eac3to - audio conversion tool [Archive] - Page 11

Abradoks

19th May 2010, 01:20

1) If FFmpeg is modified to give floating-point output will the results be closer to eac3to.
I don't know how to modify that patch to make it work with ffmpeg. If I apply it as is, ffmpeg outputs noise. So maybe someone else can check it?

3) See what the Nero decoder does compared to liba52 and libavcodec.
Nero make it even more confusing. Here are files (http://www.mediafire.com/?sharekey=01c0a12ada5691d6ed24a2875c7fa58e8c31fa5ed38dd370ea4ac78345cbe4ce). *-eac3to-nero.flac were decoded with eac3to; *-nero.flac with Nero 10 decoder and Graphedit.
Graphs:
http://img32.imageshack.us/img32/6349/5anero2.th.png (http://img32.imageshack.us/i/5anero2.png/)http://img85.imageshack.us/img85/8837/224nero2.th.png (http://img85.imageshack.us/i/224nero2.png/)http://img153.imageshack.us/img153/4343/640nero2.th.png (http://img153.imageshack.us/i/640nero2.png/)
So, eac3to-nero "looks" like ffmpeg for 5a.ac3, like liba52 for 224.ac3 and like eac3to-libav for 640.ac3.
I was surprised even more, when I compared SNR results:
5a-eac3to-libav: 27.194 dB (Gain for File B = 1.0009)
5a-eac3to-nero: 28.607 dB (Gain for File B = 0.99991)
5a-ffmpeg: 29.057 dB (Gain for File B = 0.99994)
5a-nero: 28.625 dB (Gain for File B = 1.5873)
5a-foo-liba52: 28.822 dB (Gain for File B = 0.99985)

224-eac3to-libav: 16.188 dB (Gain for File B = 0.99912)
224-eac3to-nero: 16.18 dB (Gain for File B = 0.99182)

224-ffmpeg: 13.399 dB (Gain for File B = 1.2648)
224-nero: 13.4 dB (Gain for File B = 2.0076)

224-foo-liba52: 16.166 dB (Gain for File B = 0.99174)

640-eac3to-libav: 27.809 dB (Gain for File B = 0.99982)
640-eac3to-nero: 27.813 dB (Gain for File B = 0.99982)

640-ffmpeg: 16.091 dB (Gain for File B = 1.2752)
640-nero: 16.091 dB (Gain for File B = 2.0243)

640-foo-liba52: 27.537 dB (Gain for File B = 0.99971)
Also, I've got 5a.ac3 decoded with Cyberlink PowerDVD 10. 5a-cyber.flac (http://www.mediafire.com/?ykz1wzgkylz):
http://img412.imageshack.us/img412/8210/5acyber.th.png (http://img412.imageshack.us/i/5acyber.png/)
SNR = 28.844 dB (Gain for File B = 1.5872)
As you can see, Cyberlink hasn't that problem with attenuation of high frequencies.
Can anyone check these samples with Windows 7 decoder? Or maybe even with reference decoder?

What program did you use to make the graphs? I want to use it in my comparisons as well.
Sound Forge. AFAIK baudline can provide same (http://www.baudline.com/manual/average.html) functionality.

madshi

19th May 2010, 09:04

Notwithstanding, what do you have to say about the "zero-padding" issue? :confused:
I'd need a sample.

Also: will eac3to v3.19 recognize "pure" .DTSHD files? :)
I have that on my to do list. I have a lot of things on my to do list, though.

— add switch -dtsfreq , for modifying the playback frequency of the lossy DTS streams ( most useful if someone intends to create a DTS-CD @ 1387.0828125kbps but is too lazy to deal with a hex editor :p )

— add switch -dtsrate, for modifying the apparent bitrate of .cpt files
I'm not sure what exactly you want to have changed? Just the bitstream header fields or what?

I actually have a question regarding ArcSoft decoding of DTS-HD MA streams in eac3to. Do I have to set any parameters in ASAudioHD.ax such as channel outputs or anything or does eac3to just use the engine of the dlls from ArcSoft DTS Decoder to decode the dts-hd ma streams in its original state?
No need to change anything. eac3to takes care of everything.

I have the same problem with crackling sound when converting a DTS-HD 7.1 file into a FLAC 7.1 file using the Arcsoft decoder. Only 7.1 goes wrong, 5.1 works like a charm. Does the same solution work here too? If not, is there another remedy?
Does eac3to report the 7.1 file as "(strange setup)"? If so, this is a bug in the ArcSoft decoder. Anyway, can I have a sample of the DTS-HD 7.1 file, please?

I have tested DtsDec.dll 1.0.2.2 + DTSdecoderDLL.dll 1.1.0.0, and these do NOT decode stereo DTSHD Master Audio correctly (not on my computer, at least).
I'm not aware of problems with stereo decoding. Can I have a sample, please?

Any plans to add -check capabilities to MKVs?
Does it not work yet? Will add that on my to do list...

For a particular HD-DVD, eac3to detects no subtitle track. Same thing when I open the main EVO files in MediaInfo.
But when I play the disc in my HD-DVD player, I have the possibility to switch between french audio / no subtitle and english audio / french subtitle.
eac3to detects only one video track, so I don't think there are two video tracks, one with subtitles and one without subtitles.
Can you explain where are (hidden) the subtitles please ?
Does eac3to detect the subtitles in the track listing ("eac3to c:\rippedHDDvd")?

got a 5.1 dtsma track, which only has data for the 2.0 channels, the rest is empty. so instead of keeping it as 5.1 with 4 empty channels, I'd rather turn in into a 2.0 dtsma & flac track. how do I do this? 5.1 -> wavs? and then afterwards?
Yeah, I'd export to wavs and then use wavewizard or a similar tool to combine the 2 mono wav files together. eac3to currently can't do that.

That's a good news.
I hope it means that lots of tools, and especially eac3to, could very soon support this E-AC3 extension ?
It's on my to do list.

I've recently bought Spiderman trilogy in bluray, and I successfully extracted all the audio tracks I needed from the 3 discs (in truehd format, .thd).
When I tried to convert the .thd files to mono wavs (or flac, or single wav) with eac3to & libav, no issue at all with audio from movie 1 and 3, but I experienced some problems, with the second movie's track decoding... Libav decoding in eac3to gave the repeated "Lossless check failed" warnings, even if the encoded wav files seemed to be good
Spiderman 2 is a seamless branching disc, as far as I remember. If you demux/extract the THD data first and convert it to WAVs or FLAC later, you'll get a "lossless check failed" warning for every seamless branch. That is to be expected. You should have gotten a warning from eac3to, though, when extracting the THD track in the first place. It's not a good idea to extract THD tracks from seamless branching Blu-Rays. Audio and video will drift out of sync, because eac3to can't fix the audio overlaps with THD tracks, when demuxing them. You should convert to THD/WAVs directly from the original Blu-Ray structure. Then those "lossless check failed" warnings should go away and audio and video will stay in sync.

madshi

19th May 2010, 09:04

Eac3to has too many unresolved issues to use it as decoder:
- DTS-HD 7.1 bug (http://forum.doom9.org/showthread.php?p=1363005#post1363005)
This is a bug in the ArcSoft DTS decoder and it only affects a handful of 7.1 tracks. Most DTS-HD 7.1 tracks decode just fine. Furthermore eac3to warns about the problem, when it occurs. There's nothing more I can do about it. Can you point me to a different decoder which doesn't have this problem? I'm not aware of any. So if you say eac3to can't be used as a decoder, then which DTS-HD decoder are you using instead?

- Nero DRC bug (http://forum.doom9.org/showthread.php?p=1353768#post1353768)
Hmmmm... Are you really sure about that? Do you still have files with which I can reproduce that? The files in your original post are no longer available for download.

- bug, when output is different for *.wav and *.wavs (at least with libav and ac3/dts)
That's the first time I hear about that. Can I have some more details?

Edit: Maybe you're confused by dithering? eac3to lets libav decode to floating point. For WAV/WAVs output eac3to dithers the floating decoding data down to integer. Dithering means adding random noise to the data. So it's clear that if you decode the same AC3/DTS track twice with eac3to/libav, the results will be different each time. If you want floating point WAV/WAVs files instead, use the eac3to "-full" switch. Then the files should be identical every time.

- ...
Yes? Go on, please!

0x0

19th May 2010, 09:17

Spiderman 2 is a seamless branching disc, as far as I remember. If you demux/extract the THD data first and convert it to WAVs or FLAC later, you'll get a "lossless check failed" warning for every seamless branch. That is to be expected. You should have gotten a warning from eac3to, though, when extracting the THD track in the first place. It's not a good idea to extract THD tracks from seamless branching Blu-Rays. Audio and video will drift out of sync, because eac3to can't fix the audio overlaps with THD tracks, when demuxing them. You should convert to THD/WAVs directly from the original Blu-Ray structure. Then those "lossless check failed" warnings should go away and audio and video will stay in sync.

1: Extracting directly the THD track from the bluray iso, eac3to gives no error (the lossless check failed appears only in transcoding...)
2: Tha audio of the THD is in sincro with the movie, I've checked evry 4-5 minutes, and dialogues as well as other sounds are in sincro...

3: I tried extracting the tracks in wavs format directly from bluray... no issue this time :rolleyes:

Abradoks

19th May 2010, 14:59

Hmmmm... Are you really sure about that? Do you still have files with which I can reproduce that? The files in your original post are no longer available for download.
Here (http://rghost.net/1659797) they are.
SNR:
out.10N.nero.L.wav 27.386 dB
out.10N.libav.L.wav 27.411 dB
out.10S.nero.L.wav 5.5173 dB
out.10S.libav.L.wav 27.398 dB
tartak (http://forum.doom9.org/member.php?u=169863) wrote on another forum, that with nero decoder DRC is only partially ignored: volume increasing still applies, while attenuation is ignored.

Maybe you're confused by dithering? eac3to lets libav decode to floating point. For WAV/WAVs output eac3to dithers the floating decoding data down to integer. Dithering means adding random noise to the data. So it's clear that if you decode the same AC3/DTS track twice with eac3to/libav, the results will be different each time. If you want floating point WAV/WAVs files instead, use the eac3to "-full" switch. Then the files should be identical every time.
Shame on me. That was dithering indeed. With -full everything works fine.

madshi

19th May 2010, 15:42

2: Tha audio of the THD is in sincro with the movie, I've checked evry 4-5 minutes, and dialogues as well as other sounds are in sincro...
The audio might seem to be in sync, but it can't be *perfectly* in sync. It's probably only a few milliseconds off, which you may not notice.

3: So, you recommend me to directly extract the audio track in wav format in order to avoid warnings (e.g. eac3to "G:" 1) 6: audiotrack.wavs ) ???
The warnings are not a problem. The audio sync is, unless you don't care about perfection.

Here (http://rghost.net/1659797) they are.
Thanks. That's pretty interesting. I've double checked: For some of my movie AC3 files the eac3to "Nero DRC fix" worked, but for some others it didn't. Your AC3 file falls into the "didn't work" category. Don't know why some worked and some not.

Anyway, have already found a fix for the problem, which will be included in the next build.

Thunderbolt8

19th May 2010, 16:36

^^is this problem with nero & ac3 track of importance so that we have to redo our ac3 tracks? I guess not, because libav has been default since quite a while, right?

Laurent

19th May 2010, 17:50

Does eac3to detect the subtitles in the track listing ("eac3to c:\rippedHDDvd")?

No. Only video and audio tracks. That's why I don't understand where are the subtitles if they are not in a specific track.
Or could it be a bug in the subtitle track detection ?

F:\Video>"c:\program files\eac3to\eac3to.exe" .
1) L0_mainMovie.EVO+L1_mainMovie.EVO, 1:51:18
"mainMovie"
- VC-1, 1080i (16:9)
- DTS-HD, English, 2.0, 48khz
- DTS-HD, French, 2.0, 48khz

2) trailer.EVO, 0:03:52
"trailerIntro"
- VC-1, 1080i (16:9)
- DTS-HD, French, 2.0, 48khz

3) specialFeatures.EVO, 0:00:30
"specialFeatures"
- VC-1, 1080i (16:9)
- LPCM, English, 2.0, 16 bits, 48khz

F:\Video>"c:\program files\eac3to\eac3to.exe" . 1)
EVO, 1 video track, 2 audio tracks, 1:51:25
1: Joined EVO file
2: Chapters, 12 chapters
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: DTS Master Audio, English, 2.0 channels, 16 bits, 48khz
(core: DTS, 2.0 channels, 16 bits, 768kbps, 48khz)
5: DTS, French, 2.0 channels, 16 bits, 1509kbps, 48khz

Is seamless branching a feature available for HD-DVD ? That would be very strange, but maybe seamless branching is used between video without subtitles and video with subtitles encoded in picture ?

Thunderbolt8

19th May 2010, 19:20

how does the -shutdown command work? I used it with a command line like "eac3to moviefolder 1) 2: movie.mkv 3: audio.flac 3: audio.dtsma 4: subtitles.sup -shutdown"

but my computer didn't shut down afterwards and just idled.

Snowknight26

19th May 2010, 19:22

Does it not work yet? Will add that on my to do list...

Doesn't work when there are subtitle streams (SRT for example) present. eac3to spits back 'This subtitle conversion is not supported.'

madshi

20th May 2010, 13:05

^^is this problem with nero & ac3 track of importance so that we have to redo our ac3 tracks? I guess not, because libav has been default since quite a while, right?
You won't have to redo libav decoded tracks, and also not all Nero decoded tracks. But maybe some Nero decoded tracks. Don't know which ones. Some seem to be affected, others not.

No. Only video and audio tracks. That's why I don't understand where are the subtitles if they are not in a specific track.
My best guess is that the HD DVD information records are not correct for that movie. The subtitle information may still be there, but eac3to just misses it, because it's not listed anywhere. The HD DVD menu might still be able to access it.

how does the -shutdown command work? I used it with a command line like "eac3to moviefolder 1) 2: movie.mkv 3: audio.flac 3: audio.dtsma 4: subtitles.sup -shutdown"

but my computer didn't shut down afterwards and just idled.
It should work, but maybe it needs admin rights!? Don't know for sure...

Doesn't work when there are subtitle streams (SRT for example) present. eac3to spits back 'This subtitle conversion is not supported.'
Ah, ok.

Laurent

20th May 2010, 22:29

My best guess is that the HD DVD information records are not correct for that movie. The subtitle information may still be there, but eac3to just misses it, because it's not listed anywhere. The HD DVD menu might still be able to access it.

Note that, like eac3to, MediaInfo or TsMuxer do not detect any subtitle track in the EVO file.
In the HD-DVD menu, there is only an audio menu (no subtitle choice) and the switch of audio does make appear or disappear the french subtitles. That's a kind of authoring where subtitles are forced when enabling the original english audio track. That's something we have sometimes on DVD. I don't know how it is handled on HD-DVD.

Thunderbolt8

21st May 2010, 21:09

It should work, but maybe it needs admin rights!? Don't know for sure...my user is listed with admin rights so it should work. can anyone else please try if or how the shutdown mode works?

kurt

21st May 2010, 21:23

^^

same here. shutdown doesn't do anything on win7 prof (with admin rights).

setarip_old

21st May 2010, 22:23

@madshi

Regarding "shutdown" - You might want to check with "jdobbs" (at his Doom9 sub-forum), the author of BD Rebuilder. Base3d on postings in the BD-Rebuilder thread, he has very recently resolved the identical problem...

deathlord

23rd May 2010, 18:16

Thunderbolt8

23rd May 2010, 23:14

update to the latest version, then try again.

deathlord

24th May 2010, 09:32

update to the latest version, then try again.

Yes, of course I will do that. But since the problem can not be systematically reproduced, it may take a long time for the problem to reappear.
The changelog for 3.18 does not list any bugfixes, so I would expect it to behave the same way in this respect.

Meanwhile, I have an idea what could be causing my problem. I remember running two instances of eac3to at the same time. Is this considered a bad idea? Because I have done this quite often without any trouble.
I will try to reproduce the problem this way.

GZZ

24th May 2010, 10:03

I have only had problem using two instance of eac3to when encoding to DTS using surcode, sometimes (rare cases) it happens that both instance use surcode at the same time and one instance finish before the other. Then it (sometimes) kills both process leaving a unfinished dts file. Maybe its the same that happens for you ?

You could properly check this by making a batch file with the same commandline repeated like 5-10 times and then just make it run over and over again to see if it fails at any point. If this work, then try to make 2 batch and run it at the same time. Dont use the same destination folder in both batch job, it will sure fuck it up. :)

Xorp

24th May 2010, 14:11

I'm interested in slowing down some 24.000fps Blu-rays to 23.976fps, since it's the preferred standard for my 60hz sets. Most of them are originally 16-bit though, and since slowing them down with eac3to increases their bitdepth to 24-bit, that takes some of them over the 25GB limit for SL disc burning (after converted back to DTS-MA with the DTS encoder). I'm not gonna waste a BD50 to fit on an extra 800MB or so. I know I can reduce them back down to 16-bit with the -down16 option but I'm worried about the effect all the conversions would have on quality. 16->64->24->16 is a lot of steps. Should I just live with 24fps or will there not be any noticeable sound quality difference after all the conversions?

deathlord

24th May 2010, 14:24

madshi

24th May 2010, 15:13

same here. shutdown doesn't do anything on win7 prof (with admin rights).
Ok, I'll see if I can reproduce it.

I have just encoutered a nasty problem using eac3to 3.17.
When converting 5.1 48kHz dts files to 5.1 flac, eac3to just stopped without error, leaving an unfinished file and no log. It happen only with one file out of ten.
Even worse, rerunning the identical command line once more, the problem does not appear anymore.
This has happend twice in the last couple of days.
Could be a bug in eac3to. But since nobody else has reported a similar problem yet, I'd say there's a good chance that maybe your PC is unstable in some way? Don't know...

Meanwhile, I have an idea what could be causing my problem. I remember running two instances of eac3to at the same time. Is this considered a bad idea?
I don't think so. Except maybe for Surcode encoding.

