arabic
13: Subtitle (PGS), Modern Greek
14: Subtitle (PGS), Turkish
15: Subtitle (PGS), French
16: Subtitle (PGS), German
17: Subtitle (PGS), Dutch
18: Subtitle (PGS), French
19: Subtitle (PGS), German
20: Subtitle (PGS), Dutch
[a03] Extracting audio track number 3...
[a03] Removing AC3 dialog normalization...
[a03] Removing TrueHD dialog normalization...
[a03] Creating file "C:\output.thd+ac3"...
Video track 2 contains 149977 frames.
eac3to processing took 20 minutes, 49 seconds.
Done.

The Q is where this english thd is hidden?

_ _ _ _

woah!

11th September 2008, 00:59

umm number 7 is the english thd audio isnt it?? you are asking for 3: so you get the french audio...

rica

11th September 2008, 01:02

umm number 7 is the english thd audio isnt it?? you are asking for 3: so you get the french audio...

Thanks i didn't see it really :stupid: :)

Looking but not seeing. How stupid i am :)

Edit:
OK, i got it in english:

http://img339.imageshack.us/img339/422/thdtu8.th.png (http://img339.imageshack.us/my.php?image=thdtu8.png)

_ _ _ __

JustABDFan

11th September 2008, 07:29

Madshi,

I did some checking by downloading old versions and it looks like it was working in 2.54 but broke in 2.55. I'm gonna take a guess that one of these changes from the 2.55 changelog did it. I made some guesses but i could be wrong.

* improved VOB/EVO audio delay detection algorithm
* detection and automatic skipping of invalid vob units
* options "-60i" and "-24p" are no longer supported for MPEG2 video
* improved detection of MPEG2 framerate / pulldown state / mode (my first guess:confused:)
* improved MPEG2 muxing warnings (3rd guess:confused:)
* several bugs in MPEG2 video muxing fixed (2nd guess:confused:)
* fixed interlaced VC-1 muxing with user data (Nine Inch Nails)

@madshi

I have trouble demuxing the MPEG-2 stream from VOB files. Here's what I get:

eac3to v2.62
command line: D:\eac3to\eac3to.exe vts_02_1.vob+vts_02_2.vob 2: Video.mkv
------------------------------------------------------------------------------
VOB, 1 video track, 4 audio tracks, 5 subtitle tracks, 0:43:15
1: Joined VOB file
2: MPEG2, 480p24 /1.001 (16:9) with pulldown flags
3: AC3, 5.1 channels, 384kbps, 48khz, dialnorm: -25dB
4: AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -25dB
5: AC3, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB
6: AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
7: Subtitle
8: Subtitle
9: Subtitle
10: Subtitle
11: Subtitle
[v02] This doesn't seem to be a valid MPEG2 stream.
Aborted at file position 1713938432.

I tried it with an earlier version of eac3to (I think it was 2.51) and it worked fine. Please, fix it if possible :)

Inventive Software

11th September 2008, 11:08

Try a much newer version as well. ;)

sehgal.v7

11th September 2008, 14:00

eac3to v2.62
command line: eac3to Rambo -check
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 1 subtitle track, 1:21:00
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 7.1 channels, 24 bits, 48khz
4: Subtitle (PGS), English
[a03] Extracting audio track number 3...
[s04] Extracting subtitle track number 4...
[v02] Extracting video track number 2...
[a03] Reading RAW/PCM...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] This audio track has a constant bit depth of 24 bits.
Video track 2 contains 116523 frames.
eac3to processing took 46 minutes, 15 seconds.
Done.

This is log of Blu Ray Disc, i created last. In this Swapping & Remapping being done twice, is it alright? Or i had done something wrong while remuxing??

Workaround-
Extracted Video & Audio from Original Blu Ray// Converted Audio from DTSHD-MA to PCM

eac3to v2.62
command line: eac3to Audio.dtshd Audio.pcm
------------------------------------------------------------------------------
DTS Master Audio, 7.1 channels, 24 bits, 48khz
Decoding with ArcSoft DTS Decoder...
Swapping endian...
Remapping channels...
Creating file "Audio.pcm"...
This audio track has a constant bit depth of 24 bits.
eac3to processing took 25 minutes, 42 seconds.
Done.

After this command line: Pcm2Tsmu.exe Audio.pcm Audionew.pcm -i 24 -c 8 -s 48000
And atlast creating Blu Ray Disk with TSMuxer v1.8.4(b)

Atak_Snajpera

11th September 2008, 18:02

Small request madshi.
Could you rename first analyzing process from "progress:" to "analyzing:"
So instead of two "progress:" I would like to have "analyzing:" and then when stream is really demuxed I would have "progress:"

Thanks

K1ngp1ng

13th September 2008, 18:55

i found a bug: truehd 16bit -> flac 24bit

xkodi

13th September 2008, 20:52

i found a bug: truehd 16bit -> flac 24bit

please, upload small sample somewhere, you can cut the first 5-10MB from the truehd track using hex editor.

Snowknight26

13th September 2008, 21:39

Just a little quirk:

eac3to v2.62
command line: eac3to.exe "G:\Unsorted\Click"
------------------------------------------------------------------------------
1) 00032.mpls, 00000.m2ts+00027.m2ts, 1:47:31
- MPEG2, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48khz
- RAW/PCM, English, multi-channel, 48khz
- AC3, English, stereo, 48khz

2) 00031.mpls, 00027.m2ts, 0:58:11
...[snip]...

With a trailing \, :
eac3to v2.62
command line: eac3to.exe "G:\Unsorted\Click"\
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 17 subtitle tracks, 1:47:31
1: Chapters, 16 chapters
2: MPEG2, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
4: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
5: AC3, English, 2.0 channels, 192kbps, 48khz
6: Subtitle (PGS), English
...[snip]...

n0mag!c

14th September 2008, 19:31

When appending several TS file with "+" eac3to can't process files properly.
For error reproduction I took working TS-file and splitted it (with TSsplitter) to 3221380 bytes chunks.
Here is log file:
eac3to v2.62
command line: D:\PROGRAMS\AUDIO\EAC3TO\eac3to.EXE Alien1.TSSplit.1-1434.ts+Alien1.TSSplit.2-1434.ts+Alien1.TSSplit.3-1434.ts+Alien1.TSSplit.4-1434.ts -demux
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 0:00:09
1: MPEG2, 1080i60 /1.001 (16:9)
2: AC3, 2.0 channels, 384kbps, 48khz, dialnorm: -27dB
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[v01] Creating file "Alien1.TSSplit.1-1434 - 1 - MPEG2, 1080i60.m2v"...
[v01] The pes parser received incomplete data.
[a02] The pes parser received incomplete data.
[v01] This doesn't seem to be a valid PES packet.
[a02] This doesn't seem to be a valid PES packet.
Aborted at file position 3237764.

Of course, If I join chunks back (with TSsplitter or with copy /b), resulting file is fine working.

madshi

15th September 2008, 00:24

I have trouble demuxing the MPEG-2 stream from VOB files.
I did some checking by downloading old versions and it looks like it was working in 2.54 but broke in 2.55. I'm gonna take a guess that one of these changes from the 2.55 changelog did it. I made some guesses but i could be wrong.

* detection and automatic skipping of invalid vob units
I think the skipping of invalid vob units is the problem. I've removed that as a test in the next eac3to build. Please retest whether DVD demuxing works again. Thanks!

eac3to v2.62 crashes on this dtswav file.
Thanks for the sample. Will be fixed in the next build.

1) I noticed that the libFLAC.dll that comes with Eac3to is 151,552 bytes and is dated 7/22/2007 while the one that comes in MadFLAC is 258,048 bytes and is dated 10/7/2007. I wanted to put Eac3to and Madflac into one "tools" folder so which I should use OR do I just make a folder for each and keep em separate?
To be honest, I'm not sure. You can try using the newer dll for both. If you run into no problems then it's fine.

2) Any chance of Eac3to using arcsoft decoders for (E)AC3 instead of Nero 7?
Maybe some day.

Thanks madshi for your new version, for me the standard dts is all ok decoded but:

2/2.1 - ArcSoft output is FL FR FC Empty LF BL, the BR channel is missing and need remap -0,1,3,4,2,5

And using -dtslog with Samples_dts-es (http://www.sendspace.com/file/cohrtp) encoded with DTS PRO SERIES SURROUND ENCODER (thanks Elektra999) seems there are a detect problem:

3/3.0 m 3/3.1 m 3/3.0 d 3/3.1 d
+ DTS-Core ------- ------- ------- -------
- frameSize 2013 2013 2013 2013
- DTS-ES + + + +
- channelNo 5 5 5 (?) 6
- lfe 0 1 0 1
- channelDescr 5.0 5.1 5.0 (?) 6.1
- samplingRate 48000 48000 48000 48000
- bitDepth 16 16 16 16
- bitrate 1536000 1536000 1536000 1536000
- extAudio - - XCh XCh
- samplesPerFrame 512 512 512 512
- copyHistory 1 1 1 1

With libav
3/3.1 DTS-ES 6.1 Discrete, output 5.1 with BC mixed in BL-BR, ok but need remap -1,2,0,5,3,4

And ArcSoft problems:
3/3.0 DTS-ES 6.0 Discrete, detected like 5.0 and decoded 5.0 with BC channel mixed in BL-BR like 6.0 Matrix.
3/3.1 DTS-ES 6.1 Discrete, all ok but FL and FR channels broken (missing 99% of samples)
Thanks for testing/reporting and for the new samples! The problems I could fix should be fixed in v2.63. However, ArcSoft seems to have problems with some 6.0 and 6.1 tracks. If I let ArcSoft decode them as 5.1, everything is fine. If I let it decode them as 6.0 or 6.1 the FL and FR channels are broken. Don't see how I could fix that. Funny enough Blu-Ray and HD DVD 6.1 tracks decode fine!! Because of that I can't simply block ArcSoft for 6.x decoding...

In this Swapping & Remapping being done twice, is it alright?
Yes.

Could you rename first analyzing process from "progress:" to "analyzing:"
Changed to "analyze:" and "process:".

i found a bug: truehd 16bit -> flac 24bit
Will be fixed in the next build.

Just a little quirk:

With a trailing \, :
Will be fixed in the next build.

When appending several TS file with "+" eac3to can't process files properly.
For error reproduction I took working TS-file and splitted it (with TSsplitter) to 3221380 bytes chunks.
This is a problem. eac3to contains some extra code to handle Blu-Ray seamless branching correctly. Now if I make TsSplitter splitted TS files work that may eventually harm Blu-Ray seamless branching handling. Generally if you join some TS or m2ts files which were never stored in one file you'll want eac3to to do some extra checking/processing on these files. But if you join some TS or m2ts files which were once stored in one file and which you splitted with a tool like TsSplitter the extra processing makes joining fail. I'll have to think about how to handle this properly. For now you'll have to manually join the parts with TsSplitter first to make eac3to work...

madshi

15th September 2008, 00:28

eac3to v2.63 released

http://madshi.net/eac3to.zip

* fixed: incorrect detection of 6.0 DTS tracks as 5.0
* fixed: incorrect libav DTS channel remapping for 6.x or 7.x tracks
* fixed: incorrect ArcSoft DTS channel remapping for "6.0" and "2/2.1" tracks
* fixed: v2.61+62 incorrectly decoded 16bit TrueHD tracks to 24bit FLAC/WAV/RAW
* fixed: some DTSWAV files made HDCD decoder crash
* fixed: DTSWAV and AC3WAV samplerate and bitdepth were reported incorrectly
* improved DirectShow channel configuration reporting
* undocumented option -progressnumbers now outputs "analyze:" and "process:"

rica

15th September 2008, 00:43

Hey man, you did again; anyway, hard to follow you; thanks again :)

JustABDFan

15th September 2008, 03:56

eac3to v2.63
command line: eac3to d:\video_ts\vts_01_1.vob -demux
------------------------------------------------------------------------------
VOB, 1 video track, 1 audio track, 1 subtitle track, 0:27:12
1: MPEG2, 480p24 /1.001 (16:9) with pulldown flags
2: AC3, 5.1 channels, 448kbps, 48khz
3: Subtitle
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[v01] Creating file "vts_01_1 - 1 - MPEG2, 480p24.m2v"...
[a02] Creating file "vts_01_1 - 2 - AC3, 5.1 channels, 448kbps, 48khz.ac3"...
[s03] Extracting subtitle track number 3...
[s03] Creating file "vts_01_1 - 3 - Subtitle.sup"...
[a02] The last (E-)AC3 frame is incomplete and thus gets skipped.
Video track 1 contains 39141 frames.
eac3to processing took 1 minute, 32 seconds.
Done.

It works now. :thanks:

madshi

15th September 2008, 08:21

It works now.
Good to know - thanks for the quick feedback!

deathlord

15th September 2008, 08:23

ArcSoft seems to have problems with some 6.0 and 6.1 tracks. If I let ArcSoft decode them as 5.1, everything is fine. If I let it decode them as 6.0 or 6.1 the FL and FR channels are broken. Don't see how I could fix that.

I assume the front channels are the same, no matter if you decode to 5.1 or to 6.1. You could check this on a working file. So you could take the front channels from a 5.1 decode and the rest from a 6.1 decode. Of course this is not beautiful, but it may be the best solution to treat these cases where it does not work otherwise.

While you are running tests with arcsoft, could you please also test what it does on a 6.1 track when you ask it to output 7.1?
I have more and more BDs with 6.1 audio and I can not play them back properly.

deathlord

xkodi

15th September 2008, 08:54

hubblec4

15th September 2008, 08:58

hi

madshi you are the best. your prog works very well und your work is great.

a question.

i demux my sourcefiles ever with -demux.
then i get the video-,audistream and all .sups

is there a way to demux all audiostreams and .sups but without the videostream?

hubble

nautilus7

15th September 2008, 09:10

No, automatically. You can though choose which streams to demux by using the stream numbers (1:, 2:, 3:, etc) and the proper extension for each stream.

eac3to v2.41
command line: eac3to "H:\Testosteron HD DVD 1080p VC-1 DD+ 5.1" 1) 1: c:\tes\tes.txt 4: c:\tes\tes.dtshd 5: c:\tes\tes.sup
------------------------------------------------------------------------------
EVO, 1 video track, 2 audio tracks, 1 subtitle track, 1:56:31
"MainMovie"
1: Chapters, 18 chapters with names
2: VC-1, 1080p30 /1.001 (16:9)
3: E-AC3, English, 5.1 channels, 768kbit/s, 48khz, dialnorm: -27dB
4: DTS Hi-Res, English, 5.1 channels, 16 bits, 2082kbit/s, 48khz, dialnorm: -4dB
5: Subtitle, English
Creating file "c:\tes\tes.txt"...
Extracting audio track number 4...
Removing dialog normalization...
Patching bitdepth to 24 bits...
Extracting subtitle track number 5...
Creating file "c:\tes\tes.dtshd"...
Creating file "c:\tes\tes.sup"...
eac3to processing took 14 minutes, 48 seconds.
Done.

madshi

15th September 2008, 09:15

I assume the front channels are the same, no matter if you decode to 5.1 or to 6.1. You could check this on a working file. So you could take the front channels from a 5.1 decode and the rest from a 6.1 decode. Of course this is not beautiful, but it may be the best solution to treat these cases where it does not work otherwise.
I could probably do that, but I don't really like the idea to add such complicated workarounds for clear bugs in a specific decoder which is still being worked on. Maybe ArcSoft will fix that problem sooner or later? You could report it to their support forum...

While you are running tests with arcsoft, could you please also test what it does on a 6.1 track when you ask it to output 7.1?
Already tried that. The front channels are broken if I ask ArcSoft to decode 6.x as 7.1, too.

I have more and more BDs with 6.1 audio and I can not play them back properly.
I don't see why that should be the case. In my tests the Blu-Ray 6.1 tracks decoded perfectly fine with ArcSoft (as I already said earlier).

i believe that Sonic will work on such samples, so in that case better to use Sonic instead of Arcsoft. just decode the 6.1 DTS file with both Sonic and Arcsoft and if the two decoded files are different, probably the Sonic decode is the correct one.
Correct. Sonic works perfectly fine for XCh 6.1 DTS and DTS-HD files. However, Sonic doesn't work for XXCh 6.1 tracks, and it also doesn't work for XCh and XXCh 6.0 tracks. Not much of a problem, though, since 95% of 6.x tracks are XCh 6.1.

madshi

15th September 2008, 09:20

Just noticed that I forgot to add something to the changelog:

* option "-no7doubling" is not supported anymore
* option "-double7" added which upconverts 6.1 to 7.1
So by default 6.1 tracks are not upconverted to 7.1, anymore. You have to manually activate that if you want upconvertion. Also doubling the 7th channel now also halfs the volume of the 7th channel. Which is what ArcSoft's decoder is doing, too, when it doubles the 7th channel.

nautilus7

15th September 2008, 09:45

You have also forgot to remove these options from the command list:

-stripPulldown strips the pulldown from MPEG2 video tracks
-24p/30p/60i force the use of the specified framerate for h264 muxing

:p

madshi

15th September 2008, 10:36

You have also forgot to remove these options from the command list:

-24p/30p/60i force the use of the specified framerate for h264 muxing
Right! Removed that now... :)

-stripPulldown strips the pulldown from MPEG2 video tracks
That one is still needed. MPEG2 pulldown removal is activated automatically only for m2ts and evo sources, but not for demuxed MPEG2 files nor for VOB or TS sources.

nautilus7

15th September 2008, 10:40

That one is still needed. MPEG2 pulldown removal is activated automatically only for m2ts and evo sources, but not for demuxed MPEG2 files nor for VOB or TS sources.Wasn't aware of that case. Thanks for explaining.

rica

15th September 2008, 11:58

Wasn't aware of that case. Thanks for explaining.

Neither am i; this should be added on the first post i think.

Atak_Snajpera

15th September 2008, 12:56

undocumented option -progressnumbers now outputs "analyze:" and "process:"
thank you very much!

Thunderbolt8

15th September 2008, 14:45

eac3to v2.63 released
:thanks:

nurbs

15th September 2008, 19:02

Can eac3to demux blu ray subtitles (PGS) and, if yes, what file extension do I have to use?

Snowknight26

15th September 2008, 19:28

Yes, .sup.

nurbs

15th September 2008, 19:42

Thanks. I tried .pgs and it gave me an error. Also there is nothing about it in the first post so I wasn't sure if it was supported.

rebkell

15th September 2008, 19:48

Thanks. I tried .pgs and it gave me an error. Also there is nothing about it in the first post so I wasn't sure if it was supported.

You can always use demux to see what extensions are being used, just control c when you have determined what extensions eac3to uses.

cg1200

15th September 2008, 23:04

Hi. New to this tool. Worked on one DTS file but the second one threw an error ERROR: block code look-up failed
[dca @ 68A532E0]Didn't get subframe DSYNC. It seems to create an ac3 file but I can't test right now.

Also I have Arcsoft Total Media Theater installed but eac3to does not seem to realize it and won't use that decoder. It uses libav instead. Anyway to force it to work?

rica

15th September 2008, 23:19

cg1200

16th September 2008, 01:09

Try to put " - arcsoft " at the end of the line, if it insists on not working, there must be something wrong since normally it is default decoder.

I don't even see the dts dll's on my computer. I don't get it. I downloaded it from their web site. Is there something I'm mssing? When I look at the about it says Total Media Extreme version 1.0.6.31. Do I have the wrong software?

FWIW, even though eac3to threw those errors it seemed to produce a listenable ac3 file.

JustABDFan

16th September 2008, 01:17

I don't even see the dts dll's on my computer. I don't get it. I downloaded it from their web site. Is there something I'm mssing? When I look at the about it says Total Media Extreme version 1.0.6.31. Do I have the wrong software?

FWIW, even though eac3to threw those errors it seemed to produce a listenable ac3 file.

Could be a PATH issue, this thread talks about it: http://forum.doom9.org/showthread.php?t=139946

rica

16th September 2008, 01:51

I don't even see the dts dll's on my computer. I don't get it. I downloaded it from their web site. Is there something I'm mssing? When I look at the about it says Total Media Extreme version 1.0.6.31. Do I have the wrong software?

FWIW, even though eac3to threw those errors it seemed to produce a listenable ac3 file.

I think you used TMT trial first and after it expired you installed TME, correct?
Firstly uninstall TMT and restart and reinstall TME again; i think it's gonna work for a while :)
EDIT: Arcsoft gives you 42 day trial time by TMT+TME, btw:)

cg1200

16th September 2008, 01:57

I found the proper dll's. They were hiding in a different location. I got a little further - it seemed to recognizt arcsoft but then bombed immediately:

DTS, 5.1 channels, 1:49:48, 16 bits, 1536kbps, 48khz, dialnorm: -2dB
Decoding with ArcSoft DTS Decoder...
Removing DTS dialog normalization...
Patching bitdepth to 24 bits...
Encoding AC3 <640kbps> with libAften...
Creating file "J:\ac3\Track2.ac3"...
The ArcSoft DTS Decoder reported an error while decoding.
Aborted at file position 19054592.

JustABDFan

16th September 2008, 04:03

I have TME 1.0.6.31 trial installed. I just tried to convert a dts file I found here: http://www.kellyindustries.com/sounds.html
Specifically the file named dts_the_other_side_48khz.dts , its about half way down the page in the DTS 48Khz section.
I have "C:\Program Files\Arcsoft\TotalMedia Extreme" in the PATH environmental variables in system properties.

eac3to v2.63
command line: eac3to dts_the_other_side_48khz.dts test.ac3
------------------------------------------------------------------------------
DTS, 5.1 channels, 0:02:36, 24 bits, 1536kbps, 48khz
Decoding with ArcSoft DTS Decoder...
Encoding AC3 <640kbps> with libAften...
Removing DTS zero padding...
Creating file "test.ac3"...
eac3to processing took 10 seconds.
Done.

himan2001

16th September 2008, 11:19

@Madshi:

When you use libav/decoder to decode a german/english track at the SAME time from TWO TrueHD or TWO RAW/PCM
Tracks, the encoding is often distorted/defect. I found this problem on a few machines.

When i decode the RAW/PCM streams from the german Blu-Ray "The Pursuit of Happiness" the result is broken.

When i decode one Stream after another, all is OK. Same happens with TrueHD very often.

I think it´s better to disable the use ob libAV with more then one instance :(

deathlord

16th September 2008, 12:15

I could probably do that, but I don't really like the idea to add such complicated workarounds for clear bugs in a specific decoder which is still being worked on. Maybe ArcSoft will fix that problem sooner or later? You could report it to their support forum...

That sounds reasonable. I will report it to their forum if somebody can point me to some samples showing this problem.

I don't see why that should be the case. In my tests the Blu-Ray 6.1 tracks decoded perfectly fine with ArcSoft (as I already said earlier).
I meant because of the wrong volume in the back channels when decoding to 7.1, but now you have changed this:

Just noticed that I forgot to add something to the changelog:

* option "-no7doubling" is not supported anymore
* option "-double7" added which upconverts 6.1 to 7.1
So by default 6.1 tracks are not upconverted to 7.1, anymore. You have to manually activate that if you want upconvertion. Also doubling the 7th channel now also halfs the volume of the 7th channel. Which is what ArcSoft's decoder is doing, too, when it doubles the 7th channel.

Are you sure arcsoft multiplies exactly with 0.5? Because this can't be correct, the two back speakers are not at the same position and don't radiate in the same direction, so with 0.5 they won't add to 1. Actually it should be 0.76, as I have calculated earlier (http://forum.doom9.org/showthread.php?p=1153717&highlight=tebasuna51#post1153717) using the method proposed by tebasuna51. Maybe I should report this also to arcsoft?

madshi

16th September 2008, 13:54

arcsoft but then bombed immediately
Don't know why. Maybe the DTS file is corrupt? Or maybe the ArcSoft installation is broken...

When you use libav/decoder to decode a german/english track at the SAME time from TWO TrueHD or TWO RAW/PCM Tracks, the encoding is often distorted/defect. I found this problem on a few machines.
That would be VERY bad. Do you happen to have a short sample with which I could reproduce the problem? Two raw/pcm tracks should be no problem cause libav is not involved. Two TrueHD tracks might be a problem. Or maybe the problem is not with decoding but with encoding? Which format are you encoding to?

Are you sure arcsoft multiplies exactly with 0.5?
I said half volume, which is a multiply by 0.5^2 = 0.707.

cg1200

16th September 2008, 16:09

Don't know why. Maybe the DTS file is corrupt? Or maybe the ArcSoft installation is broken...

It has worked for 2 out of 3 soundtracks for now using arcosft so I think it's ok. I think maybe that one soundtrack was corrupt. even though it threw the errors it eventually did produce an AC3 file which worked with the occasional crackle through my speakers.

BTW, this is a great tool. I have been looking high and low for an easy way to convert DTS to AC3. this is by far the easiest tool and works the best. the quality is superb and it only takes a few minutes. Thank you!!

deathlord

16th September 2008, 18:54

I said half volume, which is a multiply by 0.5^2 = 0.707.

Oops, sorry, I should have realized that.
Thanks a lot for the new "-double7" option.
:thanks:

himan2001

16th September 2008, 21:12

Don't know why. Maybe the DTS file is corrupt? Or maybe the ArcSoft installation is broken...

That would be VERY bad. Do you happen to have a short sample with which I could reproduce the problem? Two raw/pcm tracks should be no problem cause libav is not involved. Two TrueHD tracks might be a problem. Or maybe the problem is not with decoding but with encoding? Which format are you encoding to?

I said half volume, which is a multiply by 0.5^2 = 0.707.

Hmm, I have no sample at the moment :( The Blu-Ray i named before produces this problem when decoding PCM german+english at the same time to ac3 -448kbit.

I have the same problem with TrueHD, too...

Only with 2*sonic/dts -> 2*ac3/448 is working all the time.

Only when ONE file at ONE-time is encodet, the output is OK.

The Problem is not always reproduceable. Looks like uninitialized "directshow" - you know: never hope that windows can made more then one thing at the same time :)

Finaly i can say: When you made multi-audio de/enconding at the same time, ONLY Sonic/DTS works all the time.

Nero & Arcsoft can´t be used multiple, so this error never comes up here.

Looks like an directshow or threading or something-else problem...

LessThanJake

16th September 2008, 22:15

undocumented option -progressnumbers now outputs "analyze:" and "process:"

Just a little cosmetic thing.
Every new progress percentage is shown in a new line.
It would be great if it updates in the same line instead.

maybe something like this

... other code ...
cout << "analyze/process: 0%\r";
while(!inputfile.eof()){
if (counter==number) {
counter=0; cout << "analyze/process: "<< getpositioninsource()*100/getfsize() << "%" << "\r";
}
counter++;
... do something ...}
cout << "analyze/process: 100%\n";
... other code ...

with "number" = any multiplier to the size of your readingbuffer for the inputfile

example:
show update of the percentage every 50MB
buffer = 32Kb
--> number = 1600

greets
LTJ

madshi

16th September 2008, 22:40

Hmm, I have no sample at the moment :( The Blu-Ray i named before produces this problem when decoding PCM german+english at the same time to ac3 -448kbit.
If the problem ocurred with PCM then it should also occur with TrueHD, DTS and any other format, because PCM is the least sensitive of all. The problem may be caused by the AC3 encoder or maybe by an unstable PC? Have you overclocked your PC?

I need to be able to reproduce the problem, otherwise I can't do anything.

Just a little cosmetic thing.
Every new progress percentage is shown in a new line.
It would be great if it updates in the same line instead.
The progress percentage display is meant for programs which want to automate eac3to. For those programs I think it's easier to output one new line for every progress step. The option is *NOT* meant to be used by people who call eac3to directly in the command line.

rica

16th September 2008, 22:44

Hi i cut an HD-DVD EVO including a pure TrueHD with TSRemux.
When i play the recreated file no problem.

But when i demux that created file with eac3to like this:

eac3to v2.63
command line: eac3to\eac3to.exe C:\Users\rica\Desktop\out.m2ts 1: video.vc1 2: audio.thd+ac3
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 0:03:59
1: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
2: TrueHD, 5.1 channels, 48khz, dialnorm: -27dB, -761ms
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Extracting audio track number 2...
[a02] Removing TrueHD dialog normalization...
[a02] Removing TrueHD dialog normalization...
[a02] Decoding with libav/ffmpeg...
[a02] Encoding AC3 <640kbps> with libAften...
[v01] Removing VC-1 pulldown...
[v01] Creating file "video.vc1"...
[a02] Creating file "audio.thd+ac3"...
[a02] This audio track has a constant bit depth of 16 bits.
Video track 1 contains 5740 frames.
eac3to processing took 1 minute, 15 seconds.

i got a file renamed to "audio DELAY-761ms.thd+ac3 instead of audio.thd+ac3

No such kind of issues from demuxing source evo.

Even i add an +761ms at the and of the line and re-process with eac3to, it is still out of sync. (Even the reverse doesn't change the result; -761ms)

Don't know what to do in this case?

BTW, libav converts it to constant bitrate from variable?

madshi

16th September 2008, 23:03

i got a file renamed to "audio DELAY-761ms.thd+ac3 instead of audio.thd+ac3

No such kind of issues from demuxing source evo.

Even i add an +761ms at the and of the line and re-process with eac3to, it is out of sync.

Don't know what to do in this case?
eac3to cannot fix audio delays in TrueHD streams. So you need to adjust audio sync in your media player. Or when remuxing the raw streams to a new container you need to mux them in such a way that the audio tracks begins 761ms later than the first video frame. Don't ask me how to do that, I don't know. There's nothing I can do at this time. Delaying TrueHD streams is just not possible technically at this point in time. eac3to can delay all supported audio bitstreams - except TrueHD.

BTW, libav converts it to constant bitrate from variable?
That question doesn't make much sense. libav decodes the track. eac3to checks which true bitdepth the decoded data has and reports that.

rica

16th September 2008, 23:09

Thanks,

That question doesn't make much sense. libav decodes the track. eac3to checks which true bitdepth the decoded data has and reports that.

Sorry but i was told lossless BD/HD audio is variable bitrate and i thought libav decodes in that manner.

What is interesting; mediainfo shows this after remuxing with TSMuxer:

http://img135.imageshack.us/img135/9648/madshizd0.png (http://imageshack.us)
http://img135.imageshack.us/img135/madshizd0.png/1/w690.png (http://g.imageshack.us/img135/madshizd0.png/1/)

nautilus7

16th September 2008, 23:39

OMG!!! rica you 're messing bitRATE with bitDEPTH.

madshi

16th September 2008, 23:43

I thought he meant bitdepth. But it seems he really meant bitrate (which he correctly said). Anyway. Yes, lossless tracks have a variable bitrate. PCM has a constant bitrate. So asking eac3to to decode a TrueHD or DTS-HD Master Audio track to PCM or WAV results in converting variable bitrate to constant bitrate. But I don't really see what is so interesting about that.

rica

16th September 2008, 23:47

Yes i really meant bitrate :p

Yes PCM is constant bitrate :)

nautilus7

16th September 2008, 23:48

[a02] This audio track has a constant bit depth of 16 bitsThis is from his previous post. He has highlighted bitdepth. :eek:

rica

16th September 2008, 23:54

This is from his previous post. He has highlighted bitdepth. :eek:

I did say i've messed, what do you want me to say more?

nautilus7

17th September 2008, 00:00

I did say i've messed, what do you want me to say more?

Look at the time i posted that... Simultaneously with your last post.

This was headed to madshi, but i didn't get what he meant, thus i made that post.

It seems we all messed up. It's late EU hours. Time to sleep. :p

rica

17th September 2008, 00:07

Yes i realized that before you posted :)

But last thing before going to sleep:

I, this time, delayed video reverse but what happened?

Audio automatically put delay itself so nothing has changed :devil:

EDIT: OK, nautilus7 was right; with Virtual Clone Drive everything is OK after my last video delay:)

rack04

17th September 2008, 14:59

I was reading the eac3to wikibook trying to get up to speed on converting LPCM tracks from Blu-ray discs. Is there a bug in the LPCM parameter auto detection process because the wikibook gives two examples of the conversion to wavs?

nautilus7

17th September 2008, 15:05

* Convert a 16 bit 48 KHz 5.1 LPCM track to 6 mono WAV tracks, while LPCM parameter auto detection works fine:

eac3to input.pcm output.wavs

* Convert a 16 bit 48 KHz 5.1 LPCM track to 6 mono WAV tracks, while LPCM parameter auto detection doesn't work:

eac3to input.pcm output.wavs -16 -48000 -6 -little

Do you mean these? Everything works fine with the auto detection. I've never came across a pcm files which eac3to couldn't detect correctly. This is an eac3to feature... I believe it should be explained how to use it. I'll change the description a little bit.

rack04

17th September 2008, 15:10

Do you mean these? Everything works fine with the auto detection. I've never came across a pcm files which eac3to couldn't detect correctly. The examples are there in case eac3to can't find the pcm parameters.

Yes those are the ones that I'm talking about. Thanks for the clarification. BTW, what is the logic behind:

eac3to input.pcm input.wavs

Does this just seperate the LPCM channels into seperate lossless wav files?

nautilus7

17th September 2008, 15:18

Yes those are the ones that I'm talking about. Thanks for the clarification.
I edited the examples a little. :)
BTW, what is the logic behind:

eac3to input.pcm input.wavs

Does this just seperate the LPCM channels into seperate lossless wav files?Yes, plus it puts them into wav container. Don't forget that pcm is a raw format, that's why you have to somehow find what are it's parameters/characteristics. Normally this info is stored into the container, but when you demux pcm it is gone. eac3to finds them by doing some tricks. Most other programs can't.

jeffnoone

17th September 2008, 17:26

Trying to get TrueHD track off a BlueRay disc, and then remux into BlueRay structure using tsMuxeR, then play on PS3 using the AVCHD method

Anyway I cant figure out how to get the TrueHD track to play smoothly

For example, on a BR with TrueHD track in a single m2ts, I extracted three different tracks from the TrueHD
firstly the .thd+ac3, then .thd, then .ac3

Now tsMuxeR would only accept either the .thd+ac3 or the pure .ac3

went ahead and created the BR/AVCHD structure

The .ac3 track plays perfectly using either the PS3 or a Onkyo SR606 to decode

THe PS3 is completely unable to handle the .thd+ac3 - I get silence
The Onkyo plays the .thd+ac3, but with heavy stuttering/gaps

I did the extractions using default settings

So for me anyway,I can get the TrueHD tracks out using eac3to, but I cant get them playing, at least with my equipment

Is there some setting in eac3to that I am missing?
Or is tsMuxeR up to the job?
Or are the PS3/Onkyo not up to the job

Note that the TrueHD track play perfectly (and is recognised properly) by both the PS3 and the Onkyo, if I play the original disc in the PS3

Any help appreciated

Jeff Flowerday

17th September 2008, 20:34

EPiPH0NE

17th September 2008, 21:05

jeffnoone: The PS3 will only play TrueHD, or DTS-MA etc if it's on a Blu-Ray. The only audio codecs really supported in containers being streamed or played off a hd is ac3 or uncompress LPCM. Which basically explains why the AC3 track worked for you.

You could convert the TrueHD track to PCM using eac3to to get it play in all it's lossles glory on the PS3, it's a bit of a waste of space but the only alternative.

I bought a Popcorn Hour A-110 because the PS3 was limited when streaming, but as it turns out the PCH A-110 doesn't work either. It might someday with firmware updates but no official word from them. (Besides actually advertising it as being supported)

I do DTS-MA/TrueHD -> PCM all the time for my A-100 and have been for quite some time and it really doesn't even use that much space when compared to 1080p ES video, sounds perfect, and works on EVERY BluRay player. AFAIK DD+, DTS-MA and TrueHD bit streaming are working on the A-110. I am waiting for either a drop in the PS3 price or an NMT that decodes HD audio. That would be perfect and they wouldn't have to pay the BDA just Dolby/DTS which of course by itself is a lot of cash. I guy can dream..........

Jeff Flowerday

17th September 2008, 21:29

I do DTS-MA/TrueHD -> PCM all the time for my A-100 and have been for quite some time and it really doesn't even use that much space when compared to 1080p ES video, sounds perfect, and works on EVERY BluRay player. AFAIK DD+, DTS-MA and TrueHD bit streaming are working on the A-110. I am waiting for either a drop in the PS3 price or an NMT that decodes HD audio. That would be perfect and they wouldn't have to pay the BDA just Dolby/DTS which of course by itself is a lot of cash. I guy can dream..........

A-110:
DTS-MA is working for the most part. Certain movies seem to still have issues. DD+ isn't working at all and the only TrueHD that works is tsremuxed from the original file which forces you to leave the pulldown flag in. TrueHD rebuilt via tsmuxer won't work even if sent through tsremux afterwards.

EPiPH0NE

17th September 2008, 22:29

A-110:
DTS-MA is working for the most part. Certain movies seem to still have issues. DD+ isn't working at all and the only TrueHD that works is tsremuxed from the original file which forces you to leave the pulldown flag in. TrueHD rebuilt via tsmuxer won't work even if sent through tsremux afterwards.

Either you're doing something wrong or the mods over at the NMT forums are lying. Werner, has said that 7.1 DD+ from BluRay not HD-DVD is working and several others have said that TrueHD from both BluRay and HD-DVD works just fine. Either way doesn't matter for me, I'm happy with my A-100 and don't need HD audio bit streaming I need decoding.

EDIT:

I see VC-1 + TrueHD BluRay's aren't working right. I could probably fix that with Scenarist.

Anyway, sorry for the OT!

magic144

18th September 2008, 03:04

hi all,
just come across this issue of audio gap detaction with BD discs with branching or what have you

just wondering,
if I run a cmd like
eac3to L: 1) -demux

and then it creates .gaps files and asks to rerun the same command line,

does that imply that it would be better to run a much more specific command line in the first place (i.e. audio processing/output only) to avoid having eac3to churn through the huge video stream all over again... OR... will running this same command twice do something different with the video on the second pass as well???

thanks a lot!

m

fib0by

18th September 2008, 03:24

When running eac3to, it prints out these examples:

Examples:
eac3to source.pcm destination.flac
eac3to source.thd destination.flac destination.ac3
eac3to source.evo 1: chapters.txt 2: video.mkv 3: audio.flac 5: subtitle.sup
eac3to feature_1.evo+feature_2.evo movie.mkv
eac3to blurayMovieFolder movie.mkv

This might be pointless for experienced users, but for newbies it's a bit confusing: A newbie might not even realize that this application is capable of much more than that, it's capable of listing the titles on a disk - just run it on the mount point of the disk or the "drive letter" - let's say drive E, and it will give you a list of titles, complete with durations so you can identify the main title:

ec3to E:

Also, it can offer a complete list with all the tracks (video, audio, subtitles) in a title. Just indicate which title you want - let's say title number 5, and it will give you a list will all tracks in the title:

eac3to E: 5)

Finally, it can extract video, audio and subtitles tracks, while following seamless branching if needed. Let's say the previous step indicated that the video is track #2, the audio we want is track #4, and the subtitles we want are track #8. In that case, extraction works like this:

eac3to E: 5) 2: video.mkv 4: audio.ac3 8: subtitles.sup

It takes a while to figure out these features when you're not familiar with the application. Please add some indications to the help text that these features are available.
Based on the existing help, a newbie may be led to conclude that eac3to is purely an audio extraction tool, which is far from truth.

fib0by

18th September 2008, 07:03

Bug: the video track in one title is not detected correctly. See how it's initially detected for title #2 when I run eac3to on the entire disk backup, but then when I run it only on the title number 2, or only on the 00076.m2ts file, it's not detected:

C:\>eac3to E:\hd\BLAH
1) 00027.mpls, 00043.m2ts, 0:52:41
- h264/AVC, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48khz

2) 00025.mpls, 00076.m2ts, 0:39:57
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, multi-channel, 48khz
- AC3, English, multi-channel, 48khz
- AC3, French, multi-channel, 48khz
- AC3, Spanish, multi-channel, 48khz

3) 00093.mpls, 00109.m2ts, 0:24:48
- h264/AVC, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48khz

4) 00091.mpls, 00109.m2ts, 0:24:48
- h264/AVC, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48khz

C:\>eac3to E:\hd\BLAH 2)
M2TS, 4 audio tracks, 5 subtitle tracks, 0:39:57
1: Chapters, 11 chapters
2: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
3: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
4: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -28dB
5: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -30dB
6: Subtitle (PGS), English
7: Subtitle (PGS), French
8: Subtitle (PGS), Spanish
9: Subtitle (PGS), French
10: Subtitle (PGS), Spanish

C:\>eac3to E:\hd\BLAH\BDMV\STREAM\00076.m2ts
M2TS, 4 audio tracks, 5 subtitle tracks, 0:39:57
1: Chapters, 11 chapters
2: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
3: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
4: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -28dB
5: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -30dB
6: Subtitle (PGS), English
7: Subtitle (PGS), French
8: Subtitle (PGS), Spanish
9: Subtitle (PGS), French
10: Subtitle (PGS), Spanish

The 00076.m2ts file is played just fine, image and sound, by Windows Media Player.
If I open the file with tsMuxeR it detects the video track just fine. Extracting video, audio and subtitle tracks with tsMuxeR works fine.

Title 3 on the same disk is detected correctly by eac3to and I can demultiplex it just fine.
Title 1 is detected correctly. I haven't tried to demux it because I don't need it.

Let me know what do you need from me to troubleshoot it.

madshi

18th September 2008, 07:48

does that imply that it would be better to run a much more specific command line in the first place (i.e. audio processing/output only) to avoid having eac3to churn through the huge video stream all over again... OR... will running this same command twice do something different with the video on the second pass as well???
If no gaps are reported for the video then eac3to doesn't do anything different with the video in the 2nd pass. So you could use a more specific command line to redo only the audio tracks with gaps in them. But honestly all you're saving is a few minutes worth of time.

Bug: the video track in one title is not detected correctly.

[...]

Let me know what do you need from me to troubleshoot it.
It would be very nice to get a sample of 76.m2ts (first 50MB, if possible). But before you upload that, please check whether you can reproduce the problem with the 50MB sample. Thanks!!

magic144

18th September 2008, 07:59

thanks madshi...

So you could use a more specific command line to redo only the audio tracks with gaps in them. But honestly all you're saving is a few minutes worth of time.

and presumably several GB of repeated and unnecessary disk I/O?

who knows how much we want to thrash HDDs these days... since they're now 1Tb big, raking them might upset all the eggs I have in this 1 basket :-)

so you're also saying that sometimes there are video gaps that are tidied up in a subsequent pass too... (if so reported)? I did not know that, if that is also the case

madshi

18th September 2008, 08:28

so you're also saying that sometimes there are video gaps that are tidied up in a subsequent pass too... (if so reported)?
Aparrently I did. It's extremely rare, though, at least with Blu-Ray and HD DVD.

magic144

18th September 2008, 08:45

okydoke - thanks very much for the feedback, and the excellent software :-)

kurt

18th September 2008, 22:24

rica

19th September 2008, 01:39

Tegeril

19th September 2008, 02:19

Nullity

19th September 2008, 04:12

I'm having trouble converting the DTS-MA to ac3 from the 28 Days Later Blu-ray disc.

Am I missing a step (I don't have ArcSoft so it's using Sonic) and I do:

eac3to "pathToProperM2TS" 6: -640 28DaysLater.ac3

It says:
"The ArcSoft decoder doesn't seem to work, will use Sonic instead."
"Extracting audio track number 6..."
"Decoding with DirectShow (Sonic Audio Decoder)..."
"DirectShow reports 5.1 channels, 24 bits, 48khz"
"Encoding AC3 (with libAften)..."

And then it hangs, the indicator stops blinking, the eac3to process pegs the core it is using, and nothing happens.

I currently don't have ArcSoft either, so when I want to convert DTS-HD to AC3, I first extract the DTS core. Then I run the core file through eac3to to get the AC3.

Yeah, it's an extra step, but it only takes a few minutes, and it has always worked for me.

Tegeril

19th September 2008, 04:14

rack04

19th September 2008, 04:19

I got problems with processing the constantine hd-dvd:

"The format of the source file could not be detected"

sample (http://ultrashare.net/hosting/fl/448cdb9334/CONSTANTINE.test) (evo, 9 MB)

any help is appreciated :)

(I'm using latest eac3to)

I have this HD DVD. If you're still having problems I could rip it and give it a go. Just let me know.

Duppie

19th September 2008, 04:56

I just want to confirm if some of you also have this problem. Now and then I retest new eac3to versions to see if all is still OK for me. So, I always follow the same procedure to backup my BR movies but with eac3to version 2.63 there seems to be a problem with the PCM conversion. I have followed the same procedure with previous versions (2.57 and 2.59) and it always worked. What I do:

Converted BR audio track (TrueHD) to PCM: C:\eac3to 0001.m2ts 4: E:\audio.pcm. Obviously track 4 is the TrueHD track. Run PCM track through Pcm2Tsmu. Multiplex video and audio to Blu-Ray structure.

When played the sound is a bit fast forwarded and there is a lot of noise, weird sounds and cracks.

Any ideas? Am i doing something wrong? If I could assist in anyway to resolve this please let me know.

Duppie

yonta

19th September 2008, 05:01

I currently don't have ArcSoft either, so when I want to convert DTS-HD to AC3, I first extract the DTS core. Then I run the core file through eac3to to get the AC3.

Yeah, it's an extra step, but it only takes a few minutes, and it has always worked for me.

You don't need an extra step.

eac3to source.m2ts x: target.ac3 -libav

eac3to automatically extracts dtscore from DTS-HD track, decode it with libav and encode to ac3 in one step.

Nullity

19th September 2008, 05:09

You don't need an extra step.

eac3to source.m2ts x: target.ac3 -libav

eac3to automatically extracts dtscore from DTS-HD track, decode it with libav and encode to ac3 in one step.

Ah nice, thanks for the tip!

Snowknight26

19th September 2008, 05:47

Trying to convert the DTS commentary tracks on the Shoot 'Em Up Blu-ray. DTS is extracted properly and is played fine with ffdshow (libdts), but when I use eac3to to convert it to WAV using libav the end result is horrible.

http://www.stfcc.org/misc/shootemup.cut.dts

kurt

19th September 2008, 06:22

I wonder how you extracted this evo?
this is just one part of the hd-dvd. Don't know which programm was used for extracting anymore... :)
I have this HD DVD. If you're still having problems I could rip it and give it a go. Just let me know.
let's wait what madshi said. If opening it in evodemux everything seems to be fine...

I know I could just demux the vc1 (main movie uses vc1) with evodemux and process it with vc1conv. But I would like to use eac3to rather...

madshi

19th September 2008, 08:12

I got problems with processing the constantine hd-dvd:

"The format of the source file could not be detected"

sample (http://ultrashare.net/hosting/fl/448cdb9334/CONSTANTINE.test) (evo, 9 MB)
This one is clearly corrupt. Just play the EVO file and you'll see that the image breaks up at one point. You should rerip the disc.

I'm having trouble converting the DTS-MA to ac3 from the 28 Days Later Blu-ray disc.

Am I missing a step (I don't have ArcSoft so it's using Sonic) and I do:

eac3to "pathToProperM2TS" 6: -640 28DaysLater.ac3

It says:
"The ArcSoft decoder doesn't seem to work, will use Sonic instead."
"Extracting audio track number 6..."
"Decoding with DirectShow (Sonic Audio Decoder)..."
"DirectShow reports 5.1 channels, 24 bits, 48khz"
"Encoding AC3 (with libAften)..."

And then it hangs, the indicator stops blinking, the eac3to process pegs the core it is using, and nothing happens.
I guess that either the Sonic decoder isn't working properly for you (in that case use "-libav" or buy ArcSoft). Or maybe your rip is corrupt, which often makes the Sonic decoder crash or freeze. Try reripping the disc.

I just want to confirm if some of you also have this problem. Now and then I retest new eac3to versions to see if all is still OK for me. So, I always follow the same procedure to backup my BR movies but with eac3to version 2.63 there seems to be a problem with the PCM conversion. I have followed the same procedure with previous versions (2.57 and 2.59) and it always worked. What I do:

Converted BR audio track (TrueHD) to PCM: C:\eac3to 0001.m2ts 4: E:\audio.pcm. Obviously track 4 is the TrueHD track. Run PCM track through Pcm2Tsmu. Multiplex video and audio to Blu-Ray structure.

When played the sound is a bit fast forwarded and there is a lot of noise, weird sounds and cracks.
Try converting to FLAC instead of PCM and check whether the resulting FLAC file plays back in correct speed and without cracks (for playback I'd suggest the madFlac FLAC DirectShow filter). If the FLAC file plays back fine then most probably the problem is caused by either Pcm2Tsmu or tsMuxeR and not by eac3to. If the FLAC is bad, too, I'd be very interested in a sample of that m2ts file!

Trying to convert the DTS commentary tracks on the Shoot 'Em Up Blu-ray. DTS is extracted properly and is played fine with ffdshow (libdts), but when I use eac3to to convert it to WAV using libav the end result is horrible.

http://www.stfcc.org/misc/shootemup.cut.dts
Thanks for the sample. Seems to be a bug in libav. You'll have to use ArcSoft, Sonic or Nero for decoding. Tested with ArcSoft and the result is fine.

let's wait what madshi said. If opening it in evodemux everything seems to be fine...
EvoDemux doesn't care about corruption. eac3to does.

I know I could just demux the vc1 (main movie uses vc1) with evodemux and process it with vc1conv. But I would like to use eac3to rather...
You could still demux with EvoDemux and then use eac3to for everything else. eac3to can do exactly the same thing as vc1conv on raw demuxed VC-1 streams, too ("eac3to source.vc1 dest.vc1" will remove the pulldown etc). BUT your best bet is to rerip the disc. After that eac3to should properly handle the EVO file.

kurt

19th September 2008, 08:43

ok, thank you for your suggestion. I'll look into the main movie and see if it is corrupt too :)

(I'm sitting here behind a remote desktop so I didn't notice that the sample is garbled)

odin24

19th September 2008, 09:57

Duppie

19th September 2008, 10:31

@Madshi

Thanks for the information. Unfortunately I can only test PCM since the PS3 does not accept FLAC and my PC is too slow for HD playback. I also think FLAC is not supported in .m2ts containers. Is there another way I can test eac3to for you to determine if the problem is with Pcm2Tsmu or TsMuxeR? The thing I cannot understand is that it works fine with 2.57 with the same Pcm2Tsmu and TsMuxeR versions.

@Odin24

Thanks for your suggestions. I really would like to keep it as simple as possible (eac3to.exe file.m2ts audio.pcm). For now I will stick with eac3to version 2.57 until I do some more tests on 2.63.

madshi

19th September 2008, 10:49

Unfortunately I can only test PCM since the PS3 does not accept FLAC and my PC is too slow for HD playback. I also think FLAC is not supported in .m2ts containers. Is there another way I can test eac3to for you to determine if the problem is with Pcm2Tsmu or TsMuxeR? The thing I cannot understand is that it works fine with 2.57 with the same Pcm2Tsmu and TsMuxeR versions.
Ok, if it works fine with 2.57 then this looks like an eac3to bug. Can you upload a little sample of that m2ts file for me? 50MB would be great. Thanks!

BTW, I didn't suggest that you use FLAC for your movies. I only suggested that you should convert to FLAC as a *test*. Your PC may be to slow to playback HD video, but it will definitely not be too slow to play a FLAC audio track. So that's a valid test to check whether the FLAC audio track sounds ok, or whether it plays too fast, too, and with artifacts.

tebasuna51

19th September 2008, 10:51

Try using just eac3to, also use eac3to and more gui. Use the Remap to Blu ray 6 channel options as well, and output to .wav. tsMuxeR accepts it as PCM, unfortunately I have no way of testing due to reciever limitations. Please try this, I am curious to see if it works.
Decoding from TrueHD never need remap options.

Always is recommended decode to wav but, when the wav file is >4GB, tsMuxer don't work. The workaround is decode to lpcm and use Pcm2Tsmu v1.3 (http://www.sendspace.com/file/som3na) to insert the special headers required by tsMuxer.

If the parameters are correct, example:

Pcm2Tsmu input.pcm output_tsm.pcm -c 8 -i 16 -s 96000

(for a 8 channels, 16 bitdepth, samplerate 96 KHz) and tsMuxer accept the audio, can't be a Pcm2Tsmu problem.

@Duppie, if the flac file play properly like madshi say, please use the Pcm2Tsmu thread (http://forum.doom9.org/showthread.php?t=138417) to post a sample and continue the discussion.

Duppie

19th September 2008, 11:07

@Madshi

Sorry, you are right. The audio can be tested alone with no powerfull PC. I will do some more tests over the weekend. If need to be, it is my first upload, where can I dump a sample if it is really a problem of eac3to?

@tebasuna51

Thanks for that. Will do, after some tests.

nautilus7

19th September 2008, 11:41

where can I dump a sample?
sendspace, mediafire, repidshare etc. To cut a sample use a hex editor like HxD.

tebasuna51

19th September 2008, 12:06

Testing lpcm management I found a bug remapping 7 channel.

I convert a 331.wav to a 331.pcm and other time to a 331d.wav with this problem:
331.wav -> FL-FR-FC-LF-BL-BR-BC
331d.wav -> FL-FR-FC-BR-LF-BL-BC
Logs:
eac3to v2.63
command line: eac3to.exe 331.wav 331.pcm
------------------------------------------------------------------------------
WAV, 6.1 channels, 0:00:20, 16 bits, 5376kbps, 48khz
Reading WAV...
Swapping endian...
Creating file "331.pcm"...
This audio track has a constant bit depth of 16 bits.
eac3to processing took 1 second.
Done.

eac3to v2.63
command line: eac3to.exe 331.pcm 331d.wav
------------------------------------------------------------------------------
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 have 7 channels.
The RAW/PCM file seems to be big endian.
The RAW/PCM file seems to have a bitdepth of 16 bits.
RAW/PCM, 6.1 channels, 0:00:20, 16 bits, 5376kbps, 48khz
This is probably a Blu-ray PCM track. Will remap channels accordingly.
Reading RAW/PCM...
Swapping endian...
Remapping channels...
Writing WAV...
Creating file "331d.wav"...
This audio track has a constant bit depth of 16 bits.
eac3to processing took 1 second.
Done.

madshi

19th September 2008, 13:53

Testing lpcm management I found a bug remapping 7 channel.

I convert a 331.wav to a 331.pcm and other time to a 331d.wav with this problem:
331.wav -> FL-FR-FC-LF-BL-BR-BC
331d.wav -> FL-FR-FC-BR-LF-BL-BC

The problem is that eac3to does not remap the channels to Blu-Ray order when exporting to PCM. I guess it should do that?

P.S: Oh, 6.1 channels! My fault. There's just something missing. Will add that to the next eac3to build...
P.P.S: Blu-Ray does not support 6.1 LPCM. Instead it only supports 7.1 LPCM, as far as I know. So should I enforce channel doubling when exporting to PCM?

odin24

19th September 2008, 18:28

Decoding from TrueHD never need remap options.

Always is recommended decode to wav but, when the wav file is >4GB, tsMuxer don't work. The workaround is decode to lpcm and use Pcm2Tsmu v1.3 (http://www.sendspace.com/file/som3na) to insert the special headers required by tsMuxer.

If the parameters are correct, example:

Pcm2Tsmu input.pcm output_tsm.pcm -c 8 -i 16 -s 96000

(for a 8 channels, 16 bitdepth, samplerate 96 KHz) and tsMuxer accept the audio, can't be a Pcm2Tsmu problem.

@Duppie, if the flac file play properly like madshi say, please use the Pcm2Tsmu thread (http://forum.doom9.org/showthread.php?t=138417) to post a sample and continue the discussion.

When you say >4GB... tsMuxeR doesn't work. Do you mean it won't accept the .wav file at all, or there will be playback problems after muxing?

EPiPH0NE

19th September 2008, 18:58

^^^It Won't accept the file.

tebasuna51

19th September 2008, 19:19

When you say >4GB... tsMuxeR doesn't work. Do you mean it won't accept the .wav file at all, or there will be playback problems after muxing?

Please use the tsMuxer thread to talk about this problem.

tsMuxer, like other soft, accept the wav but don't play/use the full length.

Jeff Flowerday

19th September 2008, 21:49

rica

19th September 2008, 22:36

Converting TrueHD, DTS-MA and DD+ to PCM. Is one library better and more exact then the next? I don't have nero or total media installed and just planned on using the default library.

I want the most accurate conversion available what should I use?

Author has selected the most accurate decoders even you don't have them; so you don't have option B but just an option A which is available on default library :)
I mean there is no option B in default library.

Jeff Flowerday

19th September 2008, 22:52

Author has selected the most accurate decoders even you don't have them; so you don't have option B but just an option A which is available on default library :)
I mean there is no option B in default library.

Having a little trouble understanding you. Are you saying the default library does as good a job as nero or any of the other decoders?

rica

19th September 2008, 22:59

Having a little trouble understanding you. Are you saying the default library does as good a job as nero or any of the other decoders?

No.
The best decoders are unfortunatlely the commercial ones like nero or TMT which are selected by Madshi as default.
If eac3to can not find those defaults, that time it will use the default ones coming with the library.
For any particular format, you don't have two options since library gives only one default decoder for any particular format.

Jeff Flowerday

19th September 2008, 23:03

No.
The best decoders are unfortunatlely commercial ones like nero or TMT which are selected by Madshi as default.
If eac3to can not find those defaults, that time it will use the default ones coming with the library.

So for the most accurate conversion from TrueHD/DTS-MA/DD+ to PCM uncompressed I should purchase say TMT?

nautilus7

19th September 2008, 23:08

Is it sooo hard to read the 1st post?

Jeff Flowerday

19th September 2008, 23:10

Ok I see mention that the default libav works perfectly for TrueHD. What about DD+ though?

rica

19th September 2008, 23:12

So for the most accurate conversion from TrueHD/DTS-MA/DD+ to PCM uncompressed I should purchase say TMT?

The answer has been given on the first post by madshi:

default decoders used by eac3to:

MP1, MP2, MP3: libav/ffmpg (free-from library)

(E-)AC3: Nero (commercial)

AAC: Nero (commercial)

TrueHD: libav/ffmpeg (free-from library)

DTS(-HD): ArcSoft (commercial)

So finally Nero 7 + HD-Plugin (not 8 at all) is recommended for DD+ and TMT is recommended for DTS/DTS-HD.... For TrueHD, libav/ffmpeg is recommended which is coming with the library.

And, you don't have to purchase anything; it is up to you. (trust your own ears first; it is a rule.)
_ _ _ _ _

shambles

20th September 2008, 12:07

To use flac with 3/1.0 channels maybe we need split the surround channel in:
BL = 0.707 x S
BR = 0.707 x S
and use a 3/2.0 configuration supported by flac.

Yeah, sounds like a good workaround.

was this ever implemented? i have a bd with a 4ch pcm track (3/1.0 configuration) and eac3to seems to treat it as 2/2.0 when encoding it to flac

sample: http://www.sendspace.com/file/gvx7sn

It would be very nice to get a sample of 76.m2ts (first 50MB, if possible). But before you upload that, please check whether you can reproduce the problem with the 50MB sample. Thanks!!

here's a sample from another disc but with the same problem

http://www.sendspace.com/file/kifr28

madshi

20th September 2008, 14:45

was this ever implemented? i have a bd with a 4ch pcm track (3/1.0 configuration) and eac3to seems to treat it as 2/2.0 when encoding it to flac

sample: http://www.sendspace.com/file/gvx7sn
In order to make this fully work there are 3 things that need to be done:

(1) eac3to must properly detect the channel configuration. The track could be 2/2 or 3/1. The next eac3to version will detect that correctly for m2ts sources.

(2) FLAC used to not support funny channel masks. But I've seen that there's a way now to store a WAV channel mask as FLAC meta information. So storing 3/1 in FLAC should be possible now. But eac3to will have to read/write that channel mask meta information. I've put that on my to do list, but it might take a while until I find the time to implement that.