I'm interested in slowing down some 24.000fps Blu-rays to 23.976fps, since it's the preferred standard for my 60hz sets. Most of them are originally 16-bit though, and since slowing them down with eac3to increases their bitdepth to 24-bit, that takes some of them over the 25GB limit for SL disc burning (after converted back to DTS-MA with the DTS encoder). I'm not gonna waste a BD50 to fit on an extra 800MB or so. I know I can reduce them back down to 16-bit with the -down16 option but I'm worried about the effect all the conversions would have on quality. 16->64->24->16 is a lot of steps. Should I just live with 24fps or will there not be any noticeable sound quality difference after all the conversions?
The biggest potential sound quality drop will come from resampling the audio. It's hard to say how big the difference will be, though. You can try comparing the 24.000 original to the 23.976 conversion with good headphones. You might hear a small change in the audio pitch (much much less than the PAL Micky Mouse effect, and in the opposite direction), but apart from that quality shouldn't suffer too much. eac3to does not usually do 16->64->24->16. Instead it directly does 16->64->16, if you specify "-down16". Of course if you slowdown first and save that as 24bit FLAC and then downsample to 16bit in a separate step you do have 16->64->24->16. But let me say that no human being is able to hear any quality loss caused from reducing 64->24. The experts all agree on that. Some experts even say that 24->16 does not cause any audible loss in quality, as long as proper dithering is applied (which eac3to does), but not everybody agrees here. If the 24->16 conversion does have a negative impact, it can only be a very slightly higher noise floor. If you want to go safe, you can use any bitdepth value between 24 and 16 bit. Every bit you lose saves space, when using lossless compression. E.g. use "-down18" to dither down to 18bit. That way the noise floor will be 4x lower than when using 16bit. But as I said before, the biggest quality issue will be the resampling itself.

BTW, personally I'm also modding all 24.000 Blu-Ray movies to 23.976. However, for such movies I store the original audio track, too, just to be safe...

rack04

24th May 2010, 17:24

Will eac3to be updated to support muxing with x264's new --fake-interlaced?

Add "Fake interlaced" option

This encodes all frames progressively yet flags the stream as interlaced.

This makes it possible to encode valid 25p and 30p Blu-Ray streams.

Also put the pulldown help section in a more appropriate place.

madshi

24th May 2010, 17:26

Will eac3to be updated to support muxing with x264's new --fake-interlaced?
Does it not work right now?

rack04

24th May 2010, 18:03

Does it not work right now?

Actually I think I jumped the gun. eac3to detects the source at 1080i60 and adds a fps value of 30000/1001 to the mkv header. So yes it works correctly. My apologies.

eac3to v3.18
command line: eac3to C:\Personal\Videos\hd_dts_sfx_long.264 C:\Personal\Videos\hd_dts_sfx_long.mkv
------------------------------------------------------------------------------
h264/AVC, 1080i60 /1.001 (16:9)
Muxing video to Matroska...
Added fps value to MKV header.
Video track contains 573 frames.
eac3to processing took 1 second.
Done.

TinTime

25th May 2010, 16:55

madshi

I've found a peculiarity when reducing the bit depth of a file. I've got a sample here to reproduce the problem (if it is a problem).

http://www.mediafire.com/file/yynhfkqgzm2/test1 24.flac

If I try to reduce the bit depth from 24 to, say, 20 bits I get the following log:

eac3to v3.18
command line: "D:\Vtemp\programs\eac3to\eac3to.exe" "E:\Video\test\eac3to test\test1 24.flac" "E:\Video\test\eac3to test\test1 20.flac" -down20
------------------------------------------------------------------------------
FLAC, 5.1 channels, 0:00:18, 24 bits, 4786kbps, 48khz
Decoding FLAC...
Reducing depth from 24 to 20 bits...
Encoding FLAC with libFlac...
Creating file "E:\Video\test\eac3to test\test1 20.flac"...
The original audio track has a constant bit depth of 24 bits.
Processed audio track, L+LFE: constant bit depth of 20 bits.
Processed audio track, R+C+SL+SR: max 24 bits, average 20 bits.
eac3to processing took 3 seconds.
Done.

Why would there still be 24 bit samples or is the eac3to log in error?

Thank you!

TinTime

25th May 2010, 17:40

Sorry, looks like mediafire is down.

Here's another link:

http://www.tintime.talktalk.net/test1 24.flac

nibus

27th May 2010, 12:27

Hi guys, I'm trying to encode a 6.1 DTS to a 5.1 AAC track with eac3to (using NeroAacEnc.exe for the AAC conversion).

I've encoded the DTS straight to a wav:

eac3to input.dts output.wav -libav

(I can't get Arcsoft to work correctly so I figure I'll just use 5.1 with libav)

But this wav is not playable in any player, and when I try to encode it with NeroAacEnc.exe it gives me this error:

ERROR: Invalid Channel Mask

In the logs, eac3to says it is extracting the core from the DTS, so shouldn't that bring it down to a 5.1 track that Nero would recognize? How do I convert a 6.1 DTS track to a 5.1 Wav that Nero will recognize and encode properly?

I'm just looking for as close of a lossless > 5.1 AAC solution as possible, and I would prefer to use the newest version of the Nero encoder. I am hesitant to encode from a low bitrate of 448kbps AC3 and would prefer to encode straight from the DTS source file.

thanks

TinTime

27th May 2010, 12:56

Just allow eac3to to handle it all. Make sure that NeroAacEnc.exe is in the same directory as eac3to.

eac3to input.dts output.m4a -quality=value between 0.00 and 1.00

nibus

27th May 2010, 20:06

Just allow eac3to to handle it all. Make sure that NeroAacEnc.exe is in the same directory as eac3to.

eac3to input.dts output.m4a -quality=value between 0.00 and 1.00

Oh cool - I must have misunderstood the documentation. I thought it did not support the new version of the encoder. Thanks for your help.

edit: I do have a couple more questions -

1) Are most of you going with the eac3to default and removing Dynamic Range Compression? I assume this would be the most accurate reproduction of the original audio track, but in all honesty, it makes it hard to watch some movies because of the extreme difference in sound.
2) When listening to the final encoded AAC file (from 6.1 DTS) it is much quieter than when I encode to an AC3 file. I assume this is because of DirectShow filters. I wasn't able to find any volume settings in the ffdshow config to adjust volume for AAC audio. Does anyone know how to remedy this?

nurbs

28th May 2010, 14:16

1) Depending on the hardware and software you use for playback DRC can be done on the fly, so it's not that big of an issue.

b66pak

28th May 2010, 18:50

2) When listening to the final encoded AAC file (from 6.1 DTS) it is much quieter than when I encode to an AC3 file. I assume this is because of DirectShow filters. I wasn't able to find any volume settings in the ffdshow config to adjust volume for AAC audio. Does anyone know how to remedy this?

do:

eac3to input.dts output.m4a -quality=value between 0.00 and 1.00 -normalize
_

nibus

28th May 2010, 20:26

do:

eac3to input.dts output.m4a -quality=value between 0.00 and 1.00 -normalize
_

Thanks b66 :)

iSeries

28th May 2010, 21:09

Hi,

A question about the -down16 command.

In the past I have been saving my blu ray collection as MKV's with FLAC audio tracks, preserving the original bit depth. I'm now running a bit short on space and using the -down16 command gives me some real space savings. I remember that I've done some backups and eac3to says something along the lines of a 24 bit audio track really being 16 bit, and that the padding will be removed in a 2nd pass.

Now the question! If I just put in the -down16 command will eac3to behave in the same way when it comes to 24 bit padded audio tracks and just remove the padding in a 2nd pass? Or does it blindly dither down to 16 bit without checking?

nibus

30th May 2010, 12:57

I have one more quick question -

In your opinion, is buying the Arcsoft suite worth the money for the quality difference between it and regular LibAV (when encoding from DTS)? I've read that the newer versions of LibAV are just as good but I would like to hear some more input on the matter.

And no I'm not asking which is best, just what you prefer. ;)

tebasuna51

30th May 2010, 13:54

Libav don't decode DTS-HD.
BTW, you can use trial versions of ArcSoft to do the job. See this post http://forum.doom9.org/showthread.php?p=1398577#post1398577

nibus

31st May 2010, 09:38

Libav don't decode DTS-HD.
BTW, you can use trial versions of ArcSoft to do the job. See this post http://forum.doom9.org/showthread.php?p=1398577#post1398577

Thanks, I've now installed it and successfully encoded two 5.1 DTS tracks. However I have tried a 6.1 track and the encoded file plays back too slow and extremely distorted, like it's coming out of a clock radio speaker under water....

No errors or indications during the encoding process. I've tried encoding it in Windows 7 and XP, both with the same result. Is the Arcsoft decoder unable to to decode some 6.1 tracks? Nero and LibAV handle it fine but without the full information like you stated.

tebasuna51

31st May 2010, 11:44

However I have tried a 6.1 track and the encoded file plays back too slow and extremely distorted, like it's coming out of a clock radio speaker under water...
Please put the full log file.

nibus

31st May 2010, 12:42

(Long encoding time because I'm running another encoding project at 90% cpu)

eac3to v3.18
command line: eac3to.exe F:\Encode\BluRay\X3\BDMV\STREAM\input.dts F:\Encode\BluRay\X3\BDMV\STREAM\output.m4a -quality=0.40
------------------------------------------------------------------------------
DTS Master Audio, 6.1 channels, 24 bits, 48khz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48khz)
NeroAacEnc doesn't support 6.1 encoding. Will double the 7th channel.
Doubling 7th channel...
Decoding with ArcSoft DTS Decoder...
Remapping channels...
Encoding AAC <0.40> with NeroAacEnc...
The original audio track has a constant bit depth of 24 bits.
The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 29 minutes, 42 seconds.
Done.

thanks for your help!

tebasuna51

31st May 2010, 14:48

I can't reproduce your problem, works fine for me.
Maybe have you more than one encode project at same time?
Seems don't work always.

Majin3

31st May 2010, 15:28

Hello, since this is my first time dealing with DTS, I wanted to ask some questions here before doing anything wrong...
eac3to reports that my Blu-ray has "LPCM 16 bit 5.1", "DTS Master Audio 16 bit 5.1" and "LPCM 16 bit 2.0" audio tracks.
I want to extract the best quality one, but I'm a bit confused since eac3to reports that the DTS MA track is 16bit (which should decompressed be 1:1 to the first LPCM one, shouldn't it?) while its DTS core is 24bit (MediaInfo says the same).
Can that be correct? Shouldn't the DTS MA also be 24 bit or the DTS core 16 bit?

nurbs

31st May 2010, 17:04

DTS tacks don't really have a fixed bitdepth, lossy audio codecs don't in general.

Majin3

31st May 2010, 17:17

Ah, I see.
So the core is of varibale bit depth and the "Master Audio" extension patches it to 16 bit? Then I guess I'll just go with the LPCM track, thanks.

madshi

31st May 2010, 21:56

Here (http://rghost.net/1659797) they are.
SNR:
out.10N.nero.L.wav 27.386 dB
out.10N.libav.L.wav 27.411 dB
out.10S.nero.L.wav 5.5173 dB
out.10S.libav.L.wav 27.398 dB
tartak (http://forum.doom9.org/member.php?u=169863) wrote on another forum, that with nero decoder DRC is only partially ignored: volume increasing still applies, while attenuation is ignored.
Argh, what I said earlier was wrong, I haven't been able to fix yet, unfortunately... :(

I've found a peculiarity when reducing the bit depth of a file. I've got a sample here to reproduce the problem (if it is a problem).
I'd say this is probably only a cosmetic issue, but I'll look into it.

Now the question! If I just put in the -down16 command will eac3to behave in the same way when it comes to 24 bit padded audio tracks and just remove the padding in a 2nd pass? Or does it blindly dither down to 16 bit without checking?
What is the source you will be using for -down16? Was the source created with eac3to? If so, if it's really only 16bit, then eac3to should already have saved it as 16bit, so if you use -down16, nothing will happen.

If your source claims to be 24bit, and if you use -down16, then eac3to will apply dithering - even if it's only really 16bit. The reason for that is that eac3to doesn't know which bitdepth the source has until processing is complete. You surely don't want eac3to to scan the whole file first, everytime, before starting processing, do you?

madshi

31st May 2010, 22:07

eac3to v3.19 released

http://madshi.net/eac3to.zip

* added support for 3D Blu-Rays (playlists, detection & demuxing)
Just a quick addition of 3D support. For playlist browsing it looks like this:

D:\BDMV\PLAYLIST>eac3to 00001.mpls
1) 00001.mpls, 00001.m2ts, 0:00:09
- h264/AVC (left eye), 1080p24 /1.001 (16:9)
- h264/AVC (right eye), 1080p24 /1.001 (16:9)
- DTS, English, multi-channel, 48kHz
For demuxing it looks like this:

M2TS, 2 video tracks, 1 audio track, 0:00:09, 24p /1.001
1: h264/AVC (left eye), 1080p24 /1.001 (16:9)
2: h264/AVC (right eye), 1080p24 /1.001 (16:9)
3: DTS, English, 5.1 channels, 16 bits, 1509kbps, 48kHz
a03 Extracting audio track number 3...
v01 Extracting video track number 1...
v01 Extracting video track number 1...
a03 Patching bitdepth to 24 bits...
v01 Creating file "c:\desktop\test - 2 - h264 (right eye), 1080p24.h264"...
v01 Creating file "c:\desktop\test - 1 - h264 (left eye), 1080p24.h264"...
a03 Creating file "c:\desktop\test - 3 - DTS, English, 5.1 channels, 16 bits, 15
09kbps, 48kHz.dts"...
Video track 1 contains 211 frames.
Video track 2 contains 211 frames.
eac3to processing took 2 seconds.
Done.
Enjoy!

(Before anybody asks: The "left eye" stream is a normal h264 stream, which every h264 decoder supports. The "right eye" stream is a special h264 extension stream which current h264 PC decoders don't understand.)

iSeries

31st May 2010, 22:55

"If your source claims to be 24bit, and if you use -down16, then eac3to will apply dithering - even if it's only really 16bit."

So, I'll get better quality on padded sources if I first demux/convert to .wav? That way eac3to can work out if its padded or true 24bit. If its padded it'll do a second pass and remove the padding and if its 24bit it will stay as 24bit and I can then run it through eac3to again with -down16? (yeah more time consuming but its the computer doing the work, not me :))

~Revolution~

31st May 2010, 23:18

I don't get the point of padding 16bit to 24bit in the first place. What does it exactly do?

hubblec4

31st May 2010, 23:19

eac3to v3.19 released

http://madshi.net/eac3to.zip

* added support for 3D Blu-Rays (playlists, detection & demuxing)

(Before anybody asks: The "left eye" stream is a normal h264 stream, which every h264 decoder supports. The "right eye" stream is a special h264 extension stream which current h264 PC decoders don't understand.)

thats wonderful news. I hope this issue with the "right eye" stream will be fixed at someone next time.

thanks for your working on eac3to.

iSeries

31st May 2010, 23:54

@~Revolution~ - Doesnt make sense to me either!

Xorp

1st June 2010, 04:03

I don't get the point of padding 16bit to 24bit in the first place. What does it exactly do?

http://forum.doom9.org/showpost.php?p=1386209&postcount=9868

If eac3to kept it flagged as 16-bit like the source, some decoders might only decode in 16-bit, so madshi smartly made eac3to flag all DTS as 24-bit just in case :)

madshi

1st June 2010, 06:53

So, I'll get better quality on padded sources if I first demux/convert to .wav? That way eac3to can work out if its padded or true 24bit. If its padded it'll do a second pass and remove the padding and if its 24bit it will stay as 24bit and I can then run it through eac3to again with -down16?
Yeah, for max quality that would be the best approach.

mrr19121970

1st June 2010, 07:14

Would it be much work to automatically create a second file containing only the forced captions while extracting this stream?

I guess it would be possible, but I don't know how difficult it would be. I'll put it on my to do list.

Whilst eac3to is back in focus, I'm wondering if you'd had time to think about extracting forced subs into an extra file?

madshi

1st June 2010, 07:22

Whilst eac3to is back in focus, I'm wondering if you'd had time to think about extracting forced subs into an extra file?
It is on my to do list - like so many other things I'd like to add to eac3to. I'm currently lacking time, though. And the time I can find for HTPC development mostly go into madVR at this point, since madVR is still a work in progress, while eac3to is pretty stable. There probably will come a time when I come back to eac3to with more development resources, but it's not here yet right now. (3D was important and easy to add.)

dansrfe

1st June 2010, 08:18

So are there any decoders which decode non 24-bit padded 16-bit DTS properly and account for the variations in frequency domain representation when the PCM audio is recreated during decoding? Does ffdshow audio decoder properly do that with the 24-bit padding? Also I've always been curious to know why are there different outputs for the different bit integers. I have checked only 32 bit floating point because I assumed it was the "best" option and gives the decoder lots of "room" for whatever it needs to do though it might unnecessarily raise cpu usage since the decoding actually only needs 24-bit for 16-bit Digital Dolby or DTS audio.

madshi

1st June 2010, 08:28

So are there any decoders which decode non 24-bit padded 16-bit DTS properly and account for the variations in frequency domain representation when the PCM audio is recreated during decoding?
Huh? You're confusing me.

Standard DTS is never padded 16bit. It's floating point.

DTS-MA is usually also never padded 16bit. It's either 16bit, and then the decoder outputs 16bit. Or it's 24bit and then the decoder outputs 24bit.

The only situation where 16bit -> 24bit padding usually occurs is with TrueHD/MLP tracks. I've seen tons of DTS-MA tracks and their bitdepth information is usually correct. I've only ever seen one single DTS-MA track which was flagged as 24bit, but only contained 16bit data, while I've seen hundreds of DTS-MA tracks which were flagged correctly.

dansrfe

1st June 2010, 08:52

Sorry I should have been more clear in my wording. What I meant to say was that is there any decoder which will correctly decode DTS audio encoded from 16-bit and which has not been flagged as 24-bit?

madshi

1st June 2010, 10:01

That still doesn't make any sense. Are you talking about standard DTS or DTS-MA?

dansrfe

1st June 2010, 10:23

Standard DTS. The link in some posts above said that when standard dts is converted to PCM via decoding it is 8 bits more therefore some decoders need the 24-bit padding in order to have all the bits that's why I was wondering if there are any decoders that can play without the padding and still anticipate 8 bits more when decoding.

madshi

1st June 2010, 10:25

I don't know for sure which decoders are outputting what, if eac3to doesn't patch the bitdepth to 24bit. ArcSoft outputs 16bit without the patching, I think. To which bitdepth libav/ffmpeg decodes, depends on the libav/ffmpeg version and the decoder/media player which uses it. If you decode by using eac3to, you will always get full bitdepth.

TinTime

1st June 2010, 12:25

madshi

A bit more info - I've been looking further into the problem I reported with reducing the bitdepth of 24 bit audio.

It seems to be near peak values that don't necessarily have their depth reduced. Presumably dithering means that a value can be rounded up or down. However near the peak I suppose the value should always be rounded down - ie truncated.

For example the centre channel in the test file I posted contains the following sequence of samples (as wav):

24 bit in 20 bit out
EC 65 77 F0 65 77
FC FF 7F FF FF 7F
FE FF 7F FF FF 7F
FD FF 7F F0 FF 7F
51 E8 5C 50 E8 5C

I reran my test file through eac3to using...
eac3to.exe "test1 24.flac" "test1 20.flac" -1dB -down20
...and it worked correctly. In fact perhaps this is the correct approach. The bit depth reduction in this case would introduce clipping otherwise.

Nando1970

1st June 2010, 13:43

It is on my to do list - like so many other things I'd like to add to eac3to. I'm currently lacking time, though.

madshi would you consider to open-source eac3to?
There might be people out there willing to contribute development time.

Thank you anyway for your great tool

ciao -- Nando

madshi

2nd June 2010, 11:56

A bit more info - I've been looking further into the problem I reported with reducing the bitdepth of 24 bit audio.
Thanks, that's helpful!

madshi would you consider to open-source eac3to?
No.

madshi

2nd June 2010, 11:58

eac3to v3.20 released

http://madshi.net/eac3to.zip

* fixed 3D demuxing (argh)
Stupid bug, was demuxing the left eye stream for both left and right eye.

Thunderbolt8

3rd June 2010, 02:18

regarding 3d remuxing, wouldnt it be better to have both streams output in seperate .mkv files instead of the raw .h264 streams? i remember you said its better to remux to .mkv instead of demuxing the stream, because then some parameters or other stuff in the chain could get lost or something like that.

Frank K Abbott

3rd June 2010, 04:07

I'm curious to know, are there even any 3D Blu Ray's in the market?

setarip_old

3rd June 2010, 06:08

@Frank K Abbott

Apparently so:

http://forum.doom9.org/showthread.php?p=1394311#post1394311

madshi

3rd June 2010, 07:00

regarding 3d remuxing, wouldnt it be better to have both streams output in seperate .mkv files instead of the raw .h264 streams?
Haali's Matroska Muxer doesn't seem to like the right eye stream, so that's not possible.

i remember you said its better to remux to .mkv instead of demuxing the stream, because then some parameters or other stuff in the chain could get lost or something like that.
Well, muxing to a container is better only if there are overlaps or gaps in the video stream. I've never seen that with Blu-Ray yet. It might happen for broadcasts, especially when there are recording glitches, though.

volence

3rd June 2010, 10:58

I'm sorry if there had been similar question before, since the thread is too long for me to go through.
I want to know can eac3to convert 6.1 and 7.1 ch TrueHD audio to aac? I think I'm using the last free version of Nero AAC, and it tells me Nero doesn't support 6.1ch or higher. Is there any way to encode 6.1 AAC with eac3to (using freeware)?

ramicio

3rd June 2010, 22:26

would a numerical progress indicator be feasible? i have my windows larger with a different font than default so i have no idea how far along the file is.

Momber

3rd June 2010, 23:32

It's prollly just cosmetic, but why does eac3to insist there is a newer version of Haali available, when there really isn't?

http://pickhost.eu/images/0003/6578/eac3to.gif

Snowknight26

4th June 2010, 00:02

Because that's how the website (http://haali.su/mkv/) shows it.

Momber

4th June 2010, 05:41

But that's where I have it from....

Snowknight26

4th June 2010, 05:51

So what are you confused about? File date say one thing, website says another.

stasi

4th June 2010, 16:18

tormento

4th June 2010, 17:58

There is a nice Flac CUDA compression implementation here:

http://cuetools.net/doku.php/flacuda

and source too.

Do you think could be feasible to implement it in the next version? ;)

komisar

4th June 2010, 18:31

FlaCuda:
"Bits per sample must be 16."
ChannelCount must be 2.

nibus

4th June 2010, 23:40

If you have more than 6 channels, you have to encode to FLAC. Nero AAC is limited to 6 channels.

I thought AAC supported 7.1 as long as you use the Arcsoft decoder. It doesn't support 6.1 so it doubles the last track.

Snowknight26

5th June 2010, 02:10

AAC has nothing to do with ArcSoft's decoder.

Laurent

5th June 2010, 10:10

Hi

When handling HD-DVD, I discovered that the chapter timestamps are wrong. On the first chapters, the timing is ok or almost ok, and then the difference grows up with the next chapters, leading to several seconds on the last chapters. The chapter jump leads to a position before the real position.
Could it be a bug in eac3to ?
Could it be something relative to pulldown removal ? (29.976 to 23.976)
Is there a solution to correct that ?

PS: I have to mention that I notice this problem after producing AVCHD discs with multiAVCHD, so the problem could be either with eac3to or with multiAVCHD. But as I already noticed the same thing with MKV production, I think it could be more in relation with eac3to actions.

TinTime

5th June 2010, 11:44

Laurent

5th June 2010, 12:19

Do the timestamps in the chapter text file produced by eac3to match the timestamps in the HD DVD xml file? If so then the problem isn't with eac3to.

I can't verify it immediately but I will do it later. I was not aware of this XML file.

The question is: should the pulldown removal (done by eac3to) have an impact on chapter timestamps ? I believe it sould not ?

I hade done a fast computation, I don't think the problem could be a mistake between 29.970 and 23.976, but it could be between 23.976 and 24.000. Such mistake leads to 7 seconds after 2 hours.

Laurent

5th June 2010, 15:48

Do the timestamps in the chapter text file produced by eac3to match the timestamps in the HD DVD xml file? If so then the problem isn't with eac3to.

Which XML file are you talking about ? I just loaded a HD-DVD and in the HVDVD_TS directory, the only file extensions are BUP, EVO, MAP and VTI.

TinTime

5th June 2010, 17:00

I forget what they're called or where they are (sorry) but have a look in other directories on the disk. When you look at it there will be all kinds of track info, including chapters.

Laurent

5th June 2010, 17:04

How are extracted chapter timestamps from HD-DVD discs ? I mean, are they computed or are they just read somewhere on the disc ?

I checked with a Blu-Ray disc and there is no problem with the chapter timestamps.
They are wrong only when handling HD-DVD discs.

I would be curious to see if it changes something if I disable "pulldown removal". Is there an eac3to option to disable pulldown removal when video is interlaced at 29.97 Hz ?

Laurent

5th June 2010, 17:20

I forget what they're called or where they are (sorry) but have a look in other directories on the disk. When you look at it there will be all kinds of track info, including chapters.

Ok, I found the file, its name is VPLST000.XPL. Here is an extract of the file:
<ChapterList>
<Chapter id="mainMovie_ch1" titleTimeBegin="00:00:00:00" />
<Chapter id="mainMovie_ch2" titleTimeBegin="00:05:21:25" />
<Chapter id="mainMovie_ch3" titleTimeBegin="00:12:36:02" />
<Chapter id="mainMovie_ch4" titleTimeBegin="00:22:08:15" />
<Chapter id="mainMovie_ch5" titleTimeBegin="00:32:29:25" />
<Chapter id="mainMovie_ch6" titleTimeBegin="00:38:52:32" />
<Chapter id="mainMovie_ch7" titleTimeBegin="00:44:50:17" />
<Chapter id="mainMovie_ch8" titleTimeBegin="00:52:58:25" />
<Chapter id="mainMovie_ch9" titleTimeBegin="01:02:30:57" />
<Chapter id="mainMovie_ch10" titleTimeBegin="01:15:02:02" />
<Chapter id="mainMovie_ch11" titleTimeBegin="01:24:36:07" />
<Chapter id="mainMovie_ch12" titleTimeBegin="01:36:12:12" />
<Chapter id="mainMovie_ch13" titleTimeBegin="01:41:26:15" />
<Chapter id="mainMovie_ch14" titleTimeBegin="01:49:51:27" />
<Chapter id="mainMovie_ch15" titleTimeBegin="02:01:21:02" />
<Chapter id="mainMovie_ch16" titleTimeBegin="02:09:47:07" />
<Chapter id="mainMovie_ch17" titleTimeBegin="02:19:03:35" />
<Chapter id="mainMovie_ch18" titleTimeBegin="02:24:09:35" />
<Chapter id="mainMovie_ch19" titleTimeBegin="02:34:18:42" />
<Chapter id="mainMovie_ch20" titleTimeBegin="02:39:13:05" />
</ChapterList>
And here is the file Chapters_2.txt produced by eac3to:
CHAPTER01=00:00:00.000
CHAPTER01NAME=
CHAPTER02=00:05:21.416
CHAPTER02NAME=
CHAPTER03=00:12:36.033
CHAPTER03NAME=
CHAPTER04=00:22:08.250
CHAPTER04NAME=
CHAPTER05=00:32:29.416
CHAPTER05NAME=
CHAPTER06=00:38:52.533
CHAPTER06NAME=
CHAPTER07=00:44:50.283
CHAPTER07NAME=
CHAPTER08=00:52:58.416
CHAPTER08NAME=
CHAPTER09=01:02:30.950
CHAPTER09NAME=
CHAPTER10=01:15:02.033
CHAPTER10NAME=
CHAPTER11=01:24:36.116
CHAPTER11NAME=
CHAPTER12=01:36:12.200
CHAPTER12NAME=
CHAPTER13=01:41:26.250
CHAPTER13NAME=
CHAPTER14=01:49:51.450
CHAPTER14NAME=
CHAPTER15=02:01:21.033
CHAPTER15NAME=
CHAPTER16=02:09:47.116
CHAPTER16NAME=
CHAPTER17=02:19:03.583
CHAPTER17NAME=
CHAPTER18=02:24:09.583
CHAPTER18NAME=
CHAPTER19=02:34:18.700
CHAPTER19NAME=
CHAPTER20=02:39:13.083
CHAPTER20NAME=
So the timestamps are not eactly the same but the differences are not big.
I will create an AVCHD with this one and check with the PS3 how good are the chapter marks compared to the HD-DVD player.

Laurent

5th June 2010, 17:33

Is there an eac3to option to disable pulldown removal when video is interlaced at 29.97 Hz ?

Ok, found, this iis "-keepPulldown".

SomeJoe

5th June 2010, 22:47

Laurent,

When dealing with NTSC-derived frame rates (29.97, 23.976), the timecodes on HD-DVD and BD are actually non-drop-frame timecodes. But most of the multiplexing programs out there (TSMuxer, MKVMerge) treat them as real time. This leads to the 3.5 seconds per hour offset.

If you convert the timecodes from non-drop-frame to drop-frame, the chapters points will be exact.

This doesn't have anything to do with pulldown removal. The root of the problem is that the OGG chapter format doesn't have any way to specify NDF or DF timecodes, it's up to the program being used to interpret them correctly.

Laurent

5th June 2010, 23:04

Laurent,

When dealing with NTSC-derived frame rates (29.97, 23.976), the timecodes on HD-DVD and BD are actually non-drop-frame timecodes. But most of the multiplexing programs out there (TSMuxer, MKVMerge) treat them as real time. This leads to the 3.5 seconds per hour offset.

If you convert the timecodes from non-drop-frame to drop-frame, the chapters points will be exact.

Thank you for the explanation.
Can you tell me what coefficient should I apply to each timestamp to get the correct value ?

This doesn't have anything to do with pulldown removal. The root of the problem is that the OGG chapter format doesn't have any way to specify NDF or DF timecodes, it's up to the program being used to interpret them correctly.

If pulldown removal is not concerned, I don't really understand what you call drop frames ?

mczuzlak

6th June 2010, 10:47

I know I can encode to ac3 from any format with eac3to but is it possible to encode an ac3 file from multiple wavs ? (each one is a channel, like left and right)

nurbs

6th June 2010, 12:58

No, you need to merge them to a multi channel wav first. Software that can do that has been repeatedly mentioned in this thread, but I can't remember application names, sorry.

Thunderbolt8

6th June 2010, 14:46

wavewizard can combine multiple wav files into a single one (or another format)

Thunderbolt8

6th June 2010, 16:13

someone said that a dts-hd ma --> flac file created with 3.20 was a tiny bit different compared to when it was created with 3.18, can anyone confirm this?

mczuzlak

6th June 2010, 17:16

No, you need to merge them to a multi channel wav first. Software that can do that has been repeatedly mentioned in this thread, but I can't remember application names, sorry.

wavewizard can combine multiple wav files into a single one (or another format)

thx guys

krohm

7th June 2010, 04:16

im sure this has been posted before but the search function didnt yield me much of a result. i want to transcode various dts formats from bluray into a lossless format that will load in powerdirector. ive read the eac3to howto guide but it got a little confusing and only allowed me to be able to convert dts to ac3.

i think afterwards i use surcode or something to create a dts wav that can load to powerdirector... im not sure..

ive been advised i need to convert the dts file into multiple wav's then individually load the wav's into the editor. i need eac3to for this but am new to the program and dont know how to use the command line. need some command line assistance as it is a real sticking point for me atm as cant edit any bluray clips with lossless sound. i dont want to use the ac3 and wanted to know if someone could help me out with the command line to convert the dts to wav's...

djloewen

7th June 2010, 15:09

I've got three things to add to this thread. The first appears to be a bug in the program. The second is apparently a problem that not everyone is having, but certainly I am. The third is a feature request.

1.) The bug:
On the Blu-Ray "Batman: Gotham Knight" there are four bonus episodes of "Batman: The Anmated Series". These are reported by eac3to as "480p24/1.001 (3:2) with pulldown flags", and the demuxed VC1 stream is 23.976fps. This is incorrect, and results in the video being played back at the wrong speed. DGindexNV correctly reports that it is 29.970, no pulldown, no interlacing.

100mb sample here:
http://www.megaupload.com/?d=MKOFDA0W

2.) The problem:
I reported this initially in a separate thread, and in fact it's a little more widespread than I had thought. When I run eac3to on the Blu-Ray disc for "The Bad Lieutenant: Port of Call - New Orleans", it lists 3 titles. So:

eac3to E:\
shows three titles, each with a number of streams.

eac3to E:\ 1)
does nothing. it draws a bunch of "-" characters as it scans the disc, then it returns me to the prompt.

eac3to E:\BDMV\PLAYLIST\00000.mpls
lists one title with a number of streams.

E:\BDMV\PLAYLIST\00000.mpls 1)
again, does nothing.

I am able to open/rip the m2ts files directly, bypassing the playlist files. However, there are two m2ts files in this case and I want eac3to to demux them as one file.

Since first discovering the problem with that one disc, I've encountered the exact same problem with the ~5min short film extras on two separate Pixar discs. In one case it's a single m2ts file (and so I've just demuxed that), for the others it's two or three. But in all cases, the playlist files just don't show the streams. Any help would be greatly appreciated.

3.) The feature request:
I would personally find it very useful to be able to choose not a full stream/playlist, but just a range of chapters, and then demux the audio/video/subs for just that selection.

SomeJoe

7th June 2010, 15:25

Thank you for the explanation.
Can you tell me what coefficient should I apply to each timestamp to get the correct value ?

If pulldown removal is not concerned, I don't really understand what you call drop frames ?

Pulldown flags are indicators within a 23.976 fps video stream that tell the playback device to repeat certain fields in a 3:2 pattern. This causes the playback to be 29.97 fps. Telling eac3to.exe -keepPulldown will cause the video stream to be extracted with the pulldown flags intact. Normally, eac3to removes them. But this doesn't have anything to do with your problem.

Drop-frame (DF) and Non-drop-frame (NDF) timecodes are labels that are attached to each frame of a 29.97 fps video stream. I emphasize that -- they are labels that uniquely identify each frame. Despite the name, no frames are dropped from the video stream, either in the storage media or during playback.

A DF label is of the form hh;mm;ss;ff (Note semicolons).

A NDF label is of the form hh:mm:ss:ff (Note colons).

There is a one-to-one-to-one correspondence between a DF timecode label, an NDF timecode label, and a frame number.

To convert from an NDF timecode label to a DF timecode label (or vice versa), you convert the NDF timecode label to a frame number, and then convert the frame number to a DF timecode label.

The difference between an NDF label and a DF label is as follows:

NDF labels use a very simple counting mechanism. They count 0-29 on frames, 0-59 on seconds, 0-59 on minutes, and 0-any number for hours. There is no compensation for the fact that this counting mechanism assumes that there are exactly 30 frames per second, but there are not. NTSC frame rates do not play back at 30.000, they play back at 29.97. Thus, NDF timecode labels will slowly depart from real wall-clock time.

DF timecode labels use a modified counting mechanism that compensates for the fact that the playback rate is not 30.000, but is instead 29.97. In DF timecodes, frames count 0-29, seconds count 0-59, minutes count 0-59, and hours count 0-any number just like NDF, BUT:

- Frame numbers ;00 and ;01 are skipped on every 1-minute boundary, except for 10-minute boundaries.

Thus, the following:

DF Timecode NDF Timecode Frame #
00;00;59;28 00:00:59:28 1798
00;00;59;29 00:00:59:29 1799
00;01;00;02 00:01:00:00 1800
00;01;00;03 00:01:00:01 1801
00;01;00;04 00:01:00:02 1802

At the 1 hour (wall-clock time) point in the video, the DF timecode will be 1;00;00;00, whereas the NDF timecode will be 00:59:56:13.

The chapter points on the disc are NDF. But the OGG chapter format specifies chapter points in terms of wall-clock time, not timecode. To convert the NDF timecodes on the disc to wall-clock time:

Frame number = (((hh * 60) + mm) * 60 + ss) * 30 + ff

Wall-clock time (seconds) = Frame number / (30000/1001)

Example:

NDF timecode = 01:35:23:17

Since this is about an hour and half into the film, and the offset is approximately 3.5 seconds per hour, we would expect this to convert to a wall-clock time that's about 5.5 seconds later than this timecode, somewhere in the 1:35:29.xxx area.