(3) The FLAC decoder (e.g. madFlac) must be able to read and interpret the channel mask meta information. madFlac doesn't do that yet.

here's a sample from another disc but with the same problem

http://www.sendspace.com/file/kifr28
Thanks. Same problem as with the other sample. Will be fixed by the next eac3to version.

monohouse

20th September 2008, 15:28

does eac3to allow joining independent mono wav/raw files into a single multi-channel file ? because it seems there are no wave editors that are capable of reading a single 7.1 channel wav larger than 4 GB, while some maybe able to allow the one (larger than 4 GB) but that same program does not allow the other (7.1 channel) and the other way around, there may be one or two programs that do support reading such a file fully (both 4 GB size AND 7.1 channels), but they have other drawbacks (only open first 30 minutes (nero) and others open the multi-channel file and open all the channels as mono separated channels, without support for joining them into a single .wav file (cooledit))

kurt

20th September 2008, 16:49

I got a problem with converting vc1 tracks from hddvd (Full Metal Jacket) and Blu-Ray (The Prestige)

(this time the files are ripped properly :D)

eac3to v2.63
command line: eac3to feature_01.evo+feature_02.evo fullmetal.mkv
------------------------------------------------------------------------------
EVO, 1 video track, 6 audio tracks, 10 subtitle tracks, 1:56:30
1: Joined EVO file
2: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
3: E-AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
4: TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
5: E-AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: E-AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
7: E-AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
8: E-AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -9ms
9: Subtitle
10: Subtitle
11: Subtitle
12: Subtitle
13: Subtitle
14: Subtitle
15: Subtitle
16: Subtitle
17: Subtitle
18: Subtitle
[v02] Extracting video track number 2...
[v02] Removing VC-1 pulldown...
[v02] Muxing video to Matroska...
[a03] Extracting audio track number 3...
[a04] Extracting audio track number 4...
[a06] Extracting audio track number 6...
[a07] Extracting audio track number 7...
[a08] Extracting audio track number 8...
[a05] Extracting audio track number 5...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Encoding FLAC with libFlac...
[a05] Removing E-AC3 dialog normalization...
[a07] Removing E-AC3 dialog normalization...
[a03] Removing E-AC3 dialog normalization...
[a06] Removing E-AC3 dialog normalization...
[a08] Removing E-AC3 dialog normalization...
[a04] Creating file "fullmetal - 4 - TrueHD, 5.1 channels, 48khz.24bit.flac"...
[a05] Creating file "fullmetal - 5 - E-AC3, 5.1 channels, 640kbps, 48khz.eac3"...
[s15] Extracting subtitle track number 15...
[s18] Extracting subtitle track number 18...
[a03] Creating file "fullmetal - 3 - E-AC3, 5.1 channels, 448kbps, 48khz.eac3"...
[a06] Creating file "fullmetal - 6 - E-AC3, 5.1 channels, 448kbps, 48khz.eac3"...
[a07] Creating file "fullmetal - 7 - E-AC3, 5.1 channels, 448kbps, 48khz.eac3"...
[s12] Extracting subtitle track number 12...
[s11] Extracting subtitle track number 11...
[s17] Extracting subtitle track number 17...
[s16] Extracting subtitle track number 16...
[a08] Creating file "fullmetal - 8 - E-AC3, 2.0 channels, 192kbps, 48khz.eac3"...
[s14] Extracting subtitle track number 14...
[s13] Extracting subtitle track number 13...
[s10] Extracting subtitle track number 10...
[s09] Extracting subtitle track number 9...
[s12] Creating file "fullmetal - 12 - Subtitle.sup"...
[s10] Creating file "fullmetal - 10 - Subtitle.sup"...
[s18] Creating file "fullmetal - 18 - Subtitle.sup"...
[s09] Creating file "fullmetal - 9 - Subtitle.sup"...
[s11] Creating file "fullmetal - 11 - Subtitle.sup"...
[s17] Creating file "fullmetal - 17 - Subtitle.sup"...
[s16] Creating file "fullmetal - 16 - Subtitle.sup"...
[s15] Creating file "fullmetal - 15 - Subtitle.sup"...
[s14] Creating file "fullmetal - 14 - Subtitle.sup"...
[s13] Creating file "fullmetal - 13 - Subtitle.sup"...
[a04] This audio track has a constant bit depth of 16 bits.
[a04] The zero bytes were successfully removed.
Added fps value to MKV header.
Video track 2 contains 167613 frames.
eac3to processing took 1 hour, 58 minutes.
Done.

Everything seems to be fine but after muxing the mkv to TS with tsmuxer (1.8.5) the final ts file shows a different length in mpc compared with the source file (4h for Prestige, ~3h for Full Metal Jacket). Didn't have this problem before with vc1conv and different movies... any ideas what went wrong? tsmuxers fault?

I'm trying now vc1conv again, but this will take time...

Beastie Boy

20th September 2008, 17:07

Everything seems to be fine but after muxing the mkv...

Does the mkv video file play correctly? If so then I suspect eac3to is not at fault.

Cheers, Beastie

rica

20th September 2008, 17:19

I got a problem with converting vc1 tracks from hddvd (Full Metal Jacket) and Blu-Ray (The Prestige)

(this time the files are ripped properly :D)

Everything seems to be fine but after muxing the mkv to TS with tsmuxer (1.8.5) the final ts file shows a different length in mpc compared with the source file (4h for Prestige, ~3h for Full Metal Jacket). Didn't have this problem before with vc1conv and different movies... any ideas what went wrong? tsmuxers fault?

I'm trying now vc1conv again, but this will take time...

TSMuxer doesn't like vc1mkvs remuxed by haali.
Try to extract vc1 and audio with eac3to and remux with TSMuxer.
If you still want an mkv, try mkvmergegui using remuxed file.

kurt

20th September 2008, 17:28

Does the mkv video file play correctly? If so then I suspect eac3to is not at fault.

Cheers, Beastie
the file seems to be ok
TSMuxer doesn't like vc1mkvs remuxed by haali.
Try to extract vc1 and audio with eac3to and remux with TSMuxer.
If you still want an mkv, try mkvmergegui using remuxed file.
ah, ok. but how to extract the vc1 from the mkv? eac3to gives an error message (The format of the source file could not be detected) on its own mkv :D

trying now mkvextract... takes a while....

Edit: mkvextract produces an avi, which I don't expect that tsmuxer would load it....

rica

20th September 2008, 17:34

You must be making something wrong.

Extract vc1 directly from your evo; this is the best way.

(BTW i have lived no any issue extracting vc1 from the mkvs created by eac3to)

Edit: Take a look as well:

http://forum.doom9.org/showthread.php?p=1173291#post1173291

http://forum.doom9.org/showthread.php?p=1173370#post1173370

_ _ _ _ _

kurt

20th September 2008, 17:36

You must be making something wrong.

Extract vc1 directly from your evo; this is the best way.
ok, trying -demux now :)

madshi

20th September 2008, 18:53

tsMuxeR is known to dislike VC-1 MKVs created by eac3to. That's tsMuxeR's fault. So any further discussion about this should be done in the tsMuxeR thread.

madshi

20th September 2008, 18:54

does eac3to allow joining independent mono wav/raw files into a single multi-channel file ?
No, not at this time. But I think wavewizard does that. Or was it another tool? There was one freeware tool which can do that.

peterjcat

20th September 2008, 22:36

tsMuxeR is known to dislike VC-1 MKVs created by eac3to. That's tsMuxeR's fault. So any further discussion about this should be done in the tsMuxeR thread.

Madshi, is there any chance that eac3to will one day include remuxing to EVO/(M2)TS? I only ask since you're clearly a genius, eac3to now does such a great job making elementary streams but there seems no way to mux some of them back together due to the limitations of all the free muxers (talking about TrueHD). I'm sure remuxing is a lot more complex than demuxing but is it in your thoughts?

Please don't take this as a criticism of your remarkable application, rather as a sign of absolute faith in your ability :)

lexor

20th September 2008, 23:07

Madshi, is there any chance that eac3to will one day include remuxing to EVO/(M2)TS? I only ask since you're clearly a genius, eac3to now does such a great job making elementary streams but there seems no way to mux some of them back together due to the limitations of all the free muxers (talking about TrueHD). I'm sure remuxing is a lot more complex than demuxing but is it in your thoughts?

I'm not madshi, but people keep asking this and the answer is always the same. Implementing an evo/ts muxer is way to much work, madshi doesn't wish to do that.

But while we on the subject is TSMuxer really the only thing we have? Is there absolutely nothing else either free or reasonably priced?

Jeff Flowerday

20th September 2008, 23:13

rica

20th September 2008, 23:14

Lexor,
i don't know even any expensive remuxer which is able to remux HD audio formats with h264 or vc1 videos; do you?

nautilus7

20th September 2008, 23:47

rica

20th September 2008, 23:56

I totally agree with you. Sometimes i wish all audio/video editing/playback programs were written by madshi.

Madshi, we 're grateful. :thanks:

My vote is valid as well?
:thanks: in advance.

nautilus7

21st September 2008, 00:02

Just to be clear...

I didn't ask madshi to add support for m2ts/evo muxing. I know his opinion. I only expressed my faith and admiration for his abilities.

rica

21st September 2008, 00:19

Neither did i :)

tebasuna51

21st September 2008, 00:39

Jeff Flowerday

21st September 2008, 00:56

Please post your logs or a sample at appropriate threads:
tsMuxer (http://forum.doom9.org/showthread.php?t=134104)
pcm2tsmu (http://forum.doom9.org/showthread.php?t=138417)

Since there are 3 programs involved it made the most sense to start my search for a solution to the problem with the first program.

Are you saying the issue is most likely pcm2tsmu or tsmuxer?

Snowknight26

21st September 2008, 03:46

Whenever I try to mux a video track to an mkv (mostly when the Blu-ray contains multiple .m2ts files), the muxing process stalls very early on yet eac3to still sits there as if its going. The resulting .mkv is always 2,752,512 bytes.

eac3to v2.63
command line: eac3to.exe -test
------------------------------------------------------------------------------
Nero Audio Decoder (Nero 7 or older) doesn't seem to be installed
http://www.nero.com/eng/store-blu-ray.html
CAUTION: You need Nero 7. Nero 8 won't work with eac3to.
ArcSoft DTS Decoder doesn't seem to be installed
http://www.arcsoft.com/products/totalmediatheatre
Sonic Audio Decoder (2.63.0.0) doesn't seem to be installed
Haali Matroska Muxer (2008-03-29) is up to date
Nero AAC Encoder could not be located
http://www.nero.com/eng/nero-aac-codec.html
Copy NeroAacEnc.exe to the eac3to or to the Windows folder.
Surcode DTS Encoder doesn't seem to be installed
http://www.surcode.com
MkvToolnix (2.3.0.0, 2008-09-20) is up to date

Doesn't matter if its H.264, VC-1, or MPEG-2 (haven't tried the latter).

Nullity

21st September 2008, 06:58

Anyone have any issue converting the TrueHD track on the 300 HD-DVD to pcm? After running it through pcm2tsmu and then muxing it into a m2ts container. I'm only getting white noise on the pcm track. The ac3 track that I created at the same time is fine.

I tested the result in both TMC and my NMT box.

I am also having this same problem, although I have ruled out both pcm2tsmu and TsMuxer as the cause.

I have an EAC3 audio track which was ripped from an HDDVD. I have tried converting it to PCM with the default Nero, and also forcing libav. In both cases, the output file is just static noise (tested with GraphStudio).

I don't believe the source EAC3 is corrupted because I can play it back fine using GraphStudio. Although, I had to do this a weird way using ArcSoft - I was not able to connect the EAC3 source directly to either "Nero Audio Decoder 2" or "ffdshow Audio Decoder" filters. Admittedly, I do not have much experience using GraphStudio, maybe someone could inform me of the correct path to play through nero or libav?

Anyway, as you can see, the static problem occurs during the eac3to step. If a sample is needed, I would be happy to upload one somewhere. Just let me know what is needed and how to cut the files, I haven't done that before.

eac3to v2.63
command line: eac3to audio.eac3 audio.pcm
------------------------------------------------------------------------------
E-AC3, 5.1 channels, 1:58:16, 1536kbps, 48khz
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Swapping endian...
Remapping channels...
Creating file "audio.pcm"...
eac3to processing took 4 minutes, 40 seconds.
Done.

eac3to v2.63
command line: eac3to audio.eac3 audio.pcm -libav
------------------------------------------------------------------------------
E-AC3, 5.1 channels, 1:58:16, 1536kbps, 48khz
Decoding with libav/ffmpeg...
Remapping channels...
Loading white noise (needed for dithering)...
Swapping endian...
Remapping channels...
Creating file "audio.pcm"...
eac3to processing took 3 minutes, 28 seconds.
Done.

Jeff Flowerday

21st September 2008, 08:25

Here's 15 seconds from the beginning of 300, it should suffice to help find the problem if it is eac3to. I know "The Invasion" is doing something similar to me as well.

http://www.flowerdayconsulting.com/misc/sample.zip

Jeff Flowerday

21st September 2008, 08:32

I am also having this same problem, although I have ruled out both pcm2tsmu and TsMuxer as the cause.

I have an EAC3 audio track which was ripped from an HDDVD. I have tried converting it to PCM with the default Nero, and also forcing libav. In both cases, the output file is just static noise (tested with GraphStudio).

I don't believe the source EAC3 is corrupted because I can play it back fine using GraphStudio. Although, I had to do this a weird way using ArcSoft - I was not able to connect the EAC3 source directly to either "Nero Audio Decoder 2" or "ffdshow Audio Decoder" filters. Admittedly, I do not have much experience using GraphStudio, maybe someone could inform me of the correct path to play through nero or libav?

Anyway, as you can see, the static problem occurs during the eac3to step. If a sample is needed, I would be happy to upload one somewhere. Just let me know what is needed and how to cut the files, I haven't done that before.

I've only seen it happen on 2 TrueHD sources, 2 other ones were fine, the 4 DD+ I've done so far have all converted fine using nero.

If you want to clip a sample just use tsremux on the evo and tell it cut off all but 15 seconds from the end, it will package the beginning 15 seconds up in a container for you.

madshi

21st September 2008, 08:46

Please don't take this as a criticism of your remarkable application, rather as a sign of absolute faith in your ability :)
I totally agree with you. Sometimes i wish all audio/video editing/playback programs were written by madshi.
Thanks, guys... :)

Whenever I try to mux a video track to an mkv (mostly when the Blu-ray contains multiple .m2ts files), the muxing process stalls very early on yet eac3to still sits there as if its going. The resulting .mkv is always 2,752,512 bytes.
I'm not sure what is going on. It seems to work for everyone else. I'd suggest that you uninstall and reinstall the Haali Matroska Muxer. Maybe that fixes the problem?

I am also having this same problem, although I have ruled out both pcm2tsmu and TsMuxer as the cause.
How did you rule that out?

I have an EAC3 audio track which was ripped from an HDDVD. I have tried converting it to PCM with the default Nero, and also forcing libav. In both cases, the output file is just static noise (tested with GraphStudio).
Does GraphStudio support raw PCM files in the first place? If you want to check whether eac3to is doing everything correctly or not you should ask eac3to to convert the PCM file to WAV (if it's smaller than 2GB) or FLAC (if it's bigger than 2GB) and then try to play that.

Anyway, as you can see, the static problem occurs during the eac3to step.
"As you can see"? Sorry, I don't see that yet. Where is the proof?

Here's 15 seconds from the beginning of 300, it should suffice to help find the problem if it is eac3to.
Thanks for the sample. I've done this:

eac3to feature.m2ts 2: test.pcm
eac3to test.pcm test.wav

And the resulting WAV file plays perfectly fine. So as far as I can say, eac3to is doing everything right.

tebasuna51

21st September 2008, 10:23

Originally Posted by Jeff Flowerday
Here's 15 seconds from the beginning of 300, it should suffice to help find the problem if it is eac3to.
...
And the resulting WAV file plays perfectly fine. So as far as I can say, eac3to is doing everything right.

eac3to is doing everything right, problem solved (http://forum.doom9.org/showthread.php?p=1186545#post1186545).

Thanks madshi.

madshi

21st September 2008, 14:32

eac3to v2.64 released

http://madshi.net/eac3to.zip

* added channel mask reading support to Blu-Ray PCM track parser
* added channel mask reading support to TrueHD parser
* added channel mask reading & writing support to FLAC decoder / encoder
* changed 5.x channel mask from $03x to $60x
* changed 6.x channel mask from $13x to $70x
* mono wavs output now creates correct names for funny channel masks
* when transcoding 6.1 sources to PCM, 7th channel doubling is activated now
* fixed: DTS channelmask detection was incorrect for some strange configs
* fixed: sometimes the h264 video stream of a Blu-Ray m2ts was not detected

piratburner

21st September 2008, 14:52

I'm Intresting to convert TrueHD and DTS-MA to PCM

If using eac3to input.thd output.thd
Is that the correct way to get a prober MultiChanel PCM ? or do I have missed something in the command line ?

I dont want to get a bad track,

Doing this do I got a better soundtrack or should I stay with the original ones ?

So what would the best command be to convert to PCM ?

Have a nice day ALL

Results
[a03] Extracting audio track number 3...
[a03] Extracting TrueHD stream...
[a03] Removing TrueHD dialog normalization...
[a03] Decoding with libav/ffmpeg...
[a03] Swapping endian...
[a03] Creating file "l:\Soundoutput.24bit.pcm"...
[a03] This audio track contains more than 16 bits of information.

Is it not possible to get a 24bit PCM

Is a lot of questions and I hope some one can give me a short Answer :-)

EDIT
One example.... look at the command, to be a good output do I need more commands ?

L:\BT\bd-tg\BDMV\STREAM>eac3to 1)
M2TS, 1 video track, 5 audio tracks, 8 subtitle tracks, 2:57:09
1: Chapters, 23 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
4: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
5: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
7: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
8: Subtitle (PGS), English
9: Subtitle (PGS), English
10: Subtitle (PGS), French
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), Portuguese
13: Subtitle (PGS), English
14: Subtitle (PGS), French
15: Subtitle (PGS), Spanish

L:\BT\bd-tg\BDMV\STREAM>eac3to 1) 3: output.pcm

It gives me this info ... Is this right ??

[a03] Extracting audio track number 3...
[a03] Extracting TrueHD stream...
[a03] Removing TrueHD dialog normalization...
[a03] Decoding with libav/ffmpeg...
[a03] Swapping endian...
[a03] Creating file "l:\Soundoutput.24bit.pcm"...
[a03] This audio track contains more than 16 bits of information.

shambles

21st September 2008, 15:56

eac3to v2.64 released

thank you :D

lexor

21st September 2008, 16:05

Lexor,
i don't know even any expensive remuxer which is able to remux HD audio formats with h264 or vc1 videos; do you?

I know of Manzanita MP2TSME (http://lilapple.com/mp2tsme.html) (there are also MS/MM versions of it, they are useless for us without h264 support), but it must cost a metric shit-ton, since I can't even find a price for it. It does offer a full featured demo on their site, so you can try everything, but output file will be limited to 10 sec.

And a request for eac3to, if I may. Can we get something like .ini file, so we can specify a few paths to nero aac encoder and mkvtoolnix? First one I have to move into the folder every time I update eac3to, and I don't use installer for the second, so it's "unavailable".

rica

21st September 2008, 16:41

I know of Manzanita MP2TSME (http://lilapple.com/mp2tsme.html) (there are also MS/MM versions of it, they are useless for us without h264 support), but it must cost a metric shit-ton, since I can't even find a price for it. It does offer a full featured demo on their site, so you can try everything, but output file will be limited to 10 sec.

Thanks, but VC1 support is still missing?

madshi

21st September 2008, 16:58

One example.... look at the command, to be a good output do I need more commands ?

L:\BT\bd-tg\BDMV\STREAM>eac3to 1) 3: output.pcm

It gives me this info ... Is this right ??
Yes, that's just right if you want to get PCM. However, PCM in itself is not very useful. You may want to convert to WAV, FLAC, AC3, AAC or DTS instead. Or, if you plan to mux the PCM track to m2ts again, you may want to check out Pcm2Tsmu + tsMuxeR.

Can we get something like .ini file, so we can specify a few paths to nero aac encoder and mkvtoolnix? First one I have to move into the folder every time I update eac3to, and I don't use installer for the second, so it's "unavailable".
Sorry, but I don't want to add an ini file right now. IIRC you can copy the NeroAAC exe to the system32 folder. And mkvtoolnix is not used by eac3to at all. eac3to just checks whether there's a new version available, that's it. So it's not a big loss if eac3to doesn't find mkvtoolnix.

tebasuna51

21st September 2008, 19:58

eac3to v2.64 released

* changed 5.x channel mask from $03x to $60x

I think is a bad idea. We will see in the future ...

* changed 6.x channel mask from $13x to $70x

I think is a wrong idea. At least you need remap the output from ArcSoft decoder with DTS-ES discrete. I think the DTS Hi-Res, 6.1 channels also need the remap.

* when transcoding 6.1 sources to PCM, 7th channel doubling is activated now

If is automatically activated you need recover the
-no7doubling parameter.
or let the doubling optional using the active parameter
-double7

Nullity

21st September 2008, 20:13

How did you rule that out?

I as able to rule them out because I never got to those steps. I only ever used eac3to. However, it doesn't really matter because...

Does GraphStudio support raw PCM files in the first place? If you want to check whether eac3to is doing everything correctly or not you should ask eac3to to convert the PCM file to WAV (if it's smaller than 2GB) or FLAC (if it's bigger than 2GB) and then try to play that.

I did this test as you suggested by converting the PCM file to FLAC, and it played fine. So, I ran it through pcm2tsmu and then muxed it with the video stream in TsMuxer to m2ts. As I suspected, playing it back on my PC results in just static for audio. However, I tested playing it on my PS3 anyway, and it worked fine!

So it seems that you were correct about GraphStudio, I was not able to get raw PCM to play back correctly on my PC, but the PS3 can handle it fine.

I apologize for the confusion though, I wasn't exactly blaming eac3to. The main point of my previous post was just to show that the problem was not with pcm2tsmu or tsmuxer. But as it turns out, it wasn't eac3to either, just my PC.

I also want to thank you for the development of eac3to, it is definitely the best software of its kind!

Lastly, does anyone know how I can play back raw PCM files on my PC? If anyone wishes to respond, you're welcome to PM me, as I know this subject isn't on topic. Thanks!

madshi

21st September 2008, 20:44

I think is a bad idea. We will see in the future ...
Any other suggestions? Do you agree that $603 is the proper value? As you can see here:

http://msdn.microsoft.com/en-us/library/aa474707.aspx

Microsoft wants 5.1 by default to map to "FL, FR, FC, LFE, SL, SR". And IMHO that's the correct mapping. Do you disagree?

I guess that there might be some trouble. Maybe some applications will complain because they expect $3f. But should I output a wrong mask just because some apps expect a wrong value? I don't like to behave like that...

I think is a wrong idea. At least you need remap the output from ArcSoft decoder with DTS-ES discrete.
Yes, you're right about the remapping, forgot about that. But why do you think that is the wrong idea? For people who do have a 6.1 speaker setup, there's one rear speaker and two side speakers. So using $13f is clearly wrong. Or do you disagree?

If is automatically activated you need recover the -no7doubling parameter. or let the doubling optional using the active parameter -double7
The idea of the "pcm" output is to produce a Blu-Ray compatible PCM stream, similar to what e.g. xport demuxes. But the Blu-Ray spec clearly doesn't allow a 6.1 channel configuration for PCM. So what sense does it make to offer a 6.1 PCM output option?

xkodi

21st September 2008, 20:46

eac3to v2.64 released
* changed 5.x channel mask from $03x to $60x
I think is a bad idea. We will see in the future ...

look at the Microsoft document "Audio Driver Support for Home Theater Speaker Configurations":

http://xkodi.svobodno.com/SpkrConfig5.rar

currently, the only valid channel mask for 5.1 channels is 0x3f.

madshi

21st September 2008, 21:10

look at the Microsoft document "Audio Driver Support for Home Theater Speaker Configurations":

http://xkodi.svobodno.com/SpkrConfig5.rar

currently, the only valid channel mask for 5.1 channels is 0x3f.
Thanks, that a good document. However, I'm not sure if I interpret it to say that 0x3f is the only valid channel mask. The document contains this paragraph:

"In Windows XP with SP2 and Windows Server 2003 with SP1, if the system mixer produces a 5.1-channel output stream, the mixer always sets the stream's channel mask to 0x3F. The system mixer behaves this way even if it receives a 5.1-channel input stream with a channel mask of 0x60F."

So clearly the system mixer supports receiving a 0x60f channel mask.

Fortunately the document contains a Microsoft email address for questions about this topic. I'll write to that address and ask for clarification - also about 6.1.

lexor

21st September 2008, 21:17

The idea of the "pcm" output is to produce a Blu-Ray compatible PCM stream, similar to what e.g. xport demuxes. But the Blu-Ray spec clearly doesn't allow a 6.1 channel configuration for PCM. So what sense does it make to offer a 6.1 PCM output option?
For re-encoding to dts/ac3? But then I think I'd rather have 5.1 option, since people who don't have receivers to accept a lossless 7.1 probably also don't have 7.1 or 6.1 speakers.

xkodi

21st September 2008, 21:21

However, I'm not sure if I interpret it to say that 0x3f is the only valid channel mask.

Microsoft's explanation:

According to the bit definitions in Figure 3, the channel mask for recording the 5.1 stream shown on the left side of Figure 5 should be 0x60F, which assigns the six channels to the following speaker positions: FL, FR, FC, LFE, SL, and SR. (This is the side-speaker 5.1 configuration discussed earlier.) In fact, the channel mask for the 5.1 stream is 0x3F rather than 0x60F for reasons that were mentioned previously and will now be explained in detail.
In earlier versions of Windows (Windows 98/Me, Windows 2000, Windows XP with SP1, and Windows Server 2003), the interpretation of the channel mask 0x3F is that it assigns the six channels in the 5.1 format to the following speaker positions: FL, FR, FC, LFE, BL, and BR. (This is the back-speaker 5.1 configuration.) However, the interpretation in Windows XP with SP2, Windows Server 2003 with SP1, and Windows Vista is different: by convention, the 5.1 format with the channel mask 0x3F is interpreted to mean the side-speaker 5.1 configuration instead of the back-speaker 5.1 configuration.
Interpreting the channel mask in this manner eliminates the requirement to introduce a second 5.1-channel format descriptor to distinguish the side-speaker 5.1 configuration from the back-speaker 5.1 configuration. These two configurations are so similar that typical users might have difficulty distinguishing between them. Although having only a single 5.1-channel format descriptor avoids confusing users, it does require hardware vendors to remember to interpret the 0x3F channel mask to mean that channels 5 and 6 are assigned to the SL and SR speaker positions instead of the BL and BR positions. In return for having to remember this special-case interpretation of the channel mask for a 5.1 stream, vendors can spare users the difficulty of distinguishing between two very similar 5.1-channel format descriptors.

that give me the impression that the mask is always 0x3f.

madshi

21st September 2008, 21:35

Microsoft's explanation:

Interpreting the channel mask in this manner eliminates the requirement to introduce a second 5.1-channel format descriptor
that give me the impression that the mask is always 0x3f.
"eliminates the requirement to" doesn't equal "disallows the option to". But let's wait what Microsoft says. I've asked them a number of funny questions... :)

For re-encoding to dts/ac3?
Why would you ask eac3to to export to PCM if you want to reencode to DTS/AC3? That doesn't make much sense to me. If you want to reencode to DTS/AC3, you should either ask eac3to to do that directly for you. Or alternatively you could ask eac3to to output to WAV or AGM and then use an external AC3/DTS encoder. I'm not aware of any DTS/AC3 encoder which would accept a Blu-Ray style PCM file with the Blu-Ray channel ordering and handle that correctly.

lexor

21st September 2008, 21:42

Why would you ask eac3to to export to PCM if you want to reencode to DTS/AC3? That doesn't make much sense to me. If you want to reencode to DTS/AC3, you should either ask eac3to to do that directly for you. Or alternatively you could ask eac3to to output to WAV or AGM and then use an external AC3/DTS encoder. I'm not aware of any DTS/AC3 encoder which would accept a Blu-Ray style PCM file with the Blu-Ray channel ordering and handle that correctly.
Sorry, I got lost in what we were talking about. I was thinking wav all the time.

DeepBeepMeep

21st September 2008, 22:25

It seems that last version of EAC3TO has broken DTS encoding with surcode. I get the following:

C:\EAC3TO>eac3to "f:\dir" 1) 2: d:\file.mkv 4: d:\file.dts 5: d:\file.ac3 -keepdialnorm

...
[a04] Extracting audio track number 4...
[v02] Extracting video track number 2...
[a05] Extracting audio track number 5...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Writing WAVs...
[v02] Muxing video to Matroska...
[a04] Creating file "d:\file.L.wav"...
[a04] Creating file "d:\file.R.wav"...
[a04] Creating file "d:\file.LFE.wav"...
[a04] Creating file "d:\file.BR.wav"...
[a04] Creating file "d:\file.BL.wav"...
[a04] Creating file "d:\file.C.wav"...
[a05] Creating file "d:\file.ac3"...
[libav] End of stream indicated
[a04] This audio track has a constant bit depth of 16 bits.
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.23.0.
Surcode says/asks: "Invalid Wave File d:\file.SL.wav.".
Surcode says/asks: "Invalid Wave File d:\file.SR.wav.".
Surcode says/asks: "Source file sample rates differ from encoding sample rate.
Cannot encode.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

Fer

21st September 2008, 22:41

Great job madshi!

Where is on the to-do list the cross conversion between 23.976 and 24.000?

:thanks::thanks::thanks:

tebasuna51

21st September 2008, 23:24

Any other suggestions? Do you agree that $603 is the proper value? As you can see here:

http://msdn.microsoft.com/en-us/library/aa474707.aspx

Microsoft wants 5.1 by default to map to "FL, FR, FC, LFE, SL, SR". And IMHO that's the correct mapping. Do you disagree?
In the MS page I only see the 0x3F Channel Mask before and after XP SP2. Never recommend 0x603, I only see: when receive a 0x3F channel mask, and we have a system with Side and Back channels, send the output to "FL, FR, FC, LFE, SL, SR"

I guess that there might be some trouble. Maybe some applications will complain because they expect $3f. But should I output a wrong mask just because some apps expect a wrong value? I don't like to behave like that...
For what is wrong $3f? Must be accepted by old and new soft, then is the best value to use without problems.
The Surround channels (110-120º) aren't the Side (90º) or Back (150º) channels, also $603 is wrong.

Yes, you're right about the remapping, forgot about that. But why do you think that is the wrong idea? For people who do have a 6.1 speaker setup, there's one rear speaker and two side speakers. So using $13f is clearly wrong. Or do you disagree?
We can think than a 6.1 is 3 front speakers and 3 back speakers
FL-FR-FC (LF) BL-BR-BC. Isn't a name question is the channel order here

The idea of the "pcm" output is to produce a Blu-Ray compatible PCM stream, similar to what e.g. xport demuxes. But the Blu-Ray spec clearly doesn't allow a 6.1 channel configuration for PCM. So what sense does it make to offer a 6.1 PCM output option?

Only by coherence. Please, only recover the -no7doubling parameter.

rica

21st September 2008, 23:32

Hi guys, thanks for all those technical discussions since they will make SW better but as far as i remember somebody suggested some time ago opening a new thread on developing eac3to?

rica

22nd September 2008, 01:58

Is anybody who knows making it in a more practical way ? :eek:

F:\>eac3to\eac3to.exe 00000.m2ts+00001.m2ts+00002.m2ts+00006.m2ts+00007.m2ts+00008.m2ts+00009.m2ts+00010.m2ts+00011.m2ts+00012.m2ts+00013.m2ts+00014.m2ts+00015.m2ts+00016.m2ts
+00017.m2ts+00018.m2ts+00019.m2ts+00020.m2ts+00021.m2ts+00022.m2ts+00024.m2ts+00025.m2ts+00026.m2ts+00027.m2ts+00028.m2ts+00032.m2ts+00033.m2ts+00034.m2ts+00036.m2ts+00037.m2ts
+00038.m2ts+00039.m2ts+00040.m2ts+00041.m2ts+00042.m2ts+00043.m2ts+00044.m2ts+00045.m2ts+00046.m2ts+00047.m2ts
1: E:\output\video.h264 2: E:\output\audiothd.thd+ac3 3: E:\output\audiodts.dtshd 4: E:\output\subtitle_en.sup 23: E:\output\subtitle_tr.sup

M2TS, 1 video track, 2 audio tracks, 20 subtitle tracks, 2:39:54

1: h264/AVC, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, 5.1 channels, 48khz
3: DTS Master Audio, 5.1 channels, 24 bits, 48khz
4: Subtitle (PGS)
********
23: Subtitle (PGS)
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a03] Extracting audio track number 3...
[s04] Extracting subtitle track number 4...
[s23] Extracting subtitle track number 23...
[v01] Creating file "E:\output\video.h264"...
[a03] Creating file "E:\output\audiodts.dtshd"...
[a02] Creating file "E:\output\audiothd.thd+ac3"...
[s04] Creating file "E:\output\subtitle_en.sup"...
[s23] Creating file "E:\output\subtitle_tr.sup"...
-------------------------------------

rica

22nd September 2008, 02:26

The rest of the log; what should i do now?

Please don't tell me this:

Please rerun the same eac3to command line. That will correct the gaps/overlaps.

[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a03] Extracting audio track number 3...
[s04] Extracting subtitle track number 4...
[s23] Extracting subtitle track number 23...
[v01] Creating file "E:\output\video.h264"...
[a03] Creating file "E:\output\audiodts.dtshd"...
[a02] Creating file "E:\output\audiothd.thd+ac3"...
[s04] Creating file "E:\output\subtitle_en.sup"...
[s23] Creating file "E:\output\subtitle_tr.sup"...
[a02] Audio overlaps for 5ms at playtime 0:46:24.
[a02] Audio overlaps for 5ms at playtime 1:28:23.
[a02] Audio overlaps for 5ms at playtime 2:33:42.
[a02] The audio gaps/overlaps technically can't be removed from the TrueHD bitstream.
[a02] In order to remove them you'll have to transcode the audio to another format.
[a03] Audio overlaps for 12ms at playtime 0:17:55.
[a03] Audio overlaps for 6ms at playtime 0:26:28.
[a03] Audio overlaps for 10ms at playtime 0:28:43.
[a03] Audio overlaps for 6ms at playtime 0:31:42.
[a03] Audio overlaps for 8ms at playtime 0:32:19.
[a03] Audio overlaps for 13ms at playtime 0:37:23.
[a03] Audio overlaps for 10ms at playtime 0:38:37.
[a03] Audio overlaps for 6ms at playtime 0:42:03.
[a03] Audio overlaps for 6ms at playtime 0:45:20.
[a03] Audio overlaps for 7ms at playtime 0:50:38.
[a03] Audio overlaps for 7ms at playtime 0:54:43.
[a03] Audio overlaps for 13ms at playtime 0:58:37.
[a03] Audio overlaps for 11ms at playtime 1:04:12.
[a03] Audio overlaps for 9ms at playtime 1:17:40.
[a03] Audio overlaps for 12ms at playtime 1:26:10.
[a03] Audio overlaps for 7ms at playtime 1:32:06.
[a03] Audio overlaps for 10ms at playtime 1:34:01.
[a03] Audio overlaps for 7ms at playtime 1:35:03.
[a03] Audio overlaps for 6ms at playtime 1:36:00.
[a03] Audio overlaps for 6ms at playtime 1:40:20.
[a03] Audio overlaps for 10ms at playtime 1:43:08.
[a03] Audio overlaps for 9ms at playtime 1:55:53.
[a03] Audio overlaps for 7ms at playtime 1:56:38.
[a03] Audio overlaps for 9ms at playtime 1:57:12.
[a03] Audio overlaps for 10ms at playtime 2:33:42.
[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 1 contains 230070 frames.
eac3to processing took 3 hours, 13 minutes.
Done.

And this is what i had at the end of the process:

http://img218.imageshack.us/img218/9352/000jf9.th.png (http://img218.imageshack.us/my.php?image=000jf9.png)http://img218.imageshack.us/images/thpix.gif (http://g.imageshack.us/thpix.php)

_ _ _ __

Duppie

22nd September 2008, 06:36

Here are the results of my tests conducted over the weekend as well as the solution to my post.

http://forum.doom9.org/showpost.php?p=1185710&postcount=6251

The source I tested was BR "I am Legend" with a TrueHD track of 16bit. When I used eac3to 2.57 I also used the Pcm2Tsmu version 1.2 and the command for Pcm2Tsmu was input.pcm output.pcm with no parameters. This worked eventhough the default for Pcm2Tsmu in bit depth is 24 bits. With my tests conducted with eac3to version 2.63 the above did not work until I added the parameter -i 16 for Pcm2Tsmu. So to conclude. Eac3to and Pcm2Tsmu are not at fault and to correctly produce PCM files always use parameters for Pcm2Tsmu.
I see this was also mentioned by tebasuna51 over the weekend.

Hope it helps and clear this issue

Duppie

nautilus7

22nd September 2008, 06:38

@ rica

2 hints:

* proper way of doing these blu-ray discs is eac3to "folder" 1)
* if you press the upper arrow key the last command is shown again

madshi

22nd September 2008, 07:14

In the MS page I only see the 0x3F Channel Mask before and after XP SP2. Never recommend 0x603, I only see: when receive a 0x3F channel mask, and we have a system with Side and Back channels, send the output to "FL, FR, FC, LFE, SL, SR"

For what is wrong $3f? Must be accepted by old and new soft, then is the best value to use without problems.
The Surround channels (110-120º) aren't the Side (90º) or Back (150º) channels, also $603 is wrong.
I'll wait what MS replies about 0x3f vs. 0x60f.

But have you read the document posted by xdoki? The graphics in that document clearly show that Microsoft wants the side speakers to be used for 5.1. So $60f would definitely be the correct value, cause $600 references the side speakers while $30 references BL/BR.

We can think than a 6.1 is 3 front speakers and 3 back speakers FL-FR-FC (LF) BL-BR-BC. Isn't a name question is the channel order here
Have you ever seen a real 6.1 speaker setup? Nobody in the world sets up 3 rear speakers and no side speakers. So BL-BR-BC is as wrong as can be.

Also: What about 5.0? Microsoft has special cased 0x3f. Ok, so for 5.1 we could use 0x3f. But MS has *not* special cased 0x37! So 0x37 still references the back channels, which is wrong. So using 0x37 would be wrong for 5.0. We have to use 0x607 to reference the side speakers (which is what MS wants). What about 2/2.0 and 2/2.1? Again 0x33 and 0x3b are very clearly wrong and we must use 0x603 and 0x60b instead.

Only by coherence. Please, only recover the -no7doubling parameter.
Then tell me into which channel order 6.1 PCM has. 5.1 PCM and 7.1 PCM have a very specific channel order (which is used on Blu-Ray). When outputting 5.1 PCM and 7.1 PCM eac3to remaps the channels into this specific Blu-Ray channel order. But Blu-Ray doesn't know 6.1 PCM. So what channel order should I be using for 6.1 PCM?

I think this question shows that outputting 6.1 PCM does not really make sense. Coherence could only be achieved if there was a specific 6.1 PCM channel mapping known. But it isn't known. Of course I could simply use the WAV channel ordering. But that wouldn't be coherent to how 5.1 and 7.1 PCM export is done, either.

Thoughts?

madshi

22nd September 2008, 07:29

In the MS page I only see the 0x3F Channel Mask before and after XP SP2. Never recommend 0x603, I only see: when receive a 0x3F channel mask, and we have a system with Side and Back channels, send the output to "FL, FR, FC, LFE, SL, SR"

For what is wrong $3f? Must be accepted by old and new soft, then is the best value to use without problems.
The Surround channels (110-120º) aren't the Side (90º) or Back (150º) channels, also $603 is wrong.
Just got a reply from Microsoft. Here's what they're saying:

As of XPSP2, the recommended 5.1 and 7.1 channel masks include the side speakers rather than the back speakers.

I don't know what the 6.1 situation is.

See ksmedia.h in the SDK - you want the _SURROUND masks.

#define KSAUDIO_SPEAKER_5POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
#define KSAUDIO_SPEAKER_7POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | \
SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER)

#if ( (NTDDI_VERSION >= NTDDI_WINXPSP2) && (NTDDI_VERSION < NTDDI_WS03) ) || (NTDDI_VERSION >= NTDDI_WS03SP1)

#define KSAUDIO_SPEAKER_5POINT1_SURROUND (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT)
#define KSAUDIO_SPEAKER_7POINT1_SURROUND (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | \
SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT)
// The following are obsolete 5.1 and 7.1 settings (they lack side speakers). Note this means
// that the default 5.1 and 7.1 settings (KSAUDIO_SPEAKER_5POINT1 and KSAUDIO_SPEAKER_7POINT1 are
// similarly obsolete but are unchanged for compatibility reasons).
#define KSAUDIO_SPEAKER_5POINT1_BACK KSAUDIO_SPEAKER_5POINT1
#define KSAUDIO_SPEAKER_7POINT1_WIDE KSAUDIO_SPEAKER_7POINT1

#endif // XP SP2 and later (chronologically)
If I follow what MS just replied then I end up with 0x60f and not with 0x3f (!). I've asked for further clarification, as this seems to slightly contradict the doc that xkodi uploaded. Specifically the doc says that MS wanted to avoid having two different channel masks for 5.1. But that is exactly what ksmedia.h now contains. KSAUDIO_SPEAKER_5POINT (0x3f) is considered obsolete. And KSAUDIO_SPEAKER_5POINT1_SURROUND (0x60f) is considered the right mask to use, according to the latest ksmedia.h.

It seems that last version of EAC3TO has broken DTS encoding with surcode. I get the following:

C:\EAC3TO>eac3to "f:\dir" 1) 2: d:\file.mkv 4: d:\file.dts 5: d:\file.ac3 -keepdialnorm

Surcode says/asks: "Invalid Wave File d:\file.SL.wav.".
Surcode says/asks: "Invalid Wave File d:\file.SR.wav.".

Now that is strange. I'll look into this.

Where is on the to-do list the cross conversion between 23.976 and 24.000?
The to-do list is not sorted in any way... :)

EPiPH0NE

22nd September 2008, 07:36

Nobody in the world sets up 3 rear speakers and no side speakers.

Not true. My brother-in-law has his set up like this cause the way the room is set up there is no space for sides, just rears. Mine is similar but I have room for sides and actually pre-wired for them but my room is 5.1 set up backwards from previous tenant so I just use have speaker stands all the way at each rear end of my couch with the speakers tilted toward the center of the room.:

^ *^

/ \

madshi

22nd September 2008, 07:52

Not true. My brother-in-law has his set up like this cause the way the room is set up there is no space for sides, just rears. Mine is similar but I have room for sides and actually pre-wired for them but my room is 5.1 set up backwards from previous tenant so I just use have speaker stands all the way at each rear end of my couch with the speakers tilted toward the center of the room.
Well, ok. So nobody who has the space for side/surround speakers and knows what he's doing sets up 3 rear speakers and no side/surround speakers.
:)

K1ngp1ng

22nd September 2008, 09:18

dolby recommends 90°-110° for both 5.1 and 7.1, while ITU rec. is 110°.
also, when upmixing 5.1 to 6.1 or 7.1 (with pl2x), back srds on the stereobasis between the srds are beeing extracted,
but that wouldn't move the srds from 110° to the side.
in practice there is no difference between srd and side srd. that beeing said, i don't know about this channel masking thing.

concerning 6.1 to flac:
could you please!! add this -double7 feature to madflac.
flac is not that efficient and doing this on the fly seems to be a much more elegant way instead of wasting space.
also, this way the soundtrack could be stored bit-perfect without alteration.

tebasuna51

22nd September 2008, 11:29

...
could you please!! add this -double7 feature to madflac.
flac is not that efficient and doing this on the fly seems to be a much more elegant way instead of wasting space.
also, this way the soundtrack could be stored bit-perfect without alteration.

I think this is the right way.
The sounds must be adapted to the speakers system at play time, never before encode.

madshi

22nd September 2008, 12:07

I think this is the right way.
The sounds must be adapted to the speakers system at play time, never before encode.
Agreed. But that only works if the storage format supports the channel configuration you're looking for... :) If it doesn't, you either have to use a different storage format or convert the channels.

Anyway, madFlac should (and will) be updated. Sooner or later...

tebasuna51

22nd September 2008, 12:24

...
What about 2/2.0 and 2/2.1? Again 0x33 and 0x3b are very clearly wrong and we must use 0x603 and 0x60b instead.

I can't believe. The quadraphonic sound (http://en.wikipedia.org/wiki/Quadraphonic) was intended to use the four room corners (45º, 135º, ...), clearly the Back channels.

This is the origin of Back channels, and not the absurd BC (180º) from 6.1 or the BL-BR together (179º-181º) like some DTS configs recommend for 7.1. I don't have ears in the nape.

I'm not interested in 6.1 or 7.1 systems. I think a 5.1 system is enough to represent the surround sound (also exist 3D sounds for headphones!) in standard living rooms. More sophisticated systems are only for audiophiles with dedicated rooms or a way to spend money.

This is your tool and you decide. I only warn about possible problems.

And thanks for your job. :)

madshi

22nd September 2008, 13:51

I can't believe. The quadraphonic sound (http://en.wikipedia.org/wiki/Quadraphonic) was intended to use the four room corners (45º, 135º, ...), clearly the Back channels.
But then why do both DTS and AC3 define 2/2 as being intended for the side/surround speakers? Ok, AC3 only knows 5.1, so that's not fair. But DTS knows both surround speakers and back/rear speakers - and still DTS defines 2/2 as being intended for the surround speakers.

I've asked Microsoft about 2/2 channel mask, but the person who replied to me couldn't even give me an advice which 6.1 channel mask to use, let alone 2/2.

I'm not interested in 6.1 or 7.1 systems. I think a 5.1 system is enough to represent the surround sound (also exist 3D sounds for headphones!) in standard living rooms.
You can read in xkodi's doc that the MS system mixer simply drops the back channels when playing a 7.1 source on a 5.1 speaker setup. So in other words it seems that the MS system mixer drops any channels for which it doesn't find a directly matching speaker. That means, if I use the channel mask 0x13f for a 6.1 source with a 5.1 speaker setup, it's quite possible that the system mixer will drop all 3 back channels and leave the surround channels empty. That's one of the reasons why I prefer to use 0x70f. Besides, it's just more correct, too, IMHO.

This is your tool and you decide. I only warn about possible problems.
Yes, and I appreciate that. And I'm willing to continue discussing. I like discussions. I like people not agreeing with me and defending their opinion. Because sometimes only such discussions can end up in a better end result. But I need to be convinced to change my opinion.

madshi

22nd September 2008, 17:26

Another reply from Microsoft:

I notice the white paper [1] also doesn't mention the new _5POINT1_SURROUND constant at all; and yet the constant now exists. Perhaps it was introduced after the white paper was authored. 0x60F should be correct, and 0x3f is only for backcompat.

For 6.1 I agree that 0x70F is correct although 0x13f might be grandfathered in.

What is 2/2? Is it similar to _QUAD?

#define KSAUDIO_SPEAKER_QUAD (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
So the document posted by xdoki seems to be outdated. 0x60f is the correct mask to use for 5.1 and 0x70f is the correct mask to use for 6.1. That's according to Microsoft.

Still open for discussion seems to be 2/2. Microsoft evidently has a KSAUDIO_SPEAKER_QUAD constant which uses the back channels instead of the surround channels. BUT Microsoft does not have a comparable constant for 2/2.1. Also the DTS spec uses the surround channels instead of the back channels. I guess that might have to do with DTS being more of a home theater format and less of a music format? I mean 2/2 can be seen as Quadraphonic. But it can also be seen as 5.0 with the center channel missing. DTS seems to prefer the latter interpretation.

So which channel mask should eac3to use for 2/2 and 2/2.1 formats? Should it reference the surround channels or the back channels?

Comments?

Inventive Software

22nd September 2008, 18:00

Consider that Quadrophonic 4 or 4.1 was around before 5.1 really took off, so on that basis I think it'd be the back channels, but realistically I'd expect it to change between sources. 4 / 4.1 channels I think would be FL,FR,(LFC),BL,BR.

Atak_Snajpera

22nd September 2008, 18:01

Madshi could you add switch something like "-Automatically repeat audio demuxing if gaps are detected"?

K1ngp1ng

22nd September 2008, 18:31

So which channel mask should eac3to use for 2/2 and 2/2.1 formats? Should it reference the surround channels or the back channels?
there is only one valid multichannel setup which is described in ITU-R BS.775-1.
so 2/2/1 should be treated like 3/2/1 without center.

page 12: http://www.tonmeister.de/foren/surround/texte/SSF_01_1_E_2002_v2a.PDF

fib0by

22nd September 2008, 18:40

Madshi could you add switch something like "-Automatically repeat audio demuxing if gaps are detected"?

I wouldn't mind if that was the default behavior, unless there's something non-obvious which would make that not the best choice.

EPiPH0NE

22nd September 2008, 19:04

Madshi could you add switch something like "-Automatically repeat audio demuxing if gaps are detected"?

Eac3to_and_More_GUI does this but it would be nice to have it built-in to eac3to.

fib0by

22nd September 2008, 19:32

Eac3to_and_More_GUI does this but it would be nice to have it built-in to eac3to.

Actually, I either run it manually, or from a small Bash script that I put together one evening (running under Cygwin on Windows, since I'm too lazy to learn Windows scripting). It provides all I need in terms of demuxing.

When people use your command-line application directly (manually), you know you're doing something right. :) Good job, madshi.

xkodi

22nd September 2008, 20:22

Another reply from Microsoft:

I notice the white paper [1] also doesn't mention the new _5POINT1_SURROUND constant at all;...

i assume that "white paper [1]" is the SpkrConfig5.doc and if so this isn't true at all, because KSAUDIO_SPEAKER_5POINT1_SURROUND is equal to 0x60F and that number is all over the document.

actually, AFAIU the paper states that with WinXP SP2 0x60F is always changed to 0x3F (on the fly from the kernel mixer).

Another reply from Microsoft:

...and yet the constant now exists. Perhaps it was introduced after the white paper was authored. ...

in the "Windows Driver Development Kit" (DDK) there is a sample of simple WDM audio driver and that sample contains "KSAUDIO_SPEAKER_5POINT1_SURROUND", but according to the copyright notice it's from year 2000. the SpkrConfig5.doc is from year 2006.

also, AFAIU SpkrConfig5.doc states that before WinXP SP2, there are 0x3F and 0x60F.

so, i can't see why 0x60F is newer than 0x3F, i.e. defined later than 0x3F.

Another reply from Microsoft:

... 0x60F should be correct, and 0x3f is only for backcompat. So the document posted by xkodi seems to be outdated.

because of the comments i made above i'm not so sure that these statement are correct.

the whole channel mask thing is a huge mess.

and what about if we look to the problem from the WDM audio driver point of view:
what will happen if the WDM audio driver knows about 0x60F, but doesn't know about 0x3f, which is the case with the MS sample driver from 2000, then maybe 0x3f won't work correctly and if the real WDM audio drivers are based on the old sample then that will explain why 0x60f is more better than 0x3f, but it doesn't make it more right.

EPiPH0NE

22nd September 2008, 20:56

Actually, I either run it manually, or from a small Bash script that I put together one evening (running under Cygwin on Windows, since I'm too lazy to learn Windows scripting). It provides all I need in terms of demuxing.

When people use your command-line application directly (manually), you know you're doing something right. :) Good job, madshi.

I run it manually sometimes but usually use the GUI and most people are too lazy to hit ^ -> ENTER so including a repeat switch would be ideal.

rica

22nd September 2008, 21:39

Madshi could you add switch something like "-Automatically repeat audio demuxing if gaps are detected"?

One of the best feature requests i've heard after my last night trial :)

(Btw, TrueHD gaps can not be repaired :eek: )

rica

22nd September 2008, 22:37

Second pass:

Audio gap description file detected, can't be used for TrueHD/MLP, though.
Audio gap description file detected, will be used for processing...
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a03] Extracting audio track number 3...
[a03] Realizing DTS gaps...
[v01] Creating file "E:\output\video.h264"...
[a03] Creating file "E:\output\audiodts.dtshd"...
[a02] Creating file "E:\output\audiothd.thd+ac3"...
[a02] Audio overlaps for 5ms at playtime 0:46:24.
[a02] Audio overlaps for 5ms at playtime 1:28:23.
[a02] Audio overlaps for 5ms at playtime 2:33:42.
[a02] The audio gaps/overlaps technically can't be removed from the TrueHD bitstream.
[a02] In order to remove them you'll have to transcode the audio to another format.
Video track 1 contains 230070 frames.
eac3to processing took 31 minutes, 57 seconds.
Done.

The question here is:
Shall i re-encode to another format after or before merging?

_ _ _ _ _

Henry The Ripper

23rd September 2008, 16:47

@Madshi

The TrueHD to DTS isn't working in the 2.64 version. 2.60 that I have works fine.

The new version names the surround channels as BR/BL and Surcode gets confused cause it doesn't find SR/SL named channels and closes.

himan2001

24th September 2008, 14:05

@madshi:

Can you implement a missing Option? :

-slowdown24 and -speedup24

for 24.000 -> 23.976 and 23.976 -> 24.000 conversions ;-) ?

avivahl

24th September 2008, 15:13

@madshi:

Can you implement a missing Option? :

-slowdown24 and -speedup24

for 24.000 -> 23.976 and 23.976 -> 24.000 conversions ;-) ?
+1. Would be a great feature.

Atak_Snajpera

24th September 2008, 16:50

eac3to reports insane audio delay. DGIndex demuxes audio without problems.

eac3to v2.64
command line: "C:\temp\Ripbot264\Tools\eac3to\eac3to.exe" "C:\VTS_02_1.VOB" 2: "D:\temp\RipBot264temp\job1\audio.1.ac3" -progressnumbers
------------------------------------------------------------------------------
VOB, 1 video track, 5 audio tracks, 9 subtitle tracks, 0:02:16
1: MPEG2, 576i50 (16:9)
2: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -24dB, 60525ms
3: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -28dB, 60525ms
4: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB, 60525ms
5: AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -25dB, 60525ms
6: AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -28dB, 60525ms
7: Subtitle
8: Subtitle
9: Subtitle
10: Subtitle
11: Subtitle
12: Subtitle
13: Subtitle
14: Subtitle
15: Subtitle
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Applying (E-)AC3 delay...
[a02] Creating file "D:\temp\RipBot264temp\job1\audio.1.ac3"...
[a02] The last (E-)AC3 frame is incomplete and thus gets skipped.
Video track 1 contains 1919 frames.
eac3to processing took 1 second.
Done.

source .VOB
http://www.mediafire.com/?zdymif14jyg

saint-francis

24th September 2008, 16:55

Hi guys, thanks for all those technical discussions since they will make SW better but as far as i remember somebody suggested some time ago opening a new thread on developing eac3to?

That was me. I have spent several months working on that and I think it's impossible. I have so far gone through the thread three times and compiled lists of what posts I think should go where, and they are different each time. The development posts and usage questions are inextricably linked. Really I think that we as eac3to users should just focus on making the eac3to wiki comprehensive to alleviate the enormous amount of usage questions that pop up in this thread. Or we could just start a new eac3to usage thread.

saint-francis

24th September 2008, 16:58

Eac3to_and_More_GUI does this but it would be nice to have it built-in to eac3to.

That feature is broken for me in the GUI.

rica

25th September 2008, 00:33

That was me. I have spent several months working on that and I think it's impossible. I have so far gone through the thread three times and compiled lists of what posts I think should go where, and they are different each time. The development posts and usage questions are inextricably linked. Really I think that we as eac3to users should just focus on making the eac3to wiki comprehensive to alleviate the enormous amount of usage questions that pop up in this thread. Or we could just start a new eac3to usage thread.

I do agree.... Lets start a new eac3to usage thread?
I think you have a collected FAQs on eac3to as well?
So i think you are the one who should start this thread.
I suppose nautilus7 will contribute as well.

saint-francis

25th September 2008, 04:40

I do agree.... Lets start a new eac3to usage thread?
I think you have a collected FAQs on eac3to as well?
So i think you are the one who should start this thread.
I suppose nautilus7 will contribute as well.

Then I propose a new thread for usage with an opening post linking to the WIKI and to several posts in this thread that address frequently asked questions. Unfortunately my lists of posts from this thread that I made are all about development so I'm not currently in a position to supply the list of FAQ posts. I suppose I could whip something up though.

killa_kid

25th September 2008, 06:47

i did a quick search of the thread and didn't find anything related to this, sorry if its a dupe suggestion.

When eac3to finds gaps while doing a DTS (and i believe AAC, FLAC) it outputs as PCM and puts up the message about gaps being found before running the external program. I was wondering if you could add an option (maybe something like "-killOnGaps") that will not start that second part.

I have some automation that will run a 2nd pass if a .gaps file exists, except when I walk away my computer still has to spend 30 or so minutes making a DTS file from the first step that was filled with tiny gaps.

I'm not sure exactly how you designed the program, but if you could this would make things a lot easier for me. Still loving the program madshi!

va1aY

25th September 2008, 19:40

Dear Madshi,
thanks for the wonderful program eac3to which you keep improving over and over! I have a question: can this program create a FLAC file from six WAV files? And what would be the command line to do this? Regards, va1aY

rica

25th September 2008, 22:54

deathlord

26th September 2008, 06:13

madshi

I have an lpcm track (Crank blu-ray) where eac3to "sees" 7.1 channels, but only 5.1 of them seem to contain audio data. For me it is no problem, I assume I could specify the number of channels manually. I just thought maybe it could be fixed in eac3to. Tell me if you need a sample.

deathlord

rickardk

26th September 2008, 09:15

How do I get 6th channel doubled? (Creating a 8 channel FLAC from 6.1 DTS-HD Master Audio)

I searched the thread but can't find this one. I remember reading that this feature was disabled by default since a while back.

Kurtnoise

26th September 2008, 09:32

Just to have an info: is it possible that a DTS-MA stream can be stereo ?

edit: reply to myself - yes...

deathlord

26th September 2008, 14:44

yfed

27th September 2008, 02:02

Hi! I wanna report an error decoding DTS-HD MA:

Here are track specs: DTS Master Audio, 1.0 channels, 16 bits, 48khz (From "For A Few Dollars More")

Decoding with ArcSoft DTS Decoder...
The ArcSoft DTS Decoder reported an error while decoding.
Aborted at file position 16384.

Decoding with DirectShow (Sonic Audio Decoder)...
eac3to crashed...
last one crashes all the way at the end of file (seems so with the "-" reaching to the end)

ArcSoft DTS Decoder (1.1.0.0) works fine
Sonic Audio Decoder (4.3.0.169) works fine

saint-francis

27th September 2008, 04:07

Try your process with the -test switch to determine if you are working with a good rip.

yfed

27th September 2008, 04:19

what do you mean?
"-test" just checks if the external filters are installed & working

I cant use "-check" because I no longer have m2ts. But I remember that everything from it demuxed successfully

yfed

27th September 2008, 08:19

Well, I've tried a different program to convert DTS-MA to DTS (TsMuxer). That worked fine. Then I opened *.DTS in Adobe Audition and saved it as *.WAV.
Then did "eac3to input.wav output.ac3 -override -1 -16 -192 -48000" (since I don't really need DTS/high bitrate AC3 for 1965 mono sound)

I think there's something fishy about 1ch(mono) dts-ma. Maybe Arcsoft/Sonic couldn't handle it, although I have no idea what TsMuxer uses :confused:

madshi

27th September 2008, 10:14

The TrueHD to DTS isn't working in the 2.64 version. 2.60 that I have works fine.

The new version names the surround channels as BR/BL and Surcode gets confused cause it doesn't find SR/SL named channels and closes.
Will be fixed in the next build.

Can you implement a missing Option? :

-slowdown24 and -speedup24

for 24.000 -> 23.976 and 23.976 -> 24.000 conversions ;-) ?
It's already on my to do list.

eac3to reports insane audio delay. DGIndex demuxes audio without problems.
Thanks for the sample, I'll have a look at this.

Madshi could you add switch something like "-Automatically repeat audio demuxing if gaps are detected"?
I wouldn't mind if that was the default behavior, unless there's something non-obvious which would make that not the best choice.
When eac3to finds gaps while doing a DTS (and i believe AAC, FLAC) it outputs as PCM and puts up the message about gaps being found before running the external program. I was wondering if you could add an option (maybe something like "-killOnGaps") that will not start that second part.
I'm planning to automate the gap removal for audio in a future build.

can this program create a FLAC file from six WAV files?
Multiple wav files are not supported as input yet. You need to use WaveWizard to combine those separate wav files into one.

I have an lpcm track (Crank blu-ray) where eac3to "sees" 7.1 channels, but only 5.1 of them seem to contain audio data.
A sample would be nice. Are you sure that the back channels are silent throughout the whole movie? I mean they may be silent for most of the movie, but maybe they're used only once in a while?

Hi! I wanna report an error decoding DTS-HD MA:

Here are track specs: DTS Master Audio, 1.0 channels, 16 bits, 48khz (From "For A Few Dollars More")
Can I have a sample of the m2ts file, please? The first 50MB would be great. Maybe 20MB would also do...

Well, I've tried a different program to convert DTS-MA to DTS (TsMuxer). That worked fine.
eac3to can do the same thing. Just use the "-core" switch. That will get rid of the DTS-HD data blocks.

Second pass:

[a02] The audio gaps/overlaps technically can't be removed from the TrueHD bitstream.
[a02] In order to remove them you'll have to transcode the audio to another format.