Frame number = (((1 * 60) + 35) * 60 + 23) * 30 + 17 = 171707.

Wall-clock time = 171707 / (30000/1001) = 5729.290 seconds.

5729.290 Mod 60 = 29.29
5700 Mod 3600 / 60 = 35
3600 / 3600 = 1

Wall-clock time = 01:35:29.290

Note that DF timecodes and wall-clock time are not exactly the same. DF timecodes use frame numbers, wall-clock time uses fractional seconds.

Ideally, eac3to should produce an OGG chapter format file with this conversion already done, since the OGG chapter format file is supposed to use wall-clock time, not an NDF timecode expressed in fractional seconds.

Note that NDF and DF timecodes are only valid for 29.97 fps streams. There is no such thing as a DF or NDF timecode for a 23.976 stream, and I think there is where the problem is coming in. The timecodes on the disc for HD-DVD and BD shouldn't be NDF in the first place, as that isn't valid for a 23.976 stream.

TinTime

7th June 2010, 16:40

I am able to open/rip the m2ts files directly, bypassing the playlist files. However, there are two m2ts files in this case and I want eac3to to demux them as one file.

I can't help with your other problems but you can join files easily enough with eac3to.

eg.
eac3to.exe 00000.m2ts+00001.m2ts+00002.m2ts 1: video.h264 2: audio.ac3 etc.

djloewen

7th June 2010, 20:21

I can't help with your other problems but you can join files easily enough with eac3to.

Is there any downside to using this method as opposed to the playlist? For example, might there end up being audio or subtitle sync problems? Or does the playlist ever specify only using part of a stream?

Atak_Snajpera

7th June 2010, 22:08

I remember that In order to have working PCM audio in AVCHD/BluRay structure we had to first:
1) decompress DTS-MA/TrueHD/LPCM audio to .wav
2) convert decompressed .wav to .w64
3) import .w64 in tsmuxer

Is this still necessary ?

TinTime

7th June 2010, 22:25

Is there any downside to using this method as opposed to the playlist? For example, might there end up being audio or subtitle sync problems? Or does the playlist ever specify only using part of a stream?

You don't get chapters or (I think) the language codes for audio / subs. There shouldn't be any sync problems though.

DoomBot

8th June 2010, 17:18

Spiderman 2 is a seamless branching disc, as far as I remember. If you demux/extract the THD data first and convert it to WAVs or FLAC later, you'll get a "lossless check failed" warning for every seamless branch. That is to be expected. You should have gotten a warning from eac3to, though, when extracting the THD track in the first place. It's not a good idea to extract THD tracks from seamless branching Blu-Rays. Audio and video will drift out of sync, because eac3to can't fix the audio overlaps with THD tracks, when demuxing them. You should convert to THD/WAVs directly from the original Blu-Ray structure. Then those "lossless check failed" warnings should go away and audio and video will stay in sync.

I thought eac3to will let you know if there is audio overlaps with THD tracks even if it can't fix the overlap or is it just a gap it warns you about? Even i did Spider-man 2 keeping the Truehd track and not converting to PCM because i thought there wasn't any overlap since eac3to didn't tell me there was. Can i just convert the truehd from the original blu-ray to pcm or just a WAV and not WAVs? Say if i convert to WAVs should eac3to tell me there was overlaps to fix?

Laurent

8th June 2010, 19:09

Ideally, eac3to should produce an OGG chapter format file with this conversion already done, since the OGG chapter format file is supposed to use wall-clock time, not an NDF timecode expressed in fractional seconds.

SomeJoe, thank you a lot for you full explanation.

If eac3to is producing NDF timecode expressed in fractional seconds, I should convert these values to wall-clock time using this formula:

Wall-clock time (seconds) = NDF timecode (fractional seconds) * 1001 / 1000

I make a try immediately and come back after.

Laurent

8th June 2010, 22:01

Snowknight26

8th June 2010, 22:42

Small bug I've come across every now and then with eac3to and NeroAacEnc:

When converting/(de)muxing multiple tracks from a random source, one instance of NeroAacEnc doesn't seem to exit, making eac3to hang at the last progress bar. Ending that hung instance of NeroAacEnc causes eac3to to flood the command prompt with
a09 [NeroAacEnc] Processed x seconds... until it it gets to
Aborted at file position x. where x is the size of the file (so the end of it).

I've never seen this happen when doing single audio conversions to AAC, so I suspect it's something to do with eac3to not continuing to feed that hung instance of NeroAacEnc even after all other track operations had finished. Or maybe not.. who knows. Redoing the same command usually fixes it, suggesting that it happens randomly.

Atak_Snajpera

8th June 2010, 23:34

@madshi
I did a small test (ArcSoft DTS Decoder installed and working, source is .m2ts)
DTSMA -> .w64
and
DTSMA -> .wav -> .w64

Somehow those two file are not identical
http://img231.imageshack.us/img231/6636/64431280.png

SomeJoe

9th June 2010, 00:55

Result of the test: that is now perfect :cool:
SomeJoe, :thanks: As you said, eac3to should ideally produce the correct values. This discussion might give the idea to madshi to add an option to apply a coef 1001/1000 to chapter timestamps ?
If not, that means a little tool should be written to apply this correction, taking one OGG chapter file as input and producing a corrected OGG chapter file as output. If this tool does not exist, I will develop one myself, that should not be very hard to do.

Yes, I was actually thinking of developing a similar tool myself.

Such tools already exist for subtitles files, since this type of conversion (and many others, like 23.976 NTSC -> 25.000 PAL) are frequently required. But no such tool as yet exists for OGG chapter format files.

TinTime

9th June 2010, 02:00

Yes, I was actually thinking of developing a similar tool myself.

Such tools already exist for subtitles files, since this type of conversion (and many others, like 23.976 NTSC -> 25.000 PAL) are frequently required. But no such tool as yet exists for OGG chapter format files.

I knocked up a python script a while ago to do this if you and Laurent are interested. It's pretty basic but does the job for me.

Frank K Abbott

9th June 2010, 05:01

I hope this isn't the wrong thread but I tried to use eac3to to encode from .wav to .mp3 but looks like it doesn't support that type of conversion? What is the best software that I can use for MP3 encoding? Command line is preferred as long as it is the latest. I went to the LAME website but I can't find the command line encoder of it anywhere :(

Midzuki

9th June 2010, 05:13

I went to the LAME website but I can't find the command line encoder of it anywhere :(

Try this:

http://www.rarewares.org/mp3-lame-bundle.php

HTH.

Frank K Abbott

9th June 2010, 05:15

Wow, thanks Midzuki! This is exactly what I was looking for! :)

Laurent

9th June 2010, 18:04

I knocked up a python script a while ago to do this if you and Laurent are interested. It's pretty basic but does the job for me.

I am using a xcalc sheet to convert values + CSV format (":" as separator). My input and output file is not in OGG chapter format but only the list of values (exactly what is required by multiAVCHD). So this requires few steps to achieve the job.

I have started the update of my AVCHD compilation changing chapters for 3 titles. I will check the result later.

By the way, I am interested by your python script (especially if nothing special is required to run it).

TinTime

9th June 2010, 20:18

I am using a xcalc sheet to convert values + CSV format (":" as separator). My input and output file is not in OGG chapter format but only the list of values (exactly what is required by multiAVCHD). So this requires few steps to achieve the job.

I have started the update of my AVCHD compilation changing chapters for 3 titles. I will check the result later.

By the way, I am interested by your python script (especially if nothing special is required to run it).

Well, you'd need to install Python 3.1.2 (http://www.python.org/download/), but you can just install it and forget it.

Script and instructions are here (www.tintime.talktalk.net/chapters.zip).

Script chapters.py outputs an OGG chapter file. Script chapters2.py is the same but with a couple of lines removed so it just outputs the timecodes (it sounds like that would be more useful for you).

Note - it's pretty rough and ready so if you input something other than a standard OGG chapter file it's going to fail with a not very user friendly error message!

Laurent

9th June 2010, 22:31

Well, you'd need to install Python 3.1.2 (http://www.python.org/download/), but you can just install it and forget it.

Script and instructions are here (www.tintime.talktalk.net/chapters.zip).

Script chapters.py outputs an OGG chapter file. Script chapters2.py is the same but with a couple of lines removed so it just outputs the timecodes (it sounds like that would be more useful for you).

Note - it's pretty rough and ready so if you input something other than a standard OGG chapter file it's going to fail with a not very user friendly error message!

Thank you, I will check that.
Before using your script, I have to run eac3to to generate again the OGG chapter file from my AVCHD compilation. Easy.

Laurent

9th June 2010, 22:55

TinTime, I just tried your two Python scripts. Very usefull and (of course) easier in comparison to my previous method. :thanks:

shon3i

10th June 2010, 07:47

@SomeJoe, Laurent try my ChapterGen tool http://forum.doom9.org/showthread.php?t=150148&highlight=chaptergen, it have OGG Input/output, and conversion between rates. For HDDVD you need to alter from 24->23.976 to get exact chapters.

krohm

10th June 2010, 09:15

@madshi
I did a small test (ArcSoft DTS Decoder installed and working, source is .m2ts)
DTSMA -> .w64
and
DTSMA -> .wav -> .w64

this will do a lossless conversion of dts to wav right...??

i have arcsoft totalmedia extreme and theatre, will this have the decoder in it?

what command line would i use to get a dts audio file to be decoded with eac3to to lossless wavs so i can load it into a video editor...

ramicio

10th June 2010, 18:49

@madshi
I did a small test (ArcSoft DTS Decoder installed and working, source is .m2ts)
DTSMA -> .w64
and
DTSMA -> .wav -> .w64

Somehow those two file are not identical

how long is the audio?

Atak_Snajpera

10th June 2010, 19:32

Both files have identical size
http://img716.imageshack.us/img716/5265/76035020.png

however mediainfo shows something interesting
General
Complete name : C:\temp\audio.w64
Format : Wave64
File size : 5.20 GiB
Duration : 37mn 25s
Overall bit rate : 19.9 Mbps

Audio
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Unsigned
Codec ID : 00001000-0000-0100-8000-00AA00389B71
Codec ID/Hint : Microsoft
Duration : 37mn 25s
Bit rate mode : Constant
Bit rate : 4 608 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Stream size : 1.20 GiB (23%)

General
Complete name : C:\temp\audio_wav_to.w64
Format : Wave64
File size : 5.20 GiB
Duration : 2h 41mn
Overall bit rate : 4 608 Kbps

Audio
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Unsigned
Codec ID : 00001000-0000-0100-8000-00AA00389B71
Codec ID/Hint : Microsoft
Duration : 2h 41mn
Bit rate mode : Constant
Bit rate : 4 608 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Stream size : 5.20 GiB (100%)

tebasuna51

11th June 2010, 02:03

@Atak_Snajpera
I can confirm the eac3to bug when write the w64 header (work fine with wav input).
The fields RiffLength and DataLength are truncated to 32 bits instead the full 64 bits value (4 GB limit).

Tested decoding standard DTS (I don't have big DTS-MA) with ArcSoft or libav

alfixdvd

12th June 2010, 10:15

With movie Van Helsing Arcsoft Total Media Teather show me 37 subtitles, but eac3to only list 32 subtitles, 5 last ones are absent.

How can I extract them ?

Perhaps anything is wrong ?

nibus

13th June 2010, 10:17

What are the advantages of demuxing with eac3to rather than another decoder like DGIndex, DGAVCIndex, or DGdecNV?

nurbs

13th June 2010, 10:31

Audio handling when the movie comes in multiple files is better with eac3to especially if you convert the audio directly with it. eac3to detects and corrects gaps and overlaps as good as possible when you demux or entirely when you reencode.

AnryV

13th June 2010, 12:22

About decoding DTS-HD MA 6.1 and TrueHD 6.1:

1) Decoding by the Arcsoft decoder gives the garbage
2) Decoding by the Sonic decoder gives normal wavs, but the order of the surraund and back center channels is changed. After decoding it is necessary to make the following renamings: SL -> SR, SR -> BACK (BC), BC -> SL.

The decoding of TrueHD 6.1 16 bit has the same error in spite of the use of another decoder (libav).

Inspector.Gadget

13th June 2010, 15:54

With movie Van Helsing Arcsoft Total Media Teather show me 37 subtitles, but eac3to only list 32 subtitles, 5 last ones are absent.

eac3to, in my experience, is smart about detecting authoring oddities on Blu-ray discs that throw off other software. How many subtitles are actually accessible through the disc's subtitles menu? Be aware also that subtitles may correspond to images used for picture-in-picture or trivia content and not actually be "subtitles" in the traditional sense.

alfixdvd

13th June 2010, 16:34

eac3to, in my experience, is smart about detecting authoring oddities on Blu-ray discs that throw off other software. How many subtitles are actually accessible through the disc's subtitles menu? Be aware also that subtitles may correspond to images used for picture-in-picture or trivia content and not actually be "subtitles" in the traditional sense.

The disc's subtitles menu with Arcsoft Totalmedia Teather show 37 subtitles, and activate #36 that are the forced.

eac3to only shows 32, also tsMuxerR shows 32 but says there are some tracks not recognized that don't show. Perhaps they are the five subtitles streams absents.

BdInfo also shows only 32 subtitles streams, but Arcsoft Totalmedia Theater shows 37 and activate #36

subtitles may correspond to images used for picture-in-picture

How to extract them?

Inspector.Gadget

13th June 2010, 17:25

The disc's subtitles menu with Arcsoft Totalmedia Teather show 37 subtitles, and activate #36 that are the forced.

I'm asking about the graphical menu on the disc, not the Arcsoft context menu. Is that the one you're talking about?

How to extract them?

Without the disc in front of me, I can't say. Maybe someone else can check.

alfixdvd

13th June 2010, 23:52

I'm asking about the graphical menu on the disc, not the Arcsoft context menu. Is that the one you're talking about?

Yes I'm talking about Arcsoft context menu.

The graphical menu on the disk only shows 14 subtitles

eac3to, tsmuxerR and BdInfo shows 32, the Arcsoft context menu shows 37

Inspector.Gadget

14th June 2010, 00:12

Trust the disc, not the context menu.

setarip_old

14th June 2010, 06:20

@alfixdvd

Hi!

The following link will provide a multitude of what I believe to be pertinent posts made in the BD-Rebuilder sub-forum regarding this topic:

http://forum.doom9.org/search.php?searchid=5253964

alfixdvd

14th June 2010, 10:33

mrr19121970

14th June 2010, 20:52

alfixdvd is right, the mpls does have 5 extra subpath pids 0x1800 onwards. i've no idea how to help him further though.

setarip_old

14th June 2010, 21:02

@alfixdvd

Sorry about that.

Go to:

http://forum.doom9.org/showthread.php?t=143716

Click on the "Search this thread" tab and enter as one line, including quotation marks: "Mummy" and "subtitles"

alfixdvd

14th June 2010, 22:15

@alfixdvd

Sorry about that.

Go to:

http://forum.doom9.org/showthread.php?t=143716

Click on the "Search this thread" tab and enter as one line, including quotation marks: "Mummy" and "subtitles"

Thanks, reading at the thread I reached http://forum.doom9.org/showthread.php?t=146501

:):)

alfixdvd

14th June 2010, 22:18

alfixdvd is right, the mpls does have 5 extra subpath pids 0x1800 onwards. i've no idea how to help him further though.

Problem solved, thanks to deank's explanations at thread :http://forum.doom9.org/showthread.php?t=146501

setarip_old

14th June 2010, 22:58

@alfixdvd

Glad to hear that I may have steered you in the right direction ;>}

crasus

14th June 2010, 23:45

Why the difference in the second and third audio tracks, in regards to bitrate and channels upon detection?

Using latest versions of both software.

http://img815.imageshack.us/img815/4351/clip.png

vs

http://img821.imageshack.us/img821/6420/clip2.png

Wojszwillo

16th June 2010, 16:22

tebasuna51

16th June 2010, 23:03

No, but try yourself

ACrowley

17th June 2010, 09:09

No, but try yourself

No, DTS HD 6.1 via Arcsoft and TrueHD 6.1 via libav should be fine
I had few 6.1 Tracks without Problems

AnryV

17th June 2010, 09:15

Can anybody confirm this information?
I need to decode DTS-HD MA 6.1 to wav's but now im not sure...
It can be easily confirmed - try to encode DTS-HD MA 6.1 16-bit descrete by DTS-HD Master Audio Suite Encoder from some test wavs.

As example - Blu-ray "Evangelion 2.22".

tebasuna51

17th June 2010, 10:25

It can be easily confirmed - try to encode DTS-HD MA 6.1 16-bit descrete by DTS-HD Master Audio Suite Encoder from some test wavs.

Wors fine for me, for that I say: I can't confirm the problem.

AnryV

17th June 2010, 17:33

Wors fine for me, for that I say: I can't confirm the problem.

6.1 descrete sample
http://multi-up.com/291247
http://multi-up.com/291248

Try to hear and try to decode.

tebasuna51

18th June 2010, 04:50

6.1 descrete sample
http://multi-up.com/291247
http://multi-up.com/291248

Try to hear and try to decode.

I can't hear without decode.
I was searching garbage:
About decoding DTS-HD MA 6.1 and TrueHD 6.1:

1) Decoding by the Arcsoft decoder gives the garbage
...
And I can decode your samples without garbage.

There are a channel mapping problem but with:

eac3to 61-HD.dtshd 61-HD.wav -0,1,2,3,6,4,5

the output is perfect for me.

AnryV

18th June 2010, 09:03

I can't hear without decode.
:eek:
This is ordinary DTS - http://multi-up.com/291248

I was searching garbage:

And I can decode your samples without garbage.

There are a channel mapping problem but with:

eac3to 61-HD.dtshd 61-HD.wav -0,1,2,3,6,4,5

the output is perfect for me.

1. Can you upload the 61-HD.wav and the log ?
2. What version of Arcsoft decoder you used? (I was 1.1.0.7)
3. I see no indications in the description of eac3to to the manual remapping of the channels for 6.1 configuration. Why you used specifically 0,1,2,3,6,4,5?
4. I need the separate mono wav files not one interleaved file

tebasuna51

18th June 2010, 15:27

1. Can you upload the 61-HD.wav and the log ?
Here is the 61-HD.wav (http://www.sendspace.com/file/1sz9rk)

The log:
eac3to v3.20
command line: "D:\eac3to.exe" "D:\61-HD.dtshd" "D:\61-HD.wav" -0,1,2,3,6,4,5
------------------------------------------------------------------------------
DTS Master Audio, 6.1 channels, 16 bits, 48kHz
(core: DTS-ES, 6.1 channels, 16 bits, 1509kbps, 48kHz)
Decoding with ArcSoft DTS Decoder...
Remapping channels...
Writing WAV...
Creating file "D:\61-HD.wav"...
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 1 second.
Done.

2. What version of Arcsoft decoder you used? (I was 1.1.0.7)
I'm using:
"ArcSoft DTS Decoder (1.1.0.0) works fine"
from free trial TMT2.
In my system (XP SP3) eac3to don't work with 1.1.0.7 (the package in this thread)

Tested 1.1.0.7, in my laptop with Vista Home, the output is wrong (all channels have a distorted mix of all original channels).

3. I see no indications in the description of eac3to to the manual remapping of the channels for 6.1 configuration. Why you used specifically 0,1,2,3,6,4,5?
From eac3to Help:
"...
Options:
...
-0,1,2,3,4,5 remap the channels to the specified order"

About the channelmapping and the MaskChannel in WAVE_FORMAT_EXTENSIBLE headers:

There are two official valid MaskChannel for 5.1 surround:
0x003F FL FR FC LFE BL BR (old)
0x060F FL FR FC LFE SL SR (new from XP SP 2)
Note than the order isn't a problem.

And one official valid MaskChannel for 7.1 surround:
0x0637 FL FR FC LFE BL BR SL SR
Without problems.

But I don't know a official MaskChannel for 6.1 surround, there are two options:
0x013F FL FR FC LFE BL BR BC
0x070F FL FR FC LFE BC SL SR
Note here than the order is a problem, the BC channel must go after/before the Back/Side channels.

When eac3to-ArcSoft decode (without remmaping) the 6.1 put the BC after the Side (maybe ArcSoft consider the channels like Back) and MaskChannel 0x070F. And this is wrong.
We must change the MaskChannel or the channel order, and the -0,1,2,3,6,4,5 remap put the BC channel before the Side channels (we can't change the MaskChannel with eac3to).

4. I need the separate mono wav files not one interleaved file
Using '.wavs' in the precedent command line (with -0,1,2,3,6,4,5) the output is ok.
You can extract the monochannels from my uploaded 61-HD.wav with the same result.

AnryV

18th June 2010, 15:38

Tested 1.1.0.7, in my laptop with Vista Home, the output is wrong (all channels have a distorted mix of all original channels).

Thanks.
Looks like that is a problem of the 1.1.0.7 version.

TrueHD 6.1 need to be remapped too.

Midzuki

18th June 2010, 15:57

About the channelmapping and the MaskChannel in WAVE_FORMAT_EXTENSIBLE headers:

There are two official valid MaskChannel for 5.1 surround:
0x003F FL FR FC LFE BL BR (old)
0x060F FL FR FC LFE SL SR (new from XP SP 2)
Note than the order isn't a problem.

Apologies for my ignorance... :o
I wasn't aware of the "0x060F" one. :thanks:

And one official valid MaskChannel for 7.1 surround:
0x0637 FL FR FC LFE BL BR SL SR
Without problems.

Interesting. The Monster Audio Suite replaces the BL and BR channels with something that could be defined as two "Back Center Channels". :confused:

But I don't know a official MaskChannel for 6.1 surround, there are two options:
0x013F FL FR FC LFE BL BR BC
0x070F FL FR FC LFE BC SL SR
Note here than the order is a problem, the BC channel must go after/before the Back/Side channels.

The M.A.S. supports the creation of the "0x013F" W_F_Ext channel mask only.

P.S.: It also lists SEVEN "7.1 layouts" in total. :eek:
That's simply insane. :(

madshi

18th June 2010, 16:14

would a numerical progress indicator be feasible? i have my windows larger with a different font than default so i have no idea how far along the file is.
Sorry, but 99% of the people seem to be happy with the progress bar.

someone said that a dts-hd ma --> flac file created with 3.20 was a tiny bit different compared to when it was created with 3.18, can anyone confirm this?
I'm not aware of any change in eac3to that would result in this behaviour.

im sure this has been posted before but the search function didnt yield me much of a result. i want to transcode various dts formats from bluray into a lossless format that will load in powerdirector.
I don't know powerdirector. Tell us which format powerdirector needs and then maybe we can give you the eac3to command you need to use.

On the Blu-Ray "Batman: Gotham Knight" there are four bonus episodes of "Batman: The Anmated Series". These are reported by eac3to as "480p24/1.001 (3:2) with pulldown flags", and the demuxed VC1 stream is 23.976fps. This is incorrect
eac3to is not handling VC-1 interlaced content correctly. I'm aware of that. It's on my (very long) to do list.

I reported this initially in a separate thread, and in fact it's a little more widespread than I had thought. When I run eac3to on the Blu-Ray disc for "The Bad Lieutenant: Port of Call - New Orleans", it lists 3 titles. So:

eac3to E:\
shows three titles, each with a number of streams.

eac3to E:\ 1)
does nothing. it draws a bunch of "-" characters as it scans the disc, then it returns me to the prompt.
From what I remember, other people didn't have this problem, or did they? Can anybody confirm this problem?

Ideally, eac3to should produce an OGG chapter format file with this conversion already done, since the OGG chapter format file is supposed to use wall-clock time, not an NDF timecode expressed in fractional seconds.
I'm producing a chapter format file which mkvtoolnix accepts. Does mkvtoolnix also accept OGG chapter format files?

Note that NDF and DF timecodes are only valid for 29.97 fps streams. There is no such thing as a DF or NDF timecode for a 23.976 stream, and I think there is where the problem is coming in. The timecodes on the disc for HD-DVD and BD shouldn't be NDF in the first place, as that isn't valid for a 23.976 stream.
Well, HD-DVD streams are officially 1080i60 / 1.001, so I guess that justifies the use of NDF timecodes. And the HD DVD specification officially says that NDF timecodes are used. So eac3to should really convert...

I thought eac3to will let you know if there is audio overlaps with THD tracks even if it can't fix the overlap or is it just a gap it warns you about? Even i did Spider-man 2 keeping the Truehd track and not converting to PCM because i thought there wasn't any overlap since eac3to didn't tell me there was.
I do think that eac3to warns in any case. Maybe with Spiderman 2 the gaps are so small that eac3to didn't found it necessary to complain. TrueHD gaps are really small to begin with, and eac3to only complains if the summed up gaps is bigger than 7ms, IIRC.

Small bug I've come across every now and then with eac3to and NeroAacEnc:

When converting/(de)muxing multiple tracks from a random source, one instance of NeroAacEnc doesn't seem to exit, making eac3to hang at the last progress bar. Ending that hung instance of NeroAacEnc causes eac3to to flood the command prompt with
until it it gets to
where x is the size of the file (so the end of it).

I've never seen this happen when doing single audio conversions to AAC, so I suspect it's something to do with eac3to not continuing to feed that hung instance of NeroAacEnc even after all other track operations had finished. Or maybe not.. who knows. Redoing the same command usually fixes it, suggesting that it happens randomly.
How can I reproduce this? I've just tried "eac3to some.wav 1.aac 2.aac" and both AAC files are created just fine.

mediainfo shows something interesting
Duration : 37mn 25s

Duration : 2h 41mn
I can confirm the eac3to bug when write the w64 header (work fine with wav input).
The fields RiffLength and DataLength are truncated to 32 bits instead the full 64 bits value (4 GB limit).
Thanks, will be fixed in the next build.

Why the difference in the second and third audio tracks, in regards to bitrate and channels upon detection?
My best guess is that the Blu-Ray information records on this Blu-Ray are incorrect. tsMuxeR seems to rely on the Blu-Ray information records, while eac3to always parses the audio bitstream. So if there is a difference, usually eac3to should show the correct information.

1) Decoding by the Arcsoft decoder gives the garbage
As tebasuna51 found out, this seems to be a bug in newer versions of the ArcSoft DTS decoder. The original version 1.1.0.0 seems to handle this fine.

2) Decoding by the Sonic decoder gives normal wavs
For me, decoding your track with Sonic gives me an empty back channel. And funny enough, the runtime is shorter. The chunk of runtime, where the backchannel is supposed to be "spoken", is simply missing. Very weird...

The decoding of TrueHD 6.1 16 bit has the same error in spite of the use of another decoder (libav).
Do you have a short TrueHD 6.1 sample for me? I don't think I've seen a TrueHD 6.1 track yet!

But I don't know a official MaskChannel for 6.1 surround, there are two options:
0x013F FL FR FC LFE BL BR BC
0x070F FL FR FC LFE BC SL SR
Note here than the order is a problem, the BC channel must go after/before the Back/Side channels.

When eac3to-ArcSoft decode (without remmaping) the 6.1 put the BC after the Side (maybe ArcSoft consider the channels like Back) and MaskChannel 0x070F. And this is wrong.
You're right. Will be fixed in the next build.

madshi

18th June 2010, 16:18

The M.A.S. supports the creation of the "0x013F" W_F_Ext channel mask only.
That's really stupid, IMHO. Who has a 6.1 setup with 3 back speakers and no side speakers? That just makes no sense! Even with 7.1, you usually have 2 side speakers and 2 back speakers.

P.S.: It also lists SEVEN "7.1 layouts" in total. :eek:
That's simply insane. :(
Yes, I agree, that's insane. And the ArcSoft DTS decoder stumbles over at least 1 of those 7 layouts.

Killroy™

18th June 2010, 16:28

Originally Posted by djloewen View Post
I reported this initially in a separate thread, and in fact it's a little more widespread than I had thought. When I run eac3to on the Blu-Ray disc for "The Bad Lieutenant: Port of Call - New Orleans", it lists 3 titles. So:

eac3to E:\

shows three titles, each with a number of streams.

eac3to E:\ 1)

does nothing. it draws a bunch of "-" characters as it scans the disc, then it returns me to the prompt.

From what I remember, other people didn't have this problem, or did they? Can anybody confirm this problem?

I have seen this with other discs (can't remember which right now) and I had to manually select the .m2ts files to run. Did not notice anything wrong in the structure or .mpls files to see what cause them. It is a very very rare problem but easily overcomed by manual coding.

madshi

18th June 2010, 16:41

I have seen this with other discs (can't remember which right now) and I had to manually select the .m2ts files to run. Did not notice anything wrong in the structure or .mpls files to see what cause them. It is a very very rare problem but easily overcomed by manual coding.
If you see it again, could you please drop a short comment here (or somewhere else :))? Maybe, sooner or later, if we have luck, I have the same Blu-Ray here, anyway, then I can check out where the problem is coming from...

Inspector.Gadget

18th June 2010, 16:42

Does mkvtoolnix also accept OGG chapter format files?

Yes, it does.

AnryV

18th June 2010, 16:52

Do you have a short TrueHD 6.1 sample for me? I don't think I've seen a TrueHD 6.1 track yet!

http://rapidshare.com/files/400416105/EVA_2.22.rar.html

It's from "Evangelion 2.22".

Killroy™

18th June 2010, 16:52

If you see it again, could you please drop a short comment here (or somewhere else :))? Maybe, sooner or later, if we have luck, I have the same Blu-Ray here, anyway, then I can check out where the problem is coming from...

I will. Since I have seen it maybe two-or-three times I cannot remember when I saw it.

Snowknight26

18th June 2010, 17:51

How can I reproduce this? I've just tried "eac3to some.wav 1.aac 2.aac" and both AAC files are created just fine.

I've only ever notice it happen when converting either TrueHD or DTS-HD MA in an M2TS to FLAC while also converting a different audio track in the M2TS to AAC. Like I said, very hit or miss, but it happens often enough to be annoying. :P

Midzuki

18th June 2010, 17:55

Yes, I agree, that's insane. And the ArcSoft DTS decoder stumbles over at least 1 of those 7 layouts.

Probably because some of them are not entirely-compliant with the (already-imperfect) Wave-Format-Ext definitions...

http://forum.videohelp.com/attachments/2352-1276913498/MAS-7.1-layouts.png

Speaker Position Decimal Value
¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯ ¯¯¯¯¯¯¯¯¯¯¯¯¯

FRONT_LEFT 1
FRONT_RIGHT 2
FRONT_CENTER 4
LOW_FREQUENCY 8
BACK_LEFT 16
BACK_RIGHT 32
FRONT_LEFT_OF_CENTER 64
FRONT_RIGHT_OF_CENTER 128
BACK_CENTER 256
SIDE_LEFT 512
SIDE_RIGHT 1024
TOP_CENTER 2048
TOP_FRONT_LEFT 4096
TOP_FRONT_CENTER 8192
TOP_FRONT_RIGHT 16384
TOP_BACK_LEFT 32768
TOP_BACK_CENTER 65536
TOP_BACK_RIGHT 131072
RESERVED 262144

As the table above shows, the positions

BACK_LEFT_OF_CENTER,
BACK_RIGHT_OF_CENTER,
LEFT_WIDE,
RIGHT_WIDE,
TOP_SIDE_LEFT,
TOP_SIDE_RIGHT,

"do not exist" in the
Wave-Format-Extensible "world".

Thunderbolt8

18th June 2010, 18:15

just for clarification which tracks does the arcsoft decoder 1.1.0.7 treat in a wrong way compared to 1.1.0.0? only 6.1 tracks?

madshi

18th June 2010, 18:53

It's from "Evangelion 2.22".
Thanks. Will be fixed in the next build.

I've only ever notice it happen when converting either TrueHD or DTS-HD MA in an M2TS to FLAC while also converting a different audio track in the M2TS to AAC. Like I said, very hit or miss, but it happens often enough to be annoying. :P
Well, I need a way to reliably reproduce this, or else it will be hard to fix.

just for clarification which tracks does the arcsoft decoder 1.1.0.7 treat in a wrong way compared to 1.1.0.0? only 6.1 tracks?
AFAIK only 6.1 tracks are affected.

madshi

18th June 2010, 18:55

eac3to v3.21 released

http://madshi.net/eac3to.zip

* fixed: big w64/rf64 files sometimes had incorrect header information
* fixed: HD DVD chapter times were wrong by 3.6 seconds per hour
* fixed: 6.1 DTS decoding with ArcSoft resulted in wrong channel order
* fixed: 6.1 TrueHD decoding with libav resulted in wrong channel order
* fixed: reducing bitdepth of clipped audio produced too much bitdepth
* when clipping is automatically fixed, the applied dB gain is logged
* when normalizing an audio track, the applied dB gain is logged

Thunderbolt8

18th June 2010, 18:58

thanks :)

b66pak

18th June 2010, 19:05

thanks a lot...
_

Midzuki

18th June 2010, 19:11

Man, that was fast! :)

:thanks: :thanks: :thanks:

AnryV

18th June 2010, 20:30

THANKS!

:)
Will be anything made with the decoding of "strange setup" 7.1?

madshi

18th June 2010, 20:55

Will be anything made with the decoding of "strange setup" 7.1?
It's the fault of the ArcSoft DTS decoder. There's nothing I can do about it (at least not with any reasonable amount of work).

SomeJoe

18th June 2010, 23:08

I'm producing a chapter format file which mkvtoolnix accepts. Does mkvtoolnix also accept OGG chapter format files?

Well, HD-DVD streams are officially 1080i60 / 1.001, so I guess that justifies the use of NDF timecodes. And the HD DVD specification officially says that NDF timecodes are used. So eac3to should really convert...

As far as I know, what you're outputting from eac3to is an OGG chapter format file. MKVToolnix refers to it as "simple" chapter format. DVDDecrypter refers to it as "Chapter Information - OGG".

I noticed you fixed this for HD-DVD in build 3.21 - thanks, this works well.

Is the patch specifically only for HD-DVD? Because the chapter points are almost always off by the 1001/1000 ratio for Blu-Ray also.

tebasuna51

19th June 2010, 00:33

Thanks madshi.

MikeEby

19th June 2010, 01:37

Hi,
I'm getting very odd results from "Mary and Max" The disk should have a DTS-HD track but the first run is yeilding this.

eac3to v3.21
command line: "C:\eac3to\eac3to.exe" "D:\BDMV\STREAM\" -log="C:\Program Files\ACD\Another EAC3to GUI\log.txt"
------------------------------------------------------------------------------
1) 00000.mpls, 00023.m2ts+00000.m2ts, 1:32:26
- Chapters, 17 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz

2) 00007.mpls, 00007.m2ts, 0:22:04
- MPEG2, 480i60 /1.001 (16:9)
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz

3) 00002.mpls, 00005.m2ts, 0:15:48
- Chapters, 6 chapters
- MPEG2, 480i60 /1.001 (4:3)
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz

When I try to extract playlist 1, I get this. Notice the time it said the playlist is only 10 seconds long.

eac3to v3.21
command line: "C:\eac3to\eac3to.exe" 1) -log="C:\Program Files\ACD\Another EAC3to GUI\log.txt"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 2 subtitle tracks, 0:00:10, 24p /1.001
1: Chapters, 17 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
4: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
5: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
6: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
7: Subtitle (PGS), Spanish
8: Subtitle (PGS), English

Here is what Media Info says about 00023.m2ts

http://www.acdnow.com/anothereac3togui/MaryAndMax00023.JPG

00000.m2ts does have a DTS-MA Track. I'll try to run eac3to only on 00000.m2ts and see what results I get.

http://www.acdnow.com/anothereac3togui/MaryAndMaxMinfo.JPG

Thank!