The question here is:
Shall i re-encode to another format after or before merging?
Ask eac3to to output *.wav or *.flac or *.pcm from the get go.

i assume that "white paper [1]" is the SpkrConfig5.doc and if so this isn't true at all, because KSAUDIO_SPEAKER_5POINT1_SURROUND is equal to 0x60F and that number is all over the document.
Yes, but the constant named "KSAUDIO_SPEAKER_5POINT1_SURROUND" is not mentioned at all, while the other 3 "KSAUDIO_SPEAKER" constants are mentioned. The doc reads as if there was no constant with the name of "KSAUDIO_SPEAKER_5POINT1_SURROUND".

in the "Windows Driver Development Kit" (DDK) there is a sample of simple WDM audio driver and that sample contains "KSAUDIO_SPEAKER_5POINT1_SURROUND", but according to the copyright notice it's from year 2000. the SpkrConfig5.doc is from year 2006.
:confused:

what will happen if the WDM audio driver knows about 0x60F, but doesn't know about 0x3f, which is the case with the MS sample driver from 2000, then maybe 0x3f won't work correctly and if the real WDM audio drivers are based on the old sample then that will explain why 0x60f is more better than 0x3f, but it doesn't make it more right.
It is very clear that all DTS, Dolby and Microsoft want 5.1 to be played through the surround speakers and not through the back speakers. That makes 0x60f clearly more right compared to 0x3f. The only thing speaking for 0x3f is that doc you posted and the fact that most software out there uses 0x3f. It's really a mess, just as you say. I think I will keep using 0x60f, unless we get convincing evidence that it makes problems. From my point of view 0x60f is correct. And MS confirmed that to me via email. Maybe 0x3f is correct, too (but obsolete/outdated), maybe not. Even MS seems to be confused about that...

there is only one valid multichannel setup which is described in ITU-R BS.775-1.
so 2/2/1 should be treated like 3/2/1 without center.

page 12: http://www.tonmeister.de/foren/surround/texte/SSF_01_1_E_2002_v2a.PDF
Thanks! That's a good PDF. To be fair, this PDF doesn't seem to know the default speaker setup for 7.1. It doesn't even know that 6.1 exists at all. Probably the PDF was written before 6.1 and 7.1 were introduced for home cinema. But still the PDF clearly shows that 3/1, 2/2, 2/1 etc are all interpreted as 3/2 with some channels missing. That IMHO supports the new channel masks I'm using in the current eac3to version. Also the PDF clearly states that when duplicating the mono surround channel, volume should be lowered by 3db (or 50%), which is what the eac3to "-double7" option correctly does. So all is fine. The only thing currently wrong is that I still need to remap the channels when using 6.1 output without doubling the channels (due to the new channel mask I'm using).

tebasuna51

27th September 2008, 13:24

It is very clear that all DTS, Dolby and Microsoft want 5.1 to be played through the surround speakers and not through the back speakers.
Of course, for Surround (110º) speakers but don't say Side (90º) speakers, is near than Back (150º) but isn't the same.
That makes 0x60f clearly more right compared to 0x3f. ... From my point of view 0x60f is correct. And MS confirmed that to me via email. Maybe 0x3f is correct, too (but obsolete/outdated), maybe not. Even MS seems to be confused about that...
Not problem for me if you accept 0x3f like valid 5.1 input in your soft. I will accept also 0x60f like valid 5.1 input in my soft :)
The order channel is the same then no problem.

- A 5.1 system must send the last channels to surround speakers without modify.

- A 6.1 system must extract the common part from the last channels to BackCenter speaker. The rest for the Left and Right Side/Surround/Back (you can use any name) speakers.

- A 7.1 system with Side (90º) and Back (150º) must send 0.9 from last channel to SideRight speaker and 0.4 from last channel to BackRight speaker (equivalent for Lefts)

The only thing currently wrong is that I still need to remap the channels when using 6.1 output without doubling the channels (due to the new channel mask I'm using).

Then, when play a 6.1 directly with ArcSoft or Sonic players we have a wrong mapping channels?
If you need remap a supported format maybe you aren't at right way.

I think this can be a big problem, here isn't a name problem, here the channel order is different.

And the flac behaviour? I have a flac Feature Request (http://sourceforge.net/tracker/index.php?func=detail&aid=1949155&group_id=13478&atid=363478) from 2008-04-22 without answer. Do you now any 'official' answer from flac developers?

Atak_Snajpera

27th September 2008, 14:19

I found another problem with .vob as a source. If I let eac3to to use gaps I loose audio synchronization. Audio demuxed without using gaps has perfect synchronization.
Source: http://www.mediafire.com/file/nnnfmnx3hnm/VTS_01_1.VOB

madshi

27th September 2008, 14:36

Of course, for Surround (110º) speakers but don't say Side (90º) speakers, is near than Back (150º) but isn't the same.
Side is much nearer to Surround than Back is... ;)

Not problem for me if you accept 0x3f like valid 5.1 input in your soft.
Of course I do!

Then, when play a 6.1 directly with ArcSoft or Sonic players we have a wrong mapping channels?
ArcSoft/Sonic output a different channel order, but the channel mask they're using (0x13f) fits to that channel order.

If you need remap a supported format maybe you aren't at right way.
So ArcSoft and Sonic must always be right? ;) You don't know how many hacks I had to add to eac3to to make the decoders work correctly. Both ArcSoft and Sonic decoders make a lot of (different) mistakes. Especially with samples with funny channel configurations.

I think this can be a big problem, here isn't a name problem, here the channel order is different.
Yes, I'm aware of that. That's the main reason why eac3to used to double the 7th channel by default! Because with 7.1 there's no confusion. With 6.1 there's room for confusion.

And the flac behaviour? I have a flac Feature Request (http://sourceforge.net/tracker/index.php?func=detail&aid=1949155&group_id=13478&atid=363478) from 2008-04-22 without answer. Do you now any 'official' answer from flac developers?
Check out this thread:

http://www.hydrogenaudio.org/forums/index.php?showtopic=49182

Search for "WAVEFORMATEXTENSIBLE_CHANNEL_MASK". This new tag is already implemented in the current eac3to build! But not yet in madFlac...

madshi

27th September 2008, 14:39

I found another problem with .vob as a source. If I let eac3to to use gaps I loose audio synchronization. Audio demuxed without using gaps has perfect synchronization.
Thanks, will check this out, too.

madshi

27th September 2008, 14:41

Of course, for Surround (110º) speakers but don't say Side (90º) speakers, is near than Back (150º) but isn't the same.
Just double checked this. You are wrong!!

See here:

http://www.audioholics.com/tweaks/speaker-setup-guidelines/home-theater-speaker-layout-an-essential-guide
http://www.smartbuyspeakers.com/speakersetup
http://forum.ecoustics.com/bbs/messages/34579/129023.html

There's no difference between "side" and "surround" speakers. And there's no difference of the placement of side/surround speakers between 5.1, 6.1 and 7.1. They are always at 90°-110°.

tebasuna51

27th September 2008, 16:48

Just double checked this. You are wrong!!

See here:

http://www.audioholics.com/tweaks/speaker-setup-guidelines/home-theater-speaker-layout-an-essential-guide
http://www.smartbuyspeakers.com/speakersetup
http://forum.ecoustics.com/bbs/messages/34579/129023.html

There's no difference between "side" and "surround" speakers. And there's no difference of the placement of side/surround speakers between 5.1, 6.1 and 7.1. They are always at 90°-110°.

Why I wrong? Using your samples:
My surround speakers in room with 5.1 are at 110º, correct?
My side speakers in room with 7.1 are at 90º and the back at 135º, correct?
Are typical locations. With less speakers we need cover more space. And now the difference between surround-side is 20º and surround-back is 25º.

We need be a little more flexible, I say you surround in near to side than back, but you never can convince me that is the same.:cool:

madshi

27th September 2008, 17:08

My surround speakers in room with 5.1 are at 110º, correct?
Anywhere between 90° and 110° is correct for the surround=side speakers in a 5.1, 6.1 or 7.1 setup.

My side speakers in room with 7.1 are at 90º and the back at 135º, correct?
Anywhere between 90° and 110° is correct for the surround=side speakers in a 7.1, 6.1 or 7.1 setup.

--------------------------------

The point is: There's no difference in recommended surround/side speaker placement between 5.1, 6.1 and 7.1. That is what Dolby says. That is what the first 10 hits about speaker placement at google say. So for all intends and purposes we can say "surround = side". If you don't agree with that then you seem to be alone with your opinion, at least according to what a google search told me...

Do you have a link to an article (from a well respected source) which claims that surround speakers should be placed differently compared to side speakers?

Jeff Flowerday

27th September 2008, 18:59

Just got this message converting "There Will Be Blood" TrueHD -> pcm

[libav] End of stream indicated
[libav] Lossless check failed - expected 0, calculated 21
[a02] This audio track has a constant bit depth of 24 bits.
Video track 1 contains 227909 frames.
eac3to processing took 18 minutes, 19 seconds.
Done.

Just wondering what it means and what effect does it have on the end result?

killa_kid

27th September 2008, 20:44

fib0by

27th September 2008, 21:08

It looks like chapter track detection may not work properly in some cases.

There are 4 titles on this disk. Titles 1 and 2 are the movie. Titles 3 and 4 are the movie with "video commentary" (images from production rendered on top of the movie, pretty silly). The length of the video is the same for all titles - after all, it's the same movie.

eac3to seems to detect the chapters list just fine for titles 3 and 4, but not for 1 and 2. I can't quite believe that there are no chapters for 1 and 2 since that is the actual movie that most people would watch.

Even if I run eac3to directly on the .m2ts files, it's the same - it can't find any chapters for 00008.m2ts but it does find them for 00011.m2ts

In this particular case, I can probably extract the chapters from title 3 and use them with the video from title 1, but if 1 actually has chapters but they are not detected, it's probably a bug that needs to be squished.

E:\work>eac3to E:\hd\blah
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) 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

3) 00013.mpls, 00011.m2ts, 1:53:18
- VC-1, 1080p24 /1.001 (16:9)
- DTS, English, multi-channel, 48khz

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

E:\work>eac3to E:\hd\blah 1)
M2TS, 1 video track, 2 audio tracks, 2 subtitle tracks, 1:53:18
1: VC-1, 1080p24 /1.001 (16:9)
2: DTS Master Audio, English, 7.1 channels, 24 bits, 48khz
3: DTS, English, 2t channels, 24 bits, 256kbps, 48khz
4: Subtitle (PGS), English
5: Subtitle (PGS), Spanish

E:\work>eac3to E:\hd\blah 2)
M2TS, 1 video track, 2 audio tracks, 2 subtitle tracks, 1:53:18
1: VC-1, 1080p24 /1.001 (16:9)
2: DTS Master Audio, English, 7.1 channels, 24 bits, 48khz
3: DTS, English, 2t channels, 24 bits, 256kbps, 48khz
4: Subtitle (PGS), English
5: Subtitle (PGS), Spanish

E:\work>eac3to E:\hd\blah 3)
M2TS, 1 video track, 1 audio track, 2 subtitle tracks, 1:53:18
1: Chapters, 21 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS, English, 5.1 channels, 24 bits, 1536kbps, 48khz
4: Subtitle (PGS), English
5: Subtitle (PGS), Spanish

E:\work>eac3to E:\hd\blah 4)
M2TS, 1 video track, 1 audio track, 2 subtitle tracks, 1:53:18
1: Chapters, 21 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS, English, 5.1 channels, 24 bits, 1536kbps, 48khz
4: Subtitle (PGS), English
5: Subtitle (PGS), Spanish

E:\work>eac3to E:\hd\blah\BDMV\STREAM\00008.m2ts
M2TS, 1 video track, 2 audio tracks, 2 subtitle tracks, 1:53:18
1: VC-1, 1080p24 /1.001 (16:9)
2: DTS Master Audio, English, 7.1 channels, 24 bits, 48khz
3: DTS, English, 2t channels, 24 bits, 256kbps, 48khz
4: Subtitle (PGS), English
5: Subtitle (PGS), Spanish

E:\work>eac3to E:\hd\blah\BDMV\STREAM\00011.m2ts
M2TS, 1 video track, 1 audio track, 2 subtitle tracks, 1:53:18
1: Chapters, 21 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS, English, 5.1 channels, 24 bits, 1536kbps, 48khz
4: Subtitle (PGS), English
5: Subtitle (PGS), Spanish

madshi, what do you want me to do to help you troubleshoot it? Upload a 50MB sample of 00008.m2ts on a website somewhere? Let me know.

nautilus7

27th September 2008, 21:14

The chapter list is stored in the .mpls files (these are the playlist files). In your case 00010.mpls and 00011.mpls. You can check whether there are chapters when playing the disc with powerdvd or in your standalone player. You can also check and extract chapters with BDedit.

rack04

28th September 2008, 04:02

bigdog660

28th September 2008, 06:09

Looks like 2) 3) and 4) judging by the playing time.

rebkell

28th September 2008, 06:34

Maybe someone can help me out with the Blu-ray version of "Terminator The Sarah Connor Chronicles Season 1 Disc 1". When eac3to analyzes the disc structure it comes up with the following format:

http://i11.photobucket.com/albums/a199/rack04/untitled.jpg

I'm not sure which 3 titles to select to get the 3 individual episodes.

Actually, I think you are going to have to break them out, there is no episode 2 on the disk menu. I think the first line 00004.m2ts is episode 1, 00012.m2ts is episode 2 and 00013.m2ts is episode 3.

Playlist 2) is an extended version of episode 1.
You can't actually play episode 2 from the menu, it's not even an option. It's a strange menu, you pop in the disk and it starts playing the 2hr+ selection which I assume is all three episodes and is probably 1) ..

Ok, 4 is the first episode which is 46:40
12 is episode 2 which is 44:05
13 is episode 3 which is 43:47
15,16 and 17 are a special feature which combined equals 39:08

It's a really strange disk.

nurbs

28th September 2008, 10:39

rebkell is right. 00012.m2ts is the second(?) episode. I was worried about that myself when I encoded the disk, but there is nothing missing.

madshi

28th September 2008, 11:36

It looks like chapter track detection may not work properly in some cases.

There are 4 titles on this disk. Titles 1 and 2 are the movie. Titles 3 and 4 are the movie with "video commentary" (images from production rendered on top of the movie, pretty silly). The length of the video is the same for all titles - after all, it's the same movie.

eac3to seems to detect the chapters list just fine for titles 3 and 4, but not for 1 and 2. I can't quite believe that there are no chapters for 1 and 2 since that is the actual movie that most people would watch.

Even if I run eac3to directly on the .m2ts files, it's the same - it can't find any chapters for 00008.m2ts but it does find them for 00011.m2ts

In this particular case, I can probably extract the chapters from title 3 and use them with the video from title 1, but if 1 actually has chapters but they are not detected, it's probably a bug that needs to be squished.

E:\work>eac3to E:\hd\blah
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

3) 00013.mpls, 00011.m2ts, 1:53:18
- VC-1, 1080p24 /1.001 (16:9)
- DTS, English, multi-channel, 48khz

E:\work>eac3to E:\hd\blah 1)
M2TS, 1 video track, 2 audio tracks, 2 subtitle tracks, 1:53:18
1: VC-1, 1080p24 /1.001 (16:9)

E:\work>eac3to E:\hd\blah 3)
M2TS, 1 video track, 1 audio track, 2 subtitle tracks, 1:53:18
1: Chapters, 21 chapters
2: VC-1, 1080p24 /1.001 (16:9)

madshi, what do you want me to do to help you troubleshoot it? Upload a 50MB sample of 00008.m2ts on a website somewhere? Let me know.
Don't need any m2ts files in this case. You can upload the playlists listed in the eac3to log (*.mpls) instead.

eac3to reports insane audio delay. DGIndex demuxes audio without problems.
Checked the VOB. Actually eac3to is absolutely right. The audio begins to play only after about 1 minute. Before that there's no audio in the VOB. The reason why things didn't work as expected when using eac3to is that there's a video gap of 1 minute in the stream, too. And MPEG2 video gap/overlap fixing in eac3to v2.64 was broken. Basically this VOB looks like this:

[a few stupid video frames] [...no data for 1 minute...] [here comes the real data]
Since both audio and video have about 1 minute missing data, straightly demuxing things without correcting audio/video sync kind of works (which is what DGIndex probably does), although strictly spoken it's not correct. If you play the original VOB file you'll notice that it's playing 1 minute of black screen without audio. If you play the video/audio streams demuxed by DGIndex you'll notice that this 1 minute of black silence is gone. Now if you use eac3to v2.65 and let eac3to fix the video and audio gaps/delays, the final MKV+AC3 will play just like the VOB with correct audio sync and 1 black silence minute in the beginning.

I found another problem with .vob as a source. If I let eac3to to use gaps I loose audio synchronization. Audio demuxed without using gaps has perfect synchronization.
Again eac3to detects things correctly (there's an overlap of about 1 second in both video and audio), but the fixing of the video overlap was broken in v2.64. With v2.65 letting eac3to fix audio+video overlap will result in correct audio sync. The resulting MKV+AC3 will be about 1 second shorter if you let eac3to fix the overlaps.

madshi

28th September 2008, 11:37

eac3to v2.65 released

http://madshi.net/eac3to.zip

* automatic channel remapping for 6.1 tracks with wrong channel mask
* automatic channel remapping for ArcSoft DTS decoder 6.1 tracks
* fixed: TrueHD -> Surcode encoding didn't work, anymore
* fixed: MPEG2 + h264 video gap/overlap removal didn't work properly

nautilus7

28th September 2008, 11:50

Checked the VOB. Actually eac3to is absolutely right. The audio begins to play only after about 1 minute. Before that there's no audio in the VOB. The reason why things didn't work as expected when using eac3to is that there's a video gap of 1 minute in the stream, too. And MPEG2 video gap/overlap fixing in eac3to v2.64 was broken. Basically this VOB looks like this:

[a few stupid video frames] [...no data for 1 minute...] [here comes the real data]
Since both audio and video have about 1 minute missing data, straightly demuxing things without correcting audio/video sync kind of works (which is what DGIndex probably does), although strictly spoken it's not correct. If you play the original VOB file you'll notice that it's playing 1 minute of black screen without audio. If you play the video/audio streams demuxed by DGIndex you'll notice that this 1 minute of black silence is gone. Now if you use eac3to v2.65 and let eac3to fix the video and audio gaps/delays, the final MKV+AC3 will play just like the VOB with correct audio sync and 1 black silence minute in the beginning
I 've came across a few dvds that DGIndex reports a huge delay with. neuron2 suggests skipping a few GOPs at the beginning because they don't contain valid timecodes.

azad

28th September 2008, 11:52

I try to extract the AC3 stream of a TrueHD Audio stream, but it won't work.
eac3to always first decodes and then encodes the AC3 audio with libAften instead of just extracting the stream like it does when using a m2ts file as source.

nautilus7

28th September 2008, 11:53

madshi

28th September 2008, 12:17

I 've came across a few dvds that DGIndex reports a huge delay with. neuron2 suggests skipping a few GOPs at the beginning because they don't contain valid timecodes.
But that could result in a few valid video frames being discarded. I think a better fix would be to fully parse the DVD structure information (IFO etc) and make use of that for demuxing/remuxing. That's still on my to do list.

nautilus7

28th September 2008, 12:23

Maybe you could skip as many frames as needed until a normal delay value is reached. But that wouldn't be nice enough. Just a quick and easy (?) proposal.

azad

28th September 2008, 12:26

If your TrueHD stream is from HD DVD, then it doesn't have an AC3 track attached.

It's a TrueHD stream from BD. And it does have a AC3 Track attached, I can extract the AC3 stream if I use eac3to on the m2ts file.

nautilus7

28th September 2008, 12:29

How did you demuxed the TrueHD stream from the m2ts container? Most probably only the TrueHD part was demuxed.

madshi

28th September 2008, 12:31

Maybe you could skip as many frames as needed until a normal delay value is reached. But that wouldn't be nice enough. Just a quick and easy (?) proposal.
I don't like skipping frames without knowing what the frames are. The frames could contain important content.

It's a TrueHD stream from BD. And it does have a AC3 Track attached, I can extract the AC3 stream if I use eac3to on the m2ts file.
How about posting an eac3to log from the situation where you think there's something wrong?

kurt

28th September 2008, 12:33

It's a TrueHD stream from BD. And it does have a AC3 Track attached, I can extract the AC3 stream if I use eac3to on the m2ts file.
I'm doing this right now and it's working.

I demuxed the whole m2ts (godfather) and now extracting the ac3 core. no problems and no encoding...

eac3to v2.65
command line: eac3to english.thd+ac3 english.ac3 -core
------------------------------------------------------------------------------
TrueHD/AC3, 5.1 channels, 48khz
Extracting AC3 stream...
Creating file "englisch.ac3"...
eac3to processing took 2 minutes, 45 seconds.
Done.

btw: thx, madshi for your great work. again :)

azad

28th September 2008, 12:52

How did you demuxed the TrueHD stream from the m2ts container? Most probably only the TrueHD part was demuxed.

I guess thats it.

The log:
eac3to v2.65
command line: eac3to.exe "d:\HD Audio Temp\filename.thd" "E:\filename.ac3" -core
------------------------------------------------------------------------------
TrueHD, 5.1 channels, 48khz
Decoding with libav/ffmpeg...
Encoding AC3 <640kbps> with libAften...
Creating file "E:\test.ac3"...

In Kurt's last post regarding my TrueHD problem I saw that the TrueHD stream is detected as TrueHD/AC3. As you can see in my log, the stream is only detected as a TrueHD stream.
What did I wrong, how do I extract THD streams including the AC3 stream?
I thought that eac3to handles THD streams the same way as DTSHD with DTS core?!?

nautilus7

28th September 2008, 13:11

Again: TrueHD tracks from Blu-ray have an AC3 track attached with them. It's NOT a core like DTS-HD tracks. This means the AC3 track is not needed in order to play the TrueHD track, while you need the DTS core in order to play the HD extensions in DTS-HD tracks.

eac3to used to demux separately the TrueHD track and AC3 track from TrueHD/AC3 combo tracks. Not it is able to demux both as a combo track using the ".thd+ac3" extension.

also the -core switch you used doesn't make any difference. If there was an AC3 track attached with your TrueHD track it would have been demuxed without using that switch.

azad

28th September 2008, 13:33

Ok, thanks for your help!!!

Atak_Snajpera

28th September 2008, 13:43

But that could result in a few valid video frames being discarded. I think a better fix would be to fully parse the DVD structure information (IFO etc) and make use of that for demuxing/remuxing. That's still on my to do list.
I think it's time for DVD structure support. it would be very useful for people with episodic dvds.

matrix40

28th September 2008, 18:01

Hey does anyone else get a virus warning when download eac3to? For some reason my webroot says the hooksurcode.dll is a app/madcode-f hacktool. Ive never gotten this before with eac3to.

nautilus7

28th September 2008, 19:45

It has been reported that about hooksurcode.dll and another antivirus i think. It's false alarm.

rica

29th September 2008, 00:48

Hi guys.
I wanted to test the newest eac3to today.
I was wondering if it is able to fix the trueHD gaps.
Remember my previous post:

Audio gap description file detected, can't be used for TrueHD/MLP, though.
Audio gap description file detected, will be used for processing...
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a03] Extracting audio track number 3...
[a03] Realizing DTS gaps...
[v01] Creating file "E:\output\video.h264"...
[a03] Creating file "E:\output\audiodts.dtshd"...
[a02] Creating file "E:\output\audiothd.thd+ac3"...
[a02] Audio overlaps for 5ms at playtime 0:46:24.
[a02] Audio overlaps for 5ms at playtime 1:28:23.
[a02] Audio overlaps for 5ms at playtime 2:33:42.
[a02] The audio gaps/overlaps technically can't be removed from the TrueHD bitstream.
[a02] In order to remove them you'll have to transcode the audio to another format.
Video track 1 contains 230070 frames.
eac3to processing took 31 minutes, 57 seconds.
Done.

I've found which m2ts files are on 0:46:24 for testing.
I just got two m2ts files where eac3to couldn't repair the gaps at 0:46:24.

Remuxed m2ts files and demuxed the raw files again:

eac3to\eac3to.exe 00017.m2ts+00018.m2ts 1: G:\video.h264 2: G:\audio.thd+ac3
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 20 subtitle tracks, 0:05:18
1: h264/AVC, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, 5.1 channels, 48khz
3: DTS Master Audio, 5.1 channels, 24 bits, 48khz
4: Subtitle (PGS)
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[v01] Creating file "G:\video.h264"...
[a02] Creating file "G:\audio.thd+ac3"...
Video track 1 contains 7620 frames.
eac3to processing took 1 minute, 33 seconds.
Done.

There wasn't any problem left. (thanks to madshi :thanks: )

_ _ _ _ _

sygrup

29th September 2008, 03:31

Anyways - great work!
thaks

madshi

29th September 2008, 08:33

I was wondering if it is able to fix the trueHD gaps.
Nope.

Kurtnoise

29th September 2008, 09:04

@madshi: could you add a switch to enable/disable audio sounds (error and jobs complete) ?

madshi

29th September 2008, 09:06

@madshi: could you add a switch to enable/disable audio sounds (error and jobs complete) ?
No. You can just delete the WAV files if you don't want any sounds. eac3to doesn't mind if the WAV files aren't there.

avivahl

29th September 2008, 14:35

madshi: did you test if the new Nero 9 filters can be used by 3rd-party programs?

madshi

29th September 2008, 14:49

madshi: did you test if the new Nero 9 filters can be used by 3rd-party programs?
I don't have access to Nero 9. Whoever has access to that can test whether the DirectShow audio decoder works in GraphEdit. My best guess would be no. Why should they revert the protection they added in Nero 8?

lexor

29th September 2008, 15:18

No. You can just delete the WAV files if you don't want any sounds. eac3to doesn't mind if the WAV files aren't there.

Then how about a compromise? Can you at least use foobar2k to do WAV->WAV, but apply per-track replay gain? The bloody things are just way too damn loud as they are. When listening to replay-gained music the success sound comes like the voice of Lucifer from the bowels of Hell.

madshi

29th September 2008, 15:53

Does everybody agree that the success/error WAVs which eac3to uses are too loud?

Opinions, please!

kurt

29th September 2008, 16:27

yes, it's too loud.

markrb

29th September 2008, 16:54

I was wondering if anyone knows what format would give me the closest to the original that I could use with my ATI 4850 video card that allows for bit stream non PAP audio over HDMI to my Onkyo 805?
I have been using strictly AC3 format, but that is no better then using Spdif. I am a little confused over the different choices. I have tried to convert to PCM, but can't get it to work. Now I am trying LPCM and I will see how that works. After that I will attempt raw. Maybe it's my decoder as I am using the latest ffdshow.

Any help to find the closest audio to the Hi-Def used on Blu-Rays used would be most appreciated.

Thanks,
Mark

Chumbo

29th September 2008, 16:54

Does everybody agree that the success/error WAVs which eac3to uses are too loud?

Opinions, please!
I have these little knobs called volume controls on my speakers...they do wonders. :)

I have to say, the first time you put these in, I jumped off my chair because I had my volume pretty loud and didn't expect any sound from eac3to. LOL

Not to make more work for you, but have you considered using the pre-defined sounds from Windows? That way we can assign our own WAV files or turn them off altogether from the Control Panel. At least you won't have to embed any sounds or provide any switches for this. :)

Kurtnoise

29th September 2008, 17:00

I have to say, the first time you put these in, I jumped off my chair because I had my volume pretty loud and didn't expect any sound from eac3to.
me too...:-/

anyway, I'm fine with the removal by myself.

madshi

29th September 2008, 17:09

I was wondering if anyone knows what format would give me the closest to the original that I could use with my ATI 4850 video card that allows for bit stream non PAP audio over HDMI to my Onkyo 805?
I'd suggest transcoding to FLAC. You can use madFlac for decoding. The audio should be transported as PCM over HDMI to your Onkyo in up to 7.1 with full bitdepth, as far as I can say. However, you can't mux FLAC into a m2ts file, if that's what you're trying to do?

madshi

29th September 2008, 17:10

Not to make more work for you, but have you considered using the pre-defined sounds from Windows?
I had checked them but found none which I liked or seemed fitting.

xixi2000

29th September 2008, 17:16

good job

Boulder

29th September 2008, 17:22

piratburner

29th September 2008, 17:23

madshi

29th September 2008, 17:24

I'm not sure if this has been discussed in this thread earlier: is there any quality loss when decoding regular AC3 tracks to WAV using the Nero 7 DirectShow filter, which outputs 24bit wave files which eac3to then converts to 16bit.
eac3to doesn't convert them to 16bit (unless you ask it to). If you decode AC3 to 24bit WAV with the Nero 7 decoder there's no quality loss. If you tell eac3to to downconvert to 16bit the noise floor will ever so slightly increase. Whether you (or anybody) can hear the increased noise floor is something even the experts don't agree on. So don't expect to get a clear answer about that! The big question for me, though, is why decode the AC3 track in the first place?

madshi

29th September 2008, 17:27

Hi !

Need your answers :-)

I'm demuxing a PCM track from a M2TS file with tsMuxerGUI.
And then using mkvmerge to add that track to a MKV file.

Is this right way to do it ?

And I know this have nothing to do with eac3to , BUT you are so good at this stuff so I hope you could point me at the right direction if I'm wrong :-)
If you demux to PCM then you can't add this to MKV. I guess you're demuxing to WAV? If you do that you may run into trouble with 24bit tracks cause WAV files with more than 4GB (or even 2GB) are problematic. I'm not sure if mkvtoolnix can handle them correctly. I'm not sure if TsMuxeR can create them properly, either. If your final aim is MKV then I suggest letting eac3to directly convert the PCM track to FLAC. That saves space and still gives you 100% identical audio quality.

Boulder

29th September 2008, 17:27

eac3to doesn't convert them to 16bit (unless you ask it to). If you decode AC3 to 24bit WAV with the Nero 7 decoder there's no quality loss. If you tell eac3to to downconvert to 16bit the noise floor will ever so slightly increase. Whether you (or anybody) can hear the increased noise floor is something even the experts don't agree on. So don't expect to get a clear answer about that! The big question for me, though, is why decode the AC3 track in the first place?I need to decode them (if there's more than 2 channels) to process them to either a 2-ch AC3 or MP4, depending on the final container.

piratburner

29th September 2008, 17:34

If you demux to PCM then you can't add this to MKV. I guess you're demuxing to WAV? If you do that you may run into trouble with 24bit tracks cause WAV files with more than 4GB (or even 2GB) are problematic. I'm not sure if mkvtoolnix can handle them correctly. I'm not sure if TsMuxeR can create them properly, either. If your final aim is MKV then I suggest letting eac3to directly convert the PCM track to FLAC. That saves space and still gives you 100% identical audio quality.

Ok , But my Tvix cant handle multichanel flac, but TviX lates FW have HD-Sound passtrough :-) Yhen I would have PCM in my MKV ( If MKV allowed that) So FLAC is not a way to go for me, And I hope PCM should do it.

If PCM works in MKV how shall I get a prober PCM from a M2ts FILE, so I cam put that in a MKV file ?

Mkvmerge what completed now with NO errors.... Can I trust that

Result from MediaInfo
Audio #2
Format : PCM
Codec ID : A_PCM/INT/LIT
Duration : 1 h 50 min
Bitrateläge : Constant
Kanal(er) : 6 kanaler
Samplingstakt : 48,0 KHz
Upplösning : 16 bits
titel : PCM
Språk : Engelska

MediaInfo from the demuxed track from M2TS file
Audio
Format : FFFE
Codec ID : FFFE
Codec ID/Info : Extensible wave format
Codec ID/Hint : Extensible
Bitrate : 4 608 Kbps
Kanal(er) : 6 kanaler
Samplingstakt : 48,0 KHz
Upplösning : 16 bits

madshi

29th September 2008, 17:38

I need to decode them (if there's more than 2 channels) to process them to either a 2-ch AC3 or MP4, depending on the final container.
MP4 is a container. You mean AAC? Both AC3 and AAC are lossy codecs. So you shouldn't worry about quality loss when *decoding* the original AC3 track. The loss in audio quality is not during decoding but during the reencoding to the final AC3 or AAC track.

madshi

29th September 2008, 17:42

Ok , But my Tvix cant handle multichanel flac
Not yet. But it will probably be added sooner or later.

If PCM works in MKV how shall I get a prober PCM from a M2ts FILE, so I cam put that in a MKV file ?
mkvtoolnix doesn't accept PCM, but it accepts WAV. WAV is the same as PCM with just a little header in front of the PCM data. mkvtoolnix needs the WAV header because only that way it knows how many channels the PCM data has and which bitdepth etc...

Mkvmerge what completed now with NO errors.... Can I trust that
Looks good. But you've luck. The track is only 16bit and only 1:50 hours long. With 24bit PCM tracks or with longer 16bit tracks things might get more difficult.

Basically you can double check whether the MKV is alright by just playing the beginning and end of the movie. If you get proper sound at both beginning and end then the MKV should be fine.

piratburner

29th September 2008, 17:44

Thanks madshi :-) I will do some more testings :-)

Boulder

29th September 2008, 18:25

MP4 is a container. You mean AAC? Both AC3 and AAC are lossy codecs. So you shouldn't worry about quality loss when *decoding* the original AC3 track. The loss in audio quality is not during decoding but during the reencoding to the final AC3 or AAC track.Yes, I meant AAC.. Thanks for the info, I'll keep on using eac3to just as before :)

piratburner

29th September 2008, 20:12

Thanks madshi :-) I will do some more testings :-)

Tried to extraxt PCM 24bit with TsMuxer, The result what 2 files ???

Using eac3to = 1 file, with this messages...
[a03] Extracting audio track number 3...
[a03] Reading RAW/PCM...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Writing WAV...
[a03] Creating file "g:\bill2.wav"...
[a03] This audio track has a constant bit depth of 24 bits.
[a03] Caution: The WAV file is bigger than 4GB.
[a03] Some WAV readers might not be able to handle this file correctly.
Video track 2 contains 197030 frames.
eac3to processing took 48 minutes, 54 seconds.

This files was OK when adding to a MKV ( no error messages with mkvmerge :-) )

I'm trying now to use command -down16 and add that also to the MKV

Tomorrow I going to try this MKV:s at my TviX :-)

nautilus7

29th September 2008, 23:29

There's a problem with AGM files.

DTS Pro Encoder will accept all AGM files made from DTS-HD Master Audio tracks, but will reject every single AGM file made from TrueHD tracks (haven't tried with LPCM as source). The encoder pops up am message saying that this isn't a valid source file. This happens all the time regardless the source track is 16bit or 24 bit. In addition, mono wavs created from TrueHD tracks work fine with the AGM packer of the encoder and the actual encoder afterwards.

xkodi

30th September 2008, 02:44

but will reject every single AGM file made from TrueHD tracks

if the decoded TrueHD track is less than 4GB the problem is not there.

@madshi

just to save you time in investigating the problem:

at least when eac3to decodes TrueHD track to AGM and the decoded file is over 4GB, eac3to writes wrong value for SAMP chunk in the header.

example:

the created AGM file size: 4877833220 bytes
SAMP chunk value: 4877833220 - 260 = 0x122BDD300
eac3to writes for SAMP value: 00D3 BD22 0000 0000
instead the correct value of: 00D3 BD22 0100 0000

Joniii

30th September 2008, 04:35

Stupid question :), if I demux m2ts with eac3to which has audio delay of -9 ms, do I then need to add -9 delay in TSMuxer when I mux it back to m2ts? Or does eac3to somehow fix that delay when it demuxes it.

rack04

30th September 2008, 04:45

nautilus7

30th September 2008, 07:21

if the decoded TrueHD track is less than 4GB the problem is not there.
Maybe, haven't test that. With DTS-HD, size is not a problem though.

Kineticpoet

30th September 2008, 07:31

Hi, I've lurked here for a long time and finally registered recently. I do have a question, and I realize this might have been answered somewhere along the way, but I'm new to this and honestly all of this information can get very confusing and frustrating to a newbie. So please accept my apologies if a question like this is annoying.

I've recently learned how to back my Blu-Ray's to a .mkv container on my PC using MeGUI and a plethora of other programs. I just bought a new Blu-Ray and when I open it using tsMuxer I only see one English audio track which is in TrueHD. When I demux it, it comes out as a AC3 file, but after reading this thread I'm thinking that's not how I should extract the stream. My question is how do I remove the TrueHD stream from the .m2ts file and then how do I convert that to AC3? I've read quite a bit of this thread, I saw at one point someone said something in regards to there being an AC3 stream that's coupled with the TrueHD stream. If that's true, I would just need to extract the AC3 stream, correct? I'm just unclear on how to do that.

I was able to figure out by myself how to backup a blu-ray, which took awhile, but for some reason I'm really lost as to how I can get this to work. I searched for guides on how to do this and didn't really come up with anything, or at least anything that I really understood. I hate asking something when I'm sure 95% of everyone here will roll their eyes, but this is a last ditch effort. If anyone could help it would be appreciated. Thanks.

xkodi

30th September 2008, 10:23

Maybe, haven't test that. With DTS-HD, size is not a problem though.

DTS-HD bit-depth is known from the beginning and thus the correct size, i suspect that with TrueHD size is calculated at the end after the check if the last byte is zero or not and probably the calculation at the end is broken and assumes that the maximum possible size is 4GB (32bit) like in the WAV file, but AGM supports 64bit size.

nautilus7

30th September 2008, 11:38

Seems you have solve the issue then. :) madshi's job will be easier.

nautilus7

30th September 2008, 11:46

My question is how do I remove the TrueHD stream from the .m2ts file and then how do I convert that to AC3? I've read quite a bit of this thread, I saw at one point someone said something in regards to there being an AC3 stream that's coupled with the TrueHD stream. If that's true, I would just need to extract the AC3 stream, correct?Correct. I'm just unclear on how to do that.
I suggest this way: http://forum.doom9.org/showthread.php?p=1148640#post1148640

You just need to modify the command in the last step to correspond to the TrueHD/AC3 combo track (change the extension of the output file to .ac3). This will give the AC3 track that already exists into the Blu-ray. If you also need the TrueHD, change extension to .thd.

Mercury_22

30th September 2008, 11:51

Can you add MKV to "Supported source formats" ? PLEASE ! :helpful:

P.S. I know there are other "tools" for demux all video and audio tracks of an mkv but I'd like (as many others) to keep things simple (and not having a lot of programs installed)
:thanks:

piratburner

30th September 2008, 17:14

Tried to extraxt PCM 24bit with TsMuxer, The result what 2 files ???

Using eac3to = 1 file, with this messages...
[a03] Extracting audio track number 3...
[a03] Reading RAW/PCM...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Writing WAV...
[a03] Creating file "g:\bill2.wav"...
[a03] This audio track has a constant bit depth of 24 bits.
[a03] Caution: The WAV file is bigger than 4GB.
[a03] Some WAV readers might not be able to handle this file correctly.
Video track 2 contains 197030 frames.
eac3to processing took 48 minutes, 54 seconds.

This files was OK when adding to a MKV ( no error messages with mkvmerge :-) )

I'm trying now to use command -down16 and add that also to the MKV

Tomorrow I going to try this MKV:s at my TviX :-)

So sad TviX can't handle PCM/WAV sound inside MKV files :-(

But no problem to add this PCM/WAV to a MKV :-)

zyrill

30th September 2008, 20:40

Did anybody ever experience the following error and has found a workaround?

[a05] Removing TrueHD dialog normalization...
[a05] Decoding with libav/ffmpeg...
[libav] Substream 0 checksum failed
[libav] No restart header present in substream 1.
[a05] The libav decoder reported an error while decoding.
Aborted at file position 81920.

Apparently the sourcefile has some errors and the decoder crashes over it. Since I only have that source and PowerDVD plays it just fine I'm wondering: could a solution along the lines of null-output (for the defective chunk) be found? I could provide the sample, of course!

nautilus7

30th September 2008, 20:45

eac3to Shoot.'Em.Up.2007.Blu-ray.flac" shoot.em.up.aac
FLAC, 7.1 channels, 1:26:32, 24 bits, 5095kbps, 48khz
NeroAacEnc only supports 6 channels. Will mix the surround channels.
Decoding FLAC...
Mixing surround channels...
Loading white noise (needed for dithering)...
Encoding AAC <0.50> with NeroAacEnc...
This audio track has a constant bit depth of 24 bits.
eac3to processing took 12 minutes, 35 seconds.Nero supports 7.1 encoding. :confused:

Steel

30th September 2008, 23:12

Does everybody agree that the success/error WAVs which eac3to uses are too loud?

Opinions, please!

I just delete the wav file and I don't have to worry about it.

zyrill

1st October 2008, 02:24

Nero supports 7.1 encoding. :confused:

I don't know whether this is of any consequence, but a new version (1.3.0.0) of Nero's encoder/decoder-package came out a week ago!

EPiPH0NE

1st October 2008, 07:56

I don't know whether this is of any consequence, but a new version (1.3.0.0) of Nero's encoder/decoder-package came out a week ago!

They can't be used with eac3to only Nero 7 filters.

nautilus7

1st October 2008, 09:28

That restriction applies only to Nero directshow filters (aka the nero audio decoder 2). Neroaacenc is a command line encoder and yes i use the latest.

JustABDFan

1st October 2008, 15:30

nautilus7

1st October 2008, 18:29

madshi, it would be nice if eac3to displays the profile of the AAC file (LC, HE, HEv2), runtime and the bitrate even in vbr tracks, like it's done in flac.

fib0by

1st October 2008, 18:49

Madshi could you add switch something like "-Automatically repeat audio demuxing if gaps are detected"?

By the way, only the extraction of the audio track(s) should be repeated. It's probably slower if the entire extraction process (video, audio, subtitles) is repeated just to get audio right.

Jaja1

1st October 2008, 23:22

Does this message/warning/error/ ??? in the log means that the generated dts file is not okay?
eac3to v2.65
command line: eac3to 1) 2:i:\M.mkv 3:i:\M.dts 8:i:\M.1.sup 9:i:\M.2.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 10 subtitle tracks, 1:25:55
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
4: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: AC3, Portuguese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
7: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
8: Subtitle (PGS), English
9: Subtitle (PGS), English
10: Subtitle (PGS), French
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), Portuguese
13: Subtitle (PGS), English
14: Subtitle (PGS), French
15: Subtitle (PGS), Spanish
16: Subtitle (PGS), Portuguese
17: Subtitle (PGS), English
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[s08] Extracting subtitle track number 8...
[s09] Extracting subtitle track number 9...
[a03] Extracting TrueHD stream...
[a03] Removing TrueHD dialog normalization...
[a03] Decoding with libav/ffmpeg...
[a03] Writing WAVs...
[v02] Muxing video to Matroska...
[a03] Creating file "i:\M.R.wav"...
[a03] Creating file "i:\M.L.wav"...
[a03] Creating file "i:\M.C.wav"...
[a03] Creating file "i:\M.SR.wav"...
[a03] Creating file "i:\M.SL.wav"...
[a03] Creating file "i:\M.LFE.wav"...
[s09] Creating file "i:\M.2.sup"...
[s08] Creating file "i:\M.1.sup"...
[libav] End of stream indicated
[libav] Lossless check failed - expected 0, calculated 8b
[a03] This audio track has a constant bit depth of 24 bits.
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.21.0.
Surcode encoding successfully started. Please wait...
Closing Surcode...
Added fps value to MKV header.
Video track 2 contains 123591 frames.
eac3to processing took 27 minutes, 38 seconds.
Surcode encoding took 13 minutes, 18 seconds.
Done.

Greif

2nd October 2008, 03:47

are the filters still locked in Nero 9?

Joniii

2nd October 2008, 06:42

Does EAC3TO also fix delay in TrueHD files when demuxing HD DVD evo. When I demuxed it, EAC3TO didn't mention anything about fixing TrueHD delay, only with E-AC3 tracks.

eac3to f:\pevob_1.evo+f:\pevob_2.evo -demux
EVO, 2 video tracks, 5 audio tracks, 4 subtitle tracks, 2:20:02
1: Joined EVO file
2: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
3: VC-1, 480p30 /1.001 (3:2), 100ms
4: E-AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB, 100ms
5: TrueHD, 5.1 channels, 48khz, dialnorm: -27dB, 200ms
6: E-AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB, 100ms
7: E-AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB, 100ms
8: E-AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, 100ms
9: Subtitle
10: Subtitle
11: Subtitle
12: Subtitle
[v02] Extracting video track number 2...
[v02] Removing VC-1 pulldown...
[a04] Extracting audio track number 4...
[a06] Extracting audio track number 6...
[a07] Extracting audio track number 7...
[a07] Removing E-AC3 dialog normalization...
[a07] Applying (E-)AC3 delay...
[a04] Removing E-AC3 dialog normalization...
[a04] Applying (E-)AC3 delay...
[a06] Removing E-AC3 dialog normalization...
[a06] Applying (E-)AC3 delay...
[a05] Extracting audio track number 5...
[a05] Removing TrueHD dialog normalization...
[v02] Creating file "pevob_1 - 2 - VC-1, 1080p24.vc1"...
[a05] Creating file "pevob_1 - 5 - TrueHD, 5.1 channels, 48khz.thd"...
[a06] Creating file "pevob_1 - 6 - E-AC3, 5.1 channels, 640kbps, 48khz.eac3"...
[a07] Creating file "pevob_1 - 7 - E-AC3, 5.1 channels, 640kbps, 48khz.eac3"...
[a04] Creating file "pevob_1 - 4 - E-AC3, 5.1 channels, 640kbps, 48khz.eac3"...
[s10] Extracting subtitle track number 10...
[s12] Extracting subtitle track number 12...
[s09] Extracting subtitle track number 9...
[s11] Extracting subtitle track number 11...
[s10] Creating file "pevob_1 - 10 - Subtitle.sup"...
[s12] Creating file "pevob_1 - 12 - Subtitle.sup"...
[s09] Creating file "pevob_1 - 9 - Subtitle.sup"...
[s11] Creating file "pevob_1 - 11 - Subtitle.sup"...
Video track 2 contains 201466 frames.
Video track 3 contains 251744 frames.
eac3to processing took exactly 14 minutes.
Done.

ACrowley

2nd October 2008, 08:57

are the filters still locked in Nero 9?

i would bet 100% sure...yes

But i will try it an report later :)

nautilus7

2nd October 2008, 10:26

Does EAC3TO also fix delay in TrueHD files when demuxing HD DVD evo. When I demuxed it, EAC3TO didn't mention anything about fixing TrueHD delay, only with E-AC3 tracks.
It's been said a million of times. TrueHD bitstream can't be delayed. Only when converting TrueHD to another format it can be delayed.

sshd

2nd October 2008, 11:25

I have found a problem when using eac3to in a cygwin enviroment.

Normal Windows XP prompt:
eac3to -test
Nero Audio Decoder (Nero 7 or older) works fine
ArcSoft DTS Decoder (1.1.0.0) works fine
Sonic Audio Decoder (4.3.0.169) works fine
Haali Matroska Muxer (2008-03-29) is up to date
Nero AAC Encoder could not be located
http://www.nero.com/eng/nero-aac-codec.html
Copy NeroAacEnc.exe to the eac3to or to the Windows folder
Surcode DTS Encoder doesn't seem to be installed
http://www.surcode.com
MkvToolnix (2.3.0.0, 2008-09-08) is up to date
There's a new beta version (2.3.0.0, 2008-09-25) available
http://www.bunkus.org/videotools/mkvtoolnix/win32/pre

Cygwin prompt:
eac3to.exe -test
Nero Audio Decoder (Nero 7 or older) works fine
ArcSoft DTS Decoder doesn't seem to be installed
http://www.arcsoft.com/products/totalmediatheatre
Sonic Audio Decoder (4.3.0.169) works fine
Haali Matroska Muxer (2008-03-29) is up to date
Nero AAC Encoder could not be located
http://www.nero.com/eng/nero-aac-codec.html
Copy NeroAacEnc.exe to the eac3to or to the Windows folder
Surcode DTS Encoder doesn't seem to be installed
http://www.surcode.com
MkvToolnix (2.3.0.0, 2008-09-08) is up to date
There's a new beta version (2.3.0.0, 2008-09-25) available
http://www.bunkus.org/videotools/mkvtoolnix/win32/pre

Jeff Flowerday

2nd October 2008, 14:31

Joniii

2nd October 2008, 17:31

It's been said a million of times. TrueHD bitstream can't be delayed. Only when converting TrueHD to another format it can be delayed.

sorry

duckdown

2nd October 2008, 18:45

Hi, I already have a Nero 8 installed yet the program keeps failing to open Nero Audio Decoder 2 (I am trying to convert AAC audio into AC3) and it just keeps blaring a siren in my ear every time I try

What gives? My nero is registered.

poisondeathray

2nd October 2008, 18:56

Joniii

2nd October 2008, 20:02

I'm confused. My TrueHD track has +200ms delay. I converted it to DTS with EAC3TO. Do I now need to apply negative or positive delay with EAC3TO or leave it as it is? :confused:

sshd

2nd October 2008, 20:29

Do you have the arcsoft bin directory in your path in the cygwin environment?

I didn't, but it seems to be the cure.

nautilus7

2nd October 2008, 20:40

It's been said a million of times. TrueHD bitstream can't be delayed. Only when converting TrueHD to another format it can be delayed.

I'm confused. My TrueHD track has +200ms delay. I converted it to DTS with EAC3TO. Do I now need to apply negative or positive delay with EAC3TO or leave it as it is? :confused:If you just demux the TrueHD stream, the delay can NOT be applied.

If you decode (aka convert to another format) the TrueHD stream, then the delay IS applied.

This is the case ONLY with TrueHD. EVERY other format can be delayed.

Joniii

2nd October 2008, 20:53

If you just demux the TrueHD stream, the delay can NOT be applied.

If you decode (aka convert to another format) the TrueHD stream, then the delay IS applied.

This is the case ONLY with TrueHD. EVERY other format can be delayed.

I still dont quite understand it. I demuxed TrueHD and EAC3TO said it has +200ms delay. After demuxing I decoded it to DTS. But should I now apply delay fix for the DTS?

nautilus7

2nd October 2008, 21:49

Yes you are able to apply the delay then.

What i mean is that you can tell eac3to create the dts track from the truehd without first demuxing it. In that case the delay will automatically be applied.

eac3to input.m2ts 3: audio.dts
M2TS, 1 video track, 4 audio tracks, 1:46:27
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
4: TrueHD/AC3, French, 5.1 channels, 48khz
5: AC3, Spanish, 5.1 channels, 640kbps, 48khz
6: TrueHD/AC3, Portuguese, 5.1 channels, 48khz
[a03] Extracting audio track number 3...
[a03] Extracting TrueHD stream...
[a03] Decoding with libav/ffmpeg...
[a03] Writing WAVs...
[a03] Creating file "audio.L.wav"...
[a03] Creating file "audio.R.wav"...
[a03] Creating file "audio.SR.wav"...
[a03] Creating file "audio.LFE.wav"...
[a03] Creating file "audio.SL.wav"...
[a03] Creating file "audio.C.wav"...

But if you did:

eac3to input.m2ts 3: audio.thd

aka truehd demuxing, the delay won't be applied.

Greif

3rd October 2008, 02:10

@madshi: is there any way to host an eac3to wiki somewhere?

saint-francis

3rd October 2008, 02:24

@madshi: is there any way to host an eac3to wiki somewhere?

http://en.wikibooks.org/wiki/Eac3to

Greif

3rd October 2008, 04:16

thanks

Joniii

3rd October 2008, 04:31

nautilus7

3rd October 2008, 07:58

Thx :). Could I fix the delay in DTS with EAC3TO so I wouldn't need to do surcode encoding again, if so I would need to apply delay +200 in it or -200?

EAC3TO said that TrueHD had +200ms delay in .EVO.If you have already demuxed the truehd file do:

eac3to input.thd output.dts +200ms

If you haven't demux the truehd file do:

eac3to input.m2ts #: output.dts

where # is the number of the truehd stream.

If you don't understand it now, it might be better someone else to explain. Perhaps he/she can explain it in a different way. :)

n0mag!c

3rd October 2008, 08:15

I'm confused. My TrueHD track has +200ms delay. I converted it to DTS with EAC3TO. Do I now need to apply negative or positive delay with EAC3TO or leave it as it is? :confused:
You can apply delay (negative/positive) to DTS/AC3 streams with DELAYCUT program.

Joniii

3rd October 2008, 08:16

If you have already demuxed the truehd file do:

eac3to input.thd output.dts +200ms

If you haven't demux the truehd file do:

eac3to input.m2ts #: output.dts

where # is the number of the truehd stream.

If you don't understand it now, it might be better someone else to explain. Perhaps he/she can explain it in a different way. :)

Yeah I get it now, I need to add delay +200 to the DTS.

I was just confused if I need to add or remove delay if I do it manually.

nautilus7

3rd October 2008, 09:35

Yeah joniii, i 'm sorry. I didn't get what you asked in your last post. I understood your point right after i wrote my response, but couldn't edit it because my internet connection was giving me troubles. Sorry.

You can do eac3to iput.dts output.dts +200ms to fix your track.

ACrowley

3rd October 2008, 14:15

WOW ! Nero 9 Decoders are working outside Showtime!

You have to reg. them manually in C:\Program Files (x86)\Nero\Nero 9\Nero ShowTime

Working in Graphedit and MCP etc

Also Nero Showtime 9 works so nice on any HD Audio Format :) Great Sound ,plays TrueHD, DTSHD, EAC3 ,LPCM, DRC can be disabled in Showtime
And m2ts/EVOS are playing very smooth with dxva
eac3to with Nero 9 Decoder gives me missing MSCVR71.Dll Error. It works After downloading mscvr71.dll, but eac3to cant find the Decoder, ofcourse.
However... Nero Filters/Decoders outise Showtime have the old DRC/Dialnorm Problem ...it cant be disabled

Maybe Madshi can take a look. But its no Problme. You can install Nero9 and Nero7 at the same Time. Nero 9 DshowFilters are not reg. by default (unless you do it by yourself),so theres no Decoder confusion.

nautilus7

3rd October 2008, 14:19

What do you mean that manual registration is needed? Just installing nero 9 and the plugin isn't enough?

ACrowley

3rd October 2008, 14:51

What do you mean that manual registration is needed? Just installing nero 9 and the plugin isn't enough?

The Filter/ Decoders/.ax are not registered in the Dshow Engine.
You have to register them manually.

Nero7 Filters are registered by default

lexor

3rd October 2008, 15:13

The Filter/ Decoders/.ax are not registered in the Dshow Engine.
You have to register them manually.

Nero7 Filters are registered by default
Can you register them one at a time or all at once? If madshi ends up supporting it (with DialNorm/DRC removal, like he did for Nero 7), then I would like to get the (E)AC3 decoder (for which I currently use libav), but I don't want any other decoders to be registered at all, since I have everything else set up the way I want it and don't want to upset the balance.

So does Nero 9 allow us to pick and choose which filters we want to register and pretend the rest don't exist?

ACrowley

3rd October 2008, 15:18

Can you register them one at a time or all at once? If madshi ends up supporting it (with DialNorm/DRC removal, like he did for Nero 7), then I would like to get the (E)AC3 decoder (for which I currently use libav), but I don't want any other decoders to be registered at all, since I have everything else set up the way I want it and don't want to upset the balance.

So does Nero 9 allow us to pick and choose which filters we want to register and pretend the rest don't exist?

one at a time... there are a lot of .ax Files.
Every .ax File is one Decoder ,ofcourse. Nothing different from other Directshowfilters

NeAudio2.ax =NeroAudioDecoder2 .
It can decode´ac3,eac3,dts,dtshd,truehd,aac3,mpeg etc etc

BUT outside Showtime "with" DRC/Dialnorm. The old Problem:)

lexor

3rd October 2008, 15:27

NeAudio2.ax =NeroAudioDecoder2 .
It can decode´ac3,eac3,dts,dtshd,truehd,aac3,mpeg etc etc
aww, so it is an all in one audio decoder :(

BUT outside Showtime "with" DRC/Dialnorm. The old Problem:)
That is the case with Nero 7 too, but madshi found a way to deal with those. He might do that again.

ACrowley

3rd October 2008, 15:49

aww, so it is an all in one audio decoder :(

yes, but you can configure a Directshowplayer to use the Decoder only for your desired Formats.
Or you can tweak the registry.
But imho its useless because you get full DRC/Dialonorm outside Showtime. You cant set DRC in NeoAudioDecoder2 Properties. Only NeroShowtime has Options for DRC

That is the case with Nero 7 too, but madshi found a way to deal with those. He might do that again.
Ofcourse.....but only in eac3to:)
You still have DRC/Dialnorm in any Dshowplayer outside eac3to

I dont know if its worth it. Nero7 is fine. I dont belive the eac3 decoding quality is better in Nero9. And we dont need NEro to decoder for TrueHD,DTS(HD),Flac, etc. Only for eac3 to
So its not worth to tweak Nero 9 imho.

When you need Nero9, you can install v7 and v9 together on one System without Problems (so far you dont register Nero9 .ax Files). Nero 9 works 100% with all Functions .
i have a full install of Nero9 for burning,showtime etc and only the Showtime Part from Nero7 to work with eac3to

PS
Ive to say it again: My English is very bad. SRY

lexor

3rd October 2008, 16:30

When you need Nero9, you can install v7 and v9 together on one System without Problems (so far you dont register Nero9 .ax Files). Nero 9 works 100% with all Functions .
i have a full install of Nero9 for burning,showtime etc and only the Showtime Part from Nero7 to work with eac3to
Nero 7 isn't sold anymore though, so for those just getting into this, support for 9 is a bonus (arguably).

PS
Ive to say it again: My English is very bad. SRY
Your English is fine, don't worry about it. No one speaks proper English on the forums anyhow.

Atak_Snajpera

3rd October 2008, 23:51

By the way, only the extraction of the audio track(s) should be repeated. It's probably slower if the entire extraction process (video, audio, subtitles) is repeated just to get audio right.
Nope :) Sometimes video contains gaps as well

eac3to v2.65
command line: "C:\Program Files\RipBot264\Tools\eac3to\eac3to.exe" "D:\Beowulf (Director's Cut) Blu-Ray 1080p VC-1 DD5.1\" 4) 2:"C:\temp\RipBot264temp\job1\video.mkv" 3:"C:\temp\RipBot264temp\job1\audio.1.ac3" 10:"C:\temp\RipBot264temp\job1\subtitle.1.sup" 1:"C:\temp\RipBot264temp\job1\chapters.txt" -progressnumbers
------------------------------------------------------------------------------
M2TS, 1 video track, 7 audio tracks, 21 subtitle tracks, 0:22:01
1: Chapters, 11 chapters
2: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
3: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
4: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
5: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
6: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
7: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
8: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
9: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
10: Subtitle (PGS), English
11: Subtitle (PGS), French
12: Subtitle (PGS), Italian
13: Subtitle (PGS), Italian
14: Subtitle (PGS), German
15: Subtitle (PGS), German
16: Subtitle (PGS), Spanish
17: Subtitle (PGS), Dutch
18: Subtitle (PGS), Chinese
19: Subtitle (PGS), Korean
20: Subtitle (PGS), Spanish
21: Subtitle (PGS), Portuguese
22: Subtitle (PGS), Danish
23: Subtitle (PGS), Finnish
24: Subtitle (PGS), Norwegian
25: Subtitle (PGS), Portuguese
26: Subtitle (PGS), Swedish
27: Subtitle (PGS), French
28: Subtitle (PGS), German
29: Subtitle (PGS), Italian
30: Subtitle (PGS), Spanish
Creating file "C:\temp\RipBot264temp\job1\chapters.txt"...
[v02] Extracting video track number 2...
[s10] Extracting subtitle track number 10...
[a03] Extracting audio track number 3...
[v02] Removing VC-1 pulldown...
[v02] Muxing video to Matroska...
[a03] Removing AC3 dialog normalization...
[a03] Creating file "C:\temp\RipBot264temp\job1\audio.1.ac3"...
[s10] Creating file "C:\temp\RipBot264temp\job1\subtitle.1.sup"...
[v02] Video overlaps for 1 frames at playtime 0:00:00.
[v02] Video overlaps for 1 frames at playtime 0:00:00.
[v02] Video overlaps for 1 frames at playtime 0:00:01.
[v02] Video overlaps for 1 frames at playtime 0:00:01.
......

[v02] Video overlaps for 1 frames at playtime 0:27:26.
[v02] Video overlaps for 1 frames at playtime 0:27:26.
[v02] Video overlaps for 1 frames at playtime 0:27:27.
[v02] Video overlaps for 1 frames at playtime 0:27:27.
[v02] Video overlaps for 1 frames at playtime 0:27:27.
[v02] Video overlaps for 1 frames at playtime 0:27:27.
[v02] Video overlaps for 1 frames at playtime 0:27:27.
[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 automatically
[v02] make use of the detailed gap/overlap information.
[a03] Audio overlaps for 55ms at playtime 0:02:28.
[a03] Audio overlaps for 38ms at playtime 0:04:28.
[a03] Audio overlaps for 27ms at playtime 0:06:47.
[a03] Audio overlaps for 54ms at playtime 0:08:52.
[a03] Audio overlaps for 89ms at playtime 0:11:07.
[a03] Audio overlaps for 65ms at playtime 0:13:24.
[a03] Audio overlaps for 80ms at playtime 0:15:13.
[a03] Audio overlaps for 27ms at playtime 0:17:30.
[a03] Audio overlaps for 85ms at playtime 0:20:02.
[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.
Added fps value to MKV header.
Video track 2 contains 39626 frames.
eac3to processing took 12 minutes, 6 seconds.
Done.

ACrowley

4th October 2008, 09:40

Nero 7 isn't sold anymore though, so for those just getting into this, support for 9 is a bonus (arguably).
would be fine to have Nero9 working with eac3to.
Obviously Nero removed the Restrictions from their Decoders/Filters...theyre working properly outside Showtime ,good for us
Your English is fine, don't worry about it. No one speaks proper English on the forums anyhow.

THX :)

Joniii

4th October 2008, 10:31

Which Blu-ray tracks needed channel remapping, was it only PCM?

baudi

4th October 2008, 13:44

How can I convert a 25fps audio track to exactly 24fps.