Mike

MikeEby

19th June 2010, 03:47

I ran eac3to on just 00000.m2ts for "Mary and Max" and it was able to demux. Here is my log...I noticed I didn't get a chapter file....That might be normal when only demuxing a single file....It's been a while since I did one this way.

eac3to v3.21
command line: "C:\eac3to\eac3to.exe" "D:\BDMV\STREAM\00000.m2ts" "C:\Work Folder\MaryAndMax.mkv"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 2 subtitle tracks, 1:32:16, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: RAW/PCM, English, 2.0 channels, 24 bits, 48kHz
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
4: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
5: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
6: Subtitle (PGS), Spanish
7: Subtitle (PGS), English
[s07] Extracting subtitle track number 7...
[a02] Extracting audio track number 2...
[a04] Extracting audio track number 4...
[a02] Reading RAW/PCM...
[a04] Reading RAW/PCM...
[a02] Swapping endian...
[s06] Extracting subtitle track number 6...
[a04] Swapping endian...
[a02] Encoding FLAC with libFlac...
[a05] Extracting audio track number 5...
[a04] Encoding FLAC with libFlac...
[a05] Reading RAW/PCM...
[a05] Swapping endian...
[a03] Extracting audio track number 3...
[a05] Encoding FLAC with libFlac...
[v01] Extracting video track number 1...
[a03] Decoding with ArcSoft DTS Decoder...
[v01] Muxing video to Matroska...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "C:\Work Folder\MaryAndMax - 3 - DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz.flac"...
[a02] Creating file "C:\Work Folder\MaryAndMax - 2 - PCM, English, 2.0 channels, 24 bits, 48kHz.flac"...
[a05] Creating file "C:\Work Folder\MaryAndMax - 5 - PCM, English, 2.0 channels, 16 bits, 48kHz.flac"...
[a04] Creating file "C:\Work Folder\MaryAndMax - 4 - PCM, English, 2.0 channels, 16 bits, 48kHz.flac"...
[s07] Creating file "C:\Work Folder\MaryAndMax - 7 - Subtitle (PGS), English.sup"...
[s06] Creating file "C:\Work Folder\MaryAndMax - 6 - Subtitle (PGS), Spanish.sup"...
[a02] The original audio track has a constant bit depth of 24 bits.
[a03] The original audio track has a constant bit depth of 24 bits.
[a04] The original audio track has a constant bit depth of 16 bits.
[a05] The original audio track has a constant bit depth of 16 bits.
Added fps value (24 /1.001) to MKV header.
Video track 1 contains 132730 frames.
Subtitle track 6 contains 1120 captions.
Subtitle track 7 contains 1426 captions.
eac3to processing took 34 minutes, 43 seconds.
Done.

Mike

mrr19121970

19th June 2010, 09:46

I ran eac3to on just 00000.m2ts for "Mary and Max" and it was able to demux. Here is my log...I noticed I didn't get a chapter file....That might be normal when only demuxing a single file....It's been a while since I did one this way.

yes, that's normal. chapter info is stored in the playlist (MPLS), not the m2ts.

Hi,
I'm getting very odd results from "Mary and Max" The disk should have a DTS-HD track but the first run is yeilding this.

If you post an AnyDVD logfile then we can take a look. I suspect the mpls has been badly authored here. The playlistmarks might be incorrect & the DTS track also left out.

mrr19121970

19th June 2010, 09:51

would a numerical progress indicator be feasible? i have my windows larger with a different font than default so i have no idea how far along the file is.

Not sure why Madshi said no (http://forum.doom9.org/showpost.php?p=1409670&postcount=10149) to this, as it already exists.

-progressnumbers

madshi

19th June 2010, 10:15

As far as I know, what you're outputting from eac3to is an OGG chapter format file. MKVToolnix refers to it as "simple" chapter format. DVDDecrypter refers to it as "Chapter Information - OGG".
Yes, that's also the conclusion I came to, when looking up OGG chapters yesterday.

I noticed you fixed this for HD-DVD in build 3.21 - thanks, this works well.

Is the patch specifically only for HD-DVD? Because the chapter points are almost always off by the 1001/1000 ratio for Blu-Ray also.
The patch is only for HD DVD right now, because the HD DVD spec clearly confirms your comments about how to interpret the XML chapter times. The Blu-Ray spec is quite different. It says:

"The mark_time_stamp shall point to a presentation-time in the interval from the IN_time until the OUT_time of the PlayItem referred to by the ref_to_PlayItem_id, measured in units of a 45kHz clock."

Doesn't sound to me as if I should do * 1.001, or what do you think?

I'm getting very odd results from "Mary and Max" The disk should have a DTS-HD track but the first run is yeilding this.
I've seen some cases where the main playlist was like "movie.m2ts + someVeryShortDummy.m2ts". In such cases often the dummy m2ts file has less audio/video tracks than the movie m2ts file. eac3to is able to handle this situation, it then automatically ignores the dummy m2ts file. However, in your case it seems that the playlist *starts* with a dummy m2ts file and ends with the real movie m2ts file. I've not seen this yet. Not sure why eac3to shows a wrong runtime in this case. I wouldn't have expected that...

Not sure why Madshi said no (http://forum.doom9.org/showpost.php?p=1409670&postcount=10149) to this, as it already exists.

-progressnumbers
Yes, but that option is made specifically for GUIs. eac3to outputs one new line to the command window for every progress. I think ramicio wants a numerical indicator which updates itself in the same line, similar to what some other command tools are doing.

Laurent

19th June 2010, 11:46

As far as I know, what you're outputting from eac3to is an OGG chapter format file. MKVToolnix refers to it as "simple" chapter format. DVDDecrypter refers to it as "Chapter Information - OGG".

I noticed you fixed this for HD-DVD in build 3.21 - thanks, this works well.

I just tried and I can confirm that now it is fixed.
Thank you Madshi.

laserfan

19th June 2010, 13:10

Is the patch specifically only for HD-DVD? Because the chapter points are almost always off by the 1001/1000 ratio for Blu-Ray also.

The patch is only for HD DVD right now, because the HD DVD spec clearly confirms your comments about how to interpret the XML chapter times. The Blu-Ray spec is quite different. It says:

"The mark_time_stamp shall point to a presentation-time in the interval from the IN_time until the OUT_time of the PlayItem referred to by the ref_to_PlayItem_id, measured in units of a 45kHz clock."

Doesn't sound to me as if I should do * 1.001, or what do you think?
I'm a chapter-fanatic mysself. After I extract chapters.txt with eac3to, then convert to frames (and tsMuxeR timecodes) using shon3i's ChapterGen tool, I next examine each chapter location against the DGIndexNV-indexed video in VirtualDub, looking for discrepancies between the chapter marks and the video scene changes. In most (all?) BD cases these chapter marks fall exactly on the first frame of a new scene. But this is for h264 only. If the BD was made with VC1, I often times observe that the chapter points fall a single frame or two beyond the scene change.

Since I use the chapter info to make a qpfile.txt to incorporate into my x264 conversions (to assure there are I-frames where I need them for chapter markers), I need then to adjust the qpfile (and corresponding txMuxeR timecode info) to land exactly on that scene-change frame. Can be a real PIA if there are a lot of chapters...

But it seems I only see this with VC1 BDs, and not with EVERY VC1 BD. But it's as though the mastering tool used w/VC1 is different from h264 (dunno about MPEG2 ottomh). But like I say I don't ALWAYS have to fix these VC1 BDs, just *most* of the time.

Midzuki

19th June 2010, 13:38

madshi,

could you please read again this post (http://forum.doom9.org/showpost.php?p=1409727&postcount=10157) and then specify what is the (most) "problematic" 7.1 channel layout for the ArcSoft DTS decoder ???

-----

EDIT:

OK, I've just found this:

only 7.1 channel tracks with order "C L R Ls Rs LFE Lsr Rsr" are handled incorrectly and there is only one such track from "Qtec Hi-Definition Reference Disc".

However additional comments would be welcome of course. :)

MikeEby

19th June 2010, 13:50

I've seen some cases where the main playlist was like "movie.m2ts + someVeryShortDummy.m2ts". In such cases often the dummy m2ts file has less audio/video tracks than the movie m2ts file. eac3to is able to handle this situation, it then automatically ignores the dummy m2ts file. However, in your case it seems that the playlist *starts* with a dummy m2ts file and ends with the real movie m2ts file. I've not seen this yet. Not sure why eac3to shows a wrong runtime in this case. I wouldn't have expected that...

Madshi, thanks for the quick reply. Yes the dummy file 00023.m2ts is first file in the playlist and it is just a studio logo that is 10 seconds long. It also only has PCM audio tracks, while the second file is the main movie (00000.m2ts) has a DTS-HD track. Eac3to does create the all the files requested for the first file but seems to quits when it sees the second file 00000.m2ts that contains the DTS-HD track. See my screen shots from MediaInfo posted above.

I don't think I've seen a playlist that contains 2 files with different audio track structures, which seems to be the case on this disk. Does eac3to compare the track structure between different files in the playlist?

Mike

MikeEby

19th June 2010, 14:03

If you post an AnyDVD logfile then we can take a look. I suspect the mpls has been badly authored here. The playlistmarks might be incorrect & the DTS track also left out.

I don't think that's the issue in this case. The problem might be because the 2 files do not contain the same track structure.

Mike

Thunderbolt8

19th June 2010, 17:22

Madshi, thanks for the quick reply. Yes the dummy file 00023.m2ts is first file in the playlist and it is just a studio logo that is 10 seconds long. It also only has PCM audio tracks, while the second file is the main movie (00000.m2ts) has a DTS-HD track. Eac3to does create the all the files requested for the first file but seems to quits when it sees the second file 00000.m2ts that contains the DTS-HD track. See my screen shots from MediaInfo posted above.

I don't think I've seen a playlist that contains 2 files with different audio track structures, which seems to be the case on this disk. Does eac3to compare the track structure between different files in the playlist?

MikeIve encountered the same problem twice now, for example with saving private ryan (US) as well. and also criterion and some other studios now include their logos as seperate small .m2ts file which then preceeds the main movie more often than before. therefore it would be nice if you could come up with a solution here.

SomeJoe

19th June 2010, 18:55

The patch is only for HD DVD right now, because the HD DVD spec clearly confirms your comments about how to interpret the XML chapter times. The Blu-Ray spec is quite different. It says:

"The mark_time_stamp shall point to a presentation-time in the interval from the IN_time until the OUT_time of the PlayItem referred to by the ref_to_PlayItem_id, measured in units of a 45kHz clock."

Doesn't sound to me as if I should do * 1.001, or what do you think?

You're right, I agree that the spec seems to be very specific and shouldn't leave any doubt about where the chapter point is. If the chapter points end up being off by a 1001/1000 ratio, then you would surmise that it's essentially an authoring error on the disc -- the timecodes were NDF prior to the BD being authored, and no conversion was applied during authoring.

I'll do some further research and see if I can find some additional info on some of the BDs I have.

lchiu7

20th June 2010, 01:37

I am trying to join two evo files from the 300 HD-DVD. They had some wierd names but I worked out which was the first and which was the second. Using eac3to to extract the audio and VC1 file with this command line

eac3to.exe "E:\TEMP\HVDVD_TS\FEATURE_300NDOM6LF1VC1_HD1.EVO"+"E:\TEMP\HVDVD_TS\FEATURE_300NDOM6LF1VC1_HD2.EVO" 2: "C:\Temp\HDDVD.vc1" 5: -down2 -192 "C:\Temp\HDDVD5.thd+ac3" -stripPulldown -progressNumbers -libav

I get the message as noted above

a05 Extracting audio track number 5...
a05 Removing TrueHD dialog normalization...
a05 Decoding with libav/ffmpeg...
a05 Downmixing multi channel audio to stereo...
a05 Encoding AC3 <192kbps> with libAften...
a05 The thd ac3 joiner doesn't have the correct connections.
Aborted at file position 1048576.

Out of interest I removed ac3 as an output option (I don't need it) so the output file was "C:\Temp\HDDVD5.thd" and while it completed with no errors, the thd file isn't recognized by tsmuxer as an audio file of any type it knows. It knows about the THD+ac3 file though.

This is the version of eac3to I am using

25/01/2009 09:58 p.m. 1,779,712 eac3to.exe

Any ideas what could be going wrong? Is it possible in copying the files off the HD-DVD they got corrupted?

Thanks

[EDIT]

Never mind. While I don't know what the problem was I found a way around it. I just ran eac3to and demuxed everything in the EVO files. This created the VC1 I needed for the encode and a THD file. While tsmuxer can't read the THD file another eacto run to create a THD+AC3 create a file that tsmuxer could read so all is going okay now.

NASRMAX

20th June 2010, 14:51

Hello, I am currently in the process of testing this new version eac3to I thank Madshi for this work, the extension mkv demux for Bluray 3D works only for the left eye, having extracted without H.264 contenair, the right eye does not pass in mkvmerge, it is progressing well, hoping an update that fixes this

madshi

20th June 2010, 22:00

I'm a chapter-fanatic mysself. After I extract chapters.txt with eac3to, then convert to frames (and tsMuxeR timecodes) using shon3i's ChapterGen tool, I next examine each chapter location against the DGIndexNV-indexed video in VirtualDub, looking for discrepancies between the chapter marks and the video scene changes. In most (all?) BD cases these chapter marks fall exactly on the first frame of a new scene.
I agree that the spec seems to be very specific and shouldn't leave any doubt about where the chapter point is. If the chapter points end up being off by a 1001/1000 ratio, then you would surmise that it's essentially an authoring error on the disc -- the timecodes were NDF prior to the BD being authored, and no conversion was applied during authoring.

I'll do some further research and see if I can find some additional info on some of the BDs I have.
@SomeJoe, if you read laserfan's post, it seems that in his experience most Blu-Rays seem to have proper chapter marks (at least when using h264). I tend to agree with you that if 1001 is needed, it's probably an authoring error.

Yes the dummy file 00023.m2ts is first file in the playlist and it is just a studio logo that is 10 seconds long. It also only has PCM audio tracks, while the second file is the main movie (00000.m2ts) has a DTS-HD track. Eac3to does create the all the files requested for the first file but seems to quits when it sees the second file 00000.m2ts that contains the DTS-HD track. See my screen shots from MediaInfo posted above.
Which movie is that?

Ive encountered the same problem twice now, for example with saving private ryan (US) as well. and also criterion and some other studios now include their logos as seperate small .m2ts file which then preceeds the main movie more often than before. therefore it would be nice if you could come up with a solution here.
Please drop a comment here whenever you see this problem. Maybe we'll have luck and I have one of those Blu-Rays myself. However, I've not really stumbled over this problem myself yet.

Never mind. While I don't know what the problem was I found a way around it.
Not sure myself where the problem came from, but the eac3to you're using is pretty old.

OK, I've just found this: [...]

However additional comments would be welcome of course. :)
I think that's the only speaker combination which makes problems. At least the only one I'm aware of.

Hello, I am currently in the process of testing this new version eac3to I thank Madshi for this work, the extension mkv demux for Bluray 3D works only for the left eye, having extracted without H.264 contenair, the right eye does not pass in mkvmerge, it is progressing well, hoping an update that fixes this
The Haali MKV Muxer (which eac3to is using) does not support the right eye stream format. So that's why eac3to cannot mux the right eye stream to MKV. mkvmerge currently doesn't support it, either. I've already given Mosu (the mkvmerge creator) all the information he needs, but he says that it's not a high priority for him currently. There's not much else I can do here...

NASRMAX

20th June 2010, 22:11

thank you for your reply Madshi, I wait, it's already very good work you made, keep up the good ...;-)

TinTime

21st June 2010, 03:23

* fixed: reducing bitdepth of clipped audio produced too much bitdepth

Thanks very much for that!

MikeEby

21st June 2010, 13:43

Yes the dummy file 00023.m2ts is first file in the playlist and it is just a studio logo that is 10 seconds long. It also only has PCM audio tracks, while the second file is the main movie (00000.m2ts) has a DTS-HD track. Eac3to does create the all the files requested for the first file but seems to quits when it sees the second file 00000.m2ts that contains the DTS-HD track.

Which movie is that?

Region A (US) "Mary and Max".

Mike

Thunderbolt8

21st June 2010, 18:00

Please drop a comment here whenever you see this problem. Maybe we'll have luck and I have one of those Blu-Rays myself. However, I've not really stumbled over this problem myself yet.
might be releated, got this parsing problem where the cursor only parses the playlist up to the 2nd last marking, but then doesnt display anything, but jumps back to the directory prompt. same situation here, the big main movie is preceeded by a small studio logo .m2ts file. movie is 'happy together' by kino.

DruidCtba

21st June 2010, 19:54

Wojszwillo

21st June 2010, 20:34

Hi friends I have a problem to convert audio NTSC 29.976 fps to 23.976 because occurs one slowdown in audio, how to resolved this problem?

I used: eac3to.exe 8mm.ac3 8mm23976.ac3 -224 -29.976 -changeTo23.976

Thanky for your attention.

There exists framerates for NTSC material 24000/1001 (ie 23,976) and 30000/1001 (ie 29,97), but not 29,976...

GZZ

22nd June 2010, 07:42

might be releated, got this parsing problem where the cursor only parses the playlist up to the 2nd last marking, but then doesnt display anything, but jumps back to the directory prompt. same situation here, the big main movie is preceeded by a small studio logo .m2ts file. movie is 'happy together' by kino.

Not sure if its the same issue, but if you mount a bluray image in Virtuel Clone Drive and set the option buffered I/O, then eac3to will fail to parse a bluray, it simple quit to cmd prompt without doing anything, no error, no nothing.

DruidCtba

22nd June 2010, 17:07

Ok friend Wojszwillo then consider 30000/1001, how to resolve the problem so that no distortion occurs in the audio that has been occurring when passing from 30000/1001 to 24000/1001, how could you answer me?

madshi

22nd June 2010, 17:18

@DruidCtba, NTSC 30000/1001 is a framerate for interlaced video. Audio is 24000/1001. Your audio is not 30000/1001. There's no need to change anything. You don't need to slowdown your audio nor video. Slowdown/speedup is for converting between NTSC <-> PAL.

Momber

22nd June 2010, 23:39

DruidCtba

23rd June 2010, 02:00

OK my friend madshi thank you for your answer, now I understand it.

Midzuki

23rd June 2010, 03:15

...
The mapping of 3.0 soundtracks converted to flac is incorrect. The center channel ends up in the left surround.
...
When creating Mono-Wavs instead of flac, the three tracks are correctly titled L, R and C.