Note: I mean 25fps to 24fps, not 25fps to 24000/1001 with the -slowdown option.

Thank you.

rica

4th October 2008, 17:19

Hi guys.

I'm a little bit confused.

For Blu-rays which include multiple workflows (lets say titles)
and multiple m2ts files for every title, how do i know what title is related with what *.mpls?

For instance when i write this code:

C:\>eac3to\eac3to.exe G:\HDBACKUPS\CETK 1)

As expected i get only the track information of that title:

M2TS, 1 video track, 2 audio tracks, 20 subtitle tracks, 2:17:13
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
5: Subtitle (PGS), English
6: Subtitle (PGS), English
7: Subtitle (PGS), Dutch]

How can i see the workflow for any specific title number like this:

1) = / ? 00059.mpls [00000+00002+00006+.....].m2ts
2) = / ? 00052.mpls [00000+00001+00009+......].m2ts

Thanks.

Note: when i use bdinfo i can see the mpls files and their workflows of m2ts' but i'm not sure the first mpls is 1)

_ _ _ _ _

jwdaigle

4th October 2008, 17:23

Hi, sorry if this question has already been asked, but I got the following error from eac3to, and dont really know what to do about it.

it occurred while trying to backup a Blu Ray.

Is this a problem with this Blu Ray, or an issue with how I am using eac3to? I have backed up many other disks in my collection, and this is the first time this has happened to me.

"
Unfortunately the Haali Muxer cannot handle this source file.
It doesn't contain enough seek/recovery points.
Aborted at file position 151715840.
"

lexor

4th October 2008, 18:48

Jaja1

4th October 2008, 19:25

How can i see the workflow for any specific title number like this:
Use the command "eac3to" without any options from anywhere in the blu-ray folder and you'll get what you want.

rica

4th October 2008, 19:36

Use the command "eac3to" without any options from anywhere in the blu-ray folder and you'll get what you want.

Hey, thanks. Easy. :)

G:\>eac3to\eac3to.exe G:\HDBACKUPS\CETK (enter)

1) 00056.mpls, 2:17:14
[0+2+6+8+9+12+13+15+16+19+20+21+24+25+26+28+118+119+32+33+34+37+39+40+42+43+45+46+49].m2ts
- h264/AVC, 1080p24 /1.001 (16:9)
- TrueHD, English, multi-channel, 48khz
- DTS Master Audio, English, multi-channel, 48khz

2) 00054.mpls, 2:14:45
[0+1+6+7+9+10+13+14+16+17+20+22+24+25+26+27+118+117+32+33+34+37+38+40+41+43+44+46+47].m2ts
- h264/AVC, 1080p24 /1.001 (16:9)
- TrueHD, English, multi-channel, 48khz
- DTS Master Audio, English, multi-channel, 48khz

3) 00055.mpls, 2:12:30
[0+2+6+8+9+11+13+15+16+18+20+21+24+26+28+118+119+32+36+37+39+40+42+43+45+46+48].m2ts
- h264/AVC, 1080p24 /1.001 (16:9)
- TrueHD, English, multi-channel, 48khz
- DTS Master Audio, English, multi-channel, 48khz

And it confirms that eac3to and bdinfo are agree:

http://img442.imageshack.us/img442/9118/bdinfopw2.th.png (http://img442.imageshack.us/my.php?image=bdinfopw2.png)

__ _ __ _

rica

4th October 2008, 20:14

But the question is where these other three mpls files are?
eac3to just shows:

00056
00054 and
00055

while bdinfo shows:
00056
00054
00055
00225
00226
00101

Then where are the remainings ?
What title numbers do they have?

http://img361.imageshack.us/img361/4066/bdinfo02bd6.th.jpg (http://img361.imageshack.us/my.php?image=bdinfo02bd6.jpg)http://img361.imageshack.us/images/thpix.gif (http://g.imageshack.us/thpix.php)

_ _ _ _ _

nautilus7

4th October 2008, 20:28

rica

4th October 2008, 20:33

You can easily find what is the content of each playlist file by playing the corresponding m2ts files in a media player.

eac3to doesn't show any playlist shorter than 15 minutes. That applies to HD DVDs and Blu-ray Discs as well.

OK, thanks a lot; i've been about to get lost between 300 pages. :)

EDIT: Everyday i learn a new tip/trick; that is why i love eac3to as well. :)

rica

4th October 2008, 21:15

OK guys,
another history of mine:

today i loaded 00056.mpls to TSMuxer and i got an m2ts which including true hd and dts-hd ma.
I left true hd since eac3to doesn't correct the gaps and i got dts-hd ma corrected in second-pass; no problem.
Then as advised i directly tried to transmit true HD to PCM.
This is what i got:

:\>eac3to\eac3to.exe C:\Users\rica\Desktop\CETK.m2ts 2: C:\HD\audio.pcm

M2TS, 1 video track, 2 audio tracks, 2 subtitle tracks, 2:17:14
1: h264/AVC, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, English, 5.1 channels, 48khz, -41ms
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz, -41ms
4: Subtitle (PGS), English
5: Subtitle (PGS), Turkish
[a02] Extracting audio track number 2...
[a02] Extracting TrueHD stream...
[a02] Decoding with libav/ffmpeg...
[a02] Applying RAW/PCM delay...
[a02] Swapping endian...
[a02] Remapping channels...
[a02] Creating file "C:\HD\audio.24bit.pcm"...
[a02] This audio track contains more than 16 bits of information.
[libav] Lossless check failed - expected f9, calculated b2
[libav] Lossless check failed - expected a8, calculated d9
[libav] Lossless check failed - expected 2e, calculated a7
[libav] Lossless check failed - expected fb, calculated 35
[libav] Lossless check failed - expected 9d, calculated e
[libav] Lossless check failed - expected b7, calculated aa
[libav] Lossless check failed - expected 3e, calculated 4c
[libav] Lossless check failed - expected a7, calculated a6
[libav] Lossless check failed - expected 61, calculated 5e
[libav] Lossless check failed - expected 77, calculated 78
[libav] Lossless check failed - expected 94, calculated 9e
[libav] Lossless check failed - expected 38, calculated a1
[a02] This audio track has a constant bit depth of 24 bits.
[a02] Audio overlaps for 5ms at playtime 1:03:58.
[a02] Audio overlaps for 5ms at playtime 2:11:04.
[a02] The audio file was demuxed without making use of the gap/overlap informati
on.
[a02] Please rerun the same eac3to command line. That will correct the gaps/over
laps.
Video track 1 contains 197424 frames.
eac3to processing took 20 minutes, 17 seconds.
Done.

C:\>eac3to\eac3to.exe C:\Users\rica\Desktop\CETK.m2ts 2: C:\HD\audio.pcm

M2TS, 1 video track, 2 audio tracks, 2 subtitle tracks, 2:17:14
1: h264/AVC, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, English, 5.1 channels, 48khz, -41ms
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz, -41ms
4: Subtitle (PGS), English
5: Subtitle (PGS), Turkish
Audio gap description file detected, will be used for processing...
[a02] Extracting audio track number 2...
[a02] Extracting TrueHD stream...
[a02] Decoding with libav/ffmpeg...
[a02] Applying RAW/PCM delay...
[a02] Swapping endian...
[a02] Remapping channels...
[a02] Realizing RAW/PCM gaps...
[a02] Creating file "C:\HD\audio.24bit.pcm"...
[a02] This audio track contains more than 16 bits of information.
[libav] Lossless check failed - expected f9, calculated b2
[libav] Lossless check failed - expected a8, calculated d9
[libav] Lossless check failed - expected 2e, calculated a7
[libav] Lossless check failed - expected fb, calculated 35
[libav] Lossless check failed - expected 9d, calculated e
[libav] Lossless check failed - expected b7, calculated aa
[libav] Lossless check failed - expected 3e, calculated 4c
[libav] Lossless check failed - expected a7, calculated a6
[libav] Lossless check failed - expected 61, calculated 5e
[libav] Lossless check failed - expected 77, calculated 78
[libav] Lossless check failed - expected 94, calculated 9e
[libav] Lossless check failed - expected 38, calculated a1
[a02] This audio track has a constant bit depth of 24 bits.
Video track 1 contains 197424 frames.
eac3to processing took 33 minutes, 20 seconds.
Done.

EDIT:And this time i've been trying to correct (remove overlaps/gaps) pcm from the main source (restarted the application)

C:\>eac3to\eac3to.exe G:\HDBACKUPS\CETK\ 1) 3: G:\HDBACKUPS\audio.pcm
M2TS, 1 video track, 2 audio tracks, 20 subtitle tracks, 2:17:13
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
5: Subtitle (PGS), English
6: Subtitle (PGS), English
..........

[a03] Extracting audio track number 3...
[a03] Extracting TrueHD stream...
[a03] Decoding with libav/ffmpeg...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Creating file "G:\HDBACKUPS\audio.24bit.pcm"...
[a03] This audio track contains more than 16 bits of information.
[a03] This audio track has a constant bit depth of 24 bits.
[a03] Audio overlaps for 5ms at playtime 0:58:47.
[a03] Audio overlaps for 5ms at playtime 1:34:34.
[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.
Video track 2 contains 197424 frames.
eac3to processing took 29 minutes, 32 seconds.
Done.

C:\>eac3to\eac3to.exe G:\HDBACKUPS\CETK\ 1) 3: G:\HDBACKUPS\audio.pcm
M2TS, 1 video track, 2 audio tracks, 20 subtitle tracks, 2:17:13
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
5: Subtitle (PGS), English
6: Subtitle (PGS), English
........

Audio gap description file detected, will be used for processing...
[a03] Extracting audio track number 3...
[a03] Extracting TrueHD stream...
[a03] Decoding with libav/ffmpeg...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Realizing RAW/PCM gaps...
[a03] Creating file "G:\HDBACKUPS\audio.24bit.pcm"...
[a03] This audio track contains more than 16 bits of information.
[a03] This audio track has a constant bit depth of 24 bits.
Video track 2 contains 197424 frames.
eac3to processing took 39 minutes, 33 seconds.
Done.

Sure, those pcm files have to be re-processed by Pcm2Tsmu (by tebasuna51) so as to be recognized by TSMuxer:

C:\>Pcm2TSmu13\Pcm2Tsmu C:\HD\audio.pcm C:\HD\audio_out.pcm -i 24

_ _ ___

rica

5th October 2008, 00:16

All those experiences teach us again:
Remuxing of BDs including multiple m2ts files with TSMuxer doesn't correct or remove the gaps/overlaps; so they have to be re-processed by eac3to,
eac3to can not correct gaps/overlaps of True HD files on those BDs, so they have to be transmitted to any other uncompressed or lossless files like pcm or wav or flac.
Created re-encoded files as pcm have to be re-processed with pcm2tsmu again so as to be recognized by TSMuxer.

Greif

5th October 2008, 05:42

@madshi: what programming language is eac3to written in? I'm also curious as to the available hooks into the code. Is everything done via StdIn/StdOut?

madshi

5th October 2008, 17:32

There's a problem with AGM files.

DTS Pro Encoder will accept all AGM files made from DTS-HD Master Audio tracks, but will reject every single AGM file made from TrueHD tracks
if the decoded TrueHD track is less than 4GB the problem is not there.
Thanks, will check that out.

Can you add MKV to "Supported source formats" ? PLEASE ! :helpful:
Maybe some day. But probably not anytime soon.

Did anybody ever experience the following error and has found a workaround?

[a05] Removing TrueHD dialog normalization...
[a05] Decoding with libav/ffmpeg...
[libav] Substream 0 checksum failed
[libav] No restart header present in substream 1.
[a05] The libav decoder reported an error while decoding.
Aborted at file position 81920.

Apparently the sourcefile has some errors and the decoder crashes over it.
Correct. And your best bet is to rerip the disc. Should be no problem if you actually own the disc. Which you should... ;)

eac3to Shoot.'Em.Up.2007.Blu-ray.flac" shoot.em.up.aac
FLAC, 7.1 channels, 1:26:32, 24 bits, 5095kbps, 48khz
NeroAacEnc only supports 6 channels. Will mix the surround channels.
Nero supports 7.1 encoding. :confused:
Yes, the new version (which is just a few days old) does. The old version did not.

madshi, it would be nice if eac3to displays the profile of the AAC file (LC, HE, HEv2), runtime and the bitrate even in vbr tracks, like it's done in flac.
I don't know how to get this information. For FLAC it's easy cause all the info is in the header. But raw AAC streams don't have a header like that (AFAIK).

WOW ! Nero 9 Decoders are working outside Showtime!
Nice! Can you create an XML graph (GraphEdit) and post the XML text here, please?

Nope :) Sometimes video contains gaps as well
That's the first Blu-Ray I've seen which has video overlaps in it! Do you see any difference in the video playback smoothness when you let eac3to fix vs. not fix the overlaps?

Which Blu-ray tracks needed channel remapping, was it only PCM?
If you let eac3to do the demuxing/transcoding you don't need to worry about things like that.

How can I convert a 25fps audio track to exactly 24fps.
Currently not with eac3to. It's on my to do list.

Hi, sorry if this question has already been asked
It has. Please search for "seekToIFrames".

@madshi: what programming language is eac3to written in? I'm also curious as to the available hooks into the code. Is everything done via StdIn/StdOut?
eac3to is written in Delphi. What do you mean with "hooks into the code"?

madshi

5th October 2008, 18:11

eac3to v2.66 released

http://madshi.net/eac3to.zip

* fixed AGM creation for files bigger than 4GB
* changed eac3to to allow AAC encoding with 7.1 channels (for new Nero encoder)
* added support for Nero's new AAC Encoder download URL
* lowered volume of error/success sounds
* when there are 2 similar playlists the one with less chapters is ignored now

nautilus7

5th October 2008, 18:30

Thanks madshi.

btw any fix for the mkvtoolnix update check?

madshi

5th October 2008, 18:42

btw any fix for the mkvtoolnix update check?
What's wrong with it? Seems to work fine for me.

nautilus7

5th October 2008, 18:58

Well, it always says there's a new beta available even if i have the newest version already installed. This happens with all versions of eac3to.
And i just saw that the date of the mkvtoolnix version is not displayed correctly with v2.66, while it did with older versions.

eac3to -test
Nero Audio Decoder (Nero 7 or older) works fine
ArcSoft DTS Decoder (1.1.0.0) works fine
Sonic Audio Decoder (4.3.0.169) works fine
Haali Matroska Muxer (2008-03-29) is up to date
Nero AAC Encoder (1.3.3.0) is up to date
Surcode DTS Encoder (1.0.29.0) is installed
MkvToolnix (2.3.0.0, 2008--04) is up to date
There's a new beta version (2.3.0.0, 2008-10-05) available
http://www.bunkus.org/videotools/mkvtoolnix/win32/pre

http://img89.imageshack.us/img89/356/66516355mf4.th.png (http://img89.imageshack.us/my.php?image=66516355mf4.png)http://img89.imageshack.us/images/thpix.gif (http://g.imageshack.us/thpix.php)

madshi

5th October 2008, 21:08

Well, it always says there's a new beta available even if i have the newest version already installed. This happens with all versions of eac3to.
eac3to does not OCR the message box of the GUI, obviously, to get the version number. Instead eac3to captures what "mkvmerge --version" says. Sometimes that information is not correct or sometimes mkvmerge is simply older than the GUI. If that is the case there's nothing I can do about it. I won't add OCR to eac3to just to get the version number/date of the GUI!! It can happen that Mosu recompiled only the GUI or only mkvmerge in a new beta build. Since eac3to can only get the version number of mkvmerge, there is potential for incorrect reports. Can't do anything about that, unless Mosu adds programatically readable version information to the GUI, too.

And i just saw that the date of the mkvtoolnix version is not displayed correctly with v2.66, while it did with older versions.
That's a bug in eac3to. It doesn't understand "Oct" as a month. Will be fixed in the next build.

rebkell

5th October 2008, 21:10

Madshi, would it be possible on the discontinuities to also print the frame number in the message, in addition to the file pos?

madshi

5th October 2008, 21:21

Madshi, would it be possible on the discontinuities to also print the frame number in the message, in addition to the file pos?
That'd be hard to do because the layer of processing which detects the discontinuities doesn't know anything about audio/video frames...

rebkell

5th October 2008, 21:27

That'd be hard to do because the layer of processing which detects the discontinuities doesn't know anything about audio/video frames...

Ok, just thought I'd ask, I have them sometimes on the Hauppauge HD-PVR caps and it's difficult to figure out where the discontinuities are happening. I suspect most happen during commercials, but you never know.

madshi

5th October 2008, 21:30

Ok, just thought I'd ask, I have them sometimes on the Hauppauge HD-PVR caps and it's difficult to figure out where the discontinuities are happening. I suspect most happen during commercials, but you never know.
If you ask eac3to to ignore the discontinuities, does demuxing/transcoding succeed then? And do you then get any audio or video corruptions during playback? Just wondering whether these discontinuities are the result of a buggy broadcasting setup. Or whether there is real corruption in the data?

Jaja1

5th October 2008, 21:35

Hi Madshi,

You probably missed my post, so here is my question again. What does the message "Lossless check failed" mean? And is the resulting DTS track in someway influenced by this?

eac3to v2.65
command line: eac3to 1) 2:i:\M.mkv 3:i:\M.dts 8:i:\M.1.sup 9:i:\M.2.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 10 subtitle tracks, 1:25:55
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
4: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: AC3, Portuguese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
7: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
8: Subtitle (PGS), English
9: Subtitle (PGS), English
10: Subtitle (PGS), French
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), Portuguese
13: Subtitle (PGS), English
14: Subtitle (PGS), French
15: Subtitle (PGS), Spanish
16: Subtitle (PGS), Portuguese
17: Subtitle (PGS), English
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[s08] Extracting subtitle track number 8...
[s09] Extracting subtitle track number 9...
[a03] Extracting TrueHD stream...
[a03] Removing TrueHD dialog normalization...
[a03] Decoding with libav/ffmpeg...
[a03] Writing WAVs...
[v02] Muxing video to Matroska...
[a03] Creating file "i:\M.R.wav"...
[a03] Creating file "i:\M.L.wav"...
[a03] Creating file "i:\M.C.wav"...
[a03] Creating file "i:\M.SR.wav"...
[a03] Creating file "i:\M.SL.wav"...
[a03] Creating file "i:\M.LFE.wav"...
[s09] Creating file "i:\M.2.sup"...
[s08] Creating file "i:\M.1.sup"...
[libav] End of stream indicated
[libav] Lossless check failed - expected 0, calculated 8b
[a03] This audio track has a constant bit depth of 24 bits.
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.21.0.
Surcode encoding successfully started. Please wait...
Closing Surcode...
Added fps value to MKV header.
Video track 2 contains 123591 frames.
eac3to processing took 27 minutes, 38 seconds.
Surcode encoding took 13 minutes, 18 seconds.
Done.

rebkell

5th October 2008, 21:48

If you ask eac3to to ignore the discontinuities, does demuxing/transcoding succeed then? And do you then get any audio or video corruptions during playback? Just wondering whether these discontinuities are the result of a buggy broadcasting setup. Or whether there is real corruption in the data?

They seem to vary, I recorded Sanctuary the other night on SciFi and eac3to reported 2 discons one at
3518061672
3518061860
and another at
7045026084
7045025896

on the rerun the first set at 3518061572 were the same and thethe ones around the 7GB mark it reported them in reverse order:
7045025895
7045026084

I muxed the audio you repaired back in and the first one which I'm assuming was the same as an audio gap of 750ms reported at 54:39 didn't appear to have any problems and the audio went out of sync.

You also reported an audio gap of 745ms at 1:49:07, which would probably coincide with the discon at the 7GB mark, there was definitely a problem in that area in the original, because that was where muxing originally caused the audio/video to go out of sync, and 745ms seems probably about right.

nautilus7

5th October 2008, 21:50

eac3to does not OCR the message box of the GUI, obviously, to get the version number. Instead eac3to captures what "mkvmerge --version" says. Sometimes that information is not correct or sometimes mkvmerge is simply older than the GUI. If that is the case there's nothing I can do about it. I won't add OCR to eac3to just to get the version number/date of the GUI!! It can happen that Mosu recompiled only the GUI or only mkvmerge in a new beta build. Since eac3to can only get the version number of mkvmerge, there is potential for incorrect reports. Can't do anything about that, unless Mosu adds programatically readable version information to the GUI, too.
That explains. OCR is not needed of course, it's not important at all. Thanks!

madshi

6th October 2008, 08:16

You probably missed my post
Not really, but your question has already been answered about a dozen times earlier in this thread. Just do a search for "Lossless check failed".

I muxed the audio you repaired back in and the first one which I'm assuming was the same as an audio gap of 750ms reported at 54:39 didn't appear to have any problems and the audio went out of sync.

You also reported an audio gap of 745ms at 1:49:07, which would probably coincide with the discon at the 7GB mark, there was definitely a problem in that area in the original, because that was where muxing originally caused the audio/video to go out of sync, and 745ms seems probably about right.
Ok, thanks. Still not sure whether the broadcasting is faulty or whether your recording was faulty. But probably in the end it doesn't matter much...

That explains. OCR is not needed of course, it's not important at all. Thanks!
I've asked Mosu yesterday whether he can add a version resource to the GUI. That would solve the problem...

nautilus7

6th October 2008, 10:13

I've asked Mosu yesterday whether he can add a version resource to the GUI. That would solve the problem...I 've read your request! :p

Jaja1

6th October 2008, 12:53

Not really, but your question has already been answered about a dozen times earlier in this thread. Just do a search for "Lossless check failed".
I know this used to show up on the join spots of seamless branching movies and that this wasn't a problem anymore, but this one is a single file. I see however that this isn't clear from the log or post, stupid of me. So the same "no problem" applies to non-seamless branching movies as well?

madshi

6th October 2008, 13:12

If it's after "[libav] End of stream indicated" then it doesn't matter.

rica

6th October 2008, 22:02

Hi guys,
I've been making some trials for two days.
Finally i can say it is possible to demux directly from BD/HD Disc to HDD with eac3to; which means lots of unwanted things won't be ripped on your disk. AnyDVD-HD is working on the back stage btw.
I selected one of the most difficult ones of my collection which including many m2ts files in one playlist for test.
No issues.

getting playlist info:

C:\>eac3to\eac3to.exe E:
1) 00056.mpls, 2:17:14
[0+2+6+8+9+12+13+15+16+19+20+21+24+25+26+28+118+119+32+33+34+37+39+40+42+43+45+46+49].m2ts
- h264/AVC, 1080p24 /1.001 (16:9)
- TrueHD, English, multi-channel, 48khz
- DTS Master Audio, English, multi-channel, 48khz
**************************

getting track info:

C:\>eac3to\eac3to.exe E: 1)
M2TS, 1 video track, 2 audio tracks, 20 subtitle tracks, 2:17:13
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
5: Subtitle (PGS), English
****************
First Pass:

C:\>eac3to\eac3to.exe E: 1) 2: G:\Out\video_BD.h264 4: G:\Out\audio.dtshd

M2TS, 1 video track, 2 audio tracks, 20 subtitle tracks, 2:17:13

1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
5: Subtitle (PGS), English
**************
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[v02] Creating file "G:\Out\video_BD.h264"...
[a04] Creating file "G:\Out\audio.dtshd"...
[a04] Audio overlaps for 7ms at playtime 0:23:06.
[***********************************
[a04] Audio overlaps for 10ms at playtime 2:11:04.
[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/over
laps.
Video track 2 contains 197424 frames.
eac3to processing took 1 hour, 13 minutes.
Done.

Second Pass:

C:\>eac3to\eac3to.exe E: 1) 2: G:\Out\video_BD.h264 4: G:\Out\audio.dtshd

M2TS, 1 video track, 2 audio tracks, 20 subtitle tracks, 2:17:13
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
5: Subtitle (PGS), English

Audio gap description file detected, will be used for processing...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Realizing DTS gaps...
[v02] Creating file "G:\Out\video_BD.h264"...
[a04] Creating file "G:\Out\audio.dtshd"...
Video track 2 contains 197424 frames.
eac3to processing took 1 hour, 12 minutes.
Done.

saint-francis

7th October 2008, 00:15

rebkell

7th October 2008, 00:21

Wow, I've been doing this for the last 2 or 3 months or so, ever since Madshi got the playlists working for blu-rays, and I've never had any problems.

rica

7th October 2008, 00:25

If you look back to the early days of this thread you will see that many people attempting what you are ran into many problems. Maybe anydvd has improved since then, I don't know. The one time I tried it I got eac3to to successfully process the video but it had all kinds of crazy colors from time to time. The only solution to the problem was to rip the disk first.

No, i think you have to update.
I also asked slysoft guys.
I think the problem has been solved.
Why not give it a go?

rica

7th October 2008, 00:34

Wow, I've been doing this for the last 2 or 3 months or so, ever since Madshi got the playlists working for blu-rays, and I've never had any problems.

Thanks for the feedback but why didn't you share this before; if i've missed any post from you sorry in advance.

saint-francis

7th October 2008, 00:47

Wow, I've been doing this for the last 2 or 3 months or so, ever since Madshi got the playlists working for blu-rays, and I've never had any problems.

Crazy. I follow this thread religiously and I haven't seen any mention of people doing this since everyone generally decided that it was unsafe.

I think the problem has been solved.
Why not give it a go?

Thanks, I will!

peterjcat

7th October 2008, 06:11

I also assumed that people were just using eac3to straight from the disc with AnyDVD HD in the background! I only ever rip to HDD first when I know I'll have to do more than one pass for whatever reason. When I've had problems, ripping to HDD first has never fixed them. I think realtime AnyDVD HD must be pretty much transparent by now.

rebkell

7th October 2008, 06:28

Thanks for the feedback but why didn't you share this before; if i've missed any post from you sorry in advance.

I never intended to hold anything back, every example I've ever used always started off with eac3to d: , with d: being my Blu-ray/HD-DVD drive, I just assumed everyone was going straight from their drives. Sorry guys, I really thought everyone was just using their drive in combination with anydvd.

fib0by

7th October 2008, 06:35

EPiPH0NE

7th October 2008, 07:29

Crazy. I haven't seen any mention of people doing this since everyone generally decided that it was unsafe.

Been doing it for ages now with zero issues.

rica

7th October 2008, 19:13

Hey guys, thanks for the feedbacks.

fofwp

8th October 2008, 06:24

I have an issue with Constantine and Serenity HD-DVD's. I converted the audio for both these titles to ac3 (using libav), and it appears that at some places the audio clips.

I can upload a small clip of a particular scene in Constantine where this is most prominent. The scene is a flashback when he is a kid and he sees a demon on a bus. However, I don't know how to cut this small portion out of the main mkv. Any help?

madshi

8th October 2008, 07:46

Trying to demux files recorded with MythTV from digital HD cable. eac3to reports a very large time difference for the audio track, over 1000 ms. How is that possible?

E:\>eac3to E:\debate\1091_20080926180000.mpg
TS, 1 video track, 3 audio tracks, 1:59:57
1: MPEG2, 1080i60 /1.001 (16:9)
2: AC3, English, 2.0 channels, 384kbps, 48khz, dialnorm: -24dB, -1150ms
3: AC3, French, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -1092ms
4: AC3, esl, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -1059ms

Here's a 50 MB sample:

http://www.sendspace.com/file/b12wpo

The file is played fine by MythTV, audio and video are in sync.

I demuxed the sample with eac3to and remuxed the tracks with tsMuxeR. The final result played fine, audio and video were in sync, but a chunk at the beginning of the video, about a second long, was missing - so it looks like eac3to dropped some video to compensate for the time difference. That's fine, it doesn't bother me too much.

So all is well, more or less, but I'm puzzled by the size of the difference. Is that how HD is being transmitted? Or is it something that MythTV did?
That's how it's transmitted, sometimes. Nothing wrong with that. As long as audio sync is correct there's nothing to worry about. Although I'll check why the first video frames seem to be missing. Shouldn't really be the case. eac3to dismisses non-"I" frames at the beginning of the stream. But these non-"I" frames should not decode correctly, anyway.

P.S: Just checked: The first video frames are in the file before the file information package. So basically in order to understand the first video frames eac3to would have to first read a part of the file to find the information package and then skip back to the beginning and reread everything. That's not how eac3to works right now. But it should be possible to do it. I'll put it on my to do list.

BTW, what is that third audio track? The first two audio streams have the language identifiers "eng" (English) and "fra" (French). But the third has the language identifier "esl" which I don't know how to interpet.

I have an issue with Constantine and Serenity HD-DVD's. I converted the audio for both these titles to ac3 (using libav), and it appears that at some places the audio clips.

I can upload a small clip of a particular scene in Constantine where this is most prominent. The scene is a flashback when he is a kid and he sees a demon on a bus. However, I don't know how to cut this small portion out of the main mkv. Any help?
I'd need the original E-AC3 data for that scene. You can demux the original data by using eac3to. Then you can use delaycut to cut it down.

deathlord

8th October 2008, 14:35

A sample would be nice. Are you sure that the back channels are silent throughout the whole movie? I mean they may be silent for most of the movie, but maybe they're used only once in a while?

Sorry for taking so long. I have scanned through the file several times and the back channels are always at -inf. However, I don't know how to systematically check if there really is no signal whatsoever.
Here is a sample:
http://www.megaupload.com/?d=GLQ2UP05

deathlord

fib0by

9th October 2008, 05:47

That's how it's transmitted, sometimes. Nothing wrong with that. As long as audio sync is correct there's nothing to worry about.

Right, I was just curious.
So, you're saying, that's how it comes through the cable. Well, I just recorded another show, same channel, and the values are different:

E:\debate>eac3to debate2.mpg
TS, 1 video track, 3 audio tracks, 1:59:55
1: MPEG2, 1080i60 /1.001 (16:9)
2: AC3, English, 2.0 channels, 384kbps, 48khz, dialnorm: -24dB, -872ms
3: AC3, French, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -846ms
4: AC3, esl, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -813ms

Weird. Why would they change the delay all the time?

BTW, what is that third audio track? The first two audio streams have the language identifiers "eng" (English) and "fra" (French). But the third has the language identifier "esl" which I don't know how to interpet.

I have no idea. The content of all 3 tracks seems to be the same (english).

ACrowley

9th October 2008, 07:27

can anybody give me a link to some 7.1 samples (dtshd,truehd) ?

edit:
i have some samples now

Greif

10th October 2008, 00:23

BTW, what is that third audio track? The first two audio streams have the language identifiers "eng" (English) and "fra" (French). But the third has the language identifier "esl" which I don't know how to interpet.

I've never seen it before, but it probably refers to English as a Second Language.

fib0by

10th October 2008, 00:24

Would it be possible to have a simpler chapter listing?
Right now, from the chapters list provided by eac3to, I filter out any line containing the string "NAME"; then, from the remaining lines, I only retain everything after the "=" sign. Like this:

cat chaps.txt | grep -v NAME | cut -d = -f 2 > chapters.txt

(This is Bash shell, yes, but it's still on Windows, under Cygwin. I haven't tried to run eac3to on Linux under emulation yet.)

So, what I'm saying is, it would be nice if eac3to could simply dump the list of timestamps, one timestamp per line. No names, no equal signs, nothing else, just the "naked" timestamp values. I never had a need for anything else from the current format of the chapters list and I don't even know if there's a case when that information could be useful - it seems redundant to me.
The "naked" list could easily be imported in tsMuxeR or other applications that simply require a list of timestamps, no need to filter out other things.

Or even more generic: create a list of "naked" timestamps, and let the user choose the separator. Users of the tsMuxeR GUI version will prefer a list separated by newline, because that's how tsMuxeR GUI wants the list. Users of the command-line muxer would prefer comma as a separator, because that's the way to pass the list to the command-line muxer (-blah=timestamp1,timestamp2,timestamp3.......).

If not possible, then OK. I can continue to do the grep thing. It's just a feature that's nice to have.

Zwitterion

10th October 2008, 02:35

Would it be possible to have a simpler chapter listing?
Eac3to currently uses the OGG format which is used in MKVmerge and MP4box.
I think too many chapter formats would bloat eac3to unnecessarily. I'd prefer a simple tool which can convert between various chapter formats (is there such a thing?).

odin24

10th October 2008, 03:02

Eac3to currently uses the OGG format which is used in MKVmerge and MP4box.
I think too many chapter formats would bloat eac3to unnecessarily. I'd prefer a simple tool which can convert between various chapter formats (is there such a thing?).

Subtitle Workshop.

http://www.urusoft.net/download.php?lang=1&id=sw

rack04

10th October 2008, 04:05

Trying to work with Ratatouille. Anybody know what the difference is in the angles?

http://i11.photobucket.com/albums/a199/rack04/untitled-1.jpg

Octo-puss

10th October 2008, 08:40

What exactly does this message mean?
[a05] The libav DTS decoder doesn't decode the full DTS-HD information
It processes the file, but I wonder how big problem it is.
The first post mentions some decoders. How do I make eac3to work with them? Automatically by installing them?

Where do I obtain the libav/ffmpeg TrueHD? This one seems to be free by looks of it and since it's said it works fine, I thought it'd be the thing.

nautilus7

10th October 2008, 08:48

Trying to work with Ratatouille. Anybody know what the difference is in the angles?
You can play the different .m2ts files to figure out what's about. Most probably it has to do with language specific parts.

What exactly does this message mean?
[a05] The libav DTS decoder doesn't decode the full DTS-HD information
It processes the file, but I wonder how big problem it is.It means that only the DTS core of the track will be decoded and used as source. If you want t make a flac from this track, then it is a serious problem.

The first post mentions some decoders. How do I make eac3to work with them? Automatically by installing them? Yes, as long they are installed in your system they work with eac3to. the command eac3to -test will give a report about the installed decoders.

Where do I obtain the libav/ffmpeg TrueHD? This one seems to be free by looks of it and since it's said it works fine, I thought it'd be the thing. The libav decoder comes with eac3to package.

Octo-puss

10th October 2008, 08:59

It means that only the DTS core of the track will be decoded and used as source. If you want t make a flac from this track, then it is a serious problem.
That I don't understand. What is the DTS core?
The source is DTS Master Audio, variable bitrate from what I saw in MediaInfo. All I did was tell eac3to to save it as .dts.

Yes, as long they are installed in your system they work with eac3to. the command eac3to -test will give a report about the installed decoders.

The libav decoder comes with eac3to package.
Well, if it comes with the program, I don't understand why I got the above message, when the first post states it works perfectly well with DTS.
Doing eac3to - test provided no information about it, btw.

nautilus7

10th October 2008, 09:06

DTS core is a DTS track inside the DTS-HD track.

Libav works with fine with DTS, not DTS-HD. It is only able to decode DTS tracks thus only the DTS core of the DTS-HD track.

Libav isn't listed under the -test command, because it isn't a directshow decoder and doesn't need installation.

Octo-puss

10th October 2008, 09:13

Aha, so it's just a little bit lower quality, but nothing too bad?
I am looking at arcsoft.com and can't seem to find any word about the dts decoder. What gives?

nautilus7

10th October 2008, 09:20

Well, i wouldn't say a little bit lower, since DTS-HD master audio is a lossless format, while DTS is lossy, but all depends on your ears and sound equipment.

Have a look at the wikibooks page of eac3to (http://en.wikibooks.org/wiki/Eac3to) under the how to use section to find info about the needed decoders.

Octo-puss

10th October 2008, 09:33

thanks a lot dude

himan2001

10th October 2008, 11:21

Hello Madshi.

I have strange problems with the Seemless Branching Movie Harold & Kumar 2:

on EVERY gap-fixed Point is a Loud Spike in the AC3 encoded file. As i remember older Versions doesn´t have this Problem.

There is a special reason for using Sonic instead ArcSoft: Arcsoft produces sometimes distorted audio while downmixing from 7.1 -> 5.1:

eac3to v2.66
command line: eac3to 10) 2: d:\video.mkv 3: d:\audio.ac3 -448 -sonic
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 2 subtitle tracks, 1:47:41
1: Chapters, 23 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 channels, 24 bits, 48khz
4: DTS Master Audio, English, 2.0 channels, 24 bits, 48khz
5: DTS, English, 2.0 channels, 24 bits, 256kbps, 48khz
6: Subtitle (PGS), English
7: Subtitle (PGS), Spanish
[a03] The Sonic Audio Decoder doesn't fully decode 7.1 channels.
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Decoding with DirectShow (Sonic Audio Decoder)...
[v02] Muxing video to Matroska...
[a03] DirectShow reports 5.1 channels, 24 bits, 48khz
[a03] Encoding AC3 <448kbps> with libAften...
[a03] Creating file "d:\audio.ac3"...
[a03] This audio track has a constant bit depth of 24 bits.
[a03] Audio overlaps for 10ms at playtime 0:10:23.
[a03] Audio overlaps for 6ms at playtime 0:11:23.
[a03] Audio overlaps for 9ms at playtime 0:11:48.
[a03] Audio overlaps for 8ms at playtime 0:12:47.
[a03] Audio overlaps for 10ms at playtime 0:19:52.
[a03] Audio overlaps for 9ms at playtime 0:22:30.
[a03] Audio overlaps for 15ms at playtime 0:45:02.
[a03] Audio overlaps for 10ms at playtime 0:48:46.
[a03] Audio overlaps for 7ms at playtime 0:49:10.
[a03] Audio overlaps for 6ms at playtime 1:01:05.
[a03] Audio overlaps for 11ms at playtime 1:02:25.
[a03] Audio overlaps for 7ms at playtime 1:05:19.
[a03] Audio overlaps for 10ms at playtime 1:06:19.
[a03] Audio overlaps for 9ms at playtime 1:20:48.
[a03] Audio overlaps for 6ms at playtime 1:36:53.
[a03] Audio overlaps for 7ms at playtime 1:38:12.
[a03] Audio overlaps for 5ms at playtime 1:39:25.
[a03] Audio overlaps for 8ms at playtime 1:40:22.
[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.
Added fps value to MKV header.
Video track 2 contains 154945 frames.
eac3to processing took 37 minutes, 46 seconds.
Done.

I found out that with actual 2.66-Version it´s no longer possible to decode the audio from german HD-DVD "Terminator 2" with ARCSoft DTS-Decoder.
This HD-DVD has two 7.1 Audios.

eac3to stops with invalid Channel-Table, when using ArcSoft.
Set Deocder to -sonic then it works fine - by decoding 5.1 only.

Another issue i found by decoding "Superman-The Movie".
(5.1/AC3/640 -> 5.1/AC3/448)
The Result was defect/unusuable/crap. (an AC3-File with invalid timecode and only noise in it)

Going the manual way: .wavs -down16 -> encode the wavs to 5.1 was working fine. This Movie has 2,5 hours Playtime.
Maybe a new boundery bug (4 GB, or something) ?

I testet an older Version of eac3to (2.58) and figured out, that this issues here is not present. ?! Hmm....

Edit:

ac3to v2.66
command line: eac3to 10) 3: d:\audioARCsoft.ac3 -448
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 2 subtitle tracks, 1:47:41
1: Chapters, 23 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 channels, 24 bits, 48khz
4: DTS Master Audio, English, 2.0 channels, 24 bits, 48khz
5: DTS, English, 2.0 channels, 24 bits, 256kbps, 48khz
6: Subtitle (PGS), English
7: Subtitle (PGS), Spanish
Audio gap description file detected, will be used for processing...
[a03] AC3 encoding only supports 6 channels. Will mix the surround channels.
[a03] Extracting audio track number 3...
[a03] Mixing surround channels...
Loading white noise (needed for dithering)...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding AC3 <448kbps> with libAften...
[a03] Realizing RAW/PCM gaps...
[a03] Creating file "d:\audioARCsoft.ac3"...
[a03] This audio track has a constant bit depth of 24 bits.
Video track 2 contains 154945 frames.
eac3to processing took 27 minutes, 47 seconds.
Done.

Retested it with Arcsoft and has same result. On All fixed gaps loud sparkles...

Edit2:

Found a working Solution:

eac3to v2.66
command line: eac3to 10) 3: d:\audio.dts -core
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 2 subtitle tracks, 1:47:41
1: Chapters, 23 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 channels, 24 bits, 48khz
4: DTS Master Audio, English, 2.0 channels, 24 bits, 48khz
5: DTS, English, 2.0 channels, 24 bits, 256kbps, 48khz
6: Subtitle (PGS), English
7: Subtitle (PGS), Spanish
Audio gap description file detected, will be used for processing...
[a03] Extracting audio track number 3...
[a03] Extracting DTS core...
[a03] Realizing DTS gaps...
[a03] Creating file "d:\audio.dts"...
Video track 2 contains 154945 frames.
eac3to processing took 15 minutes, 6 seconds.
Done.

and then:

eac3to v2.66
command line: eac3to audio.dts audio.ac3 -448
------------------------------------------------------------------------------
DTS, 5.1 channels, 1:47:43, 24 bits, 1536kbps, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Encoding AC3 <448kbps> with libAften...
Creating file "audio.ac3"...
eac3to processing took 17 minutes, 17 seconds.
Done.

The DTS-Core is only 5.1 and was gap-fixed without problems.
I checked the final AC3 result on ALL gap-marks/timecodes and file is OK!

So using 7.1? or DTS-HD directly (without core) there are problems in the fill/add/appendpoint/whatever/? routine...

Can you check what is wrong when gap-fixxing the Raw DTS-HD Stream instead the core in the 2.nd fixx-run ?

i tryed the last step with version 2.65 and version 2.66 and booth encoded ac3-files are ok NOW - but in version 2.66 the final file was 2 AC3-frames shorter as when using 2.65.

frank

10th October 2008, 19:31

rica wrote:
Remuxing of BDs including multiple m2ts files with TSMuxer doesn't correct or remove the gaps/overlaps; so they have to be re-processed by eac3to,
eac3to can not correct gaps/overlaps of True HD files on those BDs, so they have to be transmitted to any other uncompressed or lossless files like pcm or wav or flac.
There is an easier way.
Load the re-encoded video stream into tsMuxer. Then load the playlist .mpls into tsMuxer, select only the wanted original audio stream and mux direct to Blu-ray.
tsMuxer respects the audio gaps!
You will get a synchronous BD.

rica

10th October 2008, 19:59

rica wrote:

There is an easier way.
Load the re-encoded video stream into tsMuxer. Then load the playlist .mpls into tsMuxer, select only the wanted audio stream and mux direct to Blu-ray.
tsMuxer respects the audio gaps!
You will get a synchronous BD.

Frank,

I did try this before, as is.
I got the playlist information from BDInfo and directly remuxed needed files.(via TSMuxer)
When i tried to demux back with eac3to, i saw the gaps/overlaps again in created m2ts under BDMV.

Is any tip/trick in loading re-encoded video; i didn't understand? Why re-encoded video?

frank

11th October 2008, 10:41

No need to reencode if you want to stay at 1920x1080.
But I only use 720p, burning on DVD-9. That's why I reencode.

Don't demux the sound track with these gaps!
If you need the separated track you have to process with eac3to that handles the gaps.

Switch to the ripped BD folder and start the command
<path> eac3to.exe .Watch the output and then make your batch.

Example:
d:\eac3\eac3to.exe 2) 4: cetk.ac3

eac3to needs some seconds to search subfolders for the needed mpls and starts demuxing the core.
Make 2 runs. The second one compensates the gaps.

It works with TrueHD!

rica

11th October 2008, 10:55

Frank,
what i am trying to say is TSMuxer remuxing doesn't correct the gaps and overlaps. So those gaps/overlaps must be corrected by eac3to before remuxing with TSMuxer.

madshi

11th October 2008, 11:03

Not necessarily. Those combined m2ts files with overlaps in them can still play correctly, if the media player is clever enough. Personally, I don't like the idea of having such "dirty" m2ts movie files, though, where you depend on the media player working around the problem...

frank

11th October 2008, 11:06

No. As I said, TsMuxeR respects the timestamps and gaps exactly if you mux the sound via playlist .mpls. Movie plays the same way as with gaps (that are only discontinuities in timestamps) on the original BD.
On remuxing you can select the AC3 core in TsMuxeR.

If you remux a demuxed track then TsMuxeR combines all sound packets continously. Gaps are not respected and the result is asynchronous.

I managed the same movie as you and all things are ok.

rica

11th October 2008, 11:09

I've been really confused :confused:

Edit:Think about i first get an m2ts remux via TSMuxer (directly from mpls)
Demux audio with TSMuxer(as is)/not with eac3to and
Remux back with TSMuxer.
What happens?
Will TSMuxer respect the timestamps this time since audio has been demuxed by itself?

himan2001

11th October 2008, 17:36

I found another small bug:

When a DTS 7.1 Track is in a m2ts-File and try to encode the -CORE 5.1 to ac3; eac3to ignore the -core setting
and will downmix the DTS-HD from 7.1 to 5.1 with Arcsoft.

Demuxing the -core from the file first and encode the resulting 5.1 core later is OK.

Greif

11th October 2008, 18:02

Try out this new GUI and let me know what you think:

http://forum.doom9.org/showthread.php?p=1193742#post1193742

http://stuff.griffore.com/stuff/HdBrStreamExtractor.jpg

shon3i

11th October 2008, 18:38

Try out this new GUI and let me know what you think:

http://forum.doom9.org/showthread.php?p=1193742#post1193742

http://stuff.griffore.com/stuff/HdBrStreamExtractor.jpg
http://img410.imageshack.us/img410/3444/29906746cs2.jpg

I think i have problem :) On mine sistem i have Net Framework 2.0 SP1 installed, i put this exe into eac3to folder and just started app and i get error :)

nautilus7

11th October 2008, 23:08

flyingernst

12th October 2008, 12:22

when I demux TrueHD Files to flac (without options), they only get decoded/encoded in 16bit? Would it be better to hang on any options?!

decode with ffmpeg:
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Encoding FLAC with libFlac...
[a04] Creating file "E:\legend\englisch.24bit.flac"...
[libav] End of stream indicated
[libav] Lossless check failed - expected 0, calculated 15
[a04] This audio track has a constant bit depth of 16 bits.
[a04] The zero bytes were successfully removed.
Video track 2 contains 149472 frames.
eac3to processing took 13 minutes, 52 seconds.
Done.

decode with nero

[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with DirectShow (Nero Audio Decoder 2)...
[a04] DirectShow reports 5.1 channels, 24 bits, 48khz
[a04] Encoding FLAC with libFlac...
[a04] Creating file "E:\legend\englisch2.24bit.flac"...
[a04] This audio track has a constant bit depth of 16 bits.
[a04] The zero bytes were successfully removed.
Video track 2 contains 149472 frames.
eac3to processing took 15 minutes, 6 seconds.
Done.

which options should i use so that a 24bit TrueHD does not get downconverted
Thanks, Greetings, Michael

tebasuna51

12th October 2008, 13:06

...
[a04] This audio track has a constant bit depth of 16 bits.
...

which options should i use so that a 24bit TrueHD does not get downconverted

Isn't downconverted, the initial source was 16 bit depth.
:search:

odin24

12th October 2008, 15:21

Guys, I'm trying to work with an HDDVD structured folder but eac3to doesn't recognize it. I get the error "HDDVD / Blu-ray disc structure not found", when in fact I havn't modified it in any way.

I also noticed the "feature_0.evo" and "feature_1.evo" are actually named feature_|0.evo and feature_|1.evo, and there is no VPLST000.XPL file, just 001 and 003.

This is my first attempt at an HDDVD, thanks for the help.

madshi

12th October 2008, 15:25

Try renaming one of the VPLST files to VPLST000.XPL.

frank

12th October 2008, 17:36

@rica
Demux audio with TSMuxer(as is)/not with eac3to and
Remux back with TSMuxer.
What happens?
Will TSMuxer respect the timestamps this time since audio has been demuxed by itself? No, you'll get asynchronous results.

nautilus7

12th October 2008, 17:52

Why do people keep talking about tsmuxer here. This i can't understand!

@madshi, I have a feature request.

Since DTS Pro Encoder is spreading more and more and we can create 6.1 DTS files, i am thinking of some additions.

-down7 switch which takes a 7.1 track and convert it to 6.1 by mixing together the BL and BR channels to BC.
-proper patch for the agm writer to support 6.1 if not already done.
-if output is agm and source 7.1 and no downmix option is specified, then automatic downmix to 6.1 should be applied as in 6/7.1 --> dts with surcode.

Also why -down6 isn't working on 7.1ch TrueHD tracks? (eac3to input.thd output.wav -down6)

rica

12th October 2008, 18:33

I don't understand also why you don't understand?
The question was, as a demuxer, finding an answer to the investigation of the advantages of eac3to over TSMuxer.
If any?
As a direct remuxer, after all those investigations TSMuxer seems more advantageous? _if you don't demux.
As a demuxer eac3to is the winner in this case?

note: Frank, thanks btw.

EDIT: Guys, please tell me if i have to correct my post here?

http://forum.doom9.org/showthread.php?p=1192816#post1192816

flyingernst

12th October 2008, 19:30

ok, last problem solved...new one:

First:
- I have a ATi48XX with 7.1 Onlyo TX-SR805. The Driver-Channeltest is correctly mapping.

- the informations how much channels the source has comes from the ffdshow audio filter:
- eac3to is converting 5.1 dts in a 7.1 Flac with 2 quiete channels
- eac3to is converting 6.1 dts-es flac in a 7.1 Flac (with 2 same back channels)
- eac3to with option "-down6" is converting 6.1 dts-es flac in a 7.1 Flac with 2 quiete channels (like 5.1, same mapping)
- for this test I play my mkv with MPC-HC, Channel-remapping is off, AudioRenderer is ffdshow with "mixer" acticated to 3/2/2 (8 Channel) + LFE

In my TEstfile there are 3 Audio Files:
1) 7.1 FLAC Stream made off the 6.1 DTS-ES Stream (converted without options)
2) original 6.1 DTS-ES Stream
3) "5.1" FLAC Stream (converted without "-down6" option)

If I play the Video (compared only the first few seconds) with
Audio 1) is playing the violin music in the sides, Backsurrounds are nearly quiete (ignore sizzling through converting)
Audio 2) (ORIGINAL FILE/STREAM) is playing the violin music in the sides, Backsurrounds are nearly quiete, like Audio 1)
Audio 3) is playing the violin music in the Backsurrounds, sides are complete quiete (ignore sizzling through converting)

The Problem is:

A 5.1 Surround System consits off Front / Center / Surrounds(also called sides today)

A 7.1 System has additional Backsurround Speakers used in EX/ES Audio (standing at the back wall positioned left and right of the middle of the wall, not on the corner)
http://http://www.thx.com/home/setup/img/dolby.jpg (red ones in the picture)

So a 5.1 File should play in the sides(Surrounds) and a 7.1 File additional in the Backsurrounds.

Now the Problem: Look at my test, the 5.1 FLAC File playes in the Backsurround.
My fix until now (to get every Source playing like it should):
- further using the ffdshow decoder
- no using of remapping tool in MPC-HC
- When I´m playing original 5.1 Files, I set the ati to 5.1 - so all 5.1 sources play fine
- When I´m playing original 7.1 Files, I set the ati to 7.1 - so all 7.1 sources play fine

Is there no way to let 7.1 always activated withoud swapped channels?
and now, who is wrong? me? my Flac Decoder? eac3to?

http://rapidshare.com/files/153354718/testvideo.mkv.html

Greetings, Michael

PS: For me it seams that DTS has changed its configuration of channels...I don´t understand it right

nautilus7

12th October 2008, 19:39

now, who is wrong? me? my Flac Decoder? eac3to?
Microsoft!

They have introduced new channel masks with windows xp sp2 which resulted in chaos.

Newer hardware (like yours - ATI) use the new channel mask, while most audio software the older. madshi, while thinks he should follow the newer masks, he hasn't made that changes to eac3to and madflac, for backwards compatibility sake.

madshi

12th October 2008, 19:46

- eac3to is converting 5.1 dts in a 7.1 Flac with 2 quiete channels
No, it doesn't, unless you ask it to. Or unless there's a new bug in eac3to.

- eac3to is converting 6.1 dts-es flac in a 7.1 Flac (with 2 same back channels)
No, the newer builds don't, unless you ask them to. Or unless there's a new bug in eac3to.

- eac3to with option "-down6" is converting 6.1 dts-es flac in a 7.1 Flac with 2 quiete channels (like 5.1, same mapping)
No, it doesn't, unless you ask it to. Or unless there's a new bug in eac3to.

Now the Problem: Look at my test, the 5.1 FLAC File playes in the Backsurround.
That's probably a bug either in Windows or in your audio driver. For 5.1 FLAC files eac3to tells Windows that it's a 5.1 track and Windows is supposed to send the channels to the right speakers. If it doesn't, there's not much I can do about it. *However*, there are 2 different ways to tell Windows that it's a 5.1 track and I'm planning to let madFlac behave different in the next madFlac build. Either way, even if that fixes the problem for you, it's not my fault to begin with.

madshi

12th October 2008, 19:47

madshi, while thinks he should follow the newer masks, he hasn't made that changes to eac3to and madflac, for backwards compatibility sake.
Actually I did change the eac3to channel masks 2 or 3 releases ago. No problems reported until now. madFlac still uses the old masks, though.

nautilus7

12th October 2008, 19:50

Oh sorry! :o

Must have forgotten it.

madshi

12th October 2008, 19:51

eac3to v2.67 released

http://madshi.net/eac3to.zip

* information about HDCD and real bitdepth is now stored into FLAC metadata
* information about real bitdepth is now read from FLAC metadata
* PTS break: PTS is increased by 1 frame (fixes some false overlap warnings)
* fixed: video gap log text was sometimes not correct (runtime information)
* added undocumented switch "-neroaacenc="c:\whatever\neroaacenc.exe""
* error log messages are now output to stderr instead of stdout
* improved "which mkvtoolnix is currently installed?" check
* fixed: mkvtoolnix version check "Oct" date was not interpreted correctly

flyingernst

12th October 2008, 20:19

No, it doesn't, unless you ask it to. Or unless there's a new bug in eac3to.

No, the newer builds don't, unless you ask them to. Or unless there's a new bug in eac3to.

No, it doesn't, unless you ask it to. Or unless there's a new bug in eac3to.

That's probably a bug either in Windows or in your audio driver. For 5.1 FLAC files eac3to tells Windows that it's a 5.1 track and Windows is supposed to send the channels to the right speakers. If it doesn't, there's not much I can do about it. *However*, there are 2 different ways to tell Windows that it's a 5.1 track and I'm planning to let madFlac behave different in the next madFlac build. Either way, even if that fixes the problem for you, it's not my fault to begin with.

You are right. 7.1 gets 7.1, 6.1 gets 6.1 and 5.1 gets 5.1...sorry I now decoded and checked with madFlac 1.8 in MPC-HC...seems ffdshow got te wrong information (showd always 8-channel)
sadly I can´t play the 7-channel FLAC from my testfile with madflac decoder in MPC-HC. Same problem is with X-Men 3 6.1 DTS-HD which I converted to flac. Only ffdshow can play the 6.1 flacs

So I made the test-row with mad flac/7.1 Ati Setting/no channelremapping in MPC-HC
With this setting the Realtek Channel test plays every speaker like it should/P-DVD Playes 5.1 sources in the surrounds(sides)/the microsoft 8-channel wav playes like it is spoken

but

Audio 2) with 6.1 DTS-ES Stream (streamed through HDMI/SPDIF) playes like it should
Audio 3) with 5.1 FLAC playes through the Backsurrounds instead of the surrounds
Audio 1) (6.1 FLAC) is not playable

Greetings Michael

PS: Microsoft Channel Test wavs: http://www.microsoft.com/windows/windowsmedia/howto/articles/Multichannel.aspx#link6
quanit is:
the 8-channel test says "Front L / center / Front R / side L / back L / Back R / side R / Subwoofer" so everything is said and played right
the 6-channel test says "Front L / center / Front R / back L / Back R / Subwoofer" and get also played in the called channels...but microsoft things a 5.1 setting has backs instead of sides...

madshi

12th October 2008, 20:25

Audio 3) with 5.1 FLAC playes through the Backsurrounds instead of the surrounds
I've already replied to that.

Audio 1) (6.1 FLAC) is not playable
Not my fault. Might depend on the audio renderer.

Further madFlac discussion please in the madFlac thread.

flyingernst

12th October 2008, 20:46

EDIT: wrote to madflac Forum

nautilus7

12th October 2008, 21:13

Ive been dealing with some 44,1KHz audio the last few hours and noticed that eac3to always detects RAW 44,1KHz as 48KHz (at least for 5.1ch). I know how this automatic detection works and that it's not 100% accurate, but i wonder if anything can be done about this.

madshi

12th October 2008, 21:27

It is nearly impossible (at least for me) to programatically detect the samplerate of raw/pcm audio.

odin24

12th October 2008, 22:48

Try renaming one of the VPLST files to VPLST000.XPL.

I tried that before posting, it didn't work. Although it did help a little with EVOdemux, I was able to view chapter points and other data, but not stream information, and still unable to demux with eac3to and EVOdemux.

flyingernst

12th October 2008, 22:57

EVO, 2 video tracks, 4 audio tracks, 8 subtitle tracks, 2:20:02
1: Joined EVO file
2: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
3: VC-1, 480p30 /1.001 (3:2), 100ms
4: E-AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB, 100ms
5: E-AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB, 100ms
6: TrueHD, 5.1 channels, 48khz, dialnorm: -27dB, 200ms
7: E-AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, 100ms

I am converting alle audio files with nero to flac. Will I have to set the delay to 100ms while muxing with MKVmerge or is the delay alreasy in the FLac-File?

(Sorry for asking so much)

EPiPH0NE

13th October 2008, 07:49

^^^Eac3to will handle delay automatically :)

madshi

13th October 2008, 08:24

I tried that before posting, it didn't work. Although it did help a little with EVOdemux, I was able to view chapter points and other data, but not stream information, and still unable to demux with eac3to and EVOdemux.
Then double check whether the feature*.evo files on your harddisk have the same name as that listed in the VPLST000.XPL file.

odin24

13th October 2008, 14:32

Then double check whether the feature*.evo files on your harddisk have the same name as that listed in the VPLST000.XPL file.

I'm not sure what I did... but I did it... sort of. EVOdemux can work with it, but still not eac3to.

Actually what I did was renamed the VPLST001.XPL to 000.XPL, and the the FEATURE_|0.evo to FEATURE_0.evo, same with the second part.

Thanks for your help.

EDIT:I can just read the EVO files directly with eac3to, it just doesn't give me stream info, which is fine.

One other thing, eac3to reads the audio track as DTSHD-MA 7.1, tsMuxeR read the same as DTS 6 channels with HD data, is that the same?

svgame

13th October 2008, 21:08

Then double check whether the feature*.evo files on your harddisk have the same name as that listed in the VPLST000.XPL file.

Can you add new effect for the new version eac3to. make it can to separate TrueHD to MLP+AC3 two files,possible？

Thunderbolt8

13th October 2008, 21:18

eac3to v2.67 released
:thanks:

nautilus7

13th October 2008, 21:29

Can you add new effect for the new version eac3to. make it can to separate TrueHD to MLP+AC3 two files,possible？
If you mean to export truehd and ac3 as separate files, then use .thd or .ac3 as extension to get what you want. eac3to can't convert truehd to mlp.

rica

13th October 2008, 23:27

Hi guys,
I've made some trials with SD DVDs tonight but i don't know how to extract subtitles since they are not pgs?

madshi

14th October 2008, 08:23

I've made some trials with SD DVDs tonight but i don't know how to extract subtitles since they are not pgs?
SD DVD subtitles are currently not supported by eac3to cause I don't know in which format I should demux them...

madshi

14th October 2008, 08:24

eac3to v2.68 released

http://madshi.net/eac3to.zip

* fixed crash when transcoding Blu-Ray/HD DVD track to FLAC

Kurtnoise

14th October 2008, 08:24

SD DVD subtitles are currently not supported by eac3to cause I don't know in which format I should demux them...

sup or vobsub...:)

madshi

14th October 2008, 08:33

sup or vobsub...:)
Do you have a technical documentation of how to transform the subtitle data of the SD DVDs into these formats? Because I don't know how to do that...

Kurtnoise

14th October 2008, 08:53

About Vobsub, you can check out these sources (http://www.digital-digest.com/software/download.php?sid=1051&ssid=0&did=3) (C++)...

and for sup format, there are SubtitleCreator (http://subtitlecreator.svn.sourceforge.net/viewvc/subtitlecreator/trunk/SUP.cs?view=markup) sources (C#).

Good luck. :)

madshi

14th October 2008, 09:11

That looks awefully complicated... :(

rica

14th October 2008, 16:51

What is interesting i can see them on bitmaps folder created by SUPRead as scaled to 720 (it's selected on options) when i extract them as *.sup via eac3to.
Edit:
Even DVDSubEdit can convert them to srt...

frogman

14th October 2008, 17:49

would be fine to have Nero9 working with eac3to.
Obviously Nero removed the Restrictions from their Decoders/Filters...they're working properly outside Showtime ,good for us

THX :)

Has madshi upgrade eac3to support Nero 9 yet, as I understand it Nero new aac encoder/decoder are now open in directshow filter system.

When I do a eac3to -test I get the simple error of use Nero 7. I have Nero 9 with BR plug-in installed and registered so I can't go back to Nero 7. Perhaps some one from Nero Tech rep could official clarify this.

Ps. How does one go about registering the these filter manually or is there a link out there with guide for the blind. Back to google.:thanks:

xkodi

14th October 2008, 18:17

Ps. How does one go about registering the these filter manually or is there a link out there with guide for the blind. Back to google.:thanks:

on the command prompt run:

regsvr32 "C:\Program Files\Nero\Nero 9\Nero ShowTime\SMC\NeSplitter.ax"

regsvr32 "C:\Program Files\Nero\Nero 9\Nero ShowTime\SMC\NeAudio2.ax"

then you can use the Nero 9 filters in GraphEdit. i believe that eac3to still doesn't recognize the Nero 9 filters.

p.s. IMHO Nero filter (all of them 7,8 and 9) are not important anymore and are useless, because:

- they don't decode DTS-HD and DTS-HD MA (decode the DTS core only)
- TrueHD libav decoder is perfect
- E-AC3 decoder of Arcsoft produce the same reference output as Nero filters and supports 7.1 channels, AFAIK Nero filters are limited to 5.1

so, better support for Arcsoft E-AC3 filter than Nero 9 filters, not to mention that Nero 9 installation is over 1 GB, which is ridiculous for me.

nautilus7

14th October 2008, 19:31

- E-AC3 decoder of Arcsoft produce the same reference output as Nero filters and supports 7.1 channels, AFAIK Nero filters are limited to 5.1
Good to know that. Thanks.

rica

14th October 2008, 21:15

Extracted by eac3to as sup and converted to srt by DVDSubEdit: (Original is SD DVD)

Compare them zooming the pictures:

Original DVD:

http://img442.imageshack.us/img442/2328/originalxi9.th.png (http://img442.imageshack.us/my.php?image=originalxi9.png)

converted to srt:

http://img442.imageshack.us/img442/448/srtes3.th.png (http://img442.imageshack.us/my.php?image=srtes3.png)

Zoom:

original:

http://img160.imageshack.us/img160/4170/orgzoomxq0.th.png (http://img160.imageshack.us/my.php?image=orgzoomxq0.png)http://img160.imageshack.us/images/thpix.gif (http://g.imageshack.us/thpix.php)

srt: (even zoom is larger than the original)

http://img442.imageshack.us/img442/1752/srtzoomaq5.th.png (http://img442.imageshack.us/my.php?image=srtzoomaq5.png)

MichaelAnders

14th October 2008, 21:36

madshi: Some of the BD's I am backing up to HDD have errors it seems, I am getting the "sync byte missing" error for audio and video stream (same error described on page 238 already in this thread).

Xport, as already mentioned in this thread, can demux these properly - it finds the errors, but doesnt stop and instead just ignores them. Would it be possible to add something like "skip errors" to eac3to as well? For me these are just a few frames that are corrupt, I can really do without those :)

Xport is a huge hazzle to use, I think eac3to is far better here especially due to FLAC encoding :D

magic144

14th October 2008, 22:32

@madshi, @MichaelAnders
I had the same issue (exact same category - sync byte error I believe) way back when once myself on a brand new disc - I can only assume an authoring error??!

In any event, I had to use tsMuxeR to extract the audio - it also complained, but finished...

the audio in question sounded fine - perhaps eac3to could indeed have a "continue on error" function (perhaps just for very specific classes of error like this one in particular) and flag the timecode of the error for later user verification?

GZZ

14th October 2008, 22:49

yea - I had a movie with a DTS-HD MA audio track. I tried to extract the core of it. But got an error: This is not a clean stream, clean it up with delaycut. HOW can I clean up a track that I cant extract with the program that gives me the error. The DTS-HD MA track was inside a M2TS file, cant use delaycut on a M2TS track, so a -SkipOnError function would be nice, then I could extract the core and then correct it with delaycut. :)

rica

14th October 2008, 22:52

Why not to try a second pass guys?

Have you seen something like this:

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

note: I've given up using guis since two/three months.

GZZ

14th October 2008, 23:01

thats not the error I get (dont use GUI) for eac3to. Just the error that my track isnt clean.

rica

14th October 2008, 23:03

Did you use command line or a gui? (you say it wasn't a gui)
If you used command what was your log?

magic144

14th October 2008, 23:08

it definitely wasn't a gap/overlap message (I know how to run the 2nd pass for those!), and no, I wasn't using a GUI either

I don't have the specific log (anymore - to be honest I forget the eac3to version and the disc title even!) - but that sync byte error sounds very familiar. Perhaps MichaelAnders has his log?
I can/will provide a log here if/when I re-encounter such an error on a title...

rica

14th October 2008, 23:15

I think madshi would expect a log.

magic144

14th October 2008, 23:22

and a sample of the erroneous audio stream...
I was merely validating another user's experience with my own and asking the question from the point of view of the concept/feasibility of offering such an ignore/override flag

no big deal if this is all ignored for now, I will provide full details if and when I ever see the problem again
all I know is that I haven't seen it since (has been several weeks and many discs) and that eac3to command-line tool is a powerful instrument indeed :-)

rica

14th October 2008, 23:27

I really wonder how you can not regenerate the problem?
Or the others?

nautilus7

14th October 2008, 23:28

rica

14th October 2008, 23:52

@ rica

Could you read more carefully and stop confusing people? It really doesn't harm doing it...

They describing their experience with damaged/corrupted m2ts files and you are talking about fixing gaps/overlaps over multiple m2ts files to correct audio/video sync.

@nautilus7

Yes, i admit i didn't read carefully but after understanding my fault i just wanted them to send the logs; that's it.
But i read carefully your bold lines as getting surprised?

kurt

15th October 2008, 07:33

got an error message regarding channel mixing:

eac3to v2.66
command line: eac3to track2.dts track2.ac3 -448
------------------------------------------------------------------------------
DTS-ES, 6.1 channels, 2:08:29, 16 bits, 1536kbps, 48khz
AC3 encoding only supports 6 channels. Will mix the surround channels.
Remapping channels...
Mixing surround channels...
The channel mixer received channels with different attributes.
Aborted at file position 16384.

Edit: v.2.68 didn't do the trick also....

bwana

15th October 2008, 19:21

Is there any place I should look in this 330 page thread to find out how to convert a blu-ray to a dvd format? I saw the part about how to make mkv files but i am confused as to how to go to a vob file.

tnx.

nautilus7

15th October 2008, 19:28

eac3to can't do that. You need to reencode the video. This isn't eac3to's job.

frogman

16th October 2008, 01:04

on the command prompt run:

regsvr32 "C:\Program Files\Nero\Nero 9\Nero ShowTime\SMC\NeSplitter.ax"

regsvr32 "C:\Program Files\Nero\Nero 9\Nero ShowTime\SMC\NeAudio2.ax"

then you can use the Nero 9 filters in GraphEdit. i believe that eac3to still doesn't recognize the Nero 9 filters.

Dam ...Alzheimer... I knew that...the Reg part anyways!
Thanks again for reminding me!

p.s. IMHO Nero filter (all of them 7,8 and 9) are not important anymore and are useless, because:

- they don't decode DTS-HD and DTS-HD MA (decode the DTS core only)
- TrueHD libav decoder is perfect
- E-AC3 decoder of Arcsoft produce the same reference output as Nero filters and supports 7.1 channels, AFAIK Nero filters are limited to 5.1

Can E-AC3 be down graded to just ac3 and if so, is the output .ac3. I guess for those without DD plus audio/video recievers.
E-AC3 output=pcm correct?

so, better support for Arcsoft E-AC3 filter than Nero 9 filters, not to mention that Nero 9 installation is over 1 GB, which is ridiculous for me.

Agree

"TrueHD libav decoder is perfect"

libav is part of mplayer is it not. I just installed that .dll in to my win/sytem32 directory...Is this correct? or due I have to install the complete mplayer for windows as well?

Thanks for the help!

Beastie Boy

16th October 2008, 07:45

libav is part of mplayer is it not. I just installed that .dll in to my win/sytem32 directory...Is this correct? or due I have to install the complete mplayer for windows as well?

Thanks for the help!

No need to install anything to decode with libav. Everything is included with eac3to.

ACrowley

16th October 2008, 10:26

so, better support for Arcsoft E-AC3 filter than Nero 9 filters, not to mention that Nero 9 installation is over 1 GB, which is ridiculous for me.

yes...you can install only Nero 9 Showtime ,should be need Discspace.. It works when youve Nero 7 and Nero 9 installed at the same Time. Theres no Conflict and eac3to can still use v7

xkodi

16th October 2008, 10:32

@xkodi

Arcsofrt EAc3 Deocdre is noz supported by eac3to...( or did i miss something?) so Nero is the best choice for eac3to DD+ decoding at the Moment.
Its better then libavcodec and its a Reference Decoder

you're completely correct.

zeropc

16th October 2008, 10:44

say, is there a good way to see what bit depth a tdh has? it would be helpful when you trancode to dts as it would save space. i mean why encode at 24 bit if the source is only 16 bt, right.

tebasuna51

16th October 2008, 10:59

say, is there a good way to see what bit depth a tdh has? it would be helpful when you trancode to dts as it would save space. i mean why encode at 24 bit if the source is only 16 bt, right.

Don't worry, dts don't have bitdepth only bitrate, select the desired bitrate: -768 or -1536 (default) for your space/quality requirements.

edit: If you have a DTS Master Audio encoder I think you need intermediate wav files and eac3to analyze the thd and output the appropriate bitdepth.

odin24

16th October 2008, 11:32

edit: If you have a DTS Master Audio encoder I think you need intermediate wav files and eac3to analyze the thd and output the appropriate bitdepth.

Is there a DTS Master audio encoder available... or just DTS?

tebasuna51

16th October 2008, 15:22

Is there a DTS Master audio encoder available... or just DTS?

Of course DTS-HD Master Audio Suite (http://www.dts.com/Pro-Audio_Software/DTS-HD_Master_Audio_Suite.aspx)

... or just DTS (http://www.minnetonkaaudioshop.eu/epages/61620177.sf)

Snowknight26

16th October 2008, 16:22

Whats the difference between a / and a . in the channels? For example:

4: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, -27ms
5: AC3, English, 3/1 channels, 448kbps, 48khz, dialnorm: -27dB, -27ms

tebasuna51

16th October 2008, 17:10

Whats the difference between a / and a . in the channels? For example:

5.1 or 3/2.1 is 3 front channels, 2 surround channels and 1 LFE channel

3/1 or 3/1.0 is 3 front channels and 1 surround channel, without LFE channel.

rica

16th October 2008, 22:47

Hi @madshi.
Splitted and remuxed back in two parts of one of my BD rips in m2ts with TSMuxer.
Demuxed audio and sup with eac3to.
Remuxed back to BD with TSMuxer.
Created UDF2.5 ISO image with ImageBurn.
Watched with Virtual Clone Drive.
The results:
You can watch sup with the first part while you can't do this with the second part.

This time i demuxed sup with TSMuxer itself and remuxed back to BD.
Created ISO.
Watched with Virtual Clone Drive.
The result; i can watch the subtitle on second part as well.

When i compared sups created by both programs i didn't see any difference?

Can you please check this?

Ytterbium

17th October 2008, 03:27

Hi, is there a way to stop the generation of a log file?

Joniii

17th October 2008, 10:47

I know eac3to automaticly detects if Blu-ray track needs channel mapping but could someone tell me which Blu-ray audio tracks need channel mapping, I can't remember if it was just PCM or was it needed for other formats too like TrueHD?

ACrowley

17th October 2008, 13:00

Of course DTS-HD Master Audio Suite (http://www.dts.com/Pro-Audio_Software/DTS-HD_Master_Audio_Suite.aspx)

... or just DTS (http://www.minnetonkaaudioshop.eu/epages/61620177.sf)

Na, i wouldnt by Surcode dts encoder. Its outdated . And you can buy the great new DTS Surround Audio Suite for the same Price(249$)
Surcode DVD DTS Pro cant do anything beyond 5.1 and the Quality is not so good

But the dts Support told me that when you have the discontinued DTS Pro Series ,therse no Reason to by the DTS Surround Audio Suite. It can do exactly the same,excpet the Streamplayer Decoder, with the same Output Quality. I have Pro Series ,but maybe i will by DTS Audio Suite soon.

jj666

17th October 2008, 13:07

Do you have anything to back up the claim on Surcode quality? Given the product is still being sold...

-jj-

ACrowley

17th October 2008, 13:11

Do you have anything to back up the claim on Surcode quality? Given the product is still being sold...

-jj-

The Optimal Encoder

ACrowley

17th October 2008, 13:15

Do you have anything to back up the claim on Surcode quality? Given the product is still being sold...

-jj-

The Optimal Encoder should be the Reference Encoder from DTS, no Doubt. As i say, you can get the new DTS Audio Surround Suite for the same Price ,249$

What i mean is that Surcode cant do DTS ES 6.1 /24/96.
So, theres no Point to by Surcode for 249$

zeropc

17th October 2008, 13:48

rica

17th October 2008, 14:02

Hi guys, whenever i try to enter the online shop, i get this:

Sorry, the DTS Online Store is closed for maintenance. We will re-open soon. Thank you!

ACrowley

17th October 2008, 14:06

that's true about the current state for dts encoding software. but if you don't need to upgrade to the new suite, then there is no need. surcode does deliver high quality dts files just only up to 5.1. heck, if i could, i'd switch to the new one but $250 is just not in it these days ;-)

When youve already dts pro series encoder ,theres no need to by dts audilo surround suite.
dts support told ,me that theres ne quality difference and you can do the same ,dts es 6.1 / 24/96, with both Encoders

But Surcode cant do dts es 6.1 ,24/96., that the Point :)
When youve already surcode and you dont need dts es 6.1, 24/96...no need to by/use dts pro series or the dts audio suite

What i mean is simple....when you plan to by a dts encoder, dont by surcode for 249$. Because you can get the dts surround audio suite for 249$ too, which can do everything except dts hd .It would be Nonsense!
Thats all what im talking about. However ,that the wrong Thread for those discussions

jj666

17th October 2008, 16:21

Ok, so the quality of Surcode output is fine, it's just lacking in certain features then.

-jj-

zeropc

17th October 2008, 16:53

that's correct jj

odin24

18th October 2008, 02:09

Guys, I didn't mean to hijack the thread... but good info about DTS and encoders.

I have a new question about eac3to. If I want to demux just one track (TrueHD) from a m2ts file would I use the following command.

eac3to <source m2ts directory> <TrueHD track #> <destination directory>audio.thd+ac3

I used this, and it worked, but I'm not sure if it demuxed, or converted :confused: I normally would user tsMuxeR for this but I really want to start using eac3to for everything.

Thanks,

Greif

18th October 2008, 02:47

What is the result if I demux a TrueHD track to:

- file.thd (is this the raw TrueHD?)
- file.thd+ac3 (??)
- file.ac3 (TrueHD transcoded to AC3?)

rica

18th October 2008, 03:07

What is the result if I demux a TrueHD track to:

- file.thd (is this the raw TrueHD?)
- file.thd+ac3 (??)
- file.ac3 (TrueHD transcoded to AC3?)

Blu-rays have to include an ac3 besides thd.
So extacting thd as alone doesn't mean anyhing since BD must have an additional ac3 as well for the backward compatibility.
Extracting ac3 does mean extracting separate ac3 without transcoding.
Extracting to thd+ac3 is needed for TSMuxer to recognize the file but TSMuxer is not enough by alone, second pass is needed by TSremux for trueHD:

http://forum.doom9.org/showthread.php?t=141125

_ _ _ _ _

florinandrei

18th October 2008, 03:53

yonta

18th October 2008, 08:09

@florinandrei

eac3to train.m2ts 1: video.mkv -seekToIFrames 2: audio.ac3

bmnot

18th October 2008, 10:09

I wanted to try -analyzeBitdepth on this Dolby TrueHD demo for laughs but it fails saying:

[a02] Extracting audio track number 2...
[a02] Extracting TrueHD stream...
[a02] Decoding with libav/ffmpeg...
[a02] Remapping channels...
[libav] Substream 0 parity check failed
[libav] Substream 0 checksum failed
[libav] Substream 0 length mismatch.
[a02] The libav decoder reported an error while decoding.
Aborted at file position 16384.

here it is: http://www.mediafire.com/?giwnn4qgtm2

nautilus7

18th October 2008, 11:15

Is this an uncut file? The problem doesn't have to do with -analyzebitdepth. It seems damaged unless libav has a bug and can't decode it, while nero can.

evdberg

18th October 2008, 12:05

I used the -analyzebitdepth option and got this result:
[a02] Bit depth analyzation: max 24 bits, average 19 bits, most common 16 bits.

What does "most common 16 bits" exactly mean? Is the LSB zero or the MSB zero (or 0xFF if the value is negative)? Because it seems to be a 24 bits tracks, my assumption is that the MSB is zero. Am I right?
In other words: how does your analyzation exactly work?

madshi

18th October 2008, 15:23

Would it be possible to add something like "skip errors" to eac3to as well?
perhaps eac3to could indeed have a "continue on error" function (perhaps just for very specific classes of error like this one in particular) and flag the timecode of the error for later user verification?
The DTS-HD MA track was inside a M2TS file, cant use delaycut on a M2TS track, so a -SkipOnError function would be nice
This is on my to do list, but it's not as easy as it may sound. All the eac3to audio and video parsers currently depend on the bitstream being alright. If the video or audio bitstream is damaged that can result in weird problems of all kinds. Before I can even think about allowing to "ignore" errors in corrupted source files I'd first have to make all audio and video bitstream parsers more resistent against errors. E.g. the AC3 parser would have to be able to handle incorrect CRCs, missing sync words and all that stuff - and might even have to fix the stream, similar to what delaycut does. Now there are a dozen of different audio bitstream parsers in eac3to and 3 video bitstream parsers. So making all this more error tolerant will take quite a while...

got an error message regarding channel mixing:

Edit: v.2.68 didn't do the trick also....
Will check that out. Do you happen to have a small sample?

Hi @madshi.
Splitted and remuxed back in two parts of one of my BD rips in m2ts with TSMuxer.
Demuxed audio and sup with eac3to.
Remuxed back to BD with TSMuxer.
Created UDF2.5 ISO image with ImageBurn.
Watched with Virtual Clone Drive.
The results:
You can watch sup with the first part while you can't do this with the second part.

This time i demuxed sup with TSMuxer itself and remuxed back to BD.
Created ISO.
Watched with Virtual Clone Drive.
The result; i can watch the subtitle on second part as well.

When i compared sups created by both programs i didn't see any difference?

Can you please check this?
No. I'm not interested at all in why a specific combination of specific things you're doing with a number of applications does not work. If you find a clear bug in the sup file produced by eac3to, let me know. If not you'll have to find out yourself why that specific combination of things you're doing doesn't work...

Hi, is there a way to stop the generation of a log file?
No.

I know eac3to automaticly detects if Blu-ray track needs channel mapping but could someone tell me which Blu-ray audio tracks need channel mapping, I can't remember if it was just PCM or was it needed for other formats too like TrueHD?
Most tracks need channel remapping. The remapping which is needed even depends on the decoder which is used.

What is the result if I demux a TrueHD track to:

- file.thd (is this the raw TrueHD?)
- file.thd+ac3 (??)
- file.ac3 (TrueHD transcoded to AC3?)
HD DVD style TrueHD tracks consist of only TrueHD frames. These are identified by eac3to as being "TrueHD". The correct extension for such tracks is "*.thd". Blu-Ray style TrueHD tracks consist of TrueHD audio frames and AC3 audio frames. Both types of frames are interweaved. eac3to identifies these tracks as "TrueHD/AC3". The correct file extension is "*.thd+ac3". The file extension "*.ac3" is self explaining.

So, if you demux a HD DVD style TrueHD track to "*.thd", the track is demuxed as it is. If you demux it to "*.thd+ac3", eac3to creates AC3 frames on the fly and interweaves them with the HD DVD data to create a Blu-Ray compatible "TrueHD/AC3" track. If you demux a HD DVD style TrueHD track to "*.ac3", the TrueHD track is transcoded to AC3.

If you demux a Blu-Ray style TrueHD track to "*.thd", the track is demuxed with the interweaved AC3 frames stripped away. If you demux it to "*.thd+ac3", the track is demuxed unmodified. If you demux a Blu-Ray style TrueHD track to "*.ac3", the AC3 frames are demuxed as they are with the TrueHD frames stripped away.

It can be even more complicated: If you demux a Blu-Ray TrueHD/AC3 track by using e.g. "eac3to source 2: audio.thd+ac3 -192", eac3to strips the original interweaved AC3 frames away, creates new AC3 frames with 192kbit/s on the fly and interweaves them with the original TrueHD frames coming from the Blu-Ray track.

I hope you're finally confused now... :D

I wanted to try -analyzeBitdepth on this Dolby TrueHD demo for laughs but it fails saying:

[a02] The libav decoder reported an error while decoding.
Is eac3to able to decode this track? I think it's simply a broken/corrupted TrueHD track.

I used the -analyzebitdepth option and got this result:
[a02] Bit depth analyzation: max 24 bits, average 19 bits, most common 16 bits.

What does "most common 16 bits" exactly mean? Is the LSB zero or the MSB zero (or 0xFF if the value is negative)? Because it seems to be a 24 bits tracks, my assumption is that the MSB is zero. Am I right?
In other words: how does your analyzation exactly work?
When people talk about Blu-Ray audio bitdepth, they usually talk about 16bit vs. 24bit. However, some Blu-Ray PCM tracks are 20bit (stored as 24bit with the 4 LSBs being zeroed out). eac3to checks the real bitdepth of the audio track. That is it searches for the first non-zero bit, beginning with the least significant bit.

The report eac3to posted in your specific case means that some samples in the audio track have a full bitdepth of 24bit (meaning no wasted zeroed out bits), while most samples are limited to 16bit only (least 8 significant bits zeroed out). I've seen this from time to time: Specific scenes have 24bit, while other scenes have the 8 LSBs zeroed out. E.g. with POTC 1 only the Walt Disney intro is 24bit, the rest of the movie is 16bit. The "average" bitdepth reported by eac3to tells you the weighted average bitdepth of all audio samples.

odin24

18th October 2008, 15:34

It can be even more complicated: If you demux a Blu-Ray TrueHD/AC3 track by using e.g. "eac3to source 2: audio.thd+ac3 -192", eac3to strips the original interweaved AC3 frames away, creates new AC3 frames with 192kbit/s on the fly and interweaves them with the original TrueHD frames coming from the Blu-Ray track.

What would be the point of this, just to save space? Would there be noticable quality loss as opposed to keeping the interweaved AC3 track at 448 or 640 kb/s?

Thanks,

rica

18th October 2008, 15:49

No. I'm not interested at all in why a specific combination of specific things you're doing with a number of applications does not work. If you find a clear bug in the sup file produced by eac3to, let me know. If not you'll have to find out yourself why that specific combination of things you're doing doesn't work...

Thanks anyway.

kurt

18th October 2008, 15:49

Will check that out. Do you happen to have a small sample?

here you are: http://rapidshare.com/files/155218023/6.1.stream.dts (~10MB)

thx for looking into this :)

madshi

18th October 2008, 15:50

What would be the point of this, just to save space?
Of course to save space. If you know for sure that you will never need that interweaved AC3 frames but still want/need to get a Blu-Ray compatible TrueHD/AC3 track, using a low bitrate for the AC3 frames makes sense.

Would there be noticable quality loss as opposed to keeping the interweaved AC3 track at 448 or 640 kb/s?
If you ever play that interweaved AC3 track then of course. If you only ever play the interweaved TrueHD frames, then no.

evdberg

18th October 2008, 16:53

When people talk about Blu-Ray audio bitdepth, they usually talk about 16bit vs. 24bit. However, some Blu-Ray PCM tracks are 20bit (stored as 24bit with the 4 LSBs being zeroed out). eac3to checks the real bitdepth of the audio track. That is it searches for the first non-zero bit, beginning with the least significant bit.

The report eac3to posted in your specific case means that some samples in the audio track have a full bitdepth of 24bit (meaning no wasted zeroed out bits), while most samples are limited to 16bit only (least 8 significant bits zeroed out). I've seen this from time to time: Specific scenes have 24bit, while other scenes have the 8 LSBs zeroed out. E.g. with POTC 1 only the Walt Disney intro is 24bit, the rest of the movie is 16bit. The "average" bitdepth reported by eac3to tells you the weighted average bitdepth of all audio samples.

Thanks for your explanation. I have some additional questions though. You state you determine the bitdepth of every sample, but do you count this separately (int count[24]) or do you categorize a sample in the 16 or 24 bits domain and count the number of samples in each domain?
If it is the first case, do you take into account the chance that a higher bitdepth sample has its lower bits zeroed? For instance if you determine that the lower 4 bits are zero, there is a 1/16 chance that this still is a 24 bits sample.

shambles

18th October 2008, 18:15

The "average" bitdepth reported by eac3to tells you the weighted average bitdepth of all audio samples.

some hd dvds released by universal have truehd tracks where centre, left, right and lfe channels are constant 16bit but surrounds are constant 24bit.. so for tracks where different channels have different bitdepths perhaps it would make sense to print out separate info for each channel

madshi

18th October 2008, 18:33

Thanks for your explanation. I have some additional questions though. You state you determine the bitdepth of every sample, but do you count this separately (int count[24]) or do you categorize a sample in the 16 or 24 bits domain and count the number of samples in each domain?
If it is the first case, do you take into account the chance that a higher bitdepth sample has its lower bits zeroed? For instance if you determine that the lower 4 bits are zero, there is a 1/16 chance that this still is a 24 bits sample.
I count each bitdepth separately (int64 count[24]). E.g. eac3to reports the "The Fifth Element" English audio track to be "constant 20 bit". That is also exactly what a Sony employee has stated about that audio track. I'm aware that some samples have their lower bits at 0 even though the bits contain valid data. I'm compensating for that "somewhat" in my final statistical calculations. Probably it's not mathematically perfect, but it seems to do the job well enough...

some hd dvds released by universal have truehd tracks where centre, left, right and lfe channels are constant 16bit but surrounds are constant 24bit.. so for tracks where different channels have different bitdepths perhaps it would make sense to print out separate info for each channel
I was not aware of that different channels sometimes have different bitdepths. That's interesting. But still I'm not sure how complicated I should make the bitdepth analyzation report. I mean I could print out 10 pages of statistics and print out some funny graphs. But actually many people are not very interested in the statistics at all. So when I developed that eac3to bitdepth statistics report I limited the output so that it always fits into one single line...

evdberg

18th October 2008, 21:09

@madshi, personally I would appreciate some more indepth statistics for the analyzebitdepth option. 10 pages is definitely not necessary, but a few lines extra would be great. The post by shambles also explains the result I got, since I tested an Universal title.

madshi

18th October 2008, 22:08

Well, what kind of statistics do you want to have? The same I'm doing now, but for every channel (in case the channels are different)? Or what are you looking for?

evdberg

18th October 2008, 23:34

Yes, statistics per channel, but if possible only when there is a difference between the channels. Also I prefer a percentage instead of terms like 'most common' when the bitdepth is not constant.

madshi

19th October 2008, 07:56

Also I prefer a percentage instead of terms like 'most common' when the bitdepth is not constant.
You mean a list of bitdepths which occur, each with the percentage number? That could get rather ugly if the bitdepth is all over the place, don't you think?

evdberg

19th October 2008, 08:14

That could indeed get ugly, although I think that soundtracks are in essence 16, 20 or 24 bits. Maybe you can divide the bitdepths into those 3 groups?

madshi

19th October 2008, 09:43

Dolby recommends to those studios which used to use 16bit only, to up that to ca. 18bit. So I don't think we can limit bitdepth analyzation to only 16, 20 and 24.

shambles

19th October 2008, 09:58

perhaps the (possible) more detailed analysis could be placed in the log while keeping the main output neat and clean

nautilus7

19th October 2008, 10:13

I think the default bit analyzation should stay as it is and for more info the -analyzebitdepth could be changed to output more detailed reports. Like -logdts.

Zwitterion

19th October 2008, 17:29

I've got a feature request:
A switch that phase-shifts the surround channels by 90°. This is needed for correct DPL II downmixing using -down2 by files that have no phase-shifted rears (especially DTS).

example (AC3 sample (http://rapidshare.com/files/155535814/_VTS_01_1_T80_3_2ch_448Kbps_DELAY_0ms.ac3.html)):
http://img410.imageshack.us/my.php?image=down2sq0.png
When using just -down2 FR and RR cancel each other out.

Another idea:
DIRAC time stretching (http://www.dspdimension.com/technology-licensing/dirac/). The LE version is free to implement.
I don't know any free tool that does high-quality timestretching, so that would definitely be another killer feature and very useful for 23.976 --> 25 audio conversions.

zeropc

19th October 2008, 17:40

erm ... why does eac3to create a gap file, even calculates what is need to fix it but then it doesn't use it afterwards. you have to run a second run which adds a tremendous amount of time.

can't this be done in the first pass or is this absolutly nessecary?

nlnl

19th October 2008, 17:44

Vista 32 + eac3to 2.67 crashed remuxing PCM 5.1 24/96 from m2ts Chris Botti: Live with Orchestra and Special Guests to Flac format.
Please help!:helpful:

menlvd

19th October 2008, 18:13

Vista 32 + eac3to 2.67 crashed remuxing PCM 5.1 24/96 from m2ts Chris Botti: Live with Orchestra and Special Guests to Flac format.
Please help!:helpful:

plz read change log for 2.68 ver.
v2.68
* fixed crash when transcoding Blu-Ray/HD DVD track to FLAC

madshi

19th October 2008, 18:46

I've got a feature request:
A switch that phase-shifts the surround channels by 90°. This is needed for correct DPL II downmixing using -down2 by files that have no phase-shifted rears (especially DTS).
Would love to, but I don't know how to do it. Does anybody know any LGPL libraries or source code which can do high quality phase shifts? (if possible floating point in/out)

Another idea:
DIRAC time stretching (http://www.dspdimension.com/technology-licensing/dirac/). The LE version is free to implement.
I don't know any free tool that does high-quality timestretching, so that would definitely be another killer feature and very useful for 23.976 --> 25 audio conversions.
Looks good! Will gave this a try...

erm ... why does eac3to create a gap file, even calculates what is need to fix it but then it doesn't use it afterwards. you have to run a second run which adds a tremendous amount of time.

can't this be done in the first pass or is this absolutly nessecary?
It can't be done in the first pass. A 2nd pass is necessary. It's not done automatically because sometimes it might be preferred to not fix the detected gaps/overlaps.

madshi

19th October 2008, 23:36

got an error message regarding channel mixing
Fixed in v2.69.

@madshi, personally I would appreciate some more indepth statistics for the analyzebitdepth option.
Have added that to my to do list. Don't know when I will get to that, though...

madshi

19th October 2008, 23:39

eac3to v2.69 released

http://madshi.net/eac3to.zip

* added high precision SSRC resampler
* resampling "-quality" now allows "low", "high" (SSRC) or "ultra" (r8brain)
* resampling quality now defaults to "high" (SSRC)
* bitdepth is now analyzed separately for original vs. processed data
* fixed: downmixing 16 bit DTS tracks to 5.1 or 2.0 didn't work
* fixed: Sonic Decoder was incorrectly assumed to decode XXCh DTS files to 6.1
* for movies the Haali Muxer can't handle "-seekToIFrames" is suggested now
Please note that I'm not sure whether SSRC or r8brain is better for resampling. r8brain is a lot slower than SSRC, so hopefully it's a little bit better, but you be the judge. Because of the dramatic speed difference the highest quality SSRC mode is now the default resampling mode (also used for PAL speedup/slowdown). SSRC is somewhat limited in which conversions it likes to do exactly, though. Some sample rate conversions might be declined by SSRC. If you stumble over such a case, just use the "-quality=ultra" option to switch to r8brain instead.

zeropc

20th October 2008, 03:47

It can't be done in the first pass. A 2nd pass is necessary. It's not done automatically because sometimes it might be preferred to not fix the detected gaps/overlaps.

thanks for the clarification :)

Boulder

20th October 2008, 03:49

Please note that I'm not sure whether SSRC or r8brain is better for resampling. r8brain is a lot slower than SSRC, so hopefully it's a little bit better, but you be the judge. Because of the dramatic speed difference the highest quality SSRC mode is now the default resampling mode (also used for PAL speedup/slowdown). SSRC is somewhat limited in which conversions it likes to do exactly, though. Some sample rate conversions might be declined by SSRC. If you stumble over such a case, just use the "-quality=ultra" option to switch to r8brain instead.Many seem to prefer SSRC over r8brain, for example http://src.infinitewave.ca/ has some nice test result graphs.

bmnot

20th October 2008, 05:43

Is eac3to able to decode this track? I think it's simply a broken/corrupted TrueHD track.

please look at the sample: http://www.mediafire.com/?giwnn4qgtm2

(it's actually the whole file off the disc untouched)

madshi

20th October 2008, 09:25

for example http://src.infinitewave.ca/ has some nice test result graphs.
Yep, that's a very nice site. Back when I decided to use r8brain the site already existed but I think it didn't have SSRC test results at the time. Anyway, I find it hard to judge final quality by these test graphs. Just one example: The default behaviour seems to be linear phase. However, some filters intentionally use a different phase design, which seems to have both advantages and disadvantages. So which is better now? Also the help of that site says: "It is not always possible to judge subjective quality from the presented measurement results."

Specifically "r8brain Free" is better than "SSRC High Precision" in the first two test graphs, but worse (or different?) in the other test graphs. So which sounds better now? SSRC seems to be better than almost any other resample in the "Passband" and "Transition" graphs. But I'm wondering: Is this due to a better design? Or do the other resamplers intentionally take a hit in these two things with the intention to improve something else?

Many seem to prefer SSRC over r8brain
I've searched for comparisons but found none, except the infinitewave site you linked to. Where did you read that people prefer SSRC over r8brain?

Thanks!

please look at the sample: http://www.mediafire.com/?giwnn4qgtm2

(it's actually the whole file off the disc untouched)
Hmmmm... Have you tried reripping the file, just to be sure? There's no obvious corruption in the file, but it's still possible that there's a bit fault hidden somewhere. So reripping might still help. If you're 100% sure that the file is correctly ripped we might have to submit the file to the libav guys so they can check what's wrong. FWIW, the Nero Audio Decoder seems to be able to decode the TrueHD track just fine. However, it's limited to 5.1 output. Now I'm not sure if there's a bug in the libav decoder or whether the file is actually broken in some way. It's possible that the Nero decoder is just more forgiving compared to libav...

ACrowley

20th October 2008, 09:33

eac3to v2.69 released
* fixed: Sonic Decoder was incorrectly assumed to decode XXCh DTS files to 6.1

What does it mean ? Down it mean DTS ES 6.1 was not decoded properly by SonicDecoder ?

Ah, Implementing DIRAC would be superb...its one of the best Algor. fro Timestretching

madshi

20th October 2008, 09:49

What does it mean ? Down it mean DTS ES 6.1 was not decoded properly by SonicDecoder ?
There are some DTS and DTS-HD 6.1 tracks which the Sonic Decoder correctly decodes as 6.1. But there are other such tracks which the Sonic decoder only outputs as 5.1. It depends on which DTS extension is used for the additional channel. There are two different extensions, namely "XCh" and "XXCh". Most 6.1 tracks use "XCh" which the Sonic decoder can decode. But some are using "XXCh" which the Sonic decoder does not support. Older eac3to versions thought that Sonic would be able to decode every 6.1 track as 6.1. This caused eac3to to abort processing in specific situations when the Sonic decoder unexpectedly only decoded 5.1. Now the new eac3to version knows exactly which tracks are decoded as 5.1 and which as 6.1 by the Sonic decoder. So eac3to doesn't get surprised, anymore...

Ah, Implementing DIRAC would be superb...its one of the best Algor. fro Timestretching
Good to hear that it's a good algo. I plan on implementing DIRAC support soon. The free lib version looks easy enough to add.

nwg

20th October 2008, 12:34

Can this program convert a TrueHD track to a DTS one?

nautilus7

20th October 2008, 12:37

If you have a dts encoder, yes.

nwg

20th October 2008, 12:57

If you have a dts encoder, yes.

I just realised I have the Surcode one. Will that do and if so how would I do it? I am not too good at the command line bit as I prefer the Eac3to GUI and that seems to have not been updated for a while.

nautilus7

20th October 2008, 13:15

eac3to source.thd dest.dts

nwg

20th October 2008, 13:17

Thanks.

nautilus7

20th October 2008, 21:19

kurt

20th October 2008, 21:23

Fixed in v2.69.

thx, working now :)

D:\Movies\xxx>eac3to deutsch1.dts deutsch.ac3 -448
DTS-ES, 6.1 channels, 2:08:29, 16 bits, 1536kbps, 48khz
AC3 encoding doesn't support back channels. Will mix them into the surround.
Remapping channels...
Mixing surround channels...
Loading white noise (needed for dithering)...
Decoding with ArcSoft DTS Decoder...
Patching bitdepth to 24 bits...
Encoding AC3 <448kbps> with libAften...
Creating file "deutsch.ac3"...
-------------

btw: anyone tried hibernating windows during conversion? is it safe? managed to get good results with megui. here too?

madshi

20th October 2008, 21:57

I have a question for DTS-ES Matrix tracks. Those tracks have 5.1 discrete channels and one (BC) matrixed into SL and SR, right? An A/V receiver capable of DTS-ES decoding outputs 6.1, correct?
Yes.

Shouldn't eac3to do the same?
What specific purpose would you need that for? Anyway, I don't know how to do that.

Are Sonic or Arcsoft decoders able to decode DTS-ES Matrix?
Sonic no. ArcSoft don't know. I can ask ArcSoft to output 6.1 and it does it - even for real 5.1 tracks. But I don't know which algorithms ArcSoft is using for its channel transformations. So I feel safer asking ArcSoft to output the native format. Which is 5.1 for "6.1 matrix" tracks.

nautilus7

20th October 2008, 22:13

Is the BC matrixed into SL, SR the same way Center and Surround channels were matrixed into Left and Right channels in the analog Dolby ProLogic format (I think this was done with phase shifting)?
EDIT: the answer is yes to the above.

Basically, what i'm interested in is to be able to maintain 6.1 channels if i transcode a DTS-ES Matrix track to AC3 or whatever. Will i get a Dolby Digital EX 6.1 (matrixed) then? I suppose i won't because DTS and Dolby aren't the same...

madshi

20th October 2008, 22:21

Aren't DTS and Dolby Matrix algorithms similar to each other? I don't really know, to be honest.

nautilus7

20th October 2008, 22:23

I don't know as well. I am confused... Maybe tebasuna51 can answer that.

I will try making some DTS-ES tracks and then converting to DD-EX to see what happens.

tebasuna51

21st October 2008, 11:10

I don't know as well. I am confused... Maybe tebasuna51 can answer that.

I will try making some DTS-ES tracks and then converting to DD-EX to see what happens.
I don't know very much about 6.1. Only the samples I make with DTS-ES encoder:

If I call FL,FR,FC,LF,BL,BR,BC my source 6.1 channels to encode, and FL,FR,FC,LF,SL,SR the 5.1 channels after decode with ArcSoft or libav (1) seems:

SL = j x BL + 0.707 x BC
SR = -j x BR + 0.707 x BC

where j is +90º phase shift and -j is -90º phase shift.

(1) the output of libav seems identical to ArcSoft but -0.5 dB

I can ask ArcSoft to output 6.1 and it does it - even for real 5.1 tracks. But I don't know which algorithms ArcSoft is using for its channel transformations. So I feel safer asking ArcSoft to output the native format. Which is 5.1 for "6.1 matrix" tracks.
Maybe we can know something about the algorithms if we ask ArcSoft to output 6.1 using the samples from the DTS-ES encoder previously mentioned.

Would love to, but I don't know how to do it. Does anybody know any LGPL libraries or source code which can do high quality phase shifts? (if possible floating point in/out)
Only know the FooBar2000 plugins:

- foo_dsp_downmix (http://www.hydrogenaudio.org/forums/index.php?showtopic=52887): make the dpl II downmix with or without phase shift.

- foo_dsp_fsurround (http://www.hydrogenaudio.org/forums/index.php?showtopic=52235): recover the 5.1 channels from a stereo with dpl II

Both have the sources availables but need the libfftw3f-3.dll (Fast Fourier Transform library)

3ngel

21st October 2008, 12:41

Would love to, but I don't know how to do it. Does anybody know any LGPL libraries or source code which can do high quality phase shifts?

That would be just fantastic, so it coul be implemented the DLIIx algorithm wich uses a variable phase shift over the channels, and permits to correctly downmix 7.1 to 2.0.

florinandrei

22nd October 2008, 06:44

Can't demux a file created by a Canon HF100 camcorder:

eac3to v2.68
command line: eac3to train.m2ts 1: video.mkv 2: audio.ac3
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 0:00:25
1: h264/AVC, 1080i60 /1.001 (16:9)
2: AC3, 2.0 channels, 256kbps, 48khz, -67ms
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Applying (E-)AC3 delay...
[v01] 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 32964608.

File is here, it's about 46 MB:

http://dl.getdropbox.com/u/29966/eac3to/train.m2ts

@florinandrei

eac3to train.m2ts 1: video.mkv -seekToIFrames 2: audio.ac3

Okay, that seems to pacify it. But there are some scary remarks about potential artifacts if I use this option. Under what conditions can this occur?

The source file is an honest-to-goodness AVCHD file generated by a Canon camcorder (http://www.usa.canon.com/consumer/controller?act=ModelInfoAct&fcategoryid=177&modelid=16187), it's probably clean and standards-compliant (the PS3 will happily play it, either directly or within a Blu-Ray structure), so I should not expect any surprises if I use -seekToIFrames - or should I?

salehin

22nd October 2008, 07:37

odin24

22nd October 2008, 07:49

Is it possible to edit/cut a ac3 stream using eac3to?

I need to cut about 0.5-1.0 sec of audio ( 6 channels AC3 Dolby audio at 384 kbps) to sync it with the BBC HD Video. Can you please advise the cmd line I should use.

The extra bit that I'm trying to get rid off is at the beginning.

Cheers :)

You could apply a negative delay (audio starts early). I'm not sure of the command though, it might look something like this;

eac3to audio.ac3 -1000ms

Darth Pinous

22nd October 2008, 13:39

Hi Madshi,

Don't know if someone as already reported on that, but doesn't seem so... Using latest versions of eac3to (2.6*), I encounter a problem when using the -down16 option on 24 bits soundtracks (WAV and DTS-HD, so far, not tested on trueHD or PCM).

downmixing to 16 bits is OK, but the audio is broken (weird sounds) as if the header was wrong. In fact, just passing the audio through wavewizard without any modifications fixes the problem...

Edit: just checked again with wavewizard.. the corrupt wav file says 16 bits per sample and 24 value bits per sample in its header. That may be the explanation...

xkodi

22nd October 2008, 14:24

Sonic no. ArcSoft don't know. I can ask ArcSoft to output 6.1 and it does it - even for real 5.1 tracks. But I don't know which algorithms ArcSoft is using for its channel transformations. So I feel safer asking ArcSoft to output the native format. Which is 5.1 for "6.1 matrix" tracks.

madshi, please add temporary test switch that allow to ask Arcsoft to output 6.1 channels for DTS-ES, so i can run some tests.

i used the following test to find the parameters of different DTS-HD MA track before eac3to had support for such tracks:

http://forum.doom9.org/showthread.php?p=1147644#post1147644

i believe the same test could be used to find out what number of channels should be set for correct DTS-ES decoding with Arcsoft:

1. decode the DTS-ES 6.1 to WAVs with Sonic: outputs 5.1 channels

2. decode the DTS-ES 6.1 to WAVs with Arcsoft and set output channels to 5.1: outputs 5.1 channels

3. decode the DTS-ES 6.1 to WAVs with Arcsoft and set output channels to 6.1: outputs 6.1 channels

i suspect that if compare byte by byte the first 5.1 channels from 2 and 3 they will be different, then compare the first 5.1 channels between 1 and 2 and then between 1 and 3.

if the first 5.1 channels of 1 are byte by byte identical to the first 5.1 channels of 2 then Arcsoft output channels should be set to 5.1.

if the first 5.1 channels of 1 are byte by byte identical to the first 5.1 channels of 3 then Arcsoft output channels should be set to 6.1

himan2001

22nd October 2008, 18:36

@madshi:

There are a lot more "old" BD Releases which contains seemless branching and "splitted" Releases that produce Audio-Spikes
and loud noise at the cutting/fixxing Points after timecode-rerun :(

Something is borked when using DTS, no matter if it´s Arcsoft or Sonic. With AC3-Track splitting/joining is ok.

As i can remember, the same BD´s using an older eac3to version < 2.58 doesn´t produce this errors.

A good try will be Conair/German BD. Try to assemble
the GERMAN DTS Track. On the cut point there is a loud
spike on the decoded file (destination format can be WAV or AC3 - result is the same)

Please check your audio/DTS Handling. Since the > 5.1 "fixxing" a lot of previous good working, older things, are screwed up more and more :(

The destorted Audio ONLY comes up when correcting overlaps!

On 2.67 Version a quickhelp was, not to use the HD-Part (export the -core and join). But since 2.69 this is no longer working. Now Core and HD/MA-Part produces drops&spikes.

When No other Track (AC3/TrueHD) is available I´am screwed on this overlapping BD´s...

lchiu7

23rd October 2008, 08:17

evdberg

23rd October 2008, 09:36

Have added that to my to do list. Don't know when I will get to that, though...

Thanks! Looking forward to it. Hope you can start with the bitdepth statistics per channel (interested to examine those Universal releases).

tebasuna51

23rd October 2008, 11:43

I am trying to compress files captured from our terrestrial HD broadcasts which are h.264/ac-3.

The files play fine but when the video is compressed and then muxed back with the audio the various tools (e.g. txmuxer and mkvmerge) complain about crc errors in the AC-3 stream and suggest re-sync.

Tried to pass the AC-3 streams through delaycut in order to fix these. Tried all options including ignore, silence, repair but while it identifies the errors and completes okay, when this file is muxed back with the video, it goes out of sync; presumably after the point where the first CRC error had occurred.

The original files get rejected by eac3to saying run delaycut first. I would have thought silence or ignore would re-sync the streams but in my experience it doesn't.

Any advice would be appreciated here.

Thanks

The ac3 streams captured from TV can have 2.0 content (commercials), 5.1 (movie), 2.0 (more commercials), ...

These streams are invalid and I don't know the software to manage easily the problem.
You can use DelayCut to see the problems: 'Some basic parameters changed between Frame # and this frame'
and cut the stream to separate the contents 2.0 and 5.1 (most the times).

Warning also with videotools to cancel commercial sequences because don't take care with audio frames.

Chumbo

23rd October 2008, 16:03

rebkell

23rd October 2008, 17:13

The ac3 streams captured from TV can have 2.0 content (commercials), 5.1 (movie), 2.0 (more commercials), ...

These streams are invalid and I don't know the software to manage easily the problem.
You can use DelayCut to see the problems: 'Some basic parameters changed between Frame # and this frame'
and cut the stream to separate the contents 2.0 and 5.1 (most the times).

Warning also with videotools to cancel commercial sequences because don't take care with audio frames.

these streams are invalid because they switch from 2.0 to 5.1 and back and forth?

What program are you using to extract the audio with? I've had plenty of sync problems and those are a pain to deal with, but try dgavcdec to extract the ac3 stream with, if it can be extracted I wouldn't expect any problems muxing it back in.

Sorry madshi, I know this is off topic.

flyingernst

23rd October 2008, 19:05

I think I asked that a few weeks aga but I think I didn´t got an answer

EVO, 2 video tracks, 4 audio tracks, 8 subtitle tracks, 2:20:02
"Feature Presentation"
1: Joined EVO file
2: Chapters, 40 chapters with names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: VC-1, 480p30 /1.001 (3:2), 100ms
5: E-AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB, 100ms
6: E-AC3, German, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB, 100ms
7: TrueHD, German, 5.1 channels, 48khz, dialnorm: -27dB, 200ms
8: E-AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, 100ms
9: Subtitle, English
10: Subtitle, English, "SDH"
11: Subtitle, German
12: Subtitle, German, "SDH"
13: Subtitle, German, "narrative"
14: Subtitle
15: Subtitle
16: Subtitle
[a05] Extracting audio track number 5...
[a05] Removing E-AC3 dialog normalization...
[a05] Applying (E-)AC3 delay...
[a05] Creating file "E:\english.eac3"...
Video track 3 contains 201466 frames.
Video track 4 contains 251744 frames.
eac3to processing took 6 minutes, 4 seconds.
Done.

will I have to insert a 100ms delay while muxing to MKV or M2TS?!

Thanks, Greetings Michael

whats here, here Delay while muxing?!

EVO, 2 video tracks, 4 audio tracks, 8 subtitle tracks, 2:20:02
"Feature Presentation"
1: Joined EVO file
2: Chapters, 40 chapters with names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: VC-1, 480p30 /1.001 (3:2), 100ms
5: E-AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB, 100ms
6: E-AC3, German, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB, 100ms
7: TrueHD, German, 5.1 channels, 48khz, dialnorm: -27dB, 200ms
8: E-AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, 100ms
9: Subtitle, English
10: Subtitle, English, "SDH"
11: Subtitle, German
12: Subtitle, German, "SDH"
13: Subtitle, German, "narrative"
14: Subtitle
15: Subtitle
16: Subtitle
[a07] Extracting audio track number 7...
[a07] Removing TrueHD dialog normalization...
[a07] Creating file "E:\deutsch.thd"...
Video track 3 contains 201466 frames.
Video track 4 contains 251744 frames.
eac3to processing took 6 minutes, 12 seconds.
Done.

and perhaps here not right: Why can´t I mux the *-thd (also renamed to *.ac3) mux with TSmuxer to a m2ts file, the EAC3 File works with the VC1 Videofile

flyingernst

23rd October 2008, 20:41

perhaps we can forget my quetion when you tell me how eac3to can directly change the EVO container in a m2ts container (wiki says nothing about).
ic can work with full disk Structures an the 2 mainEvos...but what comes after

"F:\Download\Brennen\eac3to\eac3to.exe" "P:" 1)

(1) ist the mainmovie
since TSmuxer has problems with importing thd files I want to change container and throw out not wanted streams with tsremuxer

Thanks

EDIT: have it: joining the files to one EVO with eac3to, then change container with TSremux

Blue_MiSfit

24th October 2008, 03:04

PHD_1976

24th October 2008, 06:25

I was using 2.68 version to split DD 5.1 EX file from DVD to 6 mono waves. I don't have nero 7.xx installed on my machine so the track was processed by ffmpeg decoder. The result was not good - too many peak level overflows so the audible crack noise is heard when the overflow happens.
I've never had problems with DD tracks+ffmpeg decoder.
Maybe this is connected with EX or I'm just unaware of some option to normalize peaks?

I've tried the very same procedure on other machine where Nero 7.10 is installed - the result was fine: clear 6 channels with no overflow at all.

Madshi, if you want a sample i'll provide one.

Thanx for your work.

tebasuna51

24th October 2008, 10:45

http://www.mediafire.com/?sharekey=32d600f0ab3cb846d2db6fb9a8902bda

Here's a strange little stream - eac3to, mkvtoolnix, and several other programs absolutely hate it, but VLC and Media Player Classic (with ac3filter) will play it perfectly.

It's 640kbps AC3 5.1ch.

Help???

~MiSfit

You can use DelayCut to see how many corrupt is this file.

The best 'player' for this file is the 'Recycle Bin', just drag and drop ... :p

Blue_MiSfit

24th October 2008, 20:20

DelayCut does indeed vomit out a waterfall of errors, but it plays perfectly in MPC or VLC ???

~MiSfit

madshi

25th October 2008, 09:25

I don't know very much about 6.1. Only the samples I make with DTS-ES encoder:

If I call FL,FR,FC,LF,BL,BR,BC my source 6.1 channels to encode, and FL,FR,FC,LF,SL,SR the 5.1 channels after decode with ArcSoft or libav (1) seems:

SL = j x BL + 0.707 x BC
SR = -j x BR + 0.707 x BC

where j is +90º phase shift and -j is -90º phase shift.
Hmmmm, that's interesting. So would you say that 7.1 downmixed to 5.1 would likely be:

SL = j x SL + BL
SR = -j x SR + BR

?

Maybe we can know something about the algorithms if we ask ArcSoft to output 6.1 using the samples from the DTS-ES encoder previously mentioned.
I'll add a switch which will allow you to ask ArcSoft to output any number of channels you want. That should open the door to check what ArcSoft is doing for 7.1 -> 6.1 and 7.1 -> 5.1, too.

Only know the FooBar2000 plugins:

- foo_dsp_downmix (http://www.hydrogenaudio.org/forums/index.php?showtopic=52887): make the dpl II downmix with or without phase shift.

- foo_dsp_fsurround (http://www.hydrogenaudio.org/forums/index.php?showtopic=52235): recover the 5.1 channels from a stereo with dpl II

Both have the sources availables but need the libfftw3f-3.dll (Fast Fourier Transform library)
Looks nice. Unfortunately they're both GPL. Since eac3to is closed source, I can't use that code. Would need LGPL... :(

Okay, that seems to pacify it. But there are some scary remarks about potential artifacts if I use this option. Under what conditions can this occur?
As long as you don't seek there shouldn't be any problems. If you seek, there may be artifacts visible for a short time (probably not much longer than a few seconds) - or not. I'd suggest to simply give it a try...

Don't know if someone as already reported on that, but doesn't seem so... Using latest versions of eac3to (2.6*), I encounter a problem when using the -down16 option on 24 bits soundtracks (WAV and DTS-HD, so far, not tested on trueHD or PCM).

downmixing to 16 bits is OK, but the audio is broken (weird sounds) as if the header was wrong. In fact, just passing the audio through wavewizard without any modifications fixes the problem...

Edit: just checked again with wavewizard.. the corrupt wav file says 16 bits per sample and 24 value bits per sample in its header. That may be the explanation...
Can you upload a little sample? What you report sounds a bit strange to me. It should be 24 bits per sample data structure and 16 valid bits per sample. Maybe wavewizard just names the fields in a strange way. But yes, newer eac3to versions write different wav files compared to older eac3to versions and that may confuse some applications or DirectShow filters. But as far as I can see, eac3to's WAV files are perfectly fine. It's not my fault if other software can't handle them. You can ask eac3to to create stupid simple WAV files by using the "-simple" option. Also if your WAV file is 16/24, you can use "eac3to source.wav dest.wav". That will then automatically strip the zero bytes and give you a smaller WAV file which is straight 16bit.

There are a lot more "old" BD Releases which contains seemless branching and "splitted" Releases that produce Audio-Spikes and loud noise at the cutting/fixxing Points after timecode-rerun :(

Something is borked when using DTS, no matter if it´s Arcsoft or Sonic. With AC3-Track splitting/joining is ok.

As i can remember, the same BD´s using an older eac3to version < 2.58 doesn´t produce this errors.

A good try will be Conair/German BD. Try to assemble
the GERMAN DTS Track. On the cut point there is a loud
spike on the decoded file (destination format can be WAV or AC3 - result is the same)

Please check your audio/DTS Handling. Since the > 5.1 "fixxing" a lot of previous good working, older things, are screwed up more and more :(

The destorted Audio ONLY comes up when correcting overlaps!

On 2.67 Version a quickhelp was, not to use the HD-Part (export the -core and join). But since 2.69 this is no longer working. Now Core and HD/MA-Part produces drops&spikes.

When No other Track (AC3/TrueHD) is available I´am screwed on this overlapping BD´s...
I guess you haven't compared v2.67 with v2.69 with the same title, have you? I don't think I've changed anything. I think the latest version should behave similar to much older versions.

I'll check out Con Air, anyway.

Tried to pass the AC-3 streams through delaycut in order to fix these. Tried all options including ignore, silence, repair but while it identifies the errors and completes okay, when this file is muxed back with the video, it goes out of sync; presumably after the point where the first CRC error had occurred.
delaycut can only do so much. If the audio stream is broken too much, audio sync may be lost. There's nothing we can do about that.

I think I asked that a few weeks aga but I think I didn´t got an answer
That's probably because the very same question has already been asked & answered about a million times in this thread... ;)

As long as you use eac3to to demux or remux or transcode the audio, you don't have to care about audio delays. eac3to will do everything automatically for you.

I was using 2.68 version to split DD 5.1 EX file from DVD to 6 mono waves. I don't have nero 7.xx installed on my machine so the track was processed by ffmpeg decoder. The result was not good - too many peak level overflows so the audible crack noise is heard when the overflow happens.
I've never had problems with DD tracks+ffmpeg decoder.
Maybe this is connected with EX or I'm just unaware of some option to normalize peaks?

I've tried the very same procedure on other machine where Nero 7.10 is installed - the result was fine: clear 6 channels with no overflow at all.

Madshi, if you want a sample i'll provide one.
Yes, a sample would be welcome. I'd forward it to the libav AC3 decoder maintainer.

DelayCut does indeed vomit out a waterfall of errors, but it plays perfectly in MPC or VLC ???
MPC and VLC just skip over broken data. That's why they're able to play these files. If you let delaycut run over the track, the end result should be similar to what MPC and VLC are doing...

xkodi

25th October 2008, 12:33

don't know if this is valuable information, just found it while looking with hex editor in DTSHDDec.dll from WinDVD9:

-dn/m instructs the decoder to perform the down-mix to the desired channel configuration e.g,:

-d1 - down-mix to mono
-d2 or -d2/o - down-mix to Lo/Ro
-d2/t - down-mix to Lt/Rt
-d2/1 - down-mix to L-R-Cs
-d2/1/.1 - down-mix to L-R-Cs-LFE
-d3 - down-mix to L-R-C
-d3/0/.1 - down-mix to L-R-C-LFE
-d2/2 - down-mix to L-R-Ls-Rs

-lx defines the mixing of the LFE data into the Front Channels of the Downmix where x is the flag 0 or 1 st.
0 -> LFE data is ignored; 1 -> LFE is mixed with 10dB boost.

It has effect only in the cases where -dm/n/.0 indicates no LFE output channel:
- for the case -d1 (-d1/0/.0) -l0 the Co=Co
- for the case -d1 (-d1/0/.0) -l1 the Co=0.5*(Co+3.16*LFE)
- for the case -d2 (-d2/0/.0) -l1 the Lo=0.5(*Lo+0.7078*3.16*LFE) and Ro=0.5*(Ro+0.7078*3.16*LFE)
- for the case -d3 (-d3/0/.0) -l1 the Lo=0.5*(Lo+0.7078*3.16*LFE), Ro=0.5*(Ro+0.7078*3.16*LFE) and Co=0.5*Co
- for the case -d2/2 (-d2/2/.0) -l1 the Lo=0.5*(Lo+0.7078*3.16*LFE), Ro=0.5*(Ro+0.7078*3.16*LFE), Lso=0.5*Lso and Rso=0.5*Rso

PHD_1976

25th October 2008, 13:33

Madshi,
Here's the 5 minutes sample of DD 5.1 EX with peak overflow when processed with ffmpeg to 6 mono wavs.
sample (http://www.sendspace.com/file/pn8z40)

Thanx.

madshi

25th October 2008, 13:38

don't know if this is valuable information, just found it while looking with hex editor in DTSHDDec.dll from WinDVD9:
It is interesting in that way that it shows that the LFE channel is boosted by 10db for downmixing. eac3to currently doesn't do that. I guess I should add the 10db boost?

madshi

25th October 2008, 13:48

Here's the 5 minutes sample of DD 5.1 EX with peak overflow when processed with ffmpeg to 6 mono wavs.
Hmmmm... In Audacity I don't see that much of a difference between libav and Nero. Both have very high peaks. It might be that with libav the peak overflows over so slightly and with Nero not. But the difference must be really small.

Don't know what to say/do about this now. Maybe adding "gain" functionality to eac3to would make sense...

PHD_1976

25th October 2008, 17:08

Well, it's not that I'm complaining of course, just wanted you to know about this issue.
With Nero decoder peaks are also high, you're right, but no audible crackling noise.

I know that besweet decoder performs downmix when overflow happens, maybe you would like to do it the same way?

Anyway, thank you for your time and sample checking :-)

Zwitterion

25th October 2008, 17:36

It is interesting in that way that it shows that the LFE channel is boosted by 10db for downmixing. eac3to currently doesn't do that. I guess I should add the 10db boost?
Does eac3to actually downmix the LFE channel using -down2? Dolby recommends to just discard it (phase issues, overdrive, you don't hear it anyway...).

madshi

25th October 2008, 17:48

Well, it's not that I'm complaining of course, just wanted you to know about this issue.
With Nero decoder peaks are also high, you're right, but no audible crackling noise.
Thanks for insisting... ;) Cause that made me find a bug in eac3to, which didn't handle peak clipping correctly, when using libav for decoding. This will be fixed in the next build. The crackling problems should be gone then...