To me it seems a design flaw of the FLAC .dll
<DEL>(OR of the Flac decoder you're using)</DEL>,
not a problem in eac3to itself.
<DEL>¿What does MediaInfo says about the "bad" Flac file?</DEL>

P.S.:

BTW, the avcodec library hasn't been updated. :(

P.P.S.:

Perhaps madshi could add a "FLAC's_problems.txt"
to the next eac3to packages ???

Momber

23rd June 2010, 16:22

What does MediaInfo says about the "bad" Flac file?

General
Complete name : Sanjuro JAP.flac
Format : FLAC
Format/Info : Free Lossless Audio Codec
File size : 947 MiB
Duration : 1h 35mn
Overall bit rate : 1 379 Kbps
WAVEFORMATEXTENSIBLE_CHANNEL_MAS : 0X7
VALID_BITS : 24
HDCD : 0

Audio
Format : FLAC
Format/Info : Free Lossless Audio Codec
Duration : 1h 35mn
Bit rate mode : Variable
Bit rate : 1 379 Kbps
Channel(s) : 3 channels
Sampling rate : 48.0 KHz
Resolution : 24 bits
Stream size : 947 MiB (100%)
Writing library : libFLAC 1.2.1 (UTC 2007-09-17)

Midzuki

23rd June 2010, 17:02

@ Momber:

the FLAC generated by eac3to seems OK;

the problem is, there are NO Flac decoders that honor the
W-F-Ext ChannelMask flag...

Momber

24th June 2010, 12:26

the problem is, there are NO Flac decoders that honor the
W-F-Ext ChannelMask flag...
If that is the case, perhaps it would be a good idea for eac3to to simply decline the encoding of 3.0 files to flac for the time being.
A little message along the lines of "3.0 FLAC channel mapping not supported - choose AC3 or DTS encoding instead".

Just an idea.

deank

24th June 2010, 19:42

@madshi: A feature request for two options:

-cutF (cut at frame)
-cutT (cut at HH:MM:SS.MLS) (or any variation using time format)

These two are so useful for testing or when one needs to truncate the output at given frame or time.

And again - thank you for the great tool, allowing for so many other applications to do their job.

Dean

AnryV

24th June 2010, 20:59

eac3to v3.21
command line: F:\HD-VIDEO\0_Progs\eac3to264\eac3to 51_-4DN.dtshd 51_-4DN.wavs
------------------------------------------------------------------------------
DTS Master Audio, 5.1 channels, 16 bits, 48kHz, dialnorm: -4dB
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48kHz, dialnorm: -4dB)
Decoding with ArcSoft DTS Decoder...
Removing DTS dialog normalization...
Writing WAVs...
The original audio track has a constant bit depth of 16 bits.
Creating file "51_-4DN.L.wav"...
Creating file "51_-4DN.R.wav"...
Creating file "51_-4DN.C.wav"...
Creating file "51_-4DN.LFE.wav"...
Creating file "51_-4DN.SL.wav"...
Creating file "51_-4DN.SR.wav"...
eac3to processing took 1 second.
Done.

eac3to don't remove the dialog normalization with Arcsoft decoder (1.1.0.1)

Test .dtshd

http://multi-up.com/295013

DNorm.rar - center before and after decoding.

This procedure also does not help.

eac3to v3.21
command line: F:\HD-VIDEO\0_Progs\eac3to264\eac3to 51_-4DN.dtshd 51_-4DN_removed.dtshd
------------------------------------------------------------------------------
DTS Master Audio, 5.1 channels, 16 bits, 48kHz, dialnorm: -4dB
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48kHz, dialnorm: -4dB)
Removing DTS dialog normalization...
Creating file "51_-4DN_removed.dtshd"...
eac3to processing took 1 second.
Done.

ACrowley

25th June 2010, 06:13

Hi

Can i automate WavAviMux.exe in any Way in the eac3to chain ?
I need 5.1 interelaved Waves muxed in AVI (4Gb+)for some Apps

@ AnryV
Interesting !
However.. i thought most DTS-HD encodes use -31db Dialnorm, so theres no Attenuattion defacto ?

I feel always most comfortable when i can decode DTS(HD) with DTS-HD Streamplayer because its the Reference Decoder. You can disable DialNorm/DRC in the Settings.
But DTS-HD Tracks demuxed from M2TS/MKV cant be used in the StreamPlayer because the additional DTS-HD Header Infos are missing (standard DTS works)
Im still hoping that somebody write a Small app (better eacto3 support) to Add/Strip the extra Bytes :)

AnryV

25th June 2010, 09:55

i thought most DTS-HD encodes use -31db Dialnorm, so theres no Attenuattion defacto ?

There are DTS-HD tracks with Dialnorm and I need to decode them correctly. Especially this concerns 7.1 tracks, which I cannot decode by using of Soniс decoder.

tebasuna51

25th June 2010, 11:44

ACrowley

25th June 2010, 11:50

I can confirm the problem, the:
"Removing DTS dialog normalization..."
only work, with dtshd input, when you use:

eac3to input.dtshd output.dts -core

And don't work with:

eac3to input.dtshd output.dtshd
eac3to input.dtshd output.wav
eac3to input.dtshd output.wav -libav
eac3to input.dtshd output.wav -core
eac3to input.dtshd output.wav -core -libav

BTW, if eac3to inform:
"DTS Master Audio, 5.1 channels, 16 bits, 48kHz, dialnorm: -4dB"

you can use this command line:

eac3to input.dtshd output.wav +4dB

Ah thank you....i think Madshi will fix it when he have the Time..
But as i say, please, can somebody write a small Tool or Batch to write the extra bytes back to a DTS-HD Header to restore the untouched DTS-HD Header :) Sadly ive no Skill to code something.

AnryV

25th June 2010, 12:14

BTW, if eac3to inform:
"DTS Master Audio, 5.1 channels, 16 bits, 48kHz, dialnorm: -4dB"

you can use this command line:

eac3to input.dtshd output.wav +4dB

Unfortunately this will lead to the fact that we will not obtain the original sound. This will be especially noticeably for 16-bit tracks.

tebasuna51

25th June 2010, 14:11

Then use:

eac3to input.dtshd output.wav +4dB -down16

Of course is a workaround, but the process work at 64 bit float, do you think can be especially noticeably?
Maybe any rounding bit.

I don't know for what you need the wav's files.

AnryV

25th June 2010, 14:25

73ChargerFan

25th June 2010, 16:29

The dts core is lossy, so 1-bit rounding errors could be considered immaterial.

ACrowley

25th June 2010, 17:46

So the DialNorm Problem doesnt exits when DTS-HD is decoded by Sonic Decoder ? Ofcourse it doesnt work on 7.1

Snowknight26

26th June 2010, 01:14

Reducing the bitdepth from 24-bit to 16-bit seems to cause tons of static. Tested on a 5.1 channel 24-bit WAV file. Should reproduce easily.

Blue_MiSfit

26th June 2010, 04:44

I'm seeing odd behavior with Battleship Potemkin, a newly restored BluRay release from Kino. http://www.kino.com/video/item.php?film_id=1076

See below, it's self explanatory!

E:\Potemkin>eac3to d:
1) 00004.mpls, 00045.m2ts+00044.m2ts, 1:12:12
- Chapters, 20 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48kHz

2) 00005.mpls, 00047.m2ts+00046.m2ts, 1:11:48
- Chapters, 20 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48kHz

3) 00006.mpls, 00048.m2ts, 0:42:24
- Chapters, 8 chapters
- h264/AVC, 1080i60 /1.001 (16:9)
- AC3, English, stereo, 48kHz

E:\Potemkin>eac3to d: 1)
-------------------------------------------------------------------------------
E:\Potemkin>

When I tell eac3to to give me the stream list from the main movie, it returns nothing :(

I can easily work around this by directly referencing the M2TS in question... Turns out there are two versions of the film on this disc, one with the original Russian cards (since it's a silent film), and another with English translated cards.

I prefer the Russian cards with English subtitles, so I just peeked at the M2TSs, and fully demuxed the one I wanted. Incidentally this was 00046.m2ts, and I was treated to the lovely PGS renderer that's now a part of MPC-HC!!! :D

I think eac3to's confusion may stem from the fact that the main movie M2TS files are preceeded by a very short M2TS, which is the Kino logo. But, I'm certainly not Madshi! Anyway, I was able to work around this no problem, but just a heads-up! Everyone should watch this BluRay. It's a great restoration of THE soviet film. Fantastic!

Cheers,
Derek

Xorp

26th June 2010, 04:53

Reducing the bitdepth from 24-bit to 16-bit seems to cause tons of static. Tested on a 5.1 channel 24-bit WAV file. Should reproduce easily.

I'm getting that too and also on a normal 16-bit 7.1 TrueHD to FLAC conversion.

Killroy™

26th June 2010, 05:02

`Orum

26th June 2010, 06:56

Been using eac3to for quite some time now and love it...but I do have a couple of minor requests. First, an option to disable saving the internal logging of the output to a file--typically when batch processing I'll log everything to one file anyway. Secondly, a way to silence the completion "uncorking" sound. It's fairly loud and very annoying when running batch scripts to do some quick remuxing/conversion on lots of files. Other than that it's perfect.

Many thanks, I can't tell you how much time this app has saved me.

Snowknight26

26th June 2010, 08:07

Rename the wav files.

`Orum

26th June 2010, 08:14

yeah I know you can do that, but I find them useful at times...just thinking a command line flag would be best.

Blue_MiSfit

26th June 2010, 10:30

@Killroy:
Yep, that's what I meant to say :)

Regarding the content, I do prefer to see both the Cyrillic characters and the English subs as well. It feels more... soviet?

Thanks though!

Derek

kypec

26th June 2010, 11:02

yeah I know you can do that, but I find them useful at times...just thinking a command line flag would be best.
If you are annoyed by the sound only in your batch processing then why not try something like this:rem This goes somewhere before the eac3to.exe is called
rename "C:\Program Files\AVTools\eac3to\error.wav" error.none
rename "C:\Program Files\AVTools\eac3to\success.wav" success.none
rem Here goes your usual processing...
rename "C:\Program Files\AVTools\eac3to\error.none" error.wav
rename "C:\Program Files\AVTools\eac3to\success.none" success.wav

Replace path to eac3to with your real path of course.

tebasuna51

26th June 2010, 11:14

Reducing the bitdepth from 24-bit to 16-bit seems to cause tons of static. Tested on a 5.1 channel 24-bit WAV file. Should reproduce easily.

Maybe?:
"reduce bitdepth of decoded audio data by using TPDF dithering"

`Orum

26th June 2010, 11:20

Yeah, the disadvantage of that is you have to be careful with your batch files. E.g., if I have one batch file, "bat_1.bat", that is
rename "C:\Program Files (x86)\eac3to\error.wav" error.none
rename "C:\Program Files (x86)\eac3to\success.wav" success.none
.\otherdir\bat_2.bat
"C:\Program Files (x86)\eac3to\eac3to.exe" foo.mkv 2:bar.flac
rename "C:\Program Files (x86)\eac3to\error.none" error.wav
rename "C:\Program Files (x86)\eac3to\success.none" success.wav
and a ".\otherdir\bat_2.bat" that is similar, e.g.
rename "C:\Program Files (x86)\eac3to\error.wav" error.none
rename "C:\Program Files (x86)\eac3to\success.wav" success.none
"C:\Program Files (x86)\eac3to\eac3to.exe" foo2.mkv 2:bar2.flac
rename "C:\Program Files (x86)\eac3to\error.none" error.wav
rename "C:\Program Files (x86)\eac3to\success.none" success.wav
I won't get the intended silent result. Just need to be really careful of the order you do things in, and where you do the renaming. A flag avoids the ordering problems.

DaFees

26th June 2010, 11:49

Hello everyone,

I hate to sound like a total noob, but I am having a big problem with eac3to. It simply will not convert TrueHD to LPCM. I'm trying to convert a bdrip for playback on my PS3. I do not know why, but it is specifically TrueHD that fails. I mean I just finished successfully converting a DTS-HD MA track to LPCM. I also know it's not a bad rip or anything because I've tried several different TrueHD tracks and none of them will convert. With a DTS-HD conversion I get the full sound, but with TrueHD I end up with no sound and really speed up and stuttery video.

If it helps I like to use multiavchd to convert my .m2ts files with DTS-HD / TrueHD to .m2ts files with LPCM. I get no errors with DTS-HD but TrueHD always throws the error that eac3to refuses to convert. Yes, I am running the latest version of both multiavchd and eac3to. I don't know what else I would need as eac3to is promoted as supporting TrueHD decoding out of the box without any additional files. I've searched high and low and I am at a loss on this one. I mean usually people have issues with DTS-HD to LPCM not TrueHD to LPCM.

Oh if it makes any difference I am running Windows 7 x64 with the latest windows 7 codec packages

tebasuna51

26th June 2010, 13:33

@DaFees
Please put the eac3to log.

DaFees

26th June 2010, 21:03

@DaFees
Please put the eac3to log.

I wish I could, but I'm not sure if I am getting a correct log. I ran multiavchd and that didn't provide a useful log. I did try running eac3to via commandline and it did give me a log, but I do not think it was helpful:

eac3to v3.21
command line: C:\multiavchd\tools\eac3to\eac3to.exe "D:\BD Wip\00001.track_4352.ac3" "D:\BD Wip\00001.wav"
------------------------------------------------------------------------------
TrueHD/AC3, 5.1 channels, 48kHz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB)
Extracting TrueHD stream...
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Writing WAV...
Creating file "D:\BD Wip\00001.wav"...
The original audio track has a constant bit depth of 16 bits.
Caution: The WAV file is bigger than 4GB. <WARNING>
Some WAV readers might not be able to handle this file correctly. <WARNING>
Superfluous zero bytes detected, will be stripped in 2nd pass.
Starting 2nd pass...
Reading WAV...
Stripping zero bytes...
Writing WAV...
Creating file "D:\BD Wip\00001.wav"...
Caution: The WAV file is bigger than 2GB. <WARNING>
Some WAV readers might not be able to handle this file correctly. <WARNING>
eac3to processing took 19 minutes, 51 seconds.
Done.

Atak_Snajpera

26th June 2010, 21:43

Try this
C:\multiavchd\tools\eac3to\eac3to.exe "D:\BD Wip\00001.track_4352.ac3" "D:\BD Wip\00001.w64"

tebasuna51

26th June 2010, 23:12

@DaFees
Of course the log help to know the problem.

eac3to works fine and decode the TrueHD without problems, I don't know what is the problem with multiavchd.
Try with the w64 format like Atak_Snajpera say you.

DaFees

26th June 2010, 23:48

Try this
C:\multiavchd\tools\eac3to\eac3to.exe "D:\BD Wip\00001.track_4352.ac3" "D:\BD Wip\00001.w64"

Well I tried this and I even remuxed with tsmuxer and when I tried to play the final video I got no sound and stuttery video, again.

Still though eac3to still told me all was well:

eac3to v3.21
command line: C:\multiavchd\tools\eac3to\eac3to.exe "D:\BD Wip\00001.track_4352.ac3" "D:\BD Wip\00001.w64"
------------------------------------------------------------------------------
TrueHD/AC3, 5.1 channels, 48kHz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB)
Extracting TrueHD stream...
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Writing W64...
Creating file "D:\BD Wip\00001.w64"...
The original audio track has a constant bit depth of 16 bits.
Superfluous zero bytes detected, will be stripped in 2nd pass.
Starting 2nd pass...
Reading W64...
Stripping zero bytes...
Writing W64...
Creating file "D:\BD Wip\00001.w64"...
eac3to processing took 19 minutes, 6 seconds.
Done.

I should add that multiavchd will play the error sound that comes with eac3to every time it refuses to convert the audio.

tebasuna51

27th June 2010, 10:22

Play D:\BD Wip\00001.wav to listen the audio, if play ok put your problem in multiavchd thread (http://forum.doom9.org/showthread.php?t=143744) because eac3to seems work fine.

DaFees

27th June 2010, 11:17

Ok well I just checked and eac3to is working fine, because the .wav file played fine on my PC and yes I will post in the multiavchd thread for help, but that still doesn't totally resolve the issue. Ok, so multiavchd won't work. Well then why won't my converted audio stream when remuxed with the video play on my PS3?

I've tried the conversion two ways: Using multiavchd and tsmuxer gui to demux the audio + eac3to to convert to lpcm + tsmuxer gui again to remux the converted audio with the video.

madshi

27th June 2010, 11:31

The mapping of 3.0 soundtracks converted to flac is incorrect. The center channel ends up in the left surround.
Can I have a short sample, please?

If that is the case, perhaps it would be a good idea for eac3to to simply decline the encoding of 3.0 files to flac for the time being.
No, I will not limit eac3to just because decoders don't behave correctly. Instead the decoders should be fixed.

A feature request for two options:

-cutF (cut at frame)
-cutT (cut at HH:MM:SS.MLS) (or any variation using time format)
Are you talking about audio or video? Cutting how much? Have you tried the "-edit" option?

eac3to don't remove the dialog normalization with Arcsoft decoder (1.1.0.1)
Can I have a DTS-HD sample with dialnorm, please? Don't know if I can fix this. Maybe yes, maybe no. Editing the DTS-HD headers is very complicated.

Reducing the bitdepth from 24-bit to 16-bit seems to cause tons of static. Tested on a 5.1 channel 24-bit WAV file. Should reproduce easily.
I'm getting that too and also on a normal 16-bit 7.1 TrueHD to FLAC conversion.
Sorry about that. One of the v3.21 fixes caused this. Will be fixed in the next build.

I'm seeing odd behavior with Battleship Potemkin
Another Blu-Ray I don't have. Would it be possible to upload the first small m2ts file and the first maybe 10MB of the movie m2ts file, together with the PLAYLIST and CLIPINF folders?

First, an option to disable saving the internal logging of the output to a file
Is the creation of the log file such a big problem? After all, you can simply delete it again!

Secondly, a way to silence the completion "uncorking" sound.
yeah I know you can do that, but I find them useful at times...just thinking a command line flag would be best.
Nope, sorry. I don't like the idea of adding tons of funny options that must be documented/explained somewhere. That makes eac3to usage only more complicated. BTW, when writing batch files you can use stuff like "if exist success.wav goto :someMark". That should fix all "order" problems.