Does eac3to actually downmix the LFE channel using -down2? Dolby recommends to just discard it (phase issues, overdrive, you don't hear it anyway...).
The LFE is not mixed in by default, but you can ask eac3to to do that by using the "-mixlfe" option.

bmnot

26th October 2008, 05:57

madshi

26th October 2008, 08:35

Instead of having to pick a version of a seamless branching movie, and then mux that single version to mkv, would it be possible to make a chapter/playlist for the separate files, convert all to mkv, then then add them all into one mkv and be able to pick the version you want to watch? I know there would be some issues with audio gaps, but it might be worth it instead of having to make a separate mkv for each version of the movie. I've seen it done for some movies.
I don't think MKV supports such a kind of thing in any reasonable way. Sure, one could mux a monster movie containing all the fragments of all branchings, but then you'd probably have to skip chapters manually. I don't think there's any kind of "playlist" for chapters in MKV. And even if there was, the jumps from one chapter to another would surely not be seamless. So I don't think that this is a good idea. I understand why you want this, but you'll have to keep using the Blu-Ray structure for seamless branching movies, if you want to keep the seamless branching alive.

kurt

26th October 2008, 10:33

madshi

26th October 2008, 14:03

I have another question about these gaps:

Do I have to rerun the exact same command line for seamless branching titles or is it only necassary for audio parts?

i.e:

1. eac3to 2) 2: video.vc1 4: english.ac3
2. eac3to 2) 4: english.ac3

that could save some time, right?
Correct, you only need to rerun those tracks which you want to have gap/overlap-fixed. It's just easier to rerun the same command line. Also it would be difficult to explain what you need to do exactly to save space. That's why eac3to simply asks to rerun the same command line.

piratburner

26th October 2008, 14:06

lexor

26th October 2008, 15:24

Is it possible to extract movie and sound of a BD and make a m2ts file of this files ( perhaps another prog.)
I working on Million Dollar Baby 2004 1080p Blu-ray GER AVC DTS-HD HR 7.1
On TsMuxeR I have no sound after I have remuxed with movie and English DTS-HD 7.1 sound

How should I do it with eac3to ?

eac3to does not produce m2ts files. it's a demuxing tool (it only stores video into mkv if you tell it to). So your best bet is to use eac3to to convet that DTS-HD into something TSmuxer can work with. Ask the guys in TSmuxer thread regarding which formats that tool is good for.

rebkell

26th October 2008, 16:06

Correct, you only need to rerun those tracks which you want to have gap/overlap-fixed. It's just easier to rerun the same command line. Also it would be difficult to explain what you need to do exactly to save space. That's why eac3to simply asks to rerun the same command line.

You really can't save space can you? You can run the same command line without the video, but on the first pass you still have to have the video included to fix the gaps, so you have to have enough space on the first pass to hold the video.

There isn't anyway to fix the gaps without demuxing the video, at least on the first pass, is there? If there is I'd be very interested in knowing how to do it.

rica

26th October 2008, 16:44

There isn't anyway to fix the gaps without demuxing the video, at least on the first pass, is there?

"0n the first pass" correct,
"without demuxing the video" incorrect.

Boulder

26th October 2008, 17:35

I've searched for comparisons but found none, except the infinitewave site you linked to. Where did you read that people prefer SSRC over r8brain?There are no exact comparisons, I'm afraid. Reading through posts at hydrogenaudio.org shows that SSRC is recommended most of the time. This topic particularly points out one problem with r8brain: http://www.hydrogenaudio.org/forums/index.php?showtopic=64989&hl=r8brain (see post 7).

By the way, KikeG recommends these command lines for SSRC: http://www.hydrogenaudio.org/forums/index.php?showtopic=7133 (see post 8)

madshi

26th October 2008, 17:50

You really can't save space can you? You can run the same command line without the video, but on the first pass you still have to have the video included to fix the gaps, so you have to have enough space on the first pass to hold the video.

There isn't anyway to fix the gaps without demuxing the video, at least on the first pass, is there? If there is I'd be very interested in knowing how to do it.
I don't understand what you're aiming it. Either you want the video or you don't want it. If you want it, you need the space on the harddisk. If you don't want it, then simply don't ask eac3to to demux it!

There are no exact comparisons, I'm afraid. Reading through posts at hydrogenaudio.org shows that SSRC is recommended most of the time. This topic particularly points out one problem with r8brain: http://www.hydrogenaudio.org/forums/index.php?showtopic=64989&hl=r8brain (see post 7).
Thanks. Well, the guys talking about r8brain there are just looking at the graphs and comment on that. That still doesn't fully satisfy me. But anyway, SSRC is default in eac3to now, while r8brain can still optionally be used. So everybody should be happy.

By the way, KikeG recommends these command lines for SSRC: http://www.hydrogenaudio.org/forums/index.php?showtopic=7133 (see post 8)
The settings listed there are mainly for dithering. That doesn't apply here because eac3to fetches the full floating point processing result from SSRC. So SSRC's dithering functionality isn't used at all.

Boulder

26th October 2008, 18:13

The settings listed there are mainly for dithering. That doesn't apply here because eac3to fetches the full floating point processing result from SSRC. So SSRC's dithering functionality isn't used at all.What about eac3to's own dithering, could the method KikeG describes for dither type 3 be used when resampling:

ssrc --rate 44100 --bits 16 --dither 3 --pdf 1 0.7 infile.wav outfile.wav

rebkell

26th October 2008, 18:20

I don't understand what you're aiming it. Either you want the video or you don't want it. If you want it, you need the space on the harddisk. If you don't want it, then simply don't ask eac3to to demux it!

What I'm aiming at, which may not be feasible anyway, is that I do a lot of capturing with the hauppauge hdpvr unit and it tends to have drop outs, which can cause sync problems(mostly gaps). I use neuron2's DGAVCDec to index the video and then re-encode with x264, so in essence(I think it will work out) I will be creating a video file straight from the .ts file, but a straight demux of the audio doesn't always match up, because of the gaps and such when it has the momentary dropouts.

The way it is now, I have to create the mkv file, then demux that to .h264 before DGAVCDec can handle the video, but it can handle the video directly from the .ts file.

73ChargerFan

26th October 2008, 21:45

I don't think MKV supports such a kind of thing in any reasonable way.
Actually, it does and it works perfectly. The pieces are stored as separate mkv files, then are linked. Haali's muxer & splitter can handle them.
VERY SLICK.

Check out these pages:
101 things you never knew you could do with Matroska (http://www.mod16.org/hurfdurf/?p=8)
MKV Ordered Chapters: User's Manual (http://forums.animesuki.com/showthread.php?t=66444)

lchiu7

26th October 2008, 23:17

Since your captured streams are h.264, you'd have to use tools like TSPE, found at http://www.bitstreamtools.com, to cut the file, i.e., remove the commercials and leave only the 2.0 or 5.1 programming which will help keep things in sync.

I use TSPE to edit out the ads. It works fine. The resultant file plays fine and in sync.

But I try to recompress the video (extracted from the cut file) and mux it back with original audio, it goes out of sync. And Txremuxer (and MKVMerge) report bad AC-3 frames in the stream.

lchiu7

26th October 2008, 23:32

these streams are invalid because they switch from 2.0 to 5.1 and back and forth?

What program are you using to extract the audio with? I've had plenty of sync problems and those are a pain to deal with, but try dgavcdec to extract the ac3 stream with, if it can be extracted I wouldn't expect any problems muxing it back in.

Sorry madshi, I know this is off topic.

No - there are two streams in the audio AC-3 (DD 5.1) and AAC-HE LATM. Txmuxer just ignores the AAC audio (says invalid) and dgavcindex offers the choice to demux either (I always choose AC-3).

Think as you say we are getting a bit OT. The on topic part is"

1. demux the audio out using tsremux or dgavcinxex
2. mux it back with the original video

Get AC-3 CRC frame errors and out of sync

Try to fix with eacc3to and it says bad data or whatever, use delaycut to fix. Try do do that and no matter which option I choose with delaycut, (skip, fix, silence), while no more errors, muxed audio/video is out of sync

madshi

26th October 2008, 23:34

What I'm aiming at, which may not be feasible anyway, is that I do a lot of capturing with the hauppauge hdpvr unit and it tends to have drop outs, which can cause sync problems(mostly gaps). I use neuron2's DGAVCDec to index the video and then re-encode with x264, so in essence(I think it will work out) I will be creating a video file straight from the .ts file, but a straight demux of the audio doesn't always match up, because of the gaps and such when it has the momentary dropouts.

The way it is now, I have to create the mkv file, then demux that to .h264 before DGAVCDec can handle the video, but it can handle the video directly from the .ts file.
eac3to "fixes" gaps/overlaps in video tracks by setting the timestamps in the MKV container accordingly. If you demux the video track from the MKV container the gap/overlap fixes get lost.

Actually, it does and it works perfectly. The pieces are stored as separate mkv files, then are linked. Haali's muxer & splitter can handle them.
VERY SLICK.

Check out these pages:
101 things you never knew you could do with Matroska (http://www.mod16.org/hurfdurf/?p=8)
MKV Ordered Chapters: User's Manual (http://forums.animesuki.com/showthread.php?t=66444)
Cool. Didn't know that!

However, I'm using the Haali Matroska Muxer and while the Haali Splitter might support all these fancy features, I don't think the Haali Muxer does. At least I wouldn't know how to activate and control these features. Also this sounds like a really tough job for a muxer and so I don't plan to seriously look into this.

nwg

26th October 2008, 23:50

rebkell

26th October 2008, 23:53

eac3to "fixes" gaps/overlaps in video tracks by setting the timestamps in the MKV container accordingly. If you demux the video track from the MKV container the gap/overlap fixes get lost.

Ahh, Ok, I didn't realize that, I always thought you actually padded the audio stream or removed some it to fit.

madshi

27th October 2008, 00:01

eac3to v2.70 released

http://madshi.net/eac3to.zip

* added floating point support to the complete audio processing chain
* added gain functionality, e.g. "-3db" or "+1db"
* bitdepth analyzation is now done separately for each channel
* fixed: when decoding lossy audio with libav, peaks were clipped incorrectly
* fixed: libav MP1/2/3 decoder output was cut down to 24bit
* fixed: with some EVO sources the AC3 track was not listed
* fixed: if no key frame was found, h264 track in m2(ts) was not listed
* fixed: video/audio data before first PAT/PMT was discarded
* Blu-Ray chapters now don't contain link points, anymore, unless necessary
* added 10db boost to LFE channel, when "-down2" and "-mixlfe" are used
* ArcSoft output can now be overwritten to "-2", "-6", "-7" or "-8" channels
Some more detailed comments:

* added floating point support to the complete audio processing chain
Some more related changes coming in a future build. E.g. currently I'm still passing 24bit integer to audio encoders. In a future build I'll pass floating point, whereever possible.

The changes I've done so far seem to work just fine, but there might be some new bugs.

* bitdepth analyzation is now done separately for each channel
I've checked about 20 tracks but found none where any of the channels differed from the others. If all channels share the same max bitdepth, eac3to will report the statistics as it always did. However, if some of the channels have different max bitdepth values, eac3to will now report the bitdepth statistics seperately for every channel group which has identical attributes.

* ArcSoft output can now be overwritten to "-2", "-6", "-7" or "-8" channels
This one is for xkodi and especially for tebasuna51. You can ask the ArcSoft decoder to output 2, 6, 7 or 8 channels, regardless of which channel configuration the source has. Interesting tests would be:

2.0 -> 6
2.0 -> 7
2.0 -> 8
5.1 -> 2
5.1 -> 7
5.1 -> 8
6.1 -> 2
6.1 -> 6
6.1 -> 8
7.1 -> 2
7.1 -> 6
7.1 -> 7

So there's a lot to test. Maybe we can figure out how down- and upmixing should ideally be done. But the big question is: Does the ArcSoft decoder do all that in an ideal way? Maybe the first check should be to see what 5.1 -> 2 does? If that uses proper Dolby Pro Logic II then maybe that's a good sign that the general up-/downmixing quality should be ok?

madshi

27th October 2008, 00:03

What does sync byte missing mean? I have been getting it on a few discs when trying to convert TruHD to Ac3 and also DTS to Ac3.

I have been using several GUI's and it looks like it is eac3to itself.
It means that the source file is corrupt. Try reripping the disc. If that doesn't help the disc might have an authoring fault.

Ahh, Ok, I didn't realize that, I always thought you actually padded the audio stream or removed some it to fit.
It does actually change the audio stream. I was talking about the VIDEO stream.

rebkell

27th October 2008, 00:12

It does actually change the audio stream. I was talking about the VIDEO stream.

Good, I was wondering how I had done any encodes with audio gaps in them. I guess that all the gap/overlap problems I've had must have been corrected with the audio portion of the program.

nwg

27th October 2008, 01:13

It means that the source file is corrupt. Try reripping the disc. If that doesn't help the disc might have an authoring fault.

I have had it on the last three blu rays and tried re-ripping and still does it. It plays fine in the BD player.

BlackJack1

27th October 2008, 02:21

Tried to remux to mkv "The Fugitive" HD-DVD:
eac3to v2.70
command line: eac3to\eac3to K:\HD\BD_THE_FUGITIVE_HDDVD\HVDVD_TS 1) 3: o:\ fugitive.mkv 4: o:\fugitive.ac3 2: o:\fugitive_chpts.txt
------------------------------------------------------------------------------
EVO, 1 video track, 4 audio tracks, 5 subtitle tracks, 2:10:14
"Feature Presentation"
1: Joined EVO file
2: Chapters, 42 chapters without names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: E-AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: E-AC3, French, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: E-AC3, German, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
7: E-AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
"Commentary"
8: Subtitle, English
9: Subtitle, French
10: Subtitle, German
11: Subtitle, English, "SDH"
12: Subtitle, German, "SDH"
This video conversion is not supported.
What I'm doing wrong...?

rebkell

27th October 2008, 02:25

Tried to remux to mkv "The Fugitive" HD-DVD:

What I'm doing wrong...?

It looks like you have a space after the first o:\

command line: eac3to\eac3to K:\HD\BD_THE_FUGITIVE_HDDVD\HVDVD_TS 1) 3: o:\ fugitive.mkv 4: o:\fugitive.ac3 2: o:\fugitive_chpts.txt

BlackJack1

27th October 2008, 02:29

Oooo shit...
Thx. My fault :(

rica

27th October 2008, 02:37

It looks like you have a space after the first o:\

command line: eac3to\eac3to K:\HD\BD_THE_FUGITIVE_HDDVD\HVDVD_TS 1) 3: o:\ fugitive.mkv 4: o:\fugitive.ac3 2: o:\fugitive_chpts.txt

Yes;
"o:\ fugitive.mkv " must have been:
"o:\fugitive.mkv "

This is one of the most common errors which might be done.

Simple but easy to be made.

(I've made lots of similars.)

itsancho

27th October 2008, 03:04

oops, just to report some small bug(s) in 2.70
eac3to 2.69
"K:\Mannen som elsket Yngve (2008) (The Man Who Loved Ynge) (NOR)Blu-ray AVC dts-HD Hi-Res\MANNEN SOM ELSKET YNGVE" 1)
M2TS, 1 video track, 4 audio tracks, 4 subtitle tracks, 1:34:23
1: Chapters, 13 chapters
2: h264/AVC, 1080i50 (16:9)
3: AC3, Norwegian, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
4: DTS Hi-Res, Norwegian, 5.1 channels, 16 bits, 1774kbps, 48khz, dialnorm: -4dB
5: AC3, Norwegian, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: AC3, Norwegian, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
7: Subtitle (PGS), Danish
8: Subtitle (PGS), Norwegian
9: Subtitle (PGS), Swedish
10: Subtitle (PGS), Finnish

eac3to 2.70
"K:\Mannen som elsket Yngve (2008) (The Man Who Loved Ynge) (NOR)Blu-ray AVC dts-HD Hi-Res\MANNEN SOM ELSKET YNGVE" 1)
TS, 1 video track, 4 audio tracks, 4 subtitle tracks, 1:34:23
1: h264/AVC, 1080i50 (16:9)
2: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
3: DTS Hi-Res, 5.1 channels, 16 bits, 1774kbps, 48khz, dialnorm: -4dB
4: AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
5: AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: Subtitle (PGS)
7: Subtitle (PGS)
8: Subtitle (PGS)
9: Subtitle (PGS) and 10x once again 4 the great tool, madshi!

jeffnoone

27th October 2008, 05:30

Extracting from Dave Matthews BR disc, which has 5.1 AC3, PCM 2-ch, and TrueHD 5.1

Used eac3to 2.69 to extract the VC1 and all three audio tracks in one run; used DOS commandline, and -i 24 -c 6 -s 48000

For the TrueHD, I just extracted using the stdout technique, and saved as 6-channel PCM

Anyway tsMuxeR will import and correctly recognize the video and all three audio tracks (after I ran the 2-channel PCM thru PCM2TSMU)

But there is some problem with the BR files I produce
When running tsMuxer, it says at one point about processing a certain number of video frames (as if it has finished video early)

When you let thing complete, there are no declared errors, but PowerDVD reports as 5.38.04 i length (way too long), and video seems to "run out" about half way thru.
The AC3 track plays fine and is synced
There is sound on the 6-ch PCM but it appears very slow

I have successfully extracted 6 and 8 channel PCM before from other formats including TrueHD, but something seems to have gone wrong here - any ideas

Thanks

Unhandled

27th October 2008, 06:02

First of all, thank you madshi. eac3to is a great piece of software. :)

I believe I found a regression in downmixing. Seems to have been introduced in v2.64. When I try to downmix a TrueHD 5.1 track to 2.0 (-down2), eac3to reports it can't downmix this channel configuration. It worked in v2.63:

C:\Video Stuff\eac3to_263>eac3to.exe f:\content\hddvd\HALO_22\HVDVD_TS 1) 4: ..\test.wav -down2
EVO, 1 video track, 3 audio tracks, 1 subtitle track, 1:32:28
"WinterTour"
1: Chapters, 21 chapters with names
2: VC-1, 1080p30 /1.001 (16:9)
3: AC3, English, 5.1 channels, 448kbps, 48khz
4: TrueHD, English, 5.1 channels, 48khz
5: AC3, English, 2.0 channels, 256kbps, 48khz
6: Subtitle, English
[a04] Extracting audio track number 4...
[a04] Decoding with libav/ffmpeg...
[a04] Downmixing multi channel audio to stereo...
Loading white noise (needed for dithering)...
[a04] Writing WAV...
[a04] This audio track contains more than 16 bits of information.
[a04] Creating file "..\test.24bit.wav"...
^C

C:\Video Stuff\eac3to_264>eac3to.exe f:\content\hddvd\HALO_22\HVDVD_TS 1) 4: ..\test.wav -down2
EVO, 1 video track, 3 audio tracks, 1 subtitle track, 1:32:28
"WinterTour"
1: Chapters, 21 chapters with names
2: VC-1, 1080p30 /1.001 (16:9)
3: AC3, English, 5.1 channels, 448kbps, 48khz
4: TrueHD, English, 5.1 channels, 48khz
5: AC3, English, 2.0 channels, 256kbps, 48khz
6: Subtitle, English
[a04] Can't downmix this channel configuration.
[a04] Internal error - unknown audio output format!

Same problem occurs when trying to downmix 5.1 WAV or 5.1 FLAC file created by eac3to v2.64 from the TrueHD track.
My source is an HD-DVD folder. The title is "Nine Inch Nails - Beside You In Time". Can anyone else confirm this behavior on similar tracks?
Thank you.

svgame

27th October 2008, 06:13

madshi i want you help me
In order to import TrueHD stream from BD to Scenarist HDMV you have to do the following:
- run TrueHD track via eac3to twice:
1. eac3to track.thd audio1.ac3
2. eac3to track.thd audio1.thd
Then rename audio1.thd to .mlp and feed MUI Generator with audio1.ac3 specifying it as Dolby Lossless.
In next window be sure, but at last the error message"External component has thrown an exception"
please help me``madshi

madshi

27th October 2008, 08:46

oops, just to report some small bug(s) in 2.70
and 10x once again 4 the great tool, madshi!
Ouch. Thanks, will look into this.

But there is some problem with the BR files I produce
When running tsMuxer
This is not the tsMuxeR support thread.

I believe I found a regression in downmixing. Seems to have been introduced in v2.64. When I try to downmix a TrueHD 5.1 track to 2.0 (-down2), eac3to reports it can't downmix this channel configuration.
Thanks for the report. Problem will be fixed in the next build.

In order to import TrueHD stream from BD to Scenarist HDMV you have to do the following
No. Just do "eac3to track.thd audio1.thd+ac3".

madshi

27th October 2008, 09:01

eac3to v2.71 released

http://madshi.net/eac3to.zip

* fixed: v2.70 detected Blu-Rays as "TS" without chapters and track languages
* fixed: TrueHD downmixing to 2.0 didn't work

evdberg

27th October 2008, 10:44

eac3to v2.70 released

* bitdepth analyzation is now done separately for each channel
Thanks!

I've checked about 20 tracks but found none where any of the channels differed from the others. If all channels share the same max bitdepth, eac3to will report the statistics as it always did. However, if some of the channels have different max bitdepth values, eac3to will now report the bitdepth statistics seperately for every channel group which has identical attributes.

I retested the same Universal movie and again got the same result:
"Original audio track: max 24 bits, average 19 bits, most common 16 bits."
So this means that the result is valid for all channels, and not some channels 16 bits and other 24 bits.

shambles

27th October 2008, 10:49

* bitdepth analyzation is now done separately for each channel
I've checked about 20 tracks but found none where any of the channels differed from the others. If all channels share the same max bitdepth, eac3to will report the statistics as it always did. However, if some of the channels have different max bitdepth values, eac3to will now report the bitdepth statistics seperately for every channel group which has identical attributes.

doesn't seem to work.. i checked out a few tracks where the channel bit depths differ and eac3to still reports the same it did before

C:\Program Files\eac3to>eac3to e:\inside.flac -analyzebitdepth
FLAC, 5.1 channels, 2:08:44, 24 bits, 2097kbps, 48khz
Decoding FLAC...
Original audio track: max 24 bits, average 19 bits, most common 16 bits.
eac3to processing took 2 minutes, 18 seconds.
Done.

C:\Program Files\eac3to>eac3to e:\inside.flac f:\inside.wavs
FLAC, 5.1 channels, 2:08:44, 24 bits, 2097kbps, 48khz
Decoding FLAC...
Writing WAVs...
Creating file "f:\inside.LFE.wav"...
Creating file "f:\inside.R.wav"...
Creating file "f:\inside.C.wav"...
Creating file "f:\inside.L.wav"...
Creating file "f:\inside.SL.wav"...
Creating file "f:\inside.SR.wav"...
Original audio track: max 24 bits, average 19 bits, most common 16 bits.
eac3to processing took 3 minutes, 15 seconds.
Done.

C:\Program Files\eac3to>eac3to f:\inside.c.wav -analyzebitdepth
WAV, 1.0 channels, 2:08:44, 24 bits, 1152kbps, 48khz
Reading WAV...
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 22 seconds.
Done.

C:\Program Files\eac3to>eac3to f:\inside.l.wav -analyzebitdepth
WAV, 1.0 channels, 2:08:44, 24 bits, 1152kbps, 48khz
Reading WAV...
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 22 seconds.
Done.

C:\Program Files\eac3to>eac3to f:\inside.r.wav -analyzebitdepth
WAV, 1.0 channels, 2:08:44, 24 bits, 1152kbps, 48khz
Reading WAV...
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 22 seconds.
Done.

C:\Program Files\eac3to>eac3to f:\inside.lfe.wav -analyzebitdepth
WAV, 1.0 channels, 2:08:44, 24 bits, 1152kbps, 48khz
Reading WAV...
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 28 seconds.
Done.

C:\Program Files\eac3to>eac3to f:\inside.sl.wav -analyzebitdepth
WAV, 1.0 channels, 2:08:44, 24 bits, 1152kbps, 48khz
Reading WAV...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 25 seconds.
Done.

C:\Program Files\eac3to>eac3to f:\inside.sr.wav -analyzebitdepth
WAV, 1.0 channels, 2:08:44, 24 bits, 1152kbps, 48khz
Reading WAV...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 30 seconds.
Done.

sample, just in case you need one: http://www.sendspace.com/file/fvrzoi

madshi

27th October 2008, 11:18

doesn't seem to work.. i checked out a few tracks where the channel bit depths differ and eac3to still reports the same it did before
Thanks for the sample. There was a bug in the bitdepth analyzer's channel splitter. Will be fixed in the next build. And this is what I'm now getting for your sample and also for Notting Hill:

eac3to "Notting Hill (1999)\Englisch.flac" -analyzebitdepth -50mb
FLAC, 5.1 channels, 2:04:03, 24 bits, 1842kbps, 48khz
Decoding FLAC...
Original audio track, L+R+C+LFE: constant bit depth of 16 bits.
Original audio track, SL+SR: constant bit depth of 24 bits.
eac3to processing took 5 seconds.
Done.

shambles

27th October 2008, 11:21

awesome, thanks! :D

madshi

27th October 2008, 11:22

BlackJack1

27th October 2008, 12:12

ACrowley

27th October 2008, 12:20

eac3to v2.72 released

http://madshi.net/eac3to.zip

* fixed: per channel bitdepth analyzation didn't work correctly
Enough updates now! Next build will have to wait until next week, unless some serious new bugs show up...

WOW ! Nice now i can get 7Ch Output from, DTS ES 6.1 Matrix Tracks.
I made a few Tests with DTS ES 6.1 Matrix and it looks right to me ?! The BC output from arcsoft -7 looks/sound Ok

Thunderbolt8

27th October 2008, 12:21

* fixed: when decoding lossy audio with libav, peaks were clipped incorrectly
* fixed: video/audio data before first PAT/PMT was discarded
2 questions regarding these: did those incorrectly clipped peaks result in loss of quality, would we have to redo these tracks?
the same for the PAT/PMT thing, did this result in some minor loss of quality or possible syncing mistakes?
thanks!

tebasuna51

27th October 2008, 13:23

...
Used eac3to 2.69 to extract the VC1 and all three audio tracks in one run; used DOS commandline, and -i 24 -c 6 -s 48000
...
There is sound on the 6-ch PCM but it appears very slow

Please put your eac3to log files and exact command lines used at this thread (http://forum.doom9.org/showthread.php?t=138417)

deank

27th October 2008, 13:46

Is there a way to make eac3to.exe to treat parameters enclosed in quotes?

So a command line: eac3to.exe "test.dts" "test.ac3" "-448" to be processed correctly?

i.e. if it is possible to make it recognise "-parameters" with quotes...

Thanks for this great tool!

madshi

27th October 2008, 13:53

I've got Sonic audio decoder 4.3.0.169 and Sonic DTS encoder 1.0.23.0 and would like to remux video and DTS-ES 6.1 24bit 48kHz 1536kbps track. Do I need Arcsoft DTS decoder or my Sonic will do the job?
You don't need any decoders for demuxing. You only need decoders for decoding.

WOW ! Nice now i can get 7Ch Output from, DTS ES 6.1 Matrix Tracks.
I made a few Tests with DTS ES 6.1 Matrix and it looks right to me ?! The BC output from arcsoft -7 looks/sound Ok
I'll wait for tebasuna51's check... :)

2 questions regarding these: did those incorrectly clipped peaks result in loss of quality, would we have to redo these tracks?
If you had a lossy (!) track which you decoded with libav and which had very high peaks then yes, the new eac3to version outputs better quality. However, this doesn't apply to TrueHD. Also it doesn't apply to Nero or Sonic decoders.

the same for the PAT/PMT thing, did this result in some minor loss of quality or possible syncing mistakes?
No sync mistakes. And no loss of quality. Just a loss of a few video frames at the beginning of the file. And this should only occur with broadcasts, if at all. It's not a very important fix.

Is there a way to make eac3to.exe to treat parameters enclosed in quotes?
What would you need that for?

PHD_1976

27th October 2008, 14:01

did those incorrectly clipped peaks result in loss of quality, would we have to redo these tracks?would we have to redo these tracks?

Madshi's already answered, just want to add a few words:

These peak overflow gives a crackle noise and its clearly audible. I found this with libav decoder and DD 5.1 EX track. So if you can hear crackles while watching a movie I guess you should redo your tracks. :-)

This evening I gonna check new build and report about clipping.

To Madshi:
1. Thanx a lot for a new build
2. Is there a way with eac3to to demux in 1 step TrueHD track directly from BD/HD into ac3 core and lossless part (for Scenarist)?
I tried it with HD DVD yesterday and i made 2 steps: 1) thd+ac3 track creation 2) demuxing to ac3 and mlp. It's not a problem for me, just curious.

BlackJack1

27th October 2008, 14:53

You don't need any decoders for demuxing. You only need decoders for decoding.
I need to remux no. 4: DTS-ES 6.1 track 1536k and 3: DD 5.1EX 640k track.
Here is the log:
eac3to v2.72
command line: eac3to "O:\# HD New\The_Untouchables_Blu-ray_1987-PerfectionHD\BDMV" 1) 1: o:\untouchables.txt 2: o:\untouchables.mkv 4: o:\untouchables.dts 3: o:\untouchables.ac3
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 4 subtitle tracks, 1:59:27
1: Chapters, 24 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
4: DTS-ES, English, 6.1 channels, 24 bits, 1536kbps, 48khz, dialnorm: -4dB
5: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
7: Subtitle (PGS), English
8: Subtitle (PGS), English
9: Subtitle (PGS), French
10: Subtitle (PGS), Spanish
Creating file "o:\untouchables.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a03] Extracting audio track number 3...
[a04] Removing DTS dialog normalization...
[a03] Removing AC3 dialog normalization...
[v02] Muxing video to Matroska...
[a04] Creating file "o:\untouchables.dts"...
[a03] Creating file "o:\untouchables.ac3"...
Added fps value to MKV header.
Video track 2 contains 171840 frames.
eac3to processing took 37 minutes, 8 seconds.
Done.

but... PDVD reports that I've got DTS-ES 5.1 1536k track and not 6.1...
Previously during playing BD movie using the same PDVD - reported: DTS-ES 6.1 1536k.
Please explain what I did wrong.

ACrowley

27th October 2008, 15:05

I'll wait for tebasuna51's check..

@madshi

Arcsoft -7 on DTS ES 6.1 "Matrix" looks very good !

I made a Test :

DTS ES 6.1 "Discrete" to .wavs with eac3to/Arcsoft. Then i made a reencode from these 7 mono waves to DTS ES 6.1 "Matrix" with DTS Pro Series.
After that i made a decoding from the new DTS ES 6.1 "Matrix" with eac3to/Arcsoft Decoder and -7

Then i compared both Waves from the Back Center Channels (the BC from original discrete and BC from the Matrix reencode) in Audacity. They look fully identical :)

http://www.abload.de/thumb/sshot-1ytcb.png (http://www.abload.de/image.php?img=sshot-1ytcb.png)
http://www.abload.de/thumb/sshot-25i2g.png (http://www.abload.de/image.php?img=sshot-25i2g.png)

itsancho

27th October 2008, 15:43

btw, madshi, is there any chance eac3to to report if AC3 track is DD-EX?
10x again and again...
;-)

svgame

27th October 2008, 16:06

thank you Madshi,just there has a problem.i make 3 step
1) use eac3to demux TrueHD track directly from BD/HD
2) type command "eac3to 00001.thd audio.thd" "eac3to 00001.thd audio.ac3",demuxing to ac3 core and lossless part
3) rename audio.thd to audio.mlp,and import to Scenarist.at 100%,its hung!!!about 2 minute，get a error message "External component has thrown an exception"
where is problem?Madshi

tebasuna51

27th October 2008, 17:45

Originally Posted by ACrowley
WOW ! Nice now i can get 7Ch Output from, DTS ES 6.1 Matrix Tracks.
I made a few Tests with DTS ES 6.1 Matrix and it looks right to me ?! The BC output from arcsoft -7 looks/sound Ok

I'll wait for tebasuna51's check... :)

Not all Ok here.
If we name source channels like Rear: RL, RR, RC and

RL' = +90º phase shift RL
RR' = -90º phase shift RR

seems the phase shift is make at encoder phase and is the same for 'matrix' and 'discrete' DTS-ES in all the tests.

The output channels after encode to 3/3.1 DTS-ES discrete and decoded with ArcSoft:

1) 3/3.1 -> 5.1 (like previous test)

SL = RL' + 0.71 x RC
SR = RR' + 0.71 x RC
Ok

2) 3/3.1 -> 6.1

BC = RC
SL = RL'
SR = RR'
Ok

3) 3/3.1 -> 7.1

SL = 0.40 x RL' + 0.60 x RC
SR = 0.40 x RR' + 0.60 x RC
BL = 0.75 x RL'
BR = 0.75 x RR'
Seems wrong: RC only in Side and most Rear part in Back, I think must be the other way, Side <-> Back

----------------------------
The output channels after encode to 3/3.1 DTS-ES matrix and decoded with ArcSoft:

1) 3/3.1 -> 5.1

SL = RL' + 0.71 x RC
SR = RR' + 0.71 x RC
Ok

2) 3/3.1 -> 7.1

SL = 0.71 x RL' + 0.50 x RC
SR = 0.71 x RR' + 0.50 x RC
BL = 0.71 x RL' + 0.50 x RC
BR = 0.71 x RR' + 0.50 x RC
For me seems ArcSoft can't recover the RC mixed in RL-RR because RC must be present only in Back channels.

3) 3/3.1 -> 6.1, eac3to crash with this bugreport.txt (http://pastebin.com/m276c45a0)

The report was made with 2.70 but is the same with 2.72, I send the 2.70 bug because was made after a reset with the minimum of soft running.

piratburner

27th October 2008, 18:48

eac3to does not produce m2ts files. it's a demuxing tool (it only stores video into mkv if you tell it to). So your best bet is to use eac3to to convet that DTS-HD into something TSmuxer can work with. Ask the guys in TSmuxer thread regarding which formats that tool is good for.

OK if I use eac3to with -demux I got all files I need, can I then use this files to make a m2ts file ? if so wich program ? TsMuxer ?

rica

27th October 2008, 18:53

OK if I use eac3to with -demux I got all files I need, can I then use this files to make a m2ts file ? if so wich program ? TsMuxer ?

Yes. eac3to+TSMuxer.

odin24

27th October 2008, 19:14

Yes. eac3to+TSMuxer.

If the -demux feature is used does this demux the entire BD/HDDVD? Or is -demux used for specific streams in a track. Normally I would specify which stream I want along with it's destiation without using -demux.

nautilus7

27th October 2008, 19:22

It will demux the entire title you set.

piratburner

27th October 2008, 19:30

It will demux the entire title you set.

I using this command...
L:\leon>eac3to h:\bt\LEON 1) -demux

Then I going to test this in TsMuxeR instead for the original m2ts files,

Its looks there are a bug in TsMuxer for 7.1 DTS-HD

odin24

27th October 2008, 19:51

I using this command...
L:\leon>eac3to h:\bt\LEON 1) -demux

Then I going to test this in TsMuxeR instead for the original m2ts files,

Its looks there are a bug in TsMuxer for 7.1 DTS-HD

DTS-HD 7.1 works fine for me, HDDVD --> BD with DTSHD-MA 7.1, using tsMuxeR. All demuxed using eac3to. Maybe try demuxing the individual track instead of the whole disc.

Leon, as in the Professional on BD? Not in North America yet :(, one of my favs!

Snowknight26

27th October 2008, 19:52

It will demux the entire title you set.

Actually, only the first of however many video streams.

piratburner

27th October 2008, 20:35

DTS-HD 7.1 works fine for me, HDDVD --> BD with DTSHD-MA 7.1, using tsMuxeR. All demuxed using eac3to. Maybe try demuxing the individual track instead of the whole disc.

Leon, as in the Professional on BD? Not in North America yet :(, one of my favs!
Yes !
Léon 1994 Ultimate Edition 1080p Blu-ray GER AVC DTS-HD MA 7.1

With English sound :)

Here is a trackname for one DTS-HD when using -demux ....
00005 - 3 - DTS Hi-Res, German, 7.1 channels, 24 bits, 2864kbps, 48khz.dtshr

When playing a original m2ts file from Léon i have sound from my TviX but after TsMuxer = No Sound :-(

piratburner

27th October 2008, 20:45

DTS-HD 7.1 works fine for me, HDDVD --> BD with DTSHD-MA 7.1, using tsMuxeR. All demuxed using eac3to. Maybe try demuxing the individual track instead of the whole disc.

Are you always using eac3to to demux a BD and then TsMuxer ??

I have always used TsMuxer from original BD and make a "movie only" m2ts, So I will see how it works when using eac3to first :-)

rica

27th October 2008, 20:45

If the -demux feature is used does this demux the entire BD/HDDVD? Or is -demux used for specific streams in a track. Normally I would specify which stream I want along with it's destiation without using -demux.

I don't use -demux option. I specify the files as well.

piratburner

27th October 2008, 21:44

Are you always using eac3to to demux a BD and then TsMuxer ??

I have always used TsMuxer from original BD and make a "movie only" m2ts, So I will see how it works when using eac3to first :-)

Ok I got this 2 files from eac3to ... using -demux
00005 - 2 - h264, 1080p24.h264
00005 - 4 - DTS Hi-Res, English, 7.1 channels, 24 bits, 2864kbps, 48khz.dtshr

Add this 2 into TsMuxeR -> M2TS muxing -> play the file from TviX And I got NO sound :devil:

Used version 1.8.8

nautilus7

27th October 2008, 21:59

Obviously the problem is NOT eac3to. Why do you keep posting here? It would be better to open a new thread and discuss your problem there.

shroomM

27th October 2008, 22:10

Hey,

first, a big thank you for the program, it is a godsend :)
But I do have a couple of quick questions...

I have a RAW/PCM, 5.1 channels, 16 bits, 48khz track and would like to encode it to Dolby Digital 5.1. I know I can encode using eac3to by just specifying the output file as .ac3, but here's what I'm more interested in...

When I demux the PCM track, here's the eac3to's output...

[a04] Reading RAW/PCM...
[a04] Swapping endian...
[a04] Remapping channels...
[a04] Swapping endian...
[a04] Remapping channels...
[a04] Creating file "D:\bluray\eng.pcm"...

If I later run eac3to with the syntax eac3to D:\bluray\eng.pcm D:\bluray\eng_flac.flac, the output is....

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, 0:08:28, 16 bits, 4608kbps, 48khz
This is probably a Blu-ray PCM track. Will remap channels accordingly.
Reading RAW/PCM...
Swapping endian...
Remapping channels...
Encoding FLAC with libFlac...

I'm now wondering ... is the ordering of channels in the FLAC file OK or is it swapped?

The second question is about downmixing 5.1 to stereo (I know it's pretty much obsolete, but I need it in this case)...

Assuming that the FLAC file above is OK (channels and all), can I just load that FLAC in i.e. meGUI (provided that I have madFlac installed). Would that downmix be OK?

Is the alternative, adding a -down2 to eac3to when generating the FLAC also OK ?

himan2001

27th October 2008, 22:47

2. Is there a way with eac3to to demux in 1 step TrueHD track directly from BD/HD into ac3 core and lossless part (for Scenarist)?
I tried it with HD DVD yesterday and i made 2 steps: 1) thd+ac3 track creation 2) demuxing to ac3 and mlp. It's not a problem for me, just curious.

Still the same "issue" with DTS and the -core option.

I personly dislike 7.1 downmixing from DTS-HD 7.1 files, when
a "clean" 5.1 core is present.

But is not possible to reencode directly in 1 step to ac3 from the -core. eac3to always sees the 7.1 MA and try to downmix from the extention. At the moment, first the -core must be exported, than you can ac3-encode from the 5.1 core.

rica

27th October 2008, 23:01

Still the same "issue" with DTS and the -core option.

I personly dislike 7.1 downmixing from DTS-HD 7.1 files, when
a "clean" 5.1 core is present.

But is not possible to reencode directly in 1 step to ac3 from the -core. eac3to always sees the 7.1 MA and try to downmix from the extention. At the moment, first the -core must be exported, than you can ac3-encode from the 5.1 core.

Core is not an independent file as in THD+ac3 (BD);
it can not be read individually since core is an obligation for DTS-HD format: system works in this manner: core+extensions.
So if you want to re-encode it to ac3, eac3to must decode and downmix before encoding in case of extension is 7.1.

EDIT:
Some points:
If you want to extract core dts (5.1) as is, use -core option.
When you try to re-encode (or to make a simple revision like convert 24 to 16), not only eac3to but any program has to decode the original file first.
If you don't ask eac3to extracting core, it is gonna directly try to decode DTS-HD which is 7.1 in your case.
But default encoder (aften) is not able to encode to 7.1; so the original has to be downconverted to 5.1 by eac3to.
Unless you use a pro encoder by Dolby, you will never able to get an 7.1 dolby re-encode; even in this case, as you guess, eac3to must firstly decode your DTS-HD.
So difference between you and the guy from whom you taken the quote is;
he is getting/or extracting a seperate ac3 (as is) from thd+ac3 combination -it is not a core,
but he has to make a second command line since it needs a lossless thd as well.
you, on the other hand, are re-encoding DTS-HD, not core.
So the way you follow;
extracting core dts
and later,
converting it to 5.1 is normal process.
BTW; shift+arrow up repeats your previous command.

tebasuna51

28th October 2008, 03:05

...
I'm now wondering ... is the ordering of channels in the FLAC file OK or is it swapped?
Must be Ok. If you detect any problem please report it.

The second question is about downmixing 5.1 to stereo (I know it's pretty much obsolete, but I need it in this case)...

Assuming that the FLAC file above is OK (channels and all), can I just load that FLAC in i.e. meGUI (provided that I have madFlac installed). Would that downmix be OK?
Why not?

Is the alternative, adding a -down2 to eac3to when generating the FLAC also OK ?
Of course. Test yourself.

If the raw/pcm track from your BD is the 'x' you can use directly:
eac3to D:\bluray x: D:\bluray\eng_flac2.flac -down2

ACrowley

28th October 2008, 09:39

Not all Ok here.
If we name source channels like Rear: RL, RR, RC and

RL' = +90º phase shift RL
RR' = -90º phase shift RR

seems the phase shift is make at encoder phase and is the same for 'matrix' and 'discrete' DTS-ES in all the tests.

The output channels after encode to 3/3.1 DTS-ES discrete and decoded with ArcSoft:

1) 3/3.1 -> 5.1 (like previous test)

SL = RL' + 0.71 x RC
SR = RR' + 0.71 x RC
Ok

2) 3/3.1 -> 6.1

BC = RC
SL = RL'
SR = RR'
Ok

3) 3/3.1 -> 7.1

SL = 0.40 x RL' + 0.60 x RC
SR = 0.40 x RR' + 0.60 x RC
BL = 0.75 x RL'
BR = 0.75 x RR'
Seems wrong: RC only in Side and most Rear part in Back, I think must be the other way, Side <-> Back

----------------------------
The output channels after encode to 3/3.1 DTS-ES matrix and decoded with ArcSoft:

1) 3/3.1 -> 5.1

SL = RL' + 0.71 x RC
SR = RR' + 0.71 x RC
Ok

2) 3/3.1 -> 7.1

SL = 0.71 x RL' + 0.50 x RC
SR = 0.71 x RR' + 0.50 x RC
BL = 0.71 x RL' + 0.50 x RC
BR = 0.71 x RR' + 0.50 x RC
For me seems ArcSoft can't recover the RC mixed in RL-RR because RC must be present only in Back channels.

3) 3/3.1 -> 6.1, eac3to crash with this bugreport.txt (http://pastebin.com/m276c45a0)

The report was made with 2.70 but is the same with 2.72, I send the 2.70 bug because was made after a reset with the minimum of soft running.

mhhh...youre right.
I compared only the BC Channels.
And the BC looks fully identical as you can see in my Post

But i think this can be fixed ?
I hope so ,because i have a lot of dts es 6.1 "Matrix" Tracks on my HDD ,waitnig for a proper Method to decode all 7 CH for a reencode ( all PAL Audio Tracks/Target is PAL to NTSC-reencode to dts es 6.1 "discrete")

On the other Side, i dont think the 6.1 Output from matrix 6.1from Arcsoft is sooooo wrong..better then nothing ?!

tebasuna51

28th October 2008, 11:39

But i think this can be fixed ?
I hope so ,because i have a lot of dts es 6.1 "Matrix" Tracks on my HDD ,waitnig for a proper Method to decode all 7 CH for a reencode ( all PAL Audio Tracks/Target is PAL to NTSC-reencode to dts es 6.1 "discrete")
Do you want reencode yours 6.1 "Matrix" to 6.1 "Discrete"?
Do you have a 6.1 audio equipment without support for 6.1 "Matrix" but yes for 6.1 "Discrete"?

Seems ArcSoft can't do the job by the moment.

On the other Side, i dont think the 6.1 Output from matrix 6.1from Arcsoft is sooooo wrong..better then nothing ?!

I can't obtain (crash) a 6.1 output from a 6.1 "Matrix"

The 7.1 output is a waste of space, with SL = BR and SR = BR, I'm sure any 7.1 audio equipment can do the job at play time.

ACrowley

28th October 2008, 12:39

Do you want reencode yours 6.1 "Matrix" to 6.1 "Discrete"?
Do you have a 6.1 audio equipment without support for 6.1 "Matrix" but yes for 6.1 "Discrete"?

Seems ArcSoft can't do the job by the moment.

I can't obtain (crash) a 6.1 output from a 6.1 "Matrix"

The 7.1 output is a waste of space, with SL = BR and SR = BR, I'm sure any 7.1 audio equipment can do the job at play time.

yes, i have this AV Receiver with a 7.1 Speaker Setup
http://www.kenwood.de/products/home/kompo/avcc/KRF-V7300D/kenwood (HDMI 1.3.a/7.1, dts-hd ,trueHD,DD+,dts es Discrete/Matrix, 7.1 PCM etc) .

I have a lot of dts es 6.1 matrix Tracks and i want reencode them to dts es 6.1 discrete.
Because : They are in PAL and i need them in NTSC. So i make a Timestretch and a reencode with dts pro series. Ofcourse i want keep the BC Channel. I get on 5.1 from dts es matrix Tracks with current Methods.

Mh.. no, i get clean output from dts es 6.1 matrix tracks with arcsoft -7. As you can see in my Post.

tebasuna51

28th October 2008, 14:45

I have a lot of dts es 6.1 matrix Tracks and i want reencode them to dts es 6.1 discrete.
Because : They are in PAL and i need them in NTSC. So i make a Timestretch and a reencode with dts pro series. Ofcourse i want keep the BC Channel. I get on 5.1 from dts es matrix Tracks with current Methods.
I have a method to extract the BC channel but is not easy, work fine with test samples (channels separated in time) but I can test real samples. Maybe you can make the test:
- Decode the 6.1 matrix to 6 monowavs (5.1)
- Merge (WaveWizard or Sox) the Back (or Side) channels to stereo.
- Decode this stereo file like DolbyPrologic with Foobar2000 and the Free Surround plugin decoder (Center=1, Dimension=-0.5, A=0, B=0).
- The FL,FR,FC from the resultant 5.1 output are your BL,BR,BC channels.

With the test file:
FL = 0.9448 x BL + 0.0004 x BC
FR = 0.9448 x BR + 0.0004 x BC
FC = 0.0235 x BL + 0.0248 x BR + BC
LF < 0.0000
SL < 0.0000
SR < 0.0000

Mh.. no, i get clean output from dts es 6.1 matrix tracks with arcsoft -7. As you can see in my Post.

I don't know what is my problem, I'm wait to madshi can see anything at bug report.

Thunderbolt8

28th October 2008, 18:46

Got a problem with a H.264 TV cap:
(actually wanted to slow it down, both video and audio, to 23.976 fps, but this doesnt work as well ofc)

eac3to v2.72
command line: eac3to G:\bla.ts G:\bla.mkv
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 0:00:05
1: h264/AVC, 1080i50 (16:9)
2: AC3, English, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB, 57ms
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[v01] Muxing video to Matroska...
[a02] Removing AC3 dialog normalization...
[a02] Applying (E-)AC3 delay...
[a02] This doesn't seem to be a valid PES packet.
Aborted at file position 1277952.

heres a 10mb sample: http://www.sendspace.com/file/yveygw

piratburner

28th October 2008, 19:01

tebasuna51

28th October 2008, 19:10

I got info about remapping channel, is that a problem or have I forgot something ?

Don't worry. Eac3to only inform you (DTS internal channel order is different than LPCM order). All is ok.

piratburner

28th October 2008, 19:19

Don't worry. Eac3to only inform you (DTS internal channel order is different than LPCM order). All is ok.

Thanks :-)

I have a problem with DTS-HD 7.1 with tsMuxer so I will give this a try instead, Love eac3to :)

Ahhh PCM are not allowed in TsMuxer so I can't do a new m2ts file, Or do I need something more with my PCM track in eac3to before I add that to TsMuxer?

EDIT

Found this little tool :-) Pcm2Tsmu :-) so now it's possible to use PCM in TsMuxer :-)

rica

28th October 2008, 20:07

tebasuna,
why not edit your signature and put pcm2tsmu link?

Octo-puss

28th October 2008, 20:17

I got a question. Is there any room for performance improvement? It looks to me like the data extracting is pretty slow (I don't understand how it works though, so it might be normal of course). The other day I took one of my movies and dug the AC3 audio out of it - it wasn't even converting from other format - and it was pretty slow...

piratburner

28th October 2008, 21:05

I trying to Convert .dtshr to PCM . I got info about remapping channel, is that a problem or have I forgot something ? Se my command below.

L:\leon>eac3to "DTS Hi-Res, English, 7.1 channels, 24 bits, 2864kbps, 48khz.dtshr" LEON.PCM
DTS Hi-Res, 7.1 channels, 2:12:50, 24 bits, 2864kbps, 48khz
Decoding with ArcSoft DTS Decoder...
Swapping endian...
Remapping channels...
Creating file "LEON.PCM"...

OK I use Pcm2Tsmu for my pcm track, but adding the file to TsMexer, then is telling me thats is a 5.1 track. And playing the m2ts file is giving me funny nois. Is it a bug in eac3to when converting to PCM

nautilus7

28th October 2008, 21:46

Octo-puss

28th October 2008, 22:01

There's logic behind that, but if the extraction goes at around under 5MB/s, I believe there's a fair bit of reserve :)

nautilus7

28th October 2008, 22:06

Well... What is your CPU? I have a Core 2 Duo 6600.

Thunderbolt8

28th October 2008, 22:12

does anyone know how to create a single channel wav file (only the center channel) of a multi channel flac/ac3/lpcm/dtshd/truehd source track?
when I do "eac3to source.track outcome.wav" then I only get one huge file, which includes all the source channels and not only the center one I want to have.

Octo-puss

28th October 2008, 22:19

Well... What is your CPU? I have a Core 2 Duo 6600.

E6750 (overclocked, but that doesn't matter much).

odin24

28th October 2008, 22:19

MichaelAnders

28th October 2008, 22:23

Using v2.72, it seems as if the channel order is corrupt?

M2TS, 1 video track, 2 audio tracks, 1:21:00
1: VC-1, 1080p24 /1.001 (16:9)
2: DTS Master Audio, German, 5.1 channels, 24 bits, 48khz
3: DTS Master Audio, English, 7.1 channels, 16 bits, 48khz
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "e.flac"...
[a03] The original audio track has a constant bit depth of 16 bits.

If I convert the German audio to FLAC (same as before, this time only the 2nd stream), everything is fine and the file sounds perfect in MPCHC and also in foobar2000:
[a02] Original audio track: max 24 bits, average 16 bits, most common 16 bits.

The English FLAC however, is totally wrong. It seems as if FL and FR are now SL and SR, the subwoofer is also on some other channel...

If I play the English stream in the M2TS file, the "properties" in MPCHC shows "Audio: DTS 48000Hz 6ch [Audio]".

I don't have any older version of eac3to to check this, but something is wrong... Maybe eac3to finds 8 channels, but it's actually 6?

Thunderbolt8

28th October 2008, 22:25

eac3to source.audio output.wavs

Then delete the ones you do not need, keeping the center channel.
d'oh, missed the plural. too easy, thanks! :D

nautilus7

28th October 2008, 22:27

If you only need center, you can use undocumented switch -mono to save some space and hdd activity.

Thunderbolt8

28th October 2008, 23:24

hm the result seems to be slightly different in that case, when I cancel the process after some mb (I only need the beginning of the file) and try to open it with wavepad it tells me that I need the ACM plugin, while the cancelled center channel from '.wavs' works fine :S

nautilus7

28th October 2008, 23:27

It should be the wav header in the begging of each file.

tebasuna51

29th October 2008, 00:28

OK I use Pcm2Tsmu for my pcm track, but adding the file to TsMexer, then is telling me thats is a 5.1 track. And playing the m2ts file is giving me funny nois. Is it a bug in eac3to when converting to PCM

Eac3to don't have a bug please go to Pcm2Tsmu thread (http://forum.doom9.org/showthread.php?p=1207732#post1207732).

Snowknight26

29th October 2008, 01:35

8: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
"Commentary By Director Jonathan Demme And Screenplay Co-Writer Daniel Pyne"
[...]
[a08] Reducing depth from 64 to 24 bits...
When converting to wav.

Thunderbolt8

29th October 2008, 01:46

a technical question: when comparing 2 source.track -> .wav files of the same channel from different source types (e.g. center channel of 5.1 ac3 and flac or left and right channel of 2.0 ac3 with left and right channel of 5.1 flac) and looking for their different delays is it technically correct then to look where the first tiny bit of sound occurs in the channel graph? what I mean is that a flac track of course has a higher quality as an ac3 track and this difference must be located somewhere. so could it be possible that the flac track has numerical more digits in the graphs at the beginning of the track compared to the ac3 track so that this kind of delay meassurement is wrong, or is this difference of sound quality only to be found in the amplitude of the graph, that it only spread in the vertical, but not the horizontal direction of the graph?

svgame

29th October 2008, 06:25

thank you Madshi
just there has a problem.i make 3 step
1) use eac3to demux TrueHD track directly from BD/HD
2) type command "eac3to 00001.thd audio.thd" "eac3to 00001.thd audio.ac3",demuxing to ac3 core and lossless part
3) rename audio.thd to audio.mlp,and import to Scenarist.at 100%,its hung!!!about 2 minute，get a error message "External component has thrown an exception"
where is problem?Madshi

ACrowley

29th October 2008, 09:12

I have a method to extract the BC channel but is not easy, work fine with test samples (channels separated in time) but I can test real samples. Maybe you can make the test:
- Decode the 6.1 matrix to 6 monowavs (5.1)
- Merge (WaveWizard or Sox) the Back (or Side) channels to stereo.
- Decode this stereo file like DolbyPrologic with Foobar2000 and the Free Surround plugin decoder (Center=1, Dimension=-0.5, A=0, B=0).
- The FL,FR,FC from the resultant 5.1 output are your BL,BR,BC channels.

With the test file:
FL = 0.9448 x BL + 0.0004 x BC
FR = 0.9448 x BR + 0.0004 x BC
FC = 0.0235 x BL + 0.0248 x BR + BC
LF < 0.0000
SL < 0.0000
SR < 0.0000

I don't know what is my problem, I'm wait to madshi can see anything at bug report.

so the outout is identical to the source BC Channel ?
Will i twork with DD EX tracks?

By the Way, i compared all Surround/Back Channles from dts es discrete with the test matrix reencode and arcsoft -7.
Theyre all looking fine and identical compared to the original dts es discrete....

However, Madshi will make it ,as always

tebasuna51

29th October 2008, 13:21

so the outout is identical to the source BC Channel ?
The analog extraction isn't perfect, like you see there are BL and BR parts (2%) added to the BC channel.

Will i twork with DD EX tracks?
Seems works with dolby-waterfal_51EX sample but I don't have channel test EX to quantify the separation.

By the Way, i compared all Surround/Back Channles from dts es discrete with the test matrix reencode and arcsoft -7.
Theyre all looking fine and identical compared to the original dts es discrete....

Then if work for you don't need nothing more.

ACrowley

29th October 2008, 16:21

Then if work for you don't need nothing more.

yeah..it should be tested more..im not sure
Strange, i cant reproduce my Test. eac3to/arcsoft -7 crashes on the dts es matrix reencode. But it works with other matrix dts tracks!

Madshi should take a closer look, he is the master:)

rica

29th October 2008, 17:39

Hi guys.

How can i shorten this:

eac3to E:\VIDEO_TS\VTS_01.vob+E:\VIDEO_TS\VTS_01_2.vob 2: C:\KB\video.m2v 6: C:\KB\dtsaudio.dts 5:

This:

eac3to E:\VIDEO_TS\[VTS_01_1+VTS_01_2].vob

or this:

eac3to E:\VIDEO_TS\[VTS_01_1.vob+VTS_01_2.vob]

doesn't work?

_ _ _ _

madshi

29th October 2008, 17:53

I need to remux no. 4: DTS-ES 6.1 track 1536k and 3: DD 5.1EX 640k track.
Here is the log:

but... PDVD reports that I've got DTS-ES 5.1 1536k track and not 6.1...
Previously during playing BD movie using the same PDVD - reported: DTS-ES 6.1 1536k.
Please explain what I did wrong.
Don't trust in what PDVD says. Run "eac3to demuxedtrack.dts". What does eac3to say?

btw, madshi, is there any chance eac3to to report if AC3 track is DD-EX?
Will add that to my to do list.

Not all Ok here.
If we name source channels like Rear: RL, RR, RC and

RL' = +90º phase shift RL
RR' = -90º phase shift RR

seems the phase shift is make at encoder phase and is the same for 'matrix' and 'discrete' DTS-ES in all the tests.
Ouch. Don't you consider that "bad"? I mean just imagine we want to speed correct (e.g. apply PAL speedup) to a DTS track. Doesn't that mean that the phase changes with every new reencode? So if I do "eac3to source.dts dest.dts -speedup" the phase of the surround channels will be different?

1) 3/3.1 -> 5.1 (like previous test)

SL = RL' + 0.71 x RC
SR = RR' + 0.71 x RC
Ok
Ok, that's fine, that's how eac3to is also doing 6.1 -> 5.1 downmixing.

3) 3/3.1 -> 7.1

SL = 0.40 x RL' + 0.60 x RC
SR = 0.40 x RR' + 0.60 x RC
BL = 0.75 x RL'
BR = 0.75 x RR'
Seems wrong: RC only in Side and most Rear part in Back, I think must be the other way, Side <-> Back
Agreed.

2) 3/3.1 -> 7.1

SL = 0.71 x RL' + 0.50 x RC
SR = 0.71 x RR' + 0.50 x RC
BL = 0.71 x RL' + 0.50 x RC
BR = 0.71 x RR' + 0.50 x RC
For me seems ArcSoft can't recover the RC mixed in RL-RR because RC must be present only in Back channels.
Agreed.

3) 3/3.1 -> 6.1, eac3to crash with this bugreport.txt (http://pastebin.com/m276c45a0)

The report was made with 2.70 but is the same with 2.72, I send the 2.70 bug because was made after a reset with the minimum of soft running.
It seems that the ArcSoft decoder crashes internally somewhere. That is probably a bug in the ArcSoft decoder. You have version 1.1.0.0.

@ACrowley, are you using a different ArcSoft decoder version? FWIW, I'm getting the same crash as tebasuna51...

Still the same "issue" with DTS and the -core option.

I personly dislike 7.1 downmixing from DTS-HD 7.1 files, when
a "clean" 5.1 core is present.

But is not possible to reencode directly in 1 step to ac3 from the -core. eac3to always sees the 7.1 MA and try to downmix from the extention. At the moment, first the -core must be exported, than you can ac3-encode from the 5.1 core.
There's a bug in eac3to. If you use the "-core" option, eac3to nevertheless thinks that it will get 7.1 from the decoder. Will have to fix that.

However, I'd strongly recommend that you reconsider your transcoding method. The "clean" 5.1 core will result in worse audio quality than the eac3to produced 7.1 downmix because the 7.1 source has a higher quality in every single channel compared to the core 5.1 track. Even if you're afraid that eac3to's 7.1 -> 5.1 downconversion is worse compared to what the studio did, still the front channels and LFE will have a higher quality if you use the full DTS-HD track. So IMHO using only the core is a bad idea.

Got a problem with a H.264 TV cap:
(actually wanted to slow it down, both video and audio, to 23.976 fps, but this doesnt work as well ofc)

eac3to v2.72
command line: eac3to G:\bla.ts G:\bla.mkv
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 0:00:05
1: h264/AVC, 1080i50 (16:9)
2: AC3, English, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB, 57ms
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[v01] Muxing video to Matroska...
[a02] Removing AC3 dialog normalization...
[a02] Applying (E-)AC3 delay...
[a02] This doesn't seem to be a valid PES packet.
Aborted at file position 1277952.

heres a 10mb sample: http://www.sendspace.com/file/yveygw
Looks like a broken/corrupted file.

does anyone know how to create a single channel wav file (only the center channel) of a multi channel flac/ac3/lpcm/dtshd/truehd source track?
when I do "eac3to source.track outcome.wav" then I only get one huge file, which includes all the source channels and not only the center one I want to have.
This will decode the first 50MB of the source file and store only the center channel into a WAV file:

"eac3to sourcefile test.wav -mono -50mb"

a technical question: when comparing 2 source.track -> .wav files of the same channel from different source types (e.g. center channel of 5.1 ac3 and flac or left and right channel of 2.0 ac3 with left and right channel of 5.1 flac) and looking for their different delays is it technically correct then to look where the first tiny bit of sound occurs in the channel graph?
I'd suggest using Audacity or a similar wave editor and compare the form of the graphs. It's very easy to see how much delay an audio track needs this way.

Using v2.72, it seems as if the channel order is corrupt?

M2TS, 1 video track, 2 audio tracks, 1:21:00
1: VC-1, 1080p24 /1.001 (16:9)
2: DTS Master Audio, German, 5.1 channels, 24 bits, 48khz
3: DTS Master Audio, English, 7.1 channels, 16 bits, 48khz
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "e.flac"...
[a03] The original audio track has a constant bit depth of 16 bits.

If I convert the German audio to FLAC (same as before, this time only the 2nd stream), everything is fine and the file sounds perfect in MPCHC and also in foobar2000:
[a02] Original audio track: max 24 bits, average 16 bits, most common 16 bits.

The English FLAC however, is totally wrong. It seems as if FL and FR are now SL and SR, the subwoofer is also on some other channel...

If I play the English stream in the M2TS file, the "properties" in MPCHC shows "Audio: DTS 48000Hz 6ch [Audio]".

I don't have any older version of eac3to to check this, but something is wrong... Maybe eac3to finds 8 channels, but it's actually 6?
Please try "eac3to English.flac Englisch.wav -50mb". That will decode the first 50MB of the FLAC file to a WAV file. Then load this WAV file in e.g. Audacity and check whether the channel order is correct or not.

Or if you want me to check it, please upload a small sample of the original 7.1 English DTS track, please?

8: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
"Commentary By Director Jonathan Demme And Screenplay Co-Writer Daniel Pyne"
[...]
[a08] Reducing depth from 64 to 24 bits...
When converting to wav.
Are you trying to test whether I can read your mind?

thank you Madshi
just there has a problem.i make 3 step
Please stop asking the same question over and over again. I've already answered it a while ago.

madshi

29th October 2008, 18:00

How can i shorten this
By calling eac3to from inside "E:\VIDEO_TS". Then you don't need to use any paths for the source files.

I got a question. Is there any room for performance improvement? It looks to me like the data extracting is pretty slow (I don't understand how it works though, so it might be normal of course). The other day I took one of my movies and dug the AC3 audio out of it - it wasn't even converting from other format - and it was pretty slow...
When extracing one AC3 track from a (M2)TS file I'm getting about 13-15 MB. That's ok, I think?

rica

29th October 2008, 18:27

By calling eac3to from inside "E:\VIDEO_TS". Then you don't need to use any paths for the source files

Thanks madshi but this time i get this:

C:\>eac3to E:\VIDEO_TS VTS_01_1.vob+VTS_01_2.vob
HD DVD / Blu-Ray disc structure not found.

Snowknight26

29th October 2008, 18:28

Are you trying to test whether I can read your mind?
No, but I didn't think a sample was necessary. I was able to convert it to wav using a previous version (can't remember which), but it doesn't work ever since you added the deeper bitdepth analysis, leading me to believe its a small bug introduced in your code only recently.

When extracing one AC3 track from a (M2)TS file I'm getting about 13-15 MB. That's ok, I think?
I have a server capable of doing 750MB/s average read with a C2Q, but it always takes eac3to about 10 minutes to demux a audio stream from your average length movie. Lets hope there is room for improvement. ;)

madshi

29th October 2008, 18:36

Thanks madshi but this time i get this:

C:\>eac3to E:\VIDEO_TS VTS_01_1.vob+VTS_01_2.vob
HD DVD / Blu-Ray disc structure not found.

I think you need to learn how to use command lines... ;)

You should have "E:\VIDEO_TS\>eac3to VTS_01_1.vob+VTS_01_2.vob".

No, but I didn't think a sample was necessary.
I'm not asking for a sample. You posted a log (which doesn't contain any error messages) without any further comment or question. What am I supposed to do with the log? I don't even know what you want from me.

Snowknight26

29th October 2008, 18:39

I'm not asking for a sample. You posted a log (which doesn't contain any error messages) without any further comment or question. What am I supposed to do with the log? I don't even know what you want from me.

[a08] Reducing depth from 64 to 24 bits...
Since when are AC3 tracks 64 bit?

rica

29th October 2008, 18:43

I think you need to learn how to use command lines... ;)

You should have "E:\VIDEO_TS\>eac3to VTS_01_1.vob+VTS_01_2.vob".

Agree :stupid: :p

madshi

29th October 2008, 18:50

Since when are AC3 tracks 64 bit?
AC3 tracks decode to floating point. The latest eac3to version internally uses 64bit floating point now.

Snowknight26

29th October 2008, 18:53

Ah, my mistake then. Good to hear though.

tebasuna51

29th October 2008, 18:58

RL' = +90º phase shift RL
RR' = -90º phase shift RR

seems the phase shift is make at encoder phase and is the same for 'matrix' and 'discrete' DTS-ES in all the tests.
Ouch. Don't you consider that "bad"? I mean just imagine we want to speed correct (e.g. apply PAL speedup) to a DTS track. Doesn't that mean that the phase changes with every new reencode? So if I do "eac3to source.dts dest.dts -speedup" the phase of the surround channels will be different?

Yes I think is "bad".
I don't know if DTS PRO ENCODER have a switch to apply or not the phase shift to surround channels (like Dolby Encoders have)

MichaelAnders

29th October 2008, 19:41

Or if you want me to check it, please upload a small sample of the original 7.1 English DTS track, please?
I don't have the tool you talked about, so here is the DTS file. I demuxed it via tsmuxergui, just the first 90 seconds. Interestingly, tsmuxergui also says it is 6 channels...

http://www.sendspace.com/file/6a3cl7

Thunderbolt8

29th October 2008, 19:45

Looks like a broken/corrupted file.This is a SkyHD broadcast and I heard from other people that eac3to might have problems with these 'new' ac3 tracks from sky. so is this really only a broken track or are there also other newer sky caps which caused problems?

I'd suggest using Audacity or a similar wave editor and compare the form of the graphs. It's very easy to see how much delay an audio track needs this way.
thats what I was refering to, I was using wavepad. but is the beginning of the ac3 track, for example the same tone or note, exactly the same beginning of a flac track as well? as it was said, flac has a higher quality/bitrate so this 'more' must be manifested somewhere in the graph?

odin24

29th October 2008, 20:32

I don't have the tool you talked about, so here is the DTS file. I demuxed it via tsmuxergui, just the first 90 seconds. Interestingly, tsmuxergui also says it is 6 channels...

http://www.sendspace.com/file/6a3cl7

eac3to reports this file as 8 channels, tsMuxeR on the other had reports as 6. I had this issue once before, after remuxing with tsMuxeR my PS3 & receiver still recognized my movie as 7.1 channels.

Hope this helps.

Octo-puss

29th October 2008, 20:39

When extracing one AC3 track from a (M2)TS file I'm getting about 13-15 MB. That's ok, I think?

Definitely. I just got the bluray directory structure ripped on HDD and work with that. Took few mins to extract simple .ac3 file :-O

MichaelAnders

29th October 2008, 21:02

eac3to reports this file as 8 channels, tsMuxeR on the other had reports as 6. I had this issue once before, after remuxing with tsMuxeR my PS3 & receiver still recognized my movie as 7.1 channels.

Hope this helps.
Thanks, but I actually wanted to FLAC the stream so that I can play it on my HTPC via HDMI to my 7.1 receiver as we can't transmit DTS HD or TrueHD directly today from a HTPC (and I am just looking for direct output from graphic card here, not some Asus Xonar stuff which can maybe also do that, not interested). And FLAC files are also aways quite a bit smaller than the original DTS or TrueHD ones ;)

I hope madshi can find the bug in eac3to, then I am more than happy :D This is also the only file which has failed so far btw...

nautilus7

29th October 2008, 21:45

I don't know if DTS PRO ENCODER have a switch to apply or not the phase shift to surround channels (like Dolby Encoders have)It has. ;)

odin24

29th October 2008, 21:56

Thanks, but I actually wanted to FLAC the stream so that I can play it on my HTPC via HDMI to my 7.1 receiver as we can't transmit DTS HD or TrueHD directly today from a HTPC (and I am just looking for direct output from graphic card here, not some Asus Xonar stuff which can maybe also do that, not interested). And FLAC files are also aways quite a bit smaller than the original DTS or TrueHD ones ;)

I hope madshi can find the bug in eac3to, then I am more than happy :D This is also the only file which has failed so far btw...

:o Oops, I guess I should have read earlier in the thread instead of assuming... didn't realize you need FLAC. Sorry then, I cannot help. Good luck.

tebasuna51

30th October 2008, 01:59

Originally Posted by tebasuna51 View Post
I don't know if DTS PRO ENCODER have a switch to apply or not the phase shift to surround channels (like Dolby Encoders have)
It has. ;)

Then, please, you can help me making a new set of dts test files using this sources (http://www.sendspace.com/file/haqwf6)?

svgame

30th October 2008, 02:53

Please stop asking the same question over and over again. I've already answered it a while ago.

i am so sorry.but there still has a problem.please help me.:helpful::thanks::thanks::thanks::thanks::thanks:

nautilus7

30th October 2008, 08:13

Then, please, you can help me making a new set of dts test files using this sources (http://www.sendspace.com/file/hjm2g3)?

I will, but i can't before this Weekend.

ACrowley

30th October 2008, 09:07

Yes I think is "bad".
I don't know if DTS PRO ENCODER have a switch to apply or not the phase shift to surround channels (like Dolby Encoders have)

Mh....as i say Arcsoft -7 on dts es matrix tracks works not with all tracks ive tested so far. eac3to/arcsoft crashes sometimes with -7. (works fine in 5.1) It crashes on all dts es matrix Tracks with phase shift Option enabled ( from dts pro series encoder)

Maybe its the phase shift in dts es matrix track and arcsoft cant handle 6.1 output in this case? I will test it when im at home. That would explain why some dts es matrix tracks can be decoded with arcsoft -7 and some other dts not (with a crash)

Anacletus

30th October 2008, 11:01

Hi all, maybe a dumb question.. i have a BluRay with a Dolby trueHD track i want to extract.. i've read a lot here but still im confused on what to do..

Should i extract the THD to flac and then convert the flac to ac3 640? or should i extract the thd+ac3 and just use the ac3?

Any suggestions will be more than appreciated :)

Anac

BlackJack1

30th October 2008, 11:58

[QUOTE=madshi;1207948]Don't trust in what PDVD says. Run "eac3to demuxedtrack.dts". What does eac3to say?/QUOTE]

It says: DTS-ES 6.1 1536k... so that's OK. PDVD must be wrong... :)
Thanks.

B4tm4n

30th October 2008, 17:03

i am so sorry.but there still has a problem.please help me.:helpful::thanks::thanks::thanks::thanks::thanks:

Read the last part of Madshi's post.
http://forum.doom9.org/showthread.php?p=1207099#post1207099

bmnot

30th October 2008, 19:27

[a02] This track is not clean. Processing aborted.
[a02] Please clean the track with delaycut and then retry eac3to.
Aborted at file position 16853820240.

madshi, can you add a switch to ignore or skip over these types of errors. The only time I ever get them is with Blu-rays, and that message doesn't occur until the VERY end of the file. The error reported above is actually the very last byte of the entire file (total size: 16,853,820,240 bytes)

If the error could be skipped over I doubt it would affect the playback sync.

73ChargerFan

30th October 2008, 21:18

IIRC, the answer given for this question was "don't worry", but worry I will...

When I demuxed a BD, the PCM "remapped" channels.
When I convert the PCM, it says "oh, this from a bd" and again it "remapped" channels.

So, under which cup is the walnut? I thought the first remap was from bd format to a standard windows format, and moved channels around. So, afterwards, they shouldn't have to be moved around again. I think the channels are now in the wrong location. ???

nwg

30th October 2008, 21:51

bmnot

30th October 2008, 23:14

[a04] The ArcSoft decoder doesn't seem to work, will use Sonic instead.
[a04] The Sonic Audio Decoder doesn't decode the XXCh channel extension..

It's a 7.1 DTS-HD HR track. I don't want to install Arcsoft so I'll stick with Surcode. The resulting flac is 5.1, what happened to the 2 extra rears, were they dropped or mixed into the surrounds?

svgame

31st October 2008, 07:52

Read the last part of Madshi's post.
http://forum.doom9.org/showthread.php?p=1207099#post1207099

i do that with Madshi's post,but there still has a problem

tebasuna51

31st October 2008, 13:45

Then, please, you can help me making a new set of dts test files using this sources (http://www.sendspace.com/file/haqwf6)?

Thanks nautilus7. With the new samples from DTS PRO ENCODER y can confirm:

- When make 'Ls Cs Rs 5.1 ES' (ES 3/3.1 matrix) or 'Ls Cs Rs 6.1 ES' (ES 3/3.1 discrete) the option 'phase shift' is available and default to 'yes' (with a standard 'Ls Rs' 5.1 we can't select 'phase shift')

- If 'phase shift' is selected the input channels Ls Rs are modified to:
Ls' = +45º phase shift Ls
Rs' = -45º phase shift Rs

- The mix with Cs is (or LS',Rs'):
Ls'' = Ls + 0.707 x Cs
Rs'' = Rs + 0.707 x Cs

- I always have a crash when decode a ES matrix with -7 (sorry ACrowley I can't confirm your post, the phase shift don't modify this behaviour for me)

frenchglen

1st November 2008, 21:43

madshi,

Just wanted to drop by and to acknowledge once more the amazing work you're doing with eac3to [and also to others who are also contributing!]. The ever-growing feature list is amazing and I regularly come back to the program for all sorts of things.

Thanks a million,
frenchglen

Snowknight26

1st November 2008, 23:10

Would it be possible to add a switch that detects playlists that are shorter in length than the current cutoff?

ACrowley

2nd November 2008, 09:48

Thanks nautilus7. With the new samples from DTS PRO ENCODER y can confirm:

- When make 'Ls Cs Rs 5.1 ES' (ES 3/3.1 matrix) or 'Ls Cs Rs 6.1 ES' (ES 3/3.1 discrete) the option 'phase shift' is available and default to 'yes' (with a standard 'Ls Rs' 5.1 we can't select 'phase shift')

- If 'phase shift' is selected the input channels Ls Rs are modified to:
Ls' = +45º phase shift Ls
Rs' = -45º phase shift Rs

- The mix with Cs is (or LS',Rs'):
Ls'' = Ls + 0.707 x Cs
Rs'' = Rs + 0.707 x Cs

- I always have a crash when decode a ES matrix with -7 (sorry ACrowley I can't confirm your post, the phase shift don't modify this behaviour for me)

yeah..as i say -7 works only on some dts es matrix Tracks. But no with encodes from dts pro series.

Generally, would you enalbe phase shift when reencoding to dts es matrix/discrete 6.1 ? Or can it be a Problem when you reencode those dts encodes with phase shift?

tebasuna51

2nd November 2008, 10:51

Generally, would you enalbe phase shift when reencoding to dts es matrix/discrete 6.1 ? Or can it be a Problem when you reencode those dts encodes with phase shift?

I think the phase shift is intended to difference the Ls Rs channel to make easy recover the Cs, like the common part in the Ls'' Rs''.

Then if Ls and Rs are similar is better apply the phase shift but if Ls Rs was generated previously with some phase shift maybe a new phase shift can be worse.

In a conversion DTS-ES -> wav -> some changes -> DTS-ES isn't recommended apply phase shift (must be already phase shifted, was extracted to Ls Rs because are different).

frenchglen

2nd November 2008, 20:29

I'm trying to extract AC3 from a TrueHD/AC3 track in a blu-ray m2ts. Can someone please tell me what I'm doing wrong?

Here's what I do and what it says:

W:\STREAM>eac3to 00010.m2ts trackno:1 track1.ac3
M2TS, 1 video track, 3 audio tracks, 8 subtitle tracks, 2:06::02
1: Chapters, 15 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
4: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: Subtitle (PGS), English
7: Subtitle (PGS), English
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), Portuguese
11: Subtitle (PGS), French
12: Subtitle (PGS), Spanish
13: Subtitle (PGS), English
Track 3 is used for destination file "1".
Track 3 is used for destination file "track1.ac3"
This audio conversion is not supported.

W:\STREAM>

Thanks for your help

nautilus7

2nd November 2008, 20:42

eac3to 00010.m2ts 3: track1.ac3

madshi

2nd November 2008, 21:07

I don't have the tool you talked about, so here is the DTS file. I demuxed it via tsmuxergui, just the first 90 seconds. Interestingly, tsmuxergui also says it is 6 channels...
I don't think that tsmuxer can reliably detect the number of channels in a DTS-HD file. AFAIK eac3to is the only freeware tool that can do that.

The DTS sample you uploaded is kind of strange. I don't have any problems with the channel order. However, the front left and front right channels seem to be plain broken if I decode the track to 7.1 with ArcSoft. I'm not sure if the track itself is broken or whether this is caused by a bug in ArcSoft. You can use the "-6" parameter to force ArcSoft into decoding the track as 5.1, only. I think this is the best you can do right now.

This is a SkyHD broadcast and I heard from other people that eac3to might have problems with these 'new' ac3 tracks from sky. so is this really only a broken track or are there also other newer sky caps which caused problems?
I've had a look at that sample you uploaded (thanks). As far as I can say the sample is not correctly authored. But since there's nothing we can do about that, I changed the way eac3to demuxes TS and m2ts files now. For non-broken files there should (hopefully) be no difference. But the sample you uploaded works fine now with the latest eac3to build.

Please note that xport fails to demux this sample, just as the old eac3to version does. That's because both xport and eac3to relied on the transport stream to be conformant to specs. TsRemuxer and tsMuxeR both demuxed it fine. It seems that they're using the same demuxing logic which the latest eac3to version now also uses. I guess that it's a bit more error tolerant.

thats what I was refering to, I was using wavepad. but is the beginning of the ac3 track, for example the same tone or note, exactly the same beginning of a flac track as well? as it was said, flac has a higher quality/bitrate so this 'more' must be manifested somewhere in the graph?
The difference between FLAC and (a good quality) AC3 track is very small in a wave editor. Volume differences or DRC show much more obvious differences in a wave editor. So it's really easy to compare audio tracks to find the right delay. It doesn't matter which format they're stored in. Of course in order to compare them you have to decode them to WAV first...

i am so sorry.but there still has a problem.
If there's still a problem then it's not a problem in eac3to, but a problem with Scenarist or with the way you're trying to use eac3to and/or Scenarist. Since I never used Scenarist before, I can't help you with that. If you still need help, please open a new thread, because this thread is not supposed to be a "how to use Scenarist" thread.

Hi all, maybe a dumb question.. i have a BluRay with a Dolby trueHD track i want to extract.. i've read a lot here but still im confused on what to do..

Should i extract the THD to flac and then convert the flac to ac3 640? or should i extract the thd+ac3 and just use the ac3?
What you should do depends on what you want to achieve. If you want a FLAC track then ask eac3to to create a FLAC track for you. If you want an AC3 track then ask eac3to to create an AC3 track for you. It's as simple as that...

[a02] This track is not clean. Processing aborted.
[a02] Please clean the track with delaycut and then retry eac3to.
Aborted at file position 16853820240.

madshi, can you add a switch to ignore or skip over these types of errors. The only time I ever get them is with Blu-rays, and that message doesn't occur until the VERY end of the file. The error reported above is actually the very last byte of the entire file (total size: 16,853,820,240 bytes)
Can you please upload the last 20-50MB of the problematic m2ts file for me to check?

IIRC, the answer given for this question was "don't worry", but worry I will...
That's your problem, not mine... ;)

[a04] The ArcSoft decoder doesn't seem to work, will use Sonic instead.
[a04] The Sonic Audio Decoder doesn't decode the XXCh channel extension..

It's a 7.1 DTS-HD HR track. I don't want to install Arcsoft so I'll stick with Surcode. The resulting flac is 5.1, what happened to the 2 extra rears, were they dropped or mixed into the surrounds?
Not sure. Sometimes Sonic mixes the surrounds, sometimes not.

Thanks nautilus7. With the new samples from DTS PRO ENCODER y can confirm:

- When make 'Ls Cs Rs 5.1 ES' (ES 3/3.1 matrix) or 'Ls Cs Rs 6.1 ES' (ES 3/3.1 discrete) the option 'phase shift' is available and default to 'yes' (with a standard 'Ls Rs' 5.1 we can't select 'phase shift')

- If 'phase shift' is selected the input channels Ls Rs are modified to:
Ls' = +45º phase shift Ls
Rs' = -45º phase shift Rs

- The mix with Cs is (or LS',Rs'):
Ls'' = Ls + 0.707 x Cs
Rs'' = Rs + 0.707 x Cs
Ouch. Never seen a 45° phase shift before. Are there other situations apart from this one (6.1 -> 5.1 matrix) where such odd phase shifts are used?

Would it be possible to add a switch that detects playlists that are shorter in length than the current cutoff?
Would you want such a switch to turn of ANY runtime checks? Or do you just want to lower the min runtime a little? I'm asking because there are Blu-Rays with hundreds of playlists!

madshi

2nd November 2008, 21:12

eac3to v2.73 released

http://madshi.net/eac3to.zip

* changed TS demuxing logic to make the broken (!) new SkyHD broadcasts work
* DTS core and "TrueHD/AC3" AC3 parameters are displayed separately now
* when using "-core" option, eac3to now bases its decisions on core parameters
* added WAV/W64/RF64 read/write support for 32bit PCM and 32/64 bit float
* option "-full" allows WAV/W64/RF64 output to be native (default <= 24bit PCM)
* Surcode DTS encoding is now done with up to 32bit float (if available)
* Aften AC3 encoding is now done with up to 64bit float (if available)
Here are some encoder bitdepth tidbits:

(1) Surcode accepts 16bit PCM, 24bit PCM, 32bit PCM and also 32bit float WAV files. But the DTS Pro encoder only accepts 16bit PCM and 24bit PCM files!
(2) The Nero AAC encoder is currently also limited to 24bit PCM input.
(3) When using libav decoders and the Aften or Surcode encoder, eac3to has a full floating point chain now from decoder to encoder, including all processing steps in between.

Question: Does anybody have any Dolby Digital "EX" samples for me?

Snowknight26

2nd November 2008, 21:12

Would you want such a switch to turn of ANY runtime checks? Or do you just want to lower the min runtime a little? I'm asking because there are Blu-Rays with hundreds of playlists!

Yes, the former, with the reason being that one of my Blu-rays has hundreds of playlists that I need to go through and check. ^^

Thunderbolt8

2nd November 2008, 21:22

eac3to v2.73 released

http://madshi.net/eac3to.zip

* changed TS demuxing logic to make the broken (!) new SkyHD broadcasts work*:thanks:

nautilus7

2nd November 2008, 21:31

eac3to v2.73 releasedThanks!

Question: Does anybody have any Dolby Digital "EX" samples for me?I may have some from the StarWars dvds. I will upload later tonight or tomorrow.

EDIT: StarWars IV AC3 (pal) (http://www.sendspace.com/file/bxg9c6)

hd1080

3rd November 2008, 01:04

i get an error with v.2.73

eac3to v2.73
command line: eac3to "L:\PURPUR_1_BD\BDMV\STREAM\00000.m2ts" 1:"j:\fluss_chapters.txt" 3:"j:\fluss_german.dts" -core 4:"j:\fluss_french.dts" -core
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 2 subtitle tracks, 1:45:48
1: Chapters, 23 chapters
2: VC-1, 1080p24 (16:9)
3: DTS Hi-Res, German, 6.1 channels, 16 bits, 2556kbps, 48khz
(core: DTS-ES, 6.1 channels, 16 bits, 1536kbps, 48khz)
4: DTS Hi-Res, French, 6.1 channels, 16 bits, 2556kbps, 48khz
(core: DTS-ES, 6.1 channels, 16 bits, 1536kbps, 48khz)
5: DTS, French, 2.0 channels, 16 bits, 1536kbps, 48khz
6: Subtitle (PGS), German
7: Subtitle (PGS), German
Creating file "j:\fluss_chapters.txt"...
[a03] Extracting audio track number 3...
[a04] Extracting audio track number 4...
[a04] Extracting DTS core...
[a03] Extracting DTS core...
[a04] Patching bitdepth to 24 bits...
[a03] Patching bitdepth to 24 bits...
[a03] Creating file "j:\fluss_german.dts"...
[a04] Creating file "j:\fluss_french.dts"...
[a03] This track is not clean. Processing aborted.
[a03] Please clean the track with delaycut and then retry eac3to.
Aborted at file position 32825124864.

but with v2.66 it works fine

eac3to v2.66
command line: eac3to "L:\PURPUR_1_BD\BDMV\STREAM\00000.m2ts" 3:"j:\fluss_german.dts" -core 4:"j:\fluss_french.dts" -core
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 2 subtitle tracks, 1:45:48
1: Chapters, 23 chapters
2: VC-1, 1080p24 (16:9)
3: DTS Hi-Res, German, 6.1 channels, 16 bits, 2556kbps, 48khz
4: DTS Hi-Res, French, 6.1 channels, 16 bits, 2556kbps, 48khz
5: DTS, French, 2.0 channels, 16 bits, 1536kbps, 48khz
6: Subtitle (PGS), German
7: Subtitle (PGS), German
[a03] Extracting audio track number 3...
[a04] Extracting audio track number 4...
[a03] Extracting DTS core...
[a04] Extracting DTS core...
[a03] Patching bitdepth to 24 bits...
[a04] Patching bitdepth to 24 bits...
[a03] Creating file "j:\fluss_german.dts"...
[a04] Creating file "j:\fluss_french.dts"...
Video track 2 contains 152350 frames.
eac3to processing took 14 minutes, 51 seconds.
Done.

tebasuna51

3rd November 2008, 02:44

Ouch. Never seen a 45° phase shift before. Are there other situations apart from this one (6.1 -> 5.1 matrix) where such odd phase shifts are used?

Well I only can demostrate the 45º phase shift at 500 Hz (see the pure tone in samples), but is a frequency enough representative.

And don't know for what, or if is used in other cicumstances.

Thanks for the new relase.

* option "-full" allows WAV/W64/RF64 output to be native (default <= 24bit PCM)

sorry, I don't understand the meaning.

madshi

3rd November 2008, 08:38

i get an error with v.2.73

but with v2.66 it works fine
Ouch. Is that the very end of the movie? Please check the file size of "00000.m2ts". Is that near to the reported 32825124864 file position?

* option "-full" allows WAV/W64/RF64 output to be native (default <= 24bit PCM)

sorry, I don't understand the meaning.
If you ask eac3to to output any audio source as WAV/W64/RF64, eac3to by default makes sure that the audio data is not any higher than 24bit PCM. If the source audio data is 32bit PCM, it gets dithered down to 24bit. If the audio data is floating point, it gets rounded to 32bit PCM and then dithered down to 24bit.

The "-full" switch tells eac3to that you don't want the bitdepth to be converted when creating WAV/W64/RF64 files. That means if the source is 32bit PCM (and you do no processing on the audio data) the destination WAV/W64/RF64 will be created as 32bit PCM. Or if the source is floating point (or if you did any processing which results in the source being converted to floating point) eac3to will create 64bit floating point WAV/W64/RF64 files.

The "-full" switch simply disables the downconvertion to max 24bit PCM. The switch only affects the creation of WAV/W64/RF64 files.

J0nThn

3rd November 2008, 10:40

I've got the same problem as reported by hd1080.

wine ~/tools/eac3to273/eac3to.exe ../TOTAL_RECALL "1)" 7:Total.Recall.Blu-Ray.23.976fps.DTS.FRENCH_2.dts 7:Total_Recall.DD5.1.French_2.ac3 13:French_2.sup

M2TS, 1 video track, 5 audio tracks, 11 subtitle tracks, 1:53:34
1: Chapters, 16 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 16 bits, 48khz
(core: DTS, 5.1 channels, 16 bits, 1536kbps, 48khz)
4: DTS Hi-Res, Portuguese, 5.1 channels, 16 bits, 2083kbps, 48khz
(core: DTS, 5.1 channels, 16 bits, 1536kbps, 48khz)
5: DTS Hi-Res, German, 5.1 channels, 16 bits, 2083kbps, 48khz
(core: DTS, 5.1 channels, 16 bits, 1536kbps, 48khz)
6: DTS, Spanish, 2.0 channels, 16 bits, 1536kbps, 48khz
7: DTS Hi-Res, French, 5.1 channels, 16 bits, 2083kbps, 48khz
(core: DTS, 5.1 channels, 16 bits, 1536kbps, 48khz)
8: Subtitle (PGS), English
9: Subtitle (PGS), Portuguese
10: Subtitle (PGS), Danish
11: Subtitle (PGS), German
12: Subtitle (PGS), Spanish
13: Subtitle (PGS), French
14: Subtitle (PGS), Spanish
15: Subtitle (PGS), Dutch
16: Subtitle (PGS), Norwegian
17: Subtitle (PGS), Finnish
18: Subtitle (PGS), Swedish

[a07] The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
[a07] The libav DTS decoder doesn't decode the full DTS-HD information.
[a07] Extracting audio track number 7...
[a07] Extracting audio track number 7...
[s13] Extracting subtitle track number 13...
[a07] Extracting DTS core...
[a07] Patching bitdepth to 24 bits...
[a07] Decoding with libav/ffmpeg...
[a07] Remapping channels...
[a07] Encoding AC3 <640kbps> with libAften...
[a07] Patching bitdepth to 24 bits...
[a07] Creating file "Total.Recall.Blu-Ray.23.976fps.DTS.FRENCH_2.dts"...
[a07] Creating file "Total_Recall.DD5.1.French_2.ac3"...
[s13] Creating file "French_2.sup"...
[a07] This track is not clean. Processing aborted.
[a07] Please clean the track with delaycut and then retry eac3to.
Aborted at file position 27470180352.

And the aborted position correspond to the end of the m2ts.

27470180352 1 nov 19:58 ../TOTAL_RECALL/BDMV/STREAM/00156.m2ts

madshi

3rd November 2008, 10:47

I've got the same problem as reported by hd1080.
From which country is that Total Recall Blu-Ray? Could you upload the last 20-50MB of 00156.m2ts for me?

donis

3rd November 2008, 14:21

Question: Does anybody have any Dolby Digital "EX" samples for me?

War.2007.DD-EX.ac3 (http://www.sendspace.com/file/jon6d3)

Saw.IV.2007.DD-EX.ac3 (http://www.sendspace.com/file/q6itmy)

Oldboy.2003.DD-EX.ac3 (http://www.sendspace.com/file/wthd6u)

All 3 are DD-EX @ 640 Kbps. :)

qyqgpower

3rd November 2008, 15:48

Could eac3to split audio tracks with chapter info or at specified time point?
I use DGAVCIndex to spilt H264 video at IDR frame and demux audio tracks, it's fine with AC3 and LPCM.
But for TrueHD, only the first demuxed one can be recognized by eac3to.

J0nThn

3rd November 2008, 16:20

From which country is that Total Recall Blu-Ray? Could you upload the last 20-50MB of 00156.m2ts for me?

It's European version.

Here is the sample you requested : End_00156.m2ts (35Mb) (http://rapidshare.com/files/160279440/end_156.m2ts.html)

hd1080

3rd November 2008, 16:26

Ouch. Is that the very end of the movie? Please check the file size of "00000.m2ts". Is that near to the reported 32825124864 file position?
yep the filesize is 32.825.124.864 Byte

madshi

3rd November 2008, 18:08

War.2007.DD-EX.ac3 (http://www.sendspace.com/file/jon6d3)

Saw.IV.2007.DD-EX.ac3 (http://www.sendspace.com/file/q6itmy)

Oldboy.2003.DD-EX.ac3 (http://www.sendspace.com/file/wthd6u)

All 3 are DD-EX @ 640 Kbps. :)
Thanks! That should help me adding DD-EX detection sooner or later...

Could eac3to split audio tracks with chapter info or at specified time point?
I use DGAVCIndex to spilt H264 video at IDR frame and demux audio tracks, it's fine with AC3 and LPCM.
But for TrueHD, only the first demuxed one can be recognized by eac3to.
I'm not sure what you mean with the last sentence. Anyway, sure I could in theory add splitting functionality to eac3to, but to be honest: I don't think there's much demand for that. I have so many things on my to do list where demand is a lot bigger...

It's European version.

Here is the sample you requested : End_00156.m2ts (35Mb) (http://rapidshare.com/files/160279440/end_156.m2ts.html)
Thanks. Will be fixed in the next build.

yep the filesize is 32.825.124.864 Byte
Should also be fixed in the next build.

madshi

3rd November 2008, 18:08

nwg

3rd November 2008, 18:13

Thanks for the new version.

odin24

3rd November 2008, 18:18

Would converting a TrueHD (48/16) track from an HDDVD to PCM (for BD playback) sound the same as the same movie on Blu-ray with with the audio already as PCM (48/16)?

All of this done with eac3to of course.

tebasuna51

3rd November 2008, 18:53

Seems the link to eac3to v2.74 is still the v2.73

J0nThn

3rd November 2008, 19:47

eac3to v2.74 released

http://madshi.net/eac3to.zip

* "-demux" failed to work for DTS-HD and "TrueHD/AC3" tracks in v2.73
* fixed: DTS-HD tracks could make processing abort at the very end of the movie

Thanks :)
Just to be sure that's the expected behavior (last DTS frame incomplete) ?

[a07] Remapping channels...
[a07] Encoding AC3 <640kbps> with libAften...
[a07] Creating file "Total.Recall.Blu-Ray.23.976fps.DTS.FRENCH_2.dts"...
[a07] Creating file "Total_Recall.DD5.1.French_2.ac3"...
[s13] Creating file "French_2.sup"...
[a07] The last DTS frame is incomplete and thus gets skipped.
[a07] The last DTS frame is incomplete and thus gets skipped.
Video track 2 contains 163366 frames.
eac3to processing took 26 minutes, 37 seconds.

Thunderbolt8

3rd November 2008, 20:18

eac3to v2.74 released

http://madshi.net/eac3to.zip

"-demux" failed to work for DTS-HD and "TrueHD/AC3" tracks in v2.73
did this generally affect demuxing those tracks, or only when explicetely using the '-demux' command (as indicated "")? because afaik I've demuxed each of those 2 tracks today together with flac conversion and video remux and I didnt encouter such a message.

another thing, this is old already, but after not thinking about it for some time I got confused again -.-
when having a movie of which eac3to reports +57ms delay for the AC3 track, what eac3to then does is adding 2x -32ms = -64ms delay to it, so that the least possible delay remaining is then -7ms, right? so in case I want to have it perfectly I'd have to add +7ms at playback then in mpc (I know this is imperceptible, but just for perferction)?

tebasuna51

3rd November 2008, 21:20

Thanks! That should help me adding DD-EX detection sooner or later....

To detect DD-EX the header must have:

Bsid = 6 (Alternate syntax)
Xbsi2e = 1 (Exist Extra BSI #2)
Dsurexmod = 2 (Dolby Surround EX encoded)

Remember the Table D2.1 (pg. 132) from a52b pdf to locate the fields in header, or see this seudo-code (http://pastebin.com/m48284188).

lithiumus

3rd November 2008, 21:21

However, I'd strongly recommend that you reconsider your transcoding method. The "clean" 5.1 core will result in worse audio quality than the eac3to produced 7.1 downmix because the 7.1 source has a higher quality in every single channel compared to the core 5.1 track. Even if you're afraid that eac3to's 7.1 -> 5.1 downconversion is worse compared to what the studio did, still the front channels and LFE will have a higher quality if you use the full DTS-HD track. So IMHO using only the core is a bad idea.

Hey madshi,

Does this logic apply also to say 5.1 DTS-HD/TrueHD > 5.1 DTS 1536 or 5.1 AC3 640? i.e. instead of using the available 5.1 DTS 1536 or 5.1 AC3 640 track, to transcode a new track from the DTS-HD or TrueHD tracks?

Or is that only relevant when talking about a core+data situation instead if there was separate tracks available i.e. DTS-HD (core+data) and a separate DTS 1536 track or TrueHD (core+data) and a separate AC3 640 track.

Keep up the amazing work!

madshi

3rd November 2008, 22:35

Would converting a TrueHD (48/16) track from an HDDVD to PCM (for BD playback) sound the same as the same movie on Blu-ray with with the audio already as PCM (48/16)?
If the studio used the same master for PCM and TrueHD (which is not guaranteed, but probable) then yes.

Seems the link to eac3to v2.74 is still the v2.73
Not for me. If you can't seem to be able to get v2.74, please try emptying your browser cache.

Just to be sure that's the expected behavior (last DTS frame incomplete) ?
Yes, that's ok.

did this generally affect demuxing those tracks
Yes.

when having a movie of which eac3to reports +57ms delay for the AC3 track, what eac3to then does is adding 2x -32ms = -64ms delay to it, so that the least possible delay remaining is then -7ms, right? so in case I want to have it perfectly I'd have to add +7ms at playback then in mpc?
When eac3to reports +57ms delay then a +57ms delay is needed and not a -57ms delay. So eac3to applies +64ms delay. So you will have to use -7ms delay in MPC. However, you know my point of view about this topic, right? ;)

To detect DD-EX the header must have:

Bsid = 6 (Alternate syntax)
Xbsi2e = 1 (Exist Extra BSI #2)
Dsurexmod = 2 (Dolby Surround EX encoded)

Remember the Table D2.1 (pg. 132) from a52b pdf to locate the fields in header, or see this seudo-code (http://pastebin.com/m48284188).
Will keep that in mind, thanks...

Does this logic apply also to say 5.1 DTS-HD/TrueHD > 5.1 DTS 1536 or 5.1 AC3 640? i.e. instead of using the available 5.1 DTS 1536 or 5.1 AC3 640 track, to transcode a new track from the DTS-HD or TrueHD tracks?

Or is that only relevant when talking about a core+data situation instead if there was separate tracks available i.e. DTS-HD (core+data) and a separate DTS 1536 track or TrueHD (core+data) and a separate AC3 640 track.
Generally it is BAD to do a transcode from a lossy codec to another lossy codec, because that means you'll have 2 losses which add up on each other. If you want to transcode audio you should always use the highest quality source you can get, which is (of course) lossless, if available. That's why I recommended to use DTS-HD Master Audio instead of only the core - for transcoding purposes.

The situation is different, if there already exists a core in the very format and bitdepth you're looking for. In that case transcoding is not necessary at all, but you can simply extract and use the existing core. But as soon as you need/want to transcode (which means: full decoding + reencoding) you should always choose the source with the highest possible audio quality.

odin24

3rd November 2008, 22:46

If the studio used the same master for PCM and TrueHD (which is not guaranteed, but probable) then yes.

Thanks for the quick response.

It just seemed kind of strange that watching the bitrate (HDDVD THD remuxed to BD --> on the PS3) that it only peaked at around 1.7mb/s, even during high action scenes. Where on most BDs with THD they peak around 4.5mb/s. I know it's like "comparing apples to oranges", but it's not even close.

Thunderbolt8

3rd November 2008, 23:27

madshi, I have a problem with slowing down that skyHD movie I sent you the 10mb sample to fix the skyHD ac3 tracks. its 50i and when I use -slowdown then the video turns out to be about 2-3 mins longer than it should be (audio seems to be fine though). It has video gaps, but the problem stays the same after redoing the same command line. 50 mb sample:

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

bmnot

3rd November 2008, 23:32

Can you please upload the last 20-50MB of the problematic m2ts file for me to check? get it here: http://www.sendspace.com/file/i8ak0o

Snowknight26

4th November 2008, 03:58

If (E-)AC3 tracks are decoded to 64-bit floating point, why is that when converting to WAV they're then reduced to 24-bit? Wouldn't it be best to reduce the depth to, say, 32-bit floating point?

madshi

4th November 2008, 09:40

It just seemed kind of strange that watching the bitrate (HDDVD THD remuxed to BD --> on the PS3) that it only peaked at around 1.7mb/s, even during high action scenes. Where on most BDs with THD they peak around 4.5mb/s.
If the TrueHD track is only 16bit then it's *MUCH* smaller (and has much lower peaks) compared to a track with full 24bit.

If (E-)AC3 tracks are decoded to 64-bit floating point, why is that when converting to WAV they're then reduced to 24-bit? Wouldn't it be best to reduce the depth to, say, 32-bit floating point?
First of all it depends on the decoder. I patched libav so that it gives me the full floating point decode. But Nero and Sonic decoders internally downconvert to 24bit PCM. Now with libav eac3to by default downconverts to 24bit PCM if you ask it to create a WAV file. The reason for that is that 24bit PCM is usually plenty good enough. The main purpose of keeping things in floating point is if you want to do further processing on the data or if you want to feed the data to an encoder which wants floating point data. If that is the case, simply use the "-full" switch. Then eac3to will create floating point WAV files (if floating point data is available).

get it here: http://www.sendspace.com/file/i8ak0o
This one demuxes fine for me with v2.74. Can you please retest with v2.74?

madshi, I have a problem with slowing down that skyHD movie I sent you the 10mb sample to fix the skyHD ac3 tracks. its 50i and when I use -slowdown then the video turns out to be about 2-3 mins longer than it should be (audio seems to be fine though). It has video gaps, but the problem stays the same after redoing the same command line. 50 mb sample:

http://www.sendspace.com/file/ypgilt
Thanks. Looks like a bug in eac3to. Will try to fix that in the next build. Might take a few days, though...

MichaelAnders

4th November 2008, 19:53

I don't think that tsmuxer can reliably detect the number of channels in a DTS-HD file. AFAIK eac3to is the only freeware tool that can do that.

The DTS sample you uploaded is kind of strange. I don't have any problems with the channel order. However, the front left and front right channels seem to be plain broken if I decode the track to 7.1 with ArcSoft. I'm not sure if the track itself is broken or whether this is caused by a bug in ArcSoft. You can use the "-6" parameter to force ArcSoft into decoding the track as 5.1, only. I think this is the best you can do right now.

So you mean to say that the DTS sample IS 7.1, eac3to finds this properly and then, using 6 channels, I would of course loose the data for the other two channels :( Not your fault of course, just curious... And apart from ArcSoft there seems to be nothing that can decode 7.1 channels right? Libav/ffmpeg can just do 5.1 like Nero 7 as well :(

madshi

4th November 2008, 21:22

So you mean to say that the DTS sample IS 7.1, eac3to finds this properly and then, using 6 channels, I would of course loose the data for the other two channels :( Not your fault of course, just curious... And apart from ArcSoft there seems to be nothing that can decode 7.1 channels right? Libav/ffmpeg can just do 5.1 like Nero 7 as well :(
Correct.

Well, if you absolutely must have all 8 channels then you can do "eac3to source.dts 6.wavs -6" and also "eac3to source.dts 8.wavs". That way you will get 6 WAV files for the 6 channel decoding and 8 WAV files for the 8 channel decoding. You can then use the 8 WAV files - but replace the front left and front right channels with those of the 6 channel decode. The only problem then is to convert the 8 channels back into one WAV file. eac3to cannot do that currently. But I think wavewizard can.

Thunderbolt8

4th November 2008, 22:01

got a problem with the 'how the west was won' (smilebox) blu-ray:

eac3to v2.74
command line: G:\eac3to\eac3to D:\source 1) 2: G:\west.mkv 4: G:\west.flac 4: G:\west.thd
------------------------------------------------------------------------------
M2TS, 1 video track, 7 audio tracks, 16 subtitle tracks, 2:44:41
1: Chapters, 42 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
4: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
5: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: AC3, German, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
7: AC3, Italian, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
8: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
9: AC3, Spanish, 1.0 channels, 192kbps, 48khz, dialnorm: -27dB
10: Subtitle (PGS), Japanese
11: Subtitle (PGS), English
12: Subtitle (PGS), French
13: Subtitle (PGS), German
14: Subtitle (PGS), Italian
15: Subtitle (PGS), Italian
16: Subtitle (PGS), Spanish
17: Subtitle (PGS), Dutch
18: Subtitle (PGS), Chinese
19: Subtitle (PGS), Spanish
20: Subtitle (PGS), Portuguese
21: Subtitle (PGS), Danish
22: Subtitle (PGS), Finnish
23: Subtitle (PGS), Norwegian
24: Subtitle (PGS), Portuguese
25: Subtitle (PGS), Swedish
[a04] Extracting audio track number 4...
[a04] Extracting audio track number 4...
[v02] Extracting video track number 2...
[a04] Extracting TrueHD stream...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Encoding FLAC with libFlac...
[v02] Muxing video to Matroska...
[a04] Creating file "G:\west.thd"...
[a04] Creating file "G:\west.24bit.flac"...
[libav] Lossless check failed - expected 0, calculated 80
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] The zero bytes were successfully removed.
Added fps value to MKV header.
Video track 2 contains 236898 frames.
eac3to processing took 43 minutes, 41 seconds.
Done.

So I guess there a chance now that my flac track is not 100% lossless in this case? But I'd still be on the save side when I just store the demuxed TrueHD track and then redo the flac again, when this problem is fixed?

madshi

4th November 2008, 22:10

So I guess there a chance now that my flac track is not 100% lossless in this case?
If that warning appeared exactly once then there's a chance that 0.8333ms of your track are not lossless. The other 5-10 million milliseconds of your track are still lossless.

Thunderbolt8

4th November 2008, 22:44

allright, I stick with those other million ms then

flyingernst

5th November 2008, 02:32

eac3to v2.74
command line: "F:\Download\Brennen\eac3to\eac3to.exe" "P:" 1) "E:\Hancock.m2ts"
------------------------------------------------------------------------------
M2TS, 2 video tracks, 6 audio tracks, 19 subtitle tracks, 1:42:14
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: TrueHD/AC3, English, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 448kbps, 48khz)
5: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -30dB
6: TrueHD/AC3, German, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB)
7: AC3, Russian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
8: AC3, Ukrainian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
9: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), German
13: Subtitle (PGS), Dutch
14: Subtitle (PGS), Danish
15: Subtitle (PGS), Estonian
16: Subtitle (PGS), Finnish
17: Subtitle (PGS), Modern Greek
18: Subtitle (PGS), Hindi
19: Subtitle (PGS), Lithuanian
20: Subtitle (PGS), Norwegian
21: Subtitle (PGS), Swedish
22: Subtitle (PGS), Turkish
23: Subtitle (PGS), English
24: Subtitle (PGS), German
25: Subtitle (PGS), Dutch
26: Subtitle (PGS), Russian
27: Subtitle (PGS), Ukrainian
28: Subtitle (PGS), English
Track 4 is used for destination file "Hancock.m2ts".
This audio conversion is not supported.

is there any way else to join the big Hancock Playlist?
TSmuxer destroys some informations while joining and remuxing. I Want to play it in a PCH B-110. When I remux it with TXRemux again it is okay but buggy. The problem is the joing of the Playlistfiles. 1:1 copied full Mainmovie m2ts work fine after remuxing with TSRemux, so the joining and TSmuxer is the problem.

Can eac3to do that?! Joining multiple m2ts files (like it does with evos)?

Thanks, greetings Michael

Thunderbolt8

5th November 2008, 10:49

yes, but output bill be .mkv of course, not .evo or .m2ts

lithiumus

5th November 2008, 19:01

Hey Madshi, not sure if this was suggested I tried searching...

Any thought around an option to downmix/convert DTS-HD HR/MA 7.1 or TrueHD 7.1 down to DTS-ES 6.1 or Dolby EX 6.1? I guess this would also require that we use encoders that are capable i.e. DTS Pro Encoder or I think Ulead's Videostudio can do Dolby EX or Vegas but I'm not 100% sure...

itsancho

5th November 2008, 19:11

Hi again to all ;-)
I just can't miss to say that again: Madshi, Thank You, very much for the great tool! (and in advance for DD-EX) :-)
well, and here it is some strange (at least for me) problem with avc stream
eac3to 2.74
1) 00000.mpls, 00000.m2ts, 1:25:59
- h264/AVC, 1080i60 /1.001 (16:9)
- AC3, Japanese, multi-channel, 48khz
- AC3, Japanese, stereo, 48khz
- AC3, English, stereo, 48khz

M2TS, 1 video track, 3 audio tracks, 1 subtitle track, 1:25:59
1: Chapters, 13 chapters
2: h264/AVC, 1080i60 /1.001 (16:9)
3: AC3, Japanese, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
4: AC3, Japanese, 2.0 channels, 384kbps, 48khz, dialnorm: -27dB
5: AC3, English, 2.0 channels, 384kbps, 48khz, dialnorm: -27dB, 166ms
6: Subtitle (PGS), English
Creating file "f:\Chapters.txt"...
[v02] Extracting video track number 2...
[s06] Extracting subtitle track number 6...
[a03] Extracting audio track number 3...
[v02] Muxing video to Matroska...
[a03] Removing AC3 dialog normalization...
[a03] Creating file "l:\Sister.ac3"...
[s06] Creating file "l:\EN.sup"...
Added fps value to MKV header.
Video track 2 contains 123695 frames.
eac3to processing took 6 minutes, 14 seconds.
Done. so far, so good, but the *.mkv file is about 2 minutes shorter, than the *ac3 file.

then i've open *.m2ts file with tsmuxer, the info there for video was Profile: High@4.1 Resolution: 1920:1080i Frame rate: 29.97 (pulldown) i've checked "Remove pulldown", changed fps to 24000/1001 and demux stream, then mux it with mkvmerge to *.mkv (24000/1001 FPS) and...
surpise, surprise everything was OK now, audio and video were in sync!

so... i'm just curious (and obviously ignorant) ;-)

Thunderbolt8

5th November 2008, 19:50

seems to be a bug, I have a similar problem with slowing down a h264 50i file.

menlvd

5th November 2008, 22:05

i'll have another problem with slowdown audio:
when processing progress is stun
cmd - "D:\minep\me_gui\tools\eac3_to\eac3to.exe" "I:\temp work\au\get smart T80 3_2ch 384Kbps DELAY -228ms.ac3" "I:\GSma_rus.wavs" -quality=ultra -slowdown
http://pic.ipicture.ru/uploads/081106/4347/thumbs/7Ly6JV3ySo.jpg (http://ipicture.ru/Gallery/Viewfull/8809138.html)
taskmngr
http://pic.ipicture.ru/uploads/081106/4347/thumbs/iZ8Uo4Qt5u.jpg (http://ipicture.ru/Gallery/Viewfull/8809144.html)
when simple decodinng file to *.* everything processed fine
it's only appear in slowdown and maybe -speedup and etc.
tested with latest bild 2.73 and 2.74
old ver don't have this problem :)
plz anybody UL old version 2.72 or 2.71

UPD!!!
replace -quality=ultra by -quality=high and processed fine! seems to be a bug in ultra code

edo1080

6th November 2008, 00:41

will nero9 work for directshow? Or has it the same limitation as nero8?

rica

6th November 2008, 00:49

will nero9 work for directshow? Or has it the same limitation as nero8?

It's gonna work in case you register its filters by manually.

Duppie

6th November 2008, 06:26

Madshi

I am trying to understand the following but something seems not right.

eac3to v2.74
command line: eac3to.exe THX.m2ts -check
----------------------------------------------
M2TS, 1 video track, 1 audio track, 0:00:45
1: h264/AVC, 1080p24 /1.001 (16:9)
2: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
Video track 1 contains 1079 frames.
eac3to processing took 7 seconds.
Done.

When extracting the 5.1 ac3 file I get:

eac3to v2.74
command line: eac3to.exe THX.m2ts Test5.1.ac3
------------------------------------------------
M2TS, 1 video track, 1 audio track, 0:00:45
1: h264/AVC, 1080p24 /1.001 (16:9)
2: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
Track 2 is used for destination file "Test5.1.ac3".
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Creating file "Test5.1.ac3"...
Video track 1 contains 1079 frames.
eac3to processing took 5 seconds.
Done.

Which is confirmed by:

eac3to v2.74
command line: eac3to.exe Test5.1.ac3
-----------------------------------------------
AC3, 5.1 channels, 0:00:45, 640kbps, 48khz

When converting the 5.1 ac3 file to 2.1 I get:

eac3to v2.74
command line: eac3to.exe THX.m2ts Test2.1.ac3 -down2 -mixlfe
-------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 0:00:45
1: h264/AVC, 1080p24 /1.001 (16:9)
2: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
Track 2 is used for destination file "Test2.1.ac3".
[a02] The Nero decoder doesn't seem to work, will use libav instead.
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Decoding with libav/ffmpeg...
[a02] Remapping channels...
[a02] Downmixing multi channel audio to stereo...
[a02] Encoding AC3 <640kbps> with libAften...
[a02] Creating file "Test2.1.ac3"...
Video track 1 contains 1079 frames.
eac3to processing took 9 seconds.
Done.

But eac3to indicates this file as only 2.0 and not 2.1.
Is this correct or am I missing something?

eac3to v2.74
command line: eac3to.exe Test2.1.ac3
----------------------------------------------------------
AC3, 2.0 channels, 0:00:45, 640kbps, 48khz

Also why is the size of the file for 2.0 and 5.1 the same?

My last question, the bitdepth of ac3 files is always 16bit
and will only be displayed by eac3to if they are more than 16bit?

tebasuna51

6th November 2008, 10:56

But eac3to indicates this file as only 2.0 and not 2.1.
Is this correct or am I missing something?
Is correct, the mix of LFE channel is inside the front channels
BTW, Dolby don't recommend mix the LFE channel inside a DPL mix.

Also why is the size of the file for 2.0 and 5.1 the same?
The size is determined by the bitrate, if you don't say nothing the default 640 Kb/s is applied.
For 2 channels is recommended 192 Kb/s or 224 Kb/s

My last question, the bitdepth of ac3 files is always 16bit and will only be displayed by eac3to if they are more than 16bit?
The precision measure in compressed audio is the bitrate and the bitdepth don't have sense.

If you define the bitdepth like the number of bits of an audio sample at the samplerate frecuency then an ac3 6 chan. 48 KHz. and 640 Kb/s have an average bitdepth of:
640000 / (6 x 48000) = 2.222

In DTS headers there are a field to remember the bitdepth of the source uncompressed audio (but isn't reliable), in ac3 header don't exist.

Both, AC3 and DTS, store samples in fequency domain with a precission equivalent to 20-24 uncompressed bitdepth, then when you decompress you must select at least 24 bit ignoring the source bitdepth.

Duppie

6th November 2008, 11:00

Thanks tebasuna51 :)

nautilus7

6th November 2008, 11:13

madshi any chance you update eac3to so it supports latest aften (http://code.google.com/p/wavtoac3encoder) release? The revision used currently should be about 100 revisions old. :(

:thanks:

73ChargerFan

6th November 2008, 18:24

I have a question. What is the effect/intended use for the option -seekToIFrames make all h264/AVC "I" frames seekable?
I use eac3to to demux my blu-ray rips, then I play them in mpc-hc. Should I be using that option?

nautilus7

6th November 2008, 18:30

You need it only if eac3to (haali matsoska muxer) is unable to handle the source. eac3to will prompt you to use it in that case.

MichaelAnders

6th November 2008, 21:24

madshi, quick question - indirectly related to eac3to :)

I had already told you that some of my audio streams are corrupt, and that eac3to breaks with an error.

Now, that is fine, and I can live with it :) I can demux these streams to the end with TSMuxerGUI, and then merge them into the MKV.

However! The "dialnorm" is still in the resulting stream, and I'd really love to get rid of it, e.g. in E-AC3 streams :) As I can see from the output, you don't recode this stream when I demux to ".eac3". Dialnorm is however removed. So my guess is that this is just some byte which is set somewhere in the stream... Could you give me a hint on where to look for it? Maybe I can just use a hex editor and remove it :D Thank you :)

madshi

6th November 2008, 21:31

Any thought around an option to downmix/convert DTS-HD HR/MA 7.1 or TrueHD 7.1 down to DTS-ES 6.1 or Dolby EX 6.1? I guess this would also require that we use encoders that are capable i.e. DTS Pro Encoder or I think Ulead's Videostudio can do Dolby EX or Vegas but I'm not 100% sure...
Do you know the exact downmixing matrixes needed for downconverting 7.1 to DTS-ES compatible 6.1/5.1 or Dolby EX compatible 5.1?

well, and here it is some strange (at least for me) problem with avc stream
so far, so good, but the *.mkv file is about 2 minutes shorter, than the *ac3 file.

then i've open *.m2ts file with tsmuxer, the info there for video was i've checked "Remove pulldown", changed fps to 24000/1001 and demux stream, then mux it with mkvmerge to *.mkv (24000/1001 FPS) and...
surpise, surprise everything was OK now, audio and video were in sync!
Which movie is that? Can I have a sample?

i'll have another problem with slowdown audio
You're using the "-quality=ultra" option which means that you're using the r8brain resampler. I'm not sure but I'm suspecting that r8brain eats too much memory and maybe that's why processing stalls. Try without the "-quality=ultra" option. Does it work then?

"ultra" is probably a bad name. I should rename the option to "-r8brain" because I'm not actually sure that r8brain has better quality than SSRC. It's quite possible that the opposite is true.

will nero9 work for directshow? Or has it the same limitation as nero8?
It works in graphedit, when you manually register the DirectShow filters, but eac3to currently doesn't support it.

madshi any chance you update eac3to so it supports latest aften (http://code.google.com/p/wavtoac3encoder) release? The revision used currently should be about 100 revisions old. :(
I'm using "some" 0.08 revision. Maybe there have been some new builds, but I can't see any changes in the changelog. So I'm not sure whether it's worth it updating to a newer build.

madshi

6th November 2008, 21:34

nautilus7

6th November 2008, 21:41

I'm using "some" 0.08 revision. Maybe there have been some new builds, but I can't see any changes in the changelog. So I'm not sure whether it's worth it updating to a newer build.I'll ask Justin if it worth updating then.

lithiumus

6th November 2008, 23:14

Do you know the exact downmixing matrixes needed for downconverting 7.1 to DTS-ES compatible 6.1/5.1 or Dolby EX compatible 5.1?

I wish I did so I can help you out. From a High level (let's exclude Dobly EX for now) DTS-ES can be encoded in either 1536 or 768kbs which gives the same flexibility as normal DTS does.

The rear channel or what is call Center Surround (Cs) requires a Phase shift when it is a discrete channel but does not when it's a matrixed created Phantom channel. The only encoder that I am familiar with that can do DTS-ES is the PRO Encoder directly from DTS or the Master Audio Encoder again direct from DTS.

I suspect that a simple Stereo to Mono downmix of the Left and Right Rear Surround channels is sufficient and then using that as the input to the Center Surround with the Phase Shift option enabled in the DTS Encoder will do the trick for 7.1 and just a direct input for the single rear for 6.1.

Dolby EX is a whole other story and I'm not sure logically how it's matrixed or how best to downmix to DDEX. DTS seems fairly straight forward. The PRO encoder / DTS Suite is only $250 but the Master Audio Suite is $1,500! :eek:

itsancho

7th November 2008, 00:40

Which movie is that? Can I have a sample? :-( Unfortunately I've delete the source! The movie was Onna hissatsu ken (1974) (Sister Street Fighter) and Onna hissatsu ken kiki ippatsu (1974) (Sister Street Fighter 2 Hanging by a Thread) http://www.blu-ray.com/movies/movies.php?id=637 Stupid me!

btw, if that does matter, after i've changed FPS and removed pulldown with tsmuxer, eac3to said: h264/AVC 1080i/47

fuzz!

7th November 2008, 01:28

feature request - chapter extraction from VOB.

is this easy enough?

idbirch2

7th November 2008, 13:34

I don't think VOB files ever contain chapter info. Certainly in a DVD structure, that information is in the .IFO file corresponding to the TitleSet. IFOEdit will extract chapter info from an IFO file for you.

monotube

7th November 2008, 15:18

First of all, many thanks to Madshi for his wonderful program. Now two ( probably stupid ) questions : when a m2ts file has more THD tracks what I have to digit to select only the second THD track?
Again; I have used various releases of eac3to but when I have a m2ts file like input it's never specified the language of the various audio tracks. I have tried also with -check but without success. Any help?

htpc@rischer.at

7th November 2008, 15:56

Hi!

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

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

nautilus7

7th November 2008, 18:33

command line: eac3to h: bruce.logwell, the problem is pretty obvious. Your command line is wrong and surely this is not the complete command line you use.

htpc@rischer.at

7th November 2008, 19:59

well, the problem is pretty obvious. Your command line is wrong and surely this is not the complete command line you use.

Mmhh...this was only to show the content of my bluray. I converted more than 30 bluRays and HDDVDS with eac3to - without problems.

The command I use to convert:

1)
eac3to h: 1) 3:bruce.mkv 6: bruce.ac3

2)
eac3to h: 1) 3:bruce.mkv 6: bruce.ac3 -libav

3)
eac3to h: 1) 3:bruce.mkv 6: bruce.wav

and so on.

bmnot

7th November 2008, 21:02

This one demuxes fine for me with v2.74. Can you please retest with v2.74?

The sample I made doesn't work for me in 2.74, says "The format of the source file could not be detected." but it's definitely the end of the file.

nautilus7

7th November 2008, 22:06

Mmhh...this was only to show the content of my bluray. I converted more than 30 bluRays and HDDVDS with eac3to - without problems.

The command I use to convert:

1)
eac3to h: 1) 3:bruce.mkv 6: bruce.ac3

2)
eac3to h: 1) 3:bruce.mkv 6: bruce.ac3 -libav

3)
eac3to h: 1) 3:bruce.mkv 6: bruce.wav

and so on.
Do you have nero? Does it work? What is the exact error message you get? Do you have anydvd running in the background in case the disc is protected?

htpc@rischer.at

8th November 2008, 09:12

Do you have nero? Does it work? What is the exact error message you get? Do you have anydvd running in the background in case the disc is protected?

Nero: NO

AnyDVD HD: YES

ErrorMessage:
[libav] Spectral extension is not implemented. If you want to help, update your FFmpeg version to the newest one from SVN. If the problem still occurs, it means that your file has extension which has not been tested due to a lack of samples exhibiting this feature. Upload a sample of the audio from this file to ftp://upload.mplayerhq.hu/incoming and contact the ffmpeg-devel mailing list.
[libav] error parsing the audio block

Thanks for help!
htpc
:thanks:

nautilus7

8th November 2008, 10:38

htpc@rischer.at

8th November 2008, 13:18

Ok, this pretty much explains what's wrong, huh? Libav doesn't support the track because it uses some special features of the e-ac3 standard. Nero will work though.

Anyway, for the good of our community i suggest you run the following command

eac3to h: 1) 3:bruce.mkv 6: bruce.eac3 -10mb

to create a 10MB sample of the track and upload it here, so it can be forwarded to the libav developer. :)

btw, which movie is it?

Thanks for your help.

The Blu-Ray Movie is "Enter the Dragon" in German: "Der Mann mit der Todeskralle".

I will create a sample - but how can I upload it!

btw:
I have a second problem with the Blu-Ray of "Kingdom of Heaven".

The Command is:
eac3to I: 1) 4: kh.dts

4: DTS Master Audio , German 5.1 channels 24 bits, 48 khz, (core DTS, 5.1 channels, 24 bits, 1536 kbps, 48 khz)

After a few minutes the message appears:
[a04] This track is not clean. Processing aborted.
[a04] Please clean the track with delaycut and then retry eac3to.

What does this mean?

:thanks:

nautilus7

8th November 2008, 13:24

I will create a sample - but how can I upload it!You can upload it in a file host like sendspace (http://www.sendspace.com).

[a04] This track is not clean. Processing aborted.
[a04] Please clean the track with delaycut and then retry eac3to.

What does this mean?This means that the track has some kind of corruption in it. Only possible way to fix this is to re rip the disc.
Do you need the master audio track or only the core? If the second, you can try the -core switch to see if it works.

kurt

8th November 2008, 14:02

htpc@rischer.at

8th November 2008, 14:47

You can upload it in a file host like sendspace (http://www.sendspace.com).

This means that the track has some kind of corruption in it. Only possible way to fix this is to re rip the disc.
Do you need the master audio track or only the core? If the second, you can try the -core switch to see if it works.

Mmhh...this was only to show the content of my bluray. I converted more than 30 bluRays and HDDVDS with eac3to - without problems.

The command I use to convert:

1)
eac3to h: 1) 3:bruce.mkv 6: bruce.ac3

2)
eac3to h: 1) 3:bruce.mkv 6: bruce.ac3 -libav

3)
eac3to h: 1) 3:bruce.mkv 6: bruce.wav

and so on.

Nero: NO

AnyDVD HD: YES

ErrorMessage:
[libav] Spectral extension is not implemented. If you want to help, update your FFmpeg version to the newest one from SVN. If the problem still occurs, it means that your file has extension which has not been tested due to a lack of samples exhibiting this feature. Upload a sample of the audio from this file to ftp://upload.mplayerhq.hu/incoming and contact the ffmpeg-devel mailing list.
[libav] error parsing the audio block

Here is the Sample-Download:

http://rapidshare.com/files/161812574/bruce.eac3

:thanks:

htpc@rischer.at

8th November 2008, 15:43

This means that the track has some kind of corruption in it. Only possible way to fix this is to re rip the disc.
Do you need the master audio track or only the core? If the second, you can try the -core switch to see if it works.

Damn. Obviously the corruption is on the origin blu-ray. -CORE won't work - same result.
What do you mean with re-rip? I have the Blu-Ray already on my hdd as ISO.

nautilus7

8th November 2008, 16:01

Here is the Sample-Download:

[url]http://rapidshare.com/files/161812574/bruce.eac3[/url
The sample you made is too small (only 96KB). Maybe it's not big enough (eac3to can detect its format though).
Do you have a slow connection and it's difficult to upload a bigger sample? If it's not a problem please make a 5 -10 MB sample.

Thanks.

Damn. Obviously the corruption is on the origin blu-ray. -CORE won't work - same result.
What do you mean with re-rip? I have the Blu-Ray already on my hdd as ISO.How can be sure that the corruption is in the disc aswell? Maybe something went wrong the first time you ripped the disc. Rip the disc (either as iso or as folder) and try again.

htpc@rischer.at

8th November 2008, 17:05

The sample you made is too small (only 96KB). Maybe it's not big enough (eac3to can detect its format though).
Do you have a slow connection and it's difficult to upload a bigger sample? If it's not a problem please make a 5 -10 MB sample.

Thanks.

How can be sure that the corruption is in the disc aswell? Maybe something went wrong the first time you ripped the disc. Rip the disc (either as iso or as folder) and try again.

Problem 1 (bruce-lee 1.0 ac3)
Sorry, obviously something went wrong. Here the 10mb-file:
http://rapidshare.com/files/161846249/bruce10MB.eac3

Problem 2 (kingdom of heaven)
I bought the bluray yesterday and after 2 times getting an error from eac3to by accessing to the BDROM direcly i ripped the whole blu ray to my hdd. Unfortunately, the error appears again.
So i gather, there is an error on the bluray inthe hd-dts track :devil:
But.....: playing the Blu-Ray directly from BDDVD or from the ISO with PowerDVD 8 everything wents fine! So, maybe the error is in eac3to???

Thanks!

malibu08

8th November 2008, 17:57

Hi!

I have only installed nero 9
How can i use eac3to and nero filters
How can i install the filters

Thank you !!

ACrowley

8th November 2008, 18:02

Hi!

I have only installed nero 9
How can i use eac3to and nero filters
How can i install the filters

Thank you !!

maybe you should read the 1st Site? before you ask :rolleyes:

Only Nero 7 Filters are working with eac3to!

rica

8th November 2008, 18:07

Hi!
I have only installed nero 9
How can i use eac3to and nero filters
How can i install the filters? !

You have to register Nero filters manually.
You can use Radlight Filter Manager for this purpose.
And if you want eac3to to use Nero filters, you have to put this option at the end of the line: -nero

(BTW, if you don't re-encode but just simply extract the audio (without changing it), decoders will not be used.)

madshi

8th November 2008, 18:35

Now two ( probably stupid ) questions : when a m2ts file has more THD tracks what I have to digit to select only the second THD track?
You should first ask eac3to to post a track listing (e.g. "eac3to movieFolder 1)") and then simply fetch the track index from that list. E.g. if the list shows "2: TrueHD, ..." and "3: TrueHD, ...", you can use "eac3to movieFolder 1) 3: whatever" to select the 2nd TrueHD track.

Again; I have used various releases of eac3to but when I have a m2ts file like input it's never specified the language of the various audio tracks.
The language is shown only if the full Blu-Ray folder and file structure is still intact. The language information is stored in the CLIPINF folder.

The sample I made doesn't work for me in 2.74, says "The format of the source file could not be detected." but it's definitely the end of the file.
It doesn't work because it doesn't begin correctly. If it's an M2TS file, the 5th byte must be a 0x47. If it's a TS file, the first byte must be a 0x47. So you can use a hexeditor, search for the first 0x47 byte and remove the first few bytes of the file, until the fifth (or first) byte is 0x47. If that doesn't work, just repeat the process another 2-3 times. At some time eac3to will properly detect and handle the file.

Problem 1 (bruce-lee 1.0 ac3)
Sorry, obviously something went wrong. Here the 10mb-file:
http://rapidshare.com/files/161846249/bruce10MB.eac3
Thanks. I'll forward this to the libav E-AC3 decoder maintainer.

Problem 2 (kingdom of heaven)
I bought the bluray yesterday and after 2 times getting an error from eac3to by accessing to the BDROM direcly i ripped the whole blu ray to my hdd. Unfortunately, the error appears again.
Is this with eac3to v2.74? Which country is this Blu-Ray from?

Created m2ts by TSMuxer (includig thd+ac3 demuxed by eac3to) can no be remuxed as BD by TSRemux
I wonder why people continue to ask questions about tsMuxeR/TsRemux problems in this thread? This is the eac3to thread! If tsMuxeR or TsRemux don't work correctly then you should ask in the tsMuxeR/TsRemux threads for help.

rica

8th November 2008, 18:44

I wonder why people continue to ask questions about tsMuxeR/TsRemux problems in this thread? This is the eac3to thread! If tsMuxeR or TsRemux don't work correctly then you should ask in the tsMuxeR/TsRemux threads for help.

I put that deleted question here since i haven't been sure whether it is related with eac3to or TSRemux; are you ???

htpc@rischer.at

8th November 2008, 18:46

...

Thanks. I'll forward this to the libav E-AC3 decoder maintainer.

Is this with eac3to v2.74? Which country is this Blu-Ray from?

Hi, Madhi!

NO, the version is 2.73. Now, i tried it with 2.74 and it works. Damn, only one time I missed to install the newest version and.....

malibu08

8th November 2008, 18:47

You have to register Nero filters manually.
You can use Radlight Filter Manager for this purpose.
And if you want eac3to to use Nero filters, you have to put this option at the end of the line: -nero

(BTW, if you don't re-encode but just simply extract the audio (without changing it), decoders will not be used.)

Thank you, but sorry how can i do that with the manager

rica

8th November 2008, 19:07

malibu08, check yor PM box.

Sorry; out of the thread...

madshi

8th November 2008, 23:10

I put that deleted question here since i haven't been sure whether it is related with eac3to or TSRemux; are you ???
You reported that tsMuxeR or TsRemux showed an error message. What does that have to do with eac3to? If tsMuxeR or TsRemux produce an error message then it's very probable that the problem is in tsMuxeR or TsRemux and not in eac3to. So tsMuxeR/TsRemux error messages usually don't belong into this thread.

Of course there is a (slight) chance that eac3to produces a buggy file and that tsMuxeR/TsRemux fail because of that. But as far as I know this has never happened yet. So unless there's a clear indication that eac3to is at fault, I will consider any tsMuxeR/TsRemux error message as being the fault of tsMuxeR/TsRemux and not the fault of eac3to.

rica

8th November 2008, 23:48

Because of my respect to you, i'd rather shut my mouth up.

madshi

9th November 2008, 00:51

frogman

9th November 2008, 04:06

I hope you don't misunderstand me. I'm very much willing to fix any bugs in eac3to. I'm just a bit tired of having to respond to tsMuxeR and TsRemux bug reports all the time. Why do I have to suffer from bugs in other developers' software?

As we all know the real fix to this is to clone Madshi brain, Then pimp him out to other developers. Perhaps then we all will be happy.

PS. First Dibs

Hey, Maddhi ...My sister is available next Friday night!

hehehe!

btw: Thanks for your great tool ea3to.:thanks:

malibu08

9th November 2008, 09:37

Hi!

I have manually register the nero 9 filter "neaudio2.ax" with the radlight filter manager, but it does not work

C:\Tonspuren\eac3>eac3to -test
Nero Audio Decoder (Nero 7 or older) doesn't seem to be installed
http://www.nero.com/eng/store-blu-ray.html
CAUTION: You need Nero 7. Nero 8 won't work with eac3to.
ArcSoft DTS Decoder doesn't seem to be installed
http://www.arcsoft.com/products/totalmediatheatre
Sonic Audio Decoder (2.74.0.0) doesn't seem to be installed
Haali Matroska Muxer (2008-03-29) is installed
Nero AAC Encoder could not be located
http://www.nero.com/eng/nero-aac-codec.html
Copy NeroAacEnc.exe to the eac3to or to the Windows folder.
Surcode DTS Encoder (1.0.29.0) is installed
MkvToolnix doesn't seem to be installed
http://www.bunkus.org/videotools/mkvtoolnix

C:\Tonspuren\eac3>eac3to c:\tonspuren\test.ac3 c:\tonspuren\test_neu1.dts -nero
AC3, 5.1 channels, 1:52:21, 448kbps, 48khz, dialnorm: -27dB
I'd like to use the Nero decoder, but it doesn't seem to work.

Special Thank to Rica for try to help over PN !!!
Thank you !!!

nautilus7

9th November 2008, 10:29

You need nero 7 as clearly stated in the 1st post.

lexor

9th November 2008, 15:54

You need nero 7 as clearly stated in the 1st post.
Actually Nero9 should work (the protection Nero8 had was removed), and madshi did ask for some sort of output from graphedit to help him get it to work, but the last person who had Nero9 didn't reply back with the info.

So if madshi can re-state what it is he needs, maybe malibu will be able to provide what is needed, and we'll all have Nero9 support.

rica

9th November 2008, 16:47

Lexor i agree.
Nero7 is just a default decoder and if you can use Nero9's filters in direct show, eac3to should use it when we force nero9 decoder to be used by -nero option.
I've already asked to malibu if he has bd-blu ray plugin but haven't gotten the answer.

As far as i guess malibu doesn't have this plug-in and in this case he can see and register nero audio decoder2 filter but he can not use in graphstudio which means eac3to can not use it as well.

nautilus7

9th November 2008, 16:52

Nero 9 filters can be used outside nero showtime. This has nothing to do with eac3to though. eac3to is programmed to use nero 7 filters.

Do you think that if nero 9 could be used *right now* with eac3to, the 1st post would say otherwise? And instead of speculating, why don't you try it first?

rica

9th November 2008, 17:00

nautilus, i suppose you are confusing programmed filter (to be used as default) with any other filter forced to be used.
Malibu says he hasn't nero7; in this case eac3to would want to use programmed secondary free filter coming with eac3to.
But if you force eac3to that you will use that filter , why do you think eac3to won't use it?

Edit: why i don't try? Because i have to buy nero9 and the plug-in.

qyqgpower

9th November 2008, 17:11

Can eac3to accept multiple mono wavs input? eg. L.wav, R.wav, C.wav, LFE.wav, SL.wav and SR.wav
I tried using "wavs", "*.wav" and "+" as input parameter, but all fails.
Or is there any method to join them to a multichannel wav again?

nautilus7

9th November 2008, 17:34

why do you think eac3to won't use it?Why do you think it will?

Can eac3to accept multiple mono wavs input? eg. L.wav, R.wav, C.wav, LFE.wav, SL.wav and SR.wav
I tried using "wavs", "*.wav" and "+" as input parameter, but all fails.
Or is there any method to join them to a multichannel wav again?
No, it doesn't support multiple mono wav input.

tebasuna51

9th November 2008, 17:45

Or is there any method to join them to a multichannel wav again?

You can use WaveWizard or Sox command line:

Sox -M FL.wav FR.wav FC.wav LF.wav SL.wav SR.wav multichannel.wav

Boulder

9th November 2008, 17:52

@madshi: Will the next build include feeding 32bit floating point data into the Nero AAC encoder?

madshi

9th November 2008, 17:57

@rica, nautilus7 knows eac3to very well. Normally you can trust in what he says. So instead of trying to argue with him, you should just believe him. He told you 3 times that eac3to doesn't support Nero 9 and he is absolutely right with that.

Will the next build include feeding 32bit floating point data into the Nero AAC encoder?
Yes.

madshi

9th November 2008, 20:32

eac3to v2.75 released

http://madshi.net/eac3to.zip

* added (E-)AC3 5.1 "EX" detection
* added (E-)AC3 2.0 "Surround" detection
* added (E-)AC3 2.0 "Headphone" detection
* NeroAacEnc is now fed with up to 32bit float (if available)
* resampling option "-quality=low|high|ultra" not supported, anymore
* new option "-fast" switches SSRC resampler to fast, but low quality mode
* new option "-r8brain" forces use of r8brain resampler instead of SSRC
* added support for AES3 PCM streams in TS container
* started working on encoder plugin interface
I was wrong stating that the NeroAAC encoder wouldn't support floating point input. It does.

rica

9th November 2008, 20:36

Thunderbolt8

9th November 2008, 20:46

nautilus7

9th November 2008, 21:01

eac3to v2.75 released

* started working on encoder plugin interfaceAny detail on this? Or it's a surprise? :p

Thanks for the update.

Chumbo

9th November 2008, 22:01

@madshi,
Just fyi, the version shown by 2.75 is v0.00.

@rica,
What part of "DOESN'T SUPPORT NERO 9" do you not understand? Give it a rest with Nero 9 already.

madshi

9th November 2008, 22:03

OK madshi; as you know as well, i've trusted always his knowledge on eac3to but i still have not understood why eac3to can not use nero9 filter as forced decoder?

Second, is is possible to manage eac3to to select nero9 first and if there isn't any, this time to select nero7 as secondary choice?
Almost anything is "possible" if I take the time to implement. But right now eac3to does not support Nero 9 decoders. It's as easy as that.

one question though, is the -slowdown problem of that h264 50i .ts sample fixed yet?
Didn't get around fixing that yet. But it's on top of my priority list.

Any detail on this? Or it's a surprise? :p
I've been asked to provide a way to add encoding plugins. Will there be many plugins? I've no idea...

madshi

9th November 2008, 22:05

Just fyi, the version shown by 2.75 is v0.00.
Thanks. I've re-uploaded a fixed v2.75 with no changes apart from working version information.

rica

9th November 2008, 23:22

Almost anything is "possible" if I take the time to implement.

Good news for Nero9 owners; i guess it is on your todo list.

SpaceAgeHero

10th November 2008, 01:10

Isn't it about time to build a GUI, madshi? :confused:

Thunderbolt8

10th November 2008, 01:47

what for? :D

bmnot

10th November 2008, 03:54

I tried a track from a DVD were the box states many times it's "Dolby Digital 5.1 Surround EX" but eac3to 2.75 detected it as plain ol AC3 5.1.

Here it is if you wanna check it out: http://www.sendspace.com/file/8r29v3

It doesn't work because it doesn't begin correctly. If it's an M2TS file, the 5th byte must be a 0x47. If it's a TS file, the first byte must be a 0x47. So you can use a hexeditor, search for the first 0x47 byte and remove the first few bytes of the file, until the fifth (or first) byte is 0x47. If that doesn't work, just repeat the process another 2-3 times. At some time eac3to will properly detect and handle the file. So you did this to the sample I gave you and it demuxed fine? Then I don't know how to give you an example to work with. I remember it occurred on Predator, if you have access to that disc. You couldn't simply make a switch to simple ignore problem in non-clean tracks?

madshi

10th November 2008, 09:59

I tried a track from a DVD were the box states many times it's "Dolby Digital 5.1 Surround EX" but eac3to 2.75 detected it as plain ol AC3 5.1.

Here it is if you wanna check it out: http://www.sendspace.com/file/8r29v3
That track is definitely not flagged as EX. It might still be EX, though. Some EX tracks are not flagged as such. But I can say with certainty that eac3to not showing this track as EX is correct.

So you did this to the sample I gave you and it demuxed fine?
Yes, but I only tested with v2.74.

Then I don't know how to give you an example to work with. I remember it occurred on Predator, if you have access to that disc. You couldn't simply make a switch to simple ignore problem in non-clean tracks?
I don't have Predator. Why don't you simply retry with the latest eac3to version? There was a bug which resulted in eac3to sometimes aborting processing at the very end of movies. That was fixed in v2.74. So probably Predator should work fine now.

ACrowley

10th November 2008, 11:13

That track is definitely not flagged as EX. It might still be EX, though. Some EX tracks are not flagged as such. But I can say with certainty that eac3to not showing this track as EX is correct.

Eah, i have some DD EX Tracks from PAL DVD which are not flagged correctly.

Thers a little Tool called EXme :) I can set the EX Flag with the Tool

madshi

10th November 2008, 11:55

There are a lot more "old" BD Releases which contains seemless branching and "splitted" Releases that produce Audio-Spikes and loud noise at the cutting/fixxing Points after timecode-rerun :(

Something is borked when using DTS, no matter if it´s Arcsoft or Sonic. With AC3-Track splitting/joining is ok.

As i can remember, the same BD´s using an older eac3to version < 2.58 doesn´t produce this errors.

A good try will be Conair/German BD. Try to assemble
the GERMAN DTS Track. On the cut point there is a loud
spike on the decoded file (destination format can be WAV or AC3 - result is the same)
I finally got around getting the German Con Air Blu-Ray to test this and I can't reproduce any problem with the German DTS track. No wrong loud spike for me. Ok, at the cut point there is this loud plane sound. That is some kind of spike. But it's supposed to be that way and it's that way if you play the separate m2ts parts, too. And it's in the AC3 track, too. So it's not caused by eac3to. Don't know if that is the spike you were talking about? Anyway, do you still have this spike problem with the latest eac3to version? And are you sure that it's caused by eac3to and not already in the original audio track?

Edit: Screw that, I *can* reproduce the problem, but only under specific circumstances...

madshi

10th November 2008, 11:56

Still the same "issue" with DTS and the -core option.
That should be fixed now, btw...

BlackJack1

10th November 2008, 12:23

Eah, i have some DD EX Tracks from PAL DVD which are not flagged correctly.

Thers a little Tool called EXme :) I can set the EX Flag with the Tool

Can you give me a link please. I searched google but with no result....
Thx.

tebasuna51

10th November 2008, 13:43

Originally Posted by ACrowley
Thers a little Tool called EXme I can set the EX Flag with the Tool
Can you give me a link please. I searched google but with no result....
Thx.

You don't need google is near here, in this Audio Encoding thread (http://forum.doom9.org/showthread.php?t=103382)

Only if bsid = 6 and xbsi2e = 1the tool can work safely

If bsid = 8 and Timecode2 = 1 maybe also can work but I never see Timecode2 = 1

Of course monsters.ac3 can't be EXme (bsid=8, Timecode2=0) safely.

jmonier

10th November 2008, 16:26

Isn't it about time to build a GUI, madshi? :confused:

The GUI listed in the first post works fine. Have you tried that?

Thunderbolt8

10th November 2008, 19:28

just as info, theres an eac3 commentary track on the kung fu panda blu-ray (its quite similar to the other ac3 commentary, but seems to be slightly edited differently at some places; or the frame rate is different, as in reviews only 1 commentary track is mentioned)

NorthPole

10th November 2008, 23:15

earl

11th November 2008, 00:35

is there a way to use eac3to to make all video and all audio stream from a .m2ts file in one step to put with out any change of bitrate into a .mkv file ?
or is there any other programm that can do this ?

itsancho

11th November 2008, 03:07

qyqgpower

11th November 2008, 17:05

Why do you think it will?

No, it doesn't support multiple mono wav input.
I see. Thanks

You can use WaveWizard or Sox command line:

Sox -M FL.wav FR.wav FC.wav LF.wav SL.wav SR.wav multichannel.wav
The output of sox seems not to be very compliant (foobar can't play the output wav), but eac3to is able to encode it to flac correctly, so it's fine anyway:)
:thanks:

deathlord

11th November 2008, 19:29

madshi

I have a question on the new ac3 ex feature. Dolby digital ex has a 6th channel built in for use in 6.1 or 7.1 setups. Can eac3to decode those tracks to 7.1?

nautilus7

11th November 2008, 20:09

HI!

A Problem with my new Harry Potter 5 HDDVD.

It is an original german HDDVD bought from Amazon last week.

With the following analysing command

eac3to h: hp5.log

the following appears:

eac3to v2.75
command line: eac3to h: hp5.log
------------------------------------------------------------------------------
EVO, 1 video track, 1 audio track, 7 subtitle tracks, 0:43:54
"hidden"
1: Chapters, 6 chapters without names
2: VC-1, 1080i60 /1.001 (16:9)
3: E-AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
4: Subtitle, English
5: Subtitle, French
6: Subtitle, German
7: Subtitle, Spanish
8: Subtitle, Japanese
9: Subtitle, Catalan
10: Subtitle, Portuguese
Track 3 is used for destination file "hp5.log".
This audio conversion is not supported.

The Tracks with the English 5.1 Sound and the German 5.1 Sound are missing. The time is 0:43:54 seems also wrong.
Do I have a corrupt disk?

:thanks:Unless you put the complete command line we can't help.

Type eac3to h: and then try eac3to h: ?) where ? stands for the main movie title and report back.

cavediver

11th November 2008, 21:07

Is it possible to keep a truehd track for remuxing into .ts or .m2ts or do I have to convert the truehd track to pcm?

I've been using the following command to extract the truehd track, but haven't been able to get it to play after remuxing with tsmuxer and then again with tsremux. I get no audio at all when trying to play the file on a Popcorn Hour A-110.

eac3to i:\zohan 1) 3: i:\zohan.thd+ac3

kemac

11th November 2008, 21:18

flyingernst

12th November 2008, 09:57

eac3to v2.74
command line: "C:\Program Files\Minnetonka Audio Software\SurCode DVD DTS\eac3to.exe" "K:" 1) 4: "F:\Hancock\english.dts -nero
------------------------------------------------------------------------------
M2TS, 2 video tracks, 6 audio tracks, 19 subtitle tracks, 1:42:14
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: TrueHD/AC3, English, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 448kbps, 48khz)
5: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -30dB
6: TrueHD/AC3, German, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB)
7: AC3, Russian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
8: AC3, Ukrainian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
9: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), German
13: Subtitle (PGS), Dutch
14: Subtitle (PGS), Danish
15: Subtitle (PGS), Estonian
16: Subtitle (PGS), Finnish
17: Subtitle (PGS), Modern Greek
18: Subtitle (PGS), Hindi
19: Subtitle (PGS), Lithuanian
20: Subtitle (PGS), Norwegian
21: Subtitle (PGS), Swedish
22: Subtitle (PGS), Turkish
23: Subtitle (PGS), English
24: Subtitle (PGS), German
25: Subtitle (PGS), Dutch
26: Subtitle (PGS), Russian
27: Subtitle (PGS), Ukrainian
28: Subtitle (PGS), English
This audio conversion is not supported.

why? what is my failure

odin24

12th November 2008, 10:16

why? what is my failure

Your line looks fine, except there is no closing quotation (") on the destination file.

Maybe that's it???

flyingernst

12th November 2008, 10:33

oh maaaan

madshi

12th November 2008, 10:39

Just wondering if there was any thought to adding standard input to eac3to? I was looking to decode a tta lossless stream and wanted to feed it directly into eac3to for dts encoding.
What format would the data have, you send to standard input? Generally it's a difficult thing for me to add because eac3to by design first analyzes the format of the source, and then in a 2nd step rereads the whole source file to do its work. This approach doesn't fit well to standard input where the source file can not be read twice.

is there a way to use eac3to to make all video and all audio stream from a .m2ts file in one step to put with out any change of bitrate into a .mkv file ?
or is there any other programm that can do this ?
You can do "eac3to source video.mkv". That will mux the video track to MKV and demux/transcode all audio tracks to external audio files. Lossless (TrueHD + DTS-HD Master Audio) and PCM tracks will be converted to FLAC. All other bitstream audio tracks will be kept in their original form. You can then in a 2nd step use mkvtoolnix to add the audio tracks to the video MKV.

Alternatively you can use the GUI listed in the first post of this thread. The GUI allows you to automate the 2 steps mentioned above.

Muxing TrueHD as it is into MKV is currently not possible because MKV doesn't support TrueHD yet. But the transcoding to FLAC is lossless.

I have a question on the new ac3 ex feature. Dolby digital ex has a 6th channel built in for use in 6.1 or 7.1 setups. Can eac3to decode those tracks to 7.1?
No.

Is it possible to keep a truehd track for remuxing into .ts or .m2ts or do I have to convert the truehd track to pcm?

I've been using the following command to extract the truehd track, but haven't been able to get it to play after remuxing with tsmuxer and then again with tsremux. I get no audio at all when trying to play the file on a Popcorn Hour A-110.

eac3to i:\zohan 1) 3: i:\zohan.thd+ac3
This is a problem which is most probably not caused by eac3to, but by either tsMuxeR or TsRemux or by the Popcorn Hour (don't know). So you won't get help on this thread, unfortunately, unless you find evidence that eac3to is at fault.

odin24

12th November 2008, 11:22

oh maaaan

Did that work?

nautilus7

12th November 2008, 11:25

I got the Wall E blu-ray Disc. It supposed to have DTS-HD MA 6.1 track, but eac3to reports it as 5.1 Here's the log:

eac3to wall.e.dtshd -logdts

+ DTS-Core
- frameSize 2012
- DTS-ES +
- channelNo 5
- lfe 1
- channelDescr 5.1
- samplingRate 48000
- bitDepth 24
- bitrate 1536000
- samplesPerFrame 512
- copyHistory 1

+ DTS-HD
- fullSize 2776
- headerSize 32
- refClockCode 1/48000
- frameDurationCode 1
- activeMasks [1], [[1]]
+ Asset [0]
- fullSize 2744
- headerSize 14
- corePackets Core
- extSubStrPackets XLL
- bitResolution 24
- maxSampleRate 48000
- totalNumChannels 6
- activeSpeakers C L R Ls Rs LFE ($f)

DTS Master Audio, 5.1 channels, 24 bits, 48khz
(core: DTS-ES, 5.1 channels, 24 bits, 1536kbps, 48khz)

Both Arcsoft and Sonic decoder decode it as 5.1, so i guess it's really 5.1ch, right?

I made a sample (http://www.sendspace.com/file/a7dbxu) in case you want to have a look.

madshi

12th November 2008, 11:28

I got the Wall E blu-ray Disc. It supposed to have DTS-HD MA 6.1 track, but eac3to reports it as 5.1.

Both Arcsoft and Sonic decoder decode it as 5.1, so i guess it's really 5.1ch, right?

I made a sample (http://www.sendspace.com/file/a7dbxu) in case you want to have a look.
This seems to be 5.1 ES, with the back channel(s) matrixed into the surround channels. Kind of strange, haven't seen a "DTS Master Audio ES" track yet. Why didn't they go regular 6.1 or 7.1? I've no idea...

NorthPole

12th November 2008, 16:42

What format would the data have, you send to standard input? Generally it's a difficult thing for me to add because eac3to by design first analyzes the format of the source, and then in a 2nd step rereads the whole source file to do its work. This approach doesn't fit well to standard input where the source file can not be read twice.

How about having the ability to feed in standard wav file format. I currently have material stored in TTA format. I can decode that at the command line and pipe it into eac3to.

flyingernst

12th November 2008, 17:35

Did that work?

have not tried it until now...was in the college. But my PC is still busy with the mkv .... so I have to wait

flyingernst

12th November 2008, 19:06

Hi, here is a second project:
Want a movie in m2ts with DolbyTrueHD. It is in original splittet so I have to fix that gab thing (which I dont understand) with this method:

eac3to v2.74
command line: "C:\Program Files\Minnetonka Audio Software\SurCode DVD DTS\eac3to.exe" "K:" 1) "F:\Hancock\film.mkv"
------------------------------------------------------------------------------
M2TS, 2 video tracks, 6 audio tracks, 19 subtitle tracks, 1:42:14
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: TrueHD/AC3, English, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 448kbps, 48khz)
5: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -30dB
6: TrueHD/AC3, German, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB)
7: AC3, Russian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
8: AC3, Ukrainian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
9: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), German
13: Subtitle (PGS), Dutch
14: Subtitle (PGS), Danish
15: Subtitle (PGS), Estonian
16: Subtitle (PGS), Finnish
17: Subtitle (PGS), Modern Greek
18: Subtitle (PGS), Hindi
19: Subtitle (PGS), Lithuanian
20: Subtitle (PGS), Norwegian
21: Subtitle (PGS), Swedish
22: Subtitle (PGS), Turkish
23: Subtitle (PGS), English
24: Subtitle (PGS), German
25: Subtitle (PGS), Dutch
26: Subtitle (PGS), Russian
27: Subtitle (PGS), Ukrainian
28: Subtitle (PGS), English
Creating file "F:\Hancock\film - Chapters.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a05] Extracting audio track number 5...
[a06] Extracting audio track number 6...
[a07] Extracting audio track number 7...
[a08] Extracting audio track number 8...
[s10] Extracting subtitle track number 10...
[s11] Extracting subtitle track number 11...
[s12] Extracting subtitle track number 12...
[s23] Extracting subtitle track number 23...
[s21] Extracting subtitle track number 21...
[s20] Extracting subtitle track number 20...
[s26] Extracting subtitle track number 26...
[s27] Extracting subtitle track number 27...
[s14] Extracting subtitle track number 14...
[s15] Extracting subtitle track number 15...
[s19] Extracting subtitle track number 19...
[s13] Extracting subtitle track number 13...
[s25] Extracting subtitle track number 25...
[s28] Extracting subtitle track number 28...
[s22] Extracting subtitle track number 22...
[s16] Extracting subtitle track number 16...
[s17] Extracting subtitle track number 17...
[s24] Extracting subtitle track number 24...
[s18] Extracting subtitle track number 18...
[a04] Extracting TrueHD stream...
[a04] Decoding with libav/ffmpeg...
[v02] Muxing video to Matroska...
[a06] Extracting TrueHD stream...
[a06] Decoding with libav/ffmpeg...
[a04] Encoding FLAC with libFlac...
[a06] Encoding FLAC with libFlac...
[a07] Removing AC3 dialog normalization...
[a08] Removing AC3 dialog normalization...
[a05] Removing AC3 dialog normalization...
[a06] Creating file "F:\Hancock\film - 6 - TrueHD+AC3, German, 5.1 channels, 48khz.24bit.flac"...
[a04] Creating file "F:\Hancock\film - 4 - TrueHD+AC3, English, 5.1 channels, 48khz.24bit.flac"...
[a05] Creating file "F:\Hancock\film - 5 - AC3, English, 5.1 channels, 448kbps, 48khz.ac3"...
[a08] Creating file "F:\Hancock\film - 8 - AC3, Ukrainian, 5.1 channels, 448kbps, 48khz.ac3"...
[a07] Creating file "F:\Hancock\film - 7 - AC3, Russian, 5.1 channels, 448kbps, 48khz.ac3"...
[s23] Creating file "F:\Hancock\film - 23 - Subtitle (PGS), English.sup"...
[s24] Creating file "F:\Hancock\film - 24 - Subtitle (PGS), German.sup"...
[s25] Creating file "F:\Hancock\film - 25 - Subtitle (PGS), Dutch.sup"...
[libav] Lossless check failed - expected 2e, calculated 2b
[s10] Creating file "F:\Hancock\film - 10 - Subtitle (PGS), English.sup"...
[s15] Creating file "F:\Hancock\film - 15 - Subtitle (PGS), Estonian.sup"...
[s11] Creating file "F:\Hancock\film - 11 - Subtitle (PGS), English.sup"...
[s18] Creating file "F:\Hancock\film - 18 - Subtitle (PGS), Hindi.sup"...
[s22] Creating file "F:\Hancock\film - 22 - Subtitle (PGS), Turkish.sup"...
[s19] Creating file "F:\Hancock\film - 19 - Subtitle (PGS), Lithuanian.sup"...
[s13] Creating file "F:\Hancock\film - 13 - Subtitle (PGS), Dutch.sup"...
[s12] Creating file "F:\Hancock\film - 12 - Subtitle (PGS), German.sup"...
[s17] Creating file "F:\Hancock\film - 17 - Subtitle (PGS), Modern Greek.sup"...
[s21] Creating file "F:\Hancock\film - 21 - Subtitle (PGS), Swedish.sup"...
[s14] Creating file "F:\Hancock\film - 14 - Subtitle (PGS), Danish.sup"...
[s20] Creating file "F:\Hancock\film - 20 - Subtitle (PGS), Norwegian.sup"...
[s16] Creating file "F:\Hancock\film - 16 - Subtitle (PGS), Finnish.sup"...
[s27] Creating file "F:\Hancock\film - 27 - Subtitle (PGS), Ukrainian.sup"...
[libav] Lossless check failed - expected 16, calculated 33
[libav] Lossless check failed - expected 9e, calculated bd
[libav] Lossless check failed - expected 9, calculated aa
[libav] Lossless check failed - expected 69, calculated c3
[libav] Lossless check failed - expected 3, calculated 37
[libav] Lossless check failed - expected 1c, calculated 4e
[libav] Lossless check failed - expected 3e, calculated 37
[libav] Lossless check failed - expected 5a, calculated c5
[libav] Lossless check failed - expected 9f, calculated 15
[libav] Lossless check failed - expected 22, calculated 74
[libav] Lossless check failed - expected 39, calculated 3a
[libav] Lossless check failed - expected 8a, calculated 62
[libav] Lossless check failed - expected a8, calculated 86
[libav] Lossless check failed - expected e9, calculated ec
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] The zero bytes were successfully removed.
[a05] Audio overlaps for 27ms at playtime 0:05:51.
[a05] Audio overlaps for 23ms at playtime 0:27:55.
[a05] Audio overlaps for 16ms at playtime 0:32:31.
[a05] Audio overlaps for 9ms at playtime 0:34:13.
[a05] Audio overlaps for 20ms at playtime 0:48:54.
[a05] Audio overlaps for 18ms at playtime 0:49:56.
[a05] Audio overlaps for 17ms at playtime 0:52:28.
[a05] Audio overlaps for 29ms at playtime 0:53:10.
[a05] Audio overlaps for 31ms at playtime 0:57:34.
[a05] Audio overlaps for 8ms at playtime 1:04:46.
[a05] Audio overlaps for 25ms at playtime 1:07:09.
[a05] Audio overlaps for 7ms at playtime 1:10:47.
[a05] Audio overlaps for 23ms at playtime 1:13:47.
[a05] Audio overlaps for 24ms at playtime 1:15:20.
[a05] Audio overlaps for 18ms at playtime 1:25:56.
[a05] The audio file was demuxed without making use of the gap/overlap information.
[a05] Please rerun the same eac3to command line. That will correct the gaps/overlaps.
[a06] The original audio track has a constant bit depth of 16 bits.
[a06] The zero bytes were successfully removed.
[a07] Audio overlaps for 27ms at playtime 0:05:51.
[a07] Audio overlaps for 23ms at playtime 0:27:55.
[a07] Audio overlaps for 16ms at playtime 0:32:31.
[a07] Audio overlaps for 9ms at playtime 0:34:13.
[a07] Audio overlaps for 20ms at playtime 0:48:54.
[a07] Audio overlaps for 18ms at playtime 0:49:56.
[a07] Audio overlaps for 17ms at playtime 0:52:28.
[a07] Audio overlaps for 29ms at playtime 0:53:10.
[a07] Audio overlaps for 31ms at playtime 0:57:34.
[a07] Audio overlaps for 8ms at playtime 1:04:46.
[a07] Audio overlaps for 25ms at playtime 1:07:09.
[a07] Audio overlaps for 7ms at playtime 1:10:47.
[a07] Audio overlaps for 23ms at playtime 1:13:47.
[a07] Audio overlaps for 24ms at playtime 1:15:20.
[a07] Audio overlaps for 18ms at playtime 1:25:56.
[a07] The audio file was demuxed without making use of the gap/overlap information.
[a07] Please rerun the same eac3to command line. That will correct the gaps/overlaps.
[a08] Audio overlaps for 27ms at playtime 0:05:51.
[a08] Audio overlaps for 23ms at playtime 0:27:55.
[a08] Audio overlaps for 16ms at playtime 0:32:31.
[a08] Audio overlaps for 9ms at playtime 0:34:13.
[a08] Audio overlaps for 20ms at playtime 0:48:54.
[a08] Audio overlaps for 18ms at playtime 0:49:56.
[a08] Audio overlaps for 17ms at playtime 0:52:28.
[a08] Audio overlaps for 29ms at playtime 0:53:10.
[a08] Audio overlaps for 31ms at playtime 0:57:34.
[a08] Audio overlaps for 8ms at playtime 1:04:46.
[a08] Audio overlaps for 25ms at playtime 1:07:09.
[a08] Audio overlaps for 7ms at playtime 1:10:47.
[a08] Audio overlaps for 23ms at playtime 1:13:47.
[a08] Audio overlaps for 24ms at playtime 1:15:20.
[a08] Audio overlaps for 18ms at playtime 1:25:56.
[a08] The audio file was demuxed without making use of the gap/overlap information.
[a08] Please rerun the same eac3to command line. That will correct the gaps/overlaps.
[s26] Creating file "F:\Hancock\film - 26 - Subtitle (PGS), Russian.sup"...
[s28] Creating file "F:\Hancock\film - 28 - Subtitle (PGS), English.sup"...
Added fps value to MKV header.
Video track 2 contains 147076 frames.
Video track 3 contains 147076 frames.
eac3to processing took 36 minutes, 20 seconds.
Done.

How can I use that methode and get out no flacs, but the TrueHD Files without GAP Problem
Thanks, Greetings,
Michael

madshi

12th November 2008, 19:15

How about having the ability to feed in standard wav file format. I currently have material stored in TTA format. I can decode that at the command line and pipe it into eac3to.
I'll think about that. It's not easy to implement...

Hi, here is a second project:
Want a movie in m2ts with DolbyTrueHD. It is in original splittet so I have to fix that gab thing (which I dont understand) with this method:

How can I use that methode and get out no flacs, but the TrueHD Files without GAP Problem
Thanks, Greetings,
Michael
First of all, please post logs between "[ code ]" and "[ / code ]" (without the spaces) tags. That way the logs will consume less space in your post (they'll appear in a smaller window with scroll bars).

As far as I can see from your eac3to log, the TrueHD tracks do not have any gaps/overlaps in them. eac3to only reported overlaps for the 3 AC3 tracks number 5, 7 and 8. So what you're looking for should be possible by doing this:

eac3to "K:" 1) 2: "F:\Hancock\film.mkv" 4: "F:\Hancock\Englisch.thd+ac3" 6: "F:\Hancock\Deutsch.thd+ac3"
Of course if the video MKV is already done, you can remove the "2: "F:\Hancock\film.mkv"" from the command line to save time.

Generally if there are overlaps in a TrueHD track Houston has a problem because eac3to can not properly remove overlaps from TrueHD tracks. In this specific case you're lucky cause the overlaps seem to be too small for eac3to to complain about...

flyingernst

12th November 2008, 19:22

well then thank you very much.
First I will have to see if TSmuxer will accept the TrueHD Files to import. Even when, I will have to remux the Movie with TSremux because of the PopCorn Hour wants it....

Thunderbolt8

12th November 2008, 19:26

This seems to be 5.1 ES, with the back channel(s) matrixed into the surround channels. Kind of strange, haven't seen a "DTS Master Audio ES" track yet. Why didn't they go regular 6.1 or 7.1? I've no idea...
its listed in eac3to as DTS Master Audio 5.1 with DTS-ES 5.1 core

Jeff Flowerday

12th November 2008, 19:52

Evan Almighty HD-DVD rip:

When I run eac3to on the rip it doesn't list the feature movie for some reason. All I get is the following:

C:\Users\FlowerdayJ\Videos\EVANALMIGHTY>eac3to .
1) WATERRISING_1.EVO+WATERRISING_2.EVO+RU_GIRAFFE.EVO+RU_ELEPHANT.EVO+RU_LION.EV
O+RU_GORILLA.EVO+RU_WOLVES.EVO+RU_EAGLE.EVO+RU_KOALA.EVO+RU_OTTER.EVO+RU_CHEETAH
.EVO+RU_SURFEIT.EVO+RU_AUSTRALIA.EVO+RU_CHIMP.EVO+RU_GRIZZLY.EVO+RU_LLAMA.EVO+RU
_PEACOCK.EVO+RU_CAMEL.EVO+RU_ORANGUTAN.EVO+RU_ARMADILLO.EVO+RU_GOAT.EVO+RU_ANACO
NDA.EVO+RU_EXITBEFORE.EVO+RU_OUTRO.EVO+RU_EXITAFTER.EVO+RU_END.EVO, 0:04:50
"Animal Round Up"
- VC-1, 1080p (16:9)
- E-AC3, 2.0, 48khz

nautilus7

13th November 2008, 00:45

Snowknight26

13th November 2008, 01:08

What is the difference between 2 channel AC3 Surround, 2 channel AC3 Headphone and just normal 2 channel AC3?

nautilus7

13th November 2008, 01:16

I guess dolby surround is the old "analog" dolby surround pro logic (center and surround channels matrixed into left and right), while dolby headphne is something similar (equivalent), but specially developed for headphones.

Jeff Flowerday

13th November 2008, 02:02

jruggle

13th November 2008, 04:32

Problem 1 (bruce-lee 1.0 ac3)
Sorry, obviously something went wrong. Here the 10mb-file:
http://rapidshare.com/files/161846249/bruce10MB.eac3

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

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

jruggle

13th November 2008, 05:02

I was playing around with the new floating point output and i wonder which offers better quality: AC3 decoded with nero (24 bit output) or AC3 decoded with libav to floating point?
What do you think? Is floating point output preferable over nero decoder? For sure the difference will be very small, but i don't have that much expensive equipment to test my self. Can anyone tell?
Listening tests aside, I doubt the quality is noticeably different. The internal dynamic range for AC3 is 24-bit.

First of all, we don't know what the Nero decoder does internally. It might be completely fixed point, or maybe it uses floating point everywhere and just converts to 24-bit at the end. The lavc decoder uses 24-bit fixed point for everything up to the IMDCT and downmixing in each block. If I had a 24-bit fixed-point IMDCT implementation it would not use floating point at all. The only reason the downmixing is floating point is because it's sometimes done after the IMDCT.

sehgal.v7

13th November 2008, 08:31

@Madshi
Do you want sample of Iron Man Audio in TrueHD & DTS HD-MA?
I saw sometime back you needed sample of Same track in two Lossless Formats..

madshi

13th November 2008, 08:50

What is the difference between 2 channel AC3 Surround, 2 channel AC3 Headphone and just normal 2 channel AC3?
I guess dolby surround is the old "analog" dolby surround pro logic (center and surround channels matrixed into left and right), while dolby headphne is something similar (equivalent), but specially developed for headphones.
Yep, that's correct. Dolby Headphone I believe is 5.1 downconverted to 2.0 in such a way that it is supposed to still sound like surround when played back by a headphone...

When I go straight at the drive I get the same result.

Not sure what you use to find the information but I assume the xpl files so I've zipped them up.

http://www.flowerdayconsulting.com/Misc/vpl.zip
For whatever strange reason the "VPLST000.XPL" file only contains some extra stuff. The movie is in the "VPLST001.XPL" file. Never seen that before yet! You can "fix" the problem by deleting "VPLST000.XPL" and renaming "VPLST001.XPL" to "VPLST000.XPL", because eac3to always only checks "VPLST000.XPL".

This now decodes properly with libavcodec from FFmpeg's current SVN.
Great - thanks! :)

Listening tests aside, I doubt the quality is noticeably different. The internal dynamic range for AC3 is 24-bit.

First of all, we don't know what the Nero decoder does internally. It might be completely fixed point, or maybe it uses floating point everywhere and just converts to 24-bit at the end. The lavc decoder uses 24-bit fixed point for everything up to the IMDCT and downmixing in each block. If I had a 24-bit fixed-point IMDCT implementation it would not use floating point at all. The only reason the downmixing is floating point is because it's sometimes done after the IMDCT.
Ah, that's interesting...

@Madshi
Do you want sample of Iron Man Audio in TrueHD & DTS HD-MA?
I saw sometime back you needed sample of Same track in two Lossless Formats..
Yes, why not! Such samples never hurt...

deathlord

13th November 2008, 12:55

madshi

I have a question on the new ac3 ex feature. Dolby digital ex has a 6th channel built in for use in 6.1 or 7.1 setups. Can eac3to decode those tracks to 7.1?

No.

Any plans on implementing it, or do you intentionally keep these tracks at 5.1 only?

Jeff Flowerday

13th November 2008, 15:34

htpc@rischer.at

13th November 2008, 16:01

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

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

Great work. bruce10.eac3 was my example-track.
So, when and how can I get this changes in order to convert my track?

Jeff Flowerday

13th November 2008, 19:35

For whatever strange reason the "VPLST000.XPL" file only contains some extra stuff. The movie is in the "VPLST001.XPL" file. Never seen that before yet! You can "fix" the problem by deleting "VPLST000.XPL" and renaming "VPLST001.XPL" to "VPLST000.XPL", because eac3to always only checks "VPLST000.XPL".

Just an FYI, the VPLST001.XPL is formatted weird as well, eac3to wasn't showing any chapters, so I went and deleted the <ScheduledControlList> sections in both Main Movie titles and copied the <ChapterList> section from Title 1 down into Title 27 where it was missing. After doing that eac3to showed me only one Main Movie and it had chapters.

I googled the movie and some people were having trouble with the HD-DVD player freezing on chapter operations with the particular movie. This may very well explain the problems they were having.

:rolleyes:

Boulder

13th November 2008, 19:42

What about volume normalization, is there a chance we might see that in eac3to? Sometimes tracks that are downmixed from multichannel to stereo are rather low on volume.

n0mag!c

13th November 2008, 21:33

First of all it depends on the decoder. I patched libav so that it gives me the full floating point decode. But Nero and Sonic decoders internally downconvert to 24bit PCM. Now with libav eac3to by default downconverts to 24bit PCM if you ask it to create a WAV file. The reason for that is that 24bit PCM is usually plenty good enough. The main purpose of keeping things in floating point is if you want to do further processing on the data or if you want to feed the data to an encoder which wants floating point data. If that is the case, simply use the "-full" switch. Then eac3to will create floating point WAV files (if floating point data is available).
In the educational purposes could you tell please how do you perform (for example) 24 bit to 16 bit downconverting? Just throw out 8 lower bits or doing some sort of companding (applying dynamic compression to dynamic range) similar to mu-law etc?

xkodi

13th November 2008, 22:30

before few month i reported that:

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

Sonic Audio Decoder version 5.0.0.203 (from Sonic CineVision 2.5.0) doesn't decode DTS-HD extension and decodes the DTS core only and that version 4.3.0.169 (from Sonic CineVision 2.0.2) is the last one that decodes the DTS-HD extension.

now, Sonic CineVision 2.6.1 is out, it includes version 4.3.0.230 of the decoder.

Sonic has strange version numbering:

Sonic CineVision 2.0.2: Sonic Audio Decoder version 4.3.0.169
Sonic CineVision 2.5.0: Sonic Audio Decoder version 5.0.0.203
Sonic CineVision 2.6.1: Sonic Audio Decoder version 4.3.0.230

maybe, only the last 3 digits are important and they are the build number. anyway, the new version 4.3.0.230 doesn't work with the DTS-HD extension too.

btw, while testing the new version 4.3.0.230 of Sonic i found out something that looks like a bug in eac3to: the same DTS core decoded with Sonic in eac3to and GraphEdit produce different output (not byte by byte indentical), which doesn't seem right.

odin24

13th November 2008, 23:59

I have a DTS hi-res 7.1 track @ 2084 kb/s I am about to remux for BD playback, however my HT setup is only 5.1. Is the extra kb/s only assigned to the rear surround channels or would there be "hi res" data on the other 6 channels as well?

Basically will I be getting just the core because I have a 5.1 setup? I have an HD audio amp and it is hooked up via HDMI.

Also, when demuxing a DTS HD or DTS HR track should the output be ".dtshd/hr" and would setting it as .dts write a new dts track and leave out the HD data?

Thanks.

nautilus7

14th November 2008, 00:07

HD data are used for all channels.

To demux a DTS-HD track you need .dtshd or .dts or even .dtsma/.dtshr (i have never used the last though). To get the core you need the -core switch.

odin24

14th November 2008, 00:20

HD data are used for all channels.

To demux a DTS-HD track you need .dtshd or .dts or even .dtsma/.dtshr (i have never used the last though). To get the core you need the -core switch.

So using .dts on a DTS HD/HR would in fact demux all HD data rather than just writing a new DTS track, goo to know.

I was only concerned with the fact that my setup is 5.1, and the DTS HR track I have is 7.1... and was I going to get the extra HD data.

Thanks.

Ryu77

14th November 2008, 02:09

Madshi, this may have been asked before so please excuse me if that is the case...

Since Nero 9's filters are once again accessible (as opposed to Nero 8's being locked) through direct show, do you have any plans to incorporate usage of these filters into EAC3to?

I would love to be able to upgrade my version of Nero. I am still using Nero 7, simply because I want to retain full use of your fantastic application. It just seems a shame to not be able to replace an entire multimedia suite due to this.

lithiumus

14th November 2008, 06:24

Hey Madshi,

I was wondering if you could add an option to downsize from 8 channels to 7 channels in addition to downsizing from 8 to 6 by combining the Lsr(BL) and Rsr(BR) into Cs(BC?).

When combining this option with extracting to "wavs" would take an 8 channel 7.1 stream and downsize it to 6.1 and output 7 separate wav files which you can then manually feed into a DTS-ES encoder to create a 6.1 DTS-ES track.

What do you think?

73ChargerFan

14th November 2008, 06:58

Since Nero 9's filters are once again accessible...Nero 9 sounds like a dog version, according to what I've been reading. There is no BD plug-in, and the v8 plug in won't work. BD authoring is still there, as an extra $$$ plug-in.

bmnot

14th November 2008, 08:16

I don't have Predator. Why don't you simply retry with the latest eac3to version? There was a bug which resulted in eac3to sometimes aborting processing at the very end of movies. That was fixed in v2.74. So probably Predator should work fine now.

Works with 2.75 now. Thank you very much for the fix.

Thunderbolt8

14th November 2008, 11:07

cavediver

14th November 2008, 15:14

How do I determine which playlist contains the correct movie? I ran eac3to and got the following results: Both indicate they are the same length, but use different m2ts files.

D:\Video Ripping\eac3to>eac3to i:\dieanotherday
1) 00001.mpls, 2:12:34
[130+131+133+134+136+137+139+140+142+143+145+146+148+149+151+152+154+155+157+
158+160+161+163+164+166+167+169+170+172+173+175+176+178].m2ts
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48khz
- AC3, Spanish, stereo, 48khz
- AC3, French, multi-channel, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz

2) 00099.mpls (angle 2), 2:12:34
[130+132+133+135+136+138+139+141+142+144+145+147+148+150+151+153+154+156+157+
159+160+162+163+165+166+168+169+171+172+174+175+177+178].m2ts
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48khz
- AC3, Spanish, stereo, 48khz
- AC3, French, multi-channel, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz

3) 00200.mpls, 00028.m2ts, 0:51:40
- MPEG2, 480i60 /1.001 (16:9)
- AC3, English, stereo, 48khz

4) 00201.mpls, 00029.m2ts, 0:23:35
- MPEG2, 480i60 /1.001 (16:9)
- AC3, English, stereo, 48khz

5) 00202.mpls, 00030.m2ts, 0:22:39
- MPEG2, 480i60 /1.001 (16:9)
- AC3, English, stereo, 48khz

D:\Video Ripping\eac3to>

jj666

14th November 2008, 15:21

How do I determine which playlist contains the correct movie?

Check out some of the "different" M2TS files. It's more than likely different languages in the movie titles etc.

-jj-

madshi

14th November 2008, 21:26

Great work. bruce10.eac3 was my example-track.
So, when and how can I get this changes in order to convert my track?
Justin just notified me about a bug he found in the new libav AC3 decoder code. So I can't give you any date.

What about volume normalization, is there a chance we might see that in eac3to? Sometimes tracks that are downmixed from multichannel to stereo are rather low on volume.
What exactly do you mean with that? Do you mean automatic volume increase just below clipping levels?

In the educational purposes could you tell please how do you perform (for example) 24 bit to 16 bit downconverting? Just throw out 8 lower bits or doing some sort of companding (applying dynamic compression to dynamic range) similar to mu-law etc?
Any integer to integer bitdepth reduction (e.g. 24 -> 16) is performed by doing standard TPDF dithering.

btw, while testing the new version 4.3.0.230 of Sonic i found out something that looks like a bug in eac3to: the same DTS core decoded with Sonic in eac3to and GraphEdit produce different output (not byte by byte indentical), which doesn't seem right.
Is there a volume difference? You could compare output to the ArcSoft decoder to find out whether the eac3to decode or the GraphEdit decode is correct.

do you have any plans to incorporate usage of these filters into EAC3to?
Not at this time. Maybe later... AFAIK, you can install Nero 7 and Nero 9 at the same time, btw.

Any plans on implementing it
I was wondering if you could add an option to downsize from 8 channels to 7 channels in addition to downsizing from 8 to 6 by combining the Lsr(BL) and Rsr(BR) into Cs(BC?).
I'm not sure about these feature wishes. There are a lot of channel manipulation features I could add, if I knew the exact matrix or dematrix maths. Maybe I'll find a way to do this properly, maybe not. Can't say right now...

How do I determine which playlist contains the correct movie?
This movie seems to have multiple angles. This is usually used to encode different languages in a space saving way. E.g. credits in English, French, German etc. Usually angle 1 is English, but there's no guarantee for that, so you better check out the different m2ts parts yourself to find out which angle is which language (as jj666 already said).

madshi

14th November 2008, 21:47

I have a DTS hi-res 7.1 track @ 2084 kb/s I am about to remux for BD playback, however my HT setup is only 5.1. Is the extra kb/s only assigned to the rear surround channels or would there be "hi res" data on the other 6 channels as well?
HD data are used for all channels.
Actually it's not that simple. Everything depends on what the studio did. Here are 3 very different DTS-HD High Resolution examples:

Example 1, Basic Instinct (Blu-Ray)

+ DTS-Core
- frameSize 2012
- DTS-ES +
- channelNo 6
- lfe 1
- channelDescr 6.1
- samplingRate 48000
- bitDepth 24
- bitrate 1536000
- extAudio XCh
- samplesPerFrame 512
- copyHistory 1
+ DTS-HD
- fullSize 2040
- headerSize 28
- refClockCode 1/48000
- frameDurationCode 1
- activeMasks [1], [[1]]
+ Asset [0]
- fullSize 2012
- headerSize 10
- corePackets Core+XCh
- extSubStrPackets XBR
- bitResolution 24
- maxSampleRate 48000
- totalNumChannels 7
- activeSpeakers C L R Ls Rs LFE Cs ($1f)
DTS Hi-Res, 6.1 channels, 0:00:14, 24 bits, 3093kbps, 48khz
(core: DTS-ES, 6.1 channels, 0:00:14, 24 bits, 1536kbps, 48khz)
This has a core with 6.1 discrete channels. The DTS-HD extension contains additional bitrate for all 6.1 channels. So if you extract the core, you lose bitrate for all 6.1 channels. This example matches nautilus7's explanation.

Example 2, Herbert Grönemeyer 12 live (HD DVD)

+ DTS-Core
- frameSize 2012
- DTS-ES +
- channelNo 5
- lfe 1
- channelDescr 5.1
- samplingRate 48000
- bitDepth 24
- bitrate 1536000
- samplesPerFrame 512
- copyHistory 1
+ DTS-HD
- fullSize 1400
- headerSize 28
- refClockCode 1/48000
- frameDurationCode 1
- activeMasks [1], [[1]]
+ Asset [0]
- fullSize 1372
- headerSize 12
- corePackets Core
- extSubStrPackets XXCh+X96
- bitResolution 24
- maxSampleRate 96000
- totalNumChannels 8
- activeSpeakers C L R LFE Lsr Rsr Lss Rss ($84b)
DTS Hi-Res, 7.1 channels, 0:00:16, 24 bits, 2605kbps, 96khz
(core: DTS-ES, 5.1 channels, 0:00:16, 24 bits, 1536kbps, 48khz)
This example has a core with 5.1 discrete channels. The DTS-HD information contains a channel extension to 7.1 and a sample rate extension to 96khz. But there's absolutely no added bitrate in the DTS-HD block for the 48khz version of the 5.1 channels. So if you extract the core, all you lose is the two added channels and 96khz, but you don't lose additional bitrate for the 5.1 channels (unless you want 96khz and count the added 96khz bitrate).

Example 3, House of 1000 Corpses (Blu-Ray)

+ DTS-Core
- frameSize 2012
- DTS-ES +
- channelNo 5
- lfe 1
- channelDescr 5.1
- samplingRate 48000
- bitDepth 24
- bitrate 1536000
- samplesPerFrame 512
- copyHistory 1
+ DTS-HD
- fullSize 1740
- headerSize 32
- refClockCode 1/48000
- frameDurationCode 1
- activeMasks [1], [[1]]
+ Asset [0]
- fullSize 1708
- headerSize 13
- corePackets Core
- extSubStrPackets XBR+XXCh
- bitResolution 24
- maxSampleRate 48000
- totalNumChannels 8
- activeSpeakers C L R LFE Lsr Rsr Lss Rss ($84b)
DTS Hi-Res, 7.1 channels, 0:00:14, 24 bits, 2864kbps, 48khz
(core: DTS-ES, 5.1 channels, 0:00:14, 24 bits, 1536kbps, 48khz)
Here you can see that the core contains 5.1 discrete channels. The DTS-HD part of the track contains both a bitrate extension (XBR) and a channel extension to 7.1 (XXCh). So if you extract the core, you lose the back channels and also you lose additional bitrate for the 5.1 channels.

Basically, if you want to know what you'll lose by extract the core, you should use the undocumented "-logdts" switch and then check out which extensions are stored in the DTS-HD blocks. If you strip off XCh or XXCh extensions, you're losing additional channels. If you strip off the X96 extension, you're losing 96khz. If you strip off the XBR extension, you're losing additional bitrate for the 5.1 channels. If you strip off the XLL extension, you're losing Master Audio. XCh, XXCh and X96 extensions can be contained in either the core or in the DTS-HD blocks, while XBR and XLL extensions can only be contained in the DTS-HD blocks...