Ok well I just checked and eac3to is working fine, because the .wav file played fine on my PC and yes I will post in the multiavchd thread for help, but that still doesn't totally resolve the issue. Ok, so multiavchd won't work. Well then why won't my converted audio stream when remuxed with the video play on my PS3?
So there's a bug with multiavchd and another bug with tsmuxer. Why are you asking for help about these tools in the eac3to thread? As you said yourself, eac3to seems to be working fine.

madshi

27th June 2010, 11:35

eac3to v3.22 released

http://madshi.net/eac3to.zip

* fixed: reducing bitdepth from >16bit to 16bit resulted in static (v3.21)
* updated to latest ffmpeg/libav version
* libav MP1/MP2/MP3 decoders now decode to floating point
* libav E-AC3 decoder now supports spectral extension
The updated ffmpeg/libav libraries are quite a big change. It would be nice if you guys could do some libav decoder testing with various codecs (AC3, E-AC3, DTS, MP1, MP2, MP3, MLP, TrueHD) to make sure that things still work as intended.

deathlord

27th June 2010, 15:23

Hi

I have just encoutered a nasty problem using eac3to 3.17.
When converting 5.1 48kHz dts files to 5.1 flac, eac3to just stopped without error, leaving an unfinished file and no log. It happen only with one file out of ten.
Even worse, rerunning the identical command line once more, the problem does not appear anymore.
This has happend twice in the last couple of days.

Any ideas?

Unfortunately I was unable to reproduce this problem. I could run two instances of eac3to using arcsoft at the same time.

Probably there really was (is) something wrong with my PC...

Another thing:

How can I silence the last 3 seconds, say, of a flac file? Is there a way to use the -edit function to do this without manually entering "runtime minus 3s"?

I would like to systematically silence the annoying sound at the end of tv shows.

Cheers
deathlord

AnryV

27th June 2010, 17:30

Can I have a DTS-HD sample with dialnorm, please? Don't know if I can fix this. Maybe yes, maybe no. Editing the DTS-HD headers is very complicated.

.....

Test .dtshd

http://multi-up.com/295013

RainyDog

27th June 2010, 21:19

Eac3to, or likely the ArcSoft decoder, doesn't seem to like my one channel DTS file. I want to convert it to 192kbps ac3 as one channel DTS seems like a massive waste of space. Is there anyway I can force eac3to to use a different DTS decoder or any other program I could try? Cheers in advance for any responses, the log is below.

eac3to v3.22
command line: "C:\Program Files (x86)\eac3to\eac3to.exe" "S:\Temp\RipBot264temp\job2\2_audio_Japanese.dts" "S:\Temp\Profound.ac3" -192
------------------------------------------------------------------------------
DTS, 1.0 channels, 2:53:28, 24 bits, 768kbps, 48kHz
Decoding with ArcSoft DTS Decoder...
The ArcSoft DTS Decoder reported an error while decoding. <ERROR>
Aborted at file position 262144. <ERROR>

hubblec4

27th June 2010, 21:58

Snowknight26

28th June 2010, 02:15

Is there anyway I can force eac3to to use a different DTS decoder or any other program I could try? Cheers in advance for any responses, the log is below.

Check the first post, check the readme, check by running eac3to without any parameters. Kind of hard to miss.

In this case i use Megui to encode the dts-file.

He's encoding AC3, not DTS, so that wouldn't help.

Midzuki

28th June 2010, 02:36

The (new) avcodec.dll is working correctly now
( finally! :devil: :D ),
and is much faster :)
than the outdated one:

[C:\AUDIOEXZ\LAB]
=>eac3to input447.dts output447.wav -libav -down16
DTS, 2.0 channels, 0:01:17, 24 bits, 447kbps, 48kHz
Decoding with libav/ffmpeg...
Reducing depth from 64 to 16 bits...
Writing WAV...
Creating file "output447.wav"...
Clipping detected, a 2nd pass will be necessary. <WARNING>
Starting 2nd pass...
Decoding with libav/ffmpeg...
Reducing depth from 64 to 16 bits...
Writing WAV...
Applying -0,13dB gain...
Creating file "output447.wav"...
eac3to processing took N ;-) seconds.
Done.

:thanks: a lot, madshi!

hubblec4

28th June 2010, 06:19

He's encoding AC3, not DTS, so that wouldn't help.

ok, i mean i use Megui to decode the dts-file and then encode to ac3.

ACrowley

28th June 2010, 12:10

@madshi

When you anyhow take a look into the DTS-HD Header because of the DialNorm Issue, can you add a Routine into eac3to to restore/strip the DTS-HD Header ?

..new libavcodec is faster on AC3 and EAC3, Great! EAC3/AC3/THD looks good with libav so far i can see

Jeff Flowerday

28th June 2010, 15:22

Laurent

28th June 2010, 17:53

Kurtnoise

28th June 2010, 19:43

eac3to v3.22 released

* libav E-AC3 decoder now supports spectral extension[/code]
The updated ffmpeg/libav libraries are quite a big change. It would be nice if you guys could do some libav decoder testing with various codecs (AC3, E-AC3, DTS, MP1, MP2, MP3, MLP, TrueHD) to make sure that things still work as intended.
Decoding/Transcoding E-AC3 streams within spectral extension don't seem to work properly for me using libav decoder...

1/ eac3to input.ts 2:audio.ac3

TS, 1 video track, 2 audio tracks, 2 subtitle tracks, 0:00:13, 50i
1: h264/AVC, 1440x1080 50i (16:9)
2: E-AC3, French, 5.0 channels, 256kbps, 48kHz, dialnorm: -27dB, -2140ms
3: E-AC3, qaa, 2.0 channels, 128kbps, 48kHz, dialnorm: -23dB, -2132ms
4: Subtitle (DVB), French
5: Subtitle (DVB), French
a02 The Nero decoder doesn't seem to work, will use libav instead.
a02 Extracting audio track number 2...
a02 Removing E-AC3 dialog normalization...
a02 Decoding with libav/ffmpeg...
a02 Remapping channels...
a02 Applying RAW/PCM delay...
a02 The AC3 encoder received a non-supported data format (float, 5, 64, -).
Aborted at file position 1048576.

Same thing for encoding directly from eac3 :
2 / eac3 input.eac3 output.ac3

E-AC3, 5.0 channels, 0:00:11, 256kbps, 48kHz
The Nero decoder doesn't seem to work, will use libav instead.
Decoding with libav/ffmpeg...
Remapping channels...
The AC3 encoder received a non-supported data format (float, 5, 64, -).
Aborted at file position 262144.

>> sample (http://www.mediafire.com/?wmynyimyzqx) <<

All is fine using FFmpeg w/ ffac3 or piping aften...

b66pak

28th June 2010, 20:03

Decoding/Transcoding E-AC3 streams within spectral extension don't seem to work properly for me using libav decoder...

this work perfect for me:

eac3to input.ts 2:audio.wav

the result:

General
Complete name : F:\test_eac3_spectral.ts.wav
Format : Wave
File size : 7.36 MiB
Duration : 10s 724ms
Overall bit rate : 5 760 Kbps

Audio
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Unsigned
Codec ID : 00001000-0000-0100-8000-00AA00389B71
Codec ID/Hint : Microsoft
Duration : 10s 724ms
Bit rate mode : Constant
Bit rate : 5 760 Kbps
Channel(s) : 5 channels
Channel positions : Front: L C R, Side: L R
Sampling rate : 48.0 KHz
Stream size : 7.36 MiB (100%)

the error in encoding to .ac3 is related to 5.0 channels...
_

Blue_MiSfit

28th June 2010, 20:06

Hi Madshi,

Thanks for reading my bug report, and being willing to take the time to fix the problem!

Here's the samples you requested for Battleship Potemkin:
http://www.mediafire.com/?mxdznd5gumd

Thanks again!
Derek

Kurtnoise

28th June 2010, 20:37

this work perfect for me:

eac3to input.ts 2:audio.wav

the result:

General
Complete name : F:\test_eac3_spectral.ts.wav
Format : Wave
File size : 7.36 MiB
Duration : 10s 724ms
Overall bit rate : 5 760 Kbps

Audio
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Unsigned
Codec ID : 00001000-0000-0100-8000-00AA00389B71
Codec ID/Hint : Microsoft
Duration : 10s 724ms
Bit rate mode : Constant
Bit rate : 5 760 Kbps
Channel(s) : 5 channels
Channel positions : Front: L C R, Side: L R
Sampling rate : 48.0 KHz
Stream size : 7.36 MiB (100%)

the error in encoding to .ac3 is related to 5.0 channels...
_
nope...as I said, all is fine with FFmpeg. I mean transcoding eac3 to ac3 of course.

Snowknight26

28th June 2010, 20:49

mrr19121970

28th June 2010, 21:04

I don't think that's the issue in this case. The problem might be because the 2 files do not contain the same track structure.

Mike

Will you test this for me with Mary & Max

http://forum.slysoft.com/showpost.php?p=268824&postcount=90

MikeEby

28th June 2010, 23:13

Will you test this for me with Mary & Max

http://forum.slysoft.com/showpost.php?p=268824&postcount=90

Sure, I tested and it doesn't appear to work..It does not report a DTS-HD audio track in the playlist. I didn't see that madshi fixed it in his release notes.

eac3to v3.22
command line: "C:\eac3to\eac3to.exe" "D:\BDMV\STREAM" -log="C:\Program Files\ACD\Another EAC3to GUI\log.txt"
------------------------------------------------------------------------------
1) 00000.mpls, 00023.m2ts+00000.m2ts, 1:32:26
- Chapters, 17 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz

2) 00007.mpls, 00007.m2ts, 0:22:04
- MPEG2, 480i60 /1.001 (16:9)
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz

3) 00002.mpls, 00005.m2ts, 0:15:48
- Chapters, 6 chapters
- MPEG2, 480i60 /1.001 (4:3)
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz

eac3to v3.22
command line: "C:\eac3to\eac3to.exe" 1) -log="C:\Program Files\ACD\Another EAC3to GUI\log.txt"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 2 subtitle tracks, 0:00:10, 24p /1.001
1: Chapters, 17 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
4: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
5: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
6: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
7: Subtitle (PGS), Spanish
8: Subtitle (PGS), English

Mike

Laurent

29th June 2010, 00:24

nope...as I said, all is fine with FFmpeg. I mean transcoding eac3 to ac3 of course.

Transcoding 5.1 E-AC3 track with spectral extension to 5.1 AC3 track seems to work well now with eac3to.

Transcoding 5.0 E-AC3 track with spectral extension to 5.0 AC3 track does not work. But this is probably more a problem with encoding only 5 channels than deconding 5.0 E-AC3 track with spectral extension.
If we could have a fix to be able to encode 5.0 tracks in AC3, that would be great because in the past one of our french channels used 5.0 E-AC3 audio instead of 5.1 E-AC3. So currently some of our old recordings are not yet managed by eac3to.

Laurent

29th June 2010, 00:39

this work perfect for me:

eac3to input.ts 2:audio.wav

the result:

General
Complete name : F:\test_eac3_spectral.ts.wav
Format : Wave
File size : 7.36 MiB
Duration : 10s 724ms
Overall bit rate : 5 760 Kbps

Audio
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Unsigned
Codec ID : 00001000-0000-0100-8000-00AA00389B71
Codec ID/Hint : Microsoft
Duration : 10s 724ms
Bit rate mode : Constant
Bit rate : 5 760 Kbps
Channel(s) : 5 channels
Channel positions : Front: L C R, Side: L R
Sampling rate : 48.0 KHz
Stream size : 7.36 MiB (100%)

the error in encoding to .ac3 is related to 5.0 channels...
_

Yes, same result, converting to WAV is done by eac3to without errors and the produced WAV contains 5 channels.
Error occurs when trying to encode in AC3.

Momber

29th June 2010, 02:24

Here is a 50MB sample of dtsma 3.0, as requested. The first 30 seconds are silent, this is normal.

Click me (http://fono.jp/uploader/src/file_1128.rar)

Midzuki

29th June 2010, 02:53

eac3to v3.22
command line: "C:\Program Files (x86)\eac3to\eac3to.exe" "S:\Temp\RipBot264temp\job2\2_audio_Japanese.dts" "S:\Temp\Profound.ac3" -192
------------------------------------------------------------------------------
DTS, 1.0 channels, 2:53:28, 24 bits, 768kbps, 48kHz
Decoding with ArcSoft DTS Decoder...
The ArcSoft DTS Decoder reported an error while decoding. <ERROR>
Aborted at file position 262144. <ERROR>

Apart from the "imperfections" :devil: of the ArcSoft Decoder,
and just as a side/"off-topic" note, :

whoever has created a DTS mono stream at 768kbps is a...

(place your favorite derogatory-expression here);

255kbps would have been more-than-adequate. :helpful:

P.S.:

try "-libav" instead of the ArcSoft filter,
or (as a "last resort") ffmpeg.

Kurtnoise

29th June 2010, 06:24

It's because its only 5 channels. The implementation of Aften that eac3to uses doesn't take non-standard channel masks. If you have eac3to output to stdout, you can pipe the data to standalone Aften and it'll work.

something like this ?

eac3to.exe input.ts 2:output.stdout | aften.exe -readtoeof 1 - aften.ac3

if yes, this doesn't seem to work...

Snowknight26

29th June 2010, 06:45

Try something like this instead:
eac3to input.ts 2: stdout.wav | aften -b 320 - output.ac3

jasonwc

29th June 2010, 06:55

I found odd behavior with eac3to 3.22 and the Slumdog Millionaire US AVC DTS-HD MA Blu-Ray. I converted the DTS-HD MA 24 bit track to a 16 bit FLAC with eac3to 3.22 and got a FLAC with a bitrate of 1331 Kbps. I used MKVMerge to create a MKV, and found that the video stalled on playback using ffdshow (libavcodec) to decode the FLAC. I had no such issue with madFLAC or the internal MPC-HC decoder. However, I was able to replicate the behavior in MPC-HC and Mediaporal using the ffdshow FLAC decoder.

I thought the issue might be corruption in the outputted FLAC because the video stream without audio played fine. Therefore, I demuxed the BD again and got an identically sized FLAC. Same problem.

I decided to redo the remux with eac3to 3.19 and identical settings. The FLAC now had a bitrate of 1393 Kbps. After remuxing with MKVMerge, I discovered the problem was gone. The remuxed video played back perfectly without stalling upon MPC-HC startup. I'm not sure if this is an ffdshow or eac3to problem, but it's weird that the FLAC bitrate changed, and the problem was fixed. The FLAC track was identical after both demuxes with 3.22, and all three demuxes used the same version of the FLAC encoder (1.2.1).

eac3to 3.22 log:

eac3to v3.22
command line: "C:\Users\Jason\Downloads\eac3to\eac3to.exe" "F:\In Process\Slumdog Millionaire 2008 Blu-ray 1080p AVC DTS-HD 5.1\" 1) 2: "D:\In Process\video.2.mkv" -seekToIFrames 3: "D:\In Process\audio.english.3.flac" -down16 4: "D:\In Process\audio.english.4.ac3" 5: "D:\In Process\audio.english.5.ac3" 1: "D:\In Process\chapters.txt" 7: "D:\In Process\subtitle.english.7.sup" -log="L:\Movies\Slumdog Millionaire 2008 1080p BluRay AVC FLAC-Remux\Slumdog Millionaire 2008 1080p BluRay AVC FLAC-Remux.log.txt" -progressnumbers
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 3 subtitle tracks, 2:00:38, 24p /1.001
1: Chapters, 28 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
4: AC3, English, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
5: AC3, English, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
6: AC3, French, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
7: Subtitle (PGS), English
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
Creating file "D:\In Process\chapters.txt"...
[a04] Extracting audio track number 4...
[a05] Extracting audio track number 5...
[s07] Extracting subtitle track number 7...
[a05] Removing AC3 dialog normalization...
[v02] Extracting video track number 2...
[a04] Removing AC3 dialog normalization...
[a03] Extracting audio track number 3...
[v02] Muxing video to Matroska...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Reducing depth from 24 to 16 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "D:\In Process\audio.english.3.flac"...
[a05] Creating file "D:\In Process\audio.english.5.ac3"...
[a04] Creating file "D:\In Process\audio.english.4.ac3"...
[s07] Creating file "D:\In Process\subtitle.english.7.sup"...
[a03] The original audio track has a constant bit depth of 24 bits.
[a03] The processed audio track has a constant bit depth of 16 bits.
Added fps value (24 /1.001) to MKV header.
Video track 2 contains 173544 frames.
Subtitle track 7 contains 1385 captions.
eac3to processing took 17 minutes, 37 seconds.
Done.

eac3to 3.19 log:

eac3to v3.19
command line: "C:\Users\Jason\Downloads\eac3to\eac3to.exe" "F:\In Process\Slumdog Millionaire 2008 Blu-ray 1080p AVC DTS-HD 5.1\" 1) 2: "D:\In Process\video.2.mkv" -seekToIFrames 3: "D:\In Process\audio.english.3.flac" -down16 4: "D:\In Process\audio.english.4.ac3" 5: "D:\In Process\audio.english.5.ac3" 1: "D:\In Process\chapters.txt" 7: "D:\In Process\subtitle.english.7.sup" -log="K:\Movies\Slumdog Millionaire 2008 1080p BluRay AVC FLAC-Remux\Slumdog Millionaire 2008 1080p BluRay AVC FLAC-Remux.log.txt" -progressnumbers
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 3 subtitle tracks, 2:00:38, 24p /1.001
1: Chapters, 28 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
4: AC3, English, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
5: AC3, English, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
6: AC3, French, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
7: Subtitle (PGS), English
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
Creating file "D:\In Process\chapters.txt"...
[s07] Extracting subtitle track number 7...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[a05] Extracting audio track number 5...
[a05] Removing AC3 dialog normalization...
[a04] Extracting audio track number 4...
[a04] Removing AC3 dialog normalization...
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Reducing depth from 24 to 16 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "D:\In Process\audio.english.3.flac"...
[a04] Creating file "D:\In Process\audio.english.4.ac3"...
[a05] Creating file "D:\In Process\audio.english.5.ac3"...
[s07] Creating file "D:\In Process\subtitle.english.7.sup"...
[a03] The original audio track has a constant bit depth of 24 bits.
[a03] The processed audio track has a constant bit depth of 16 bits.
Added fps value (24 /1.001) to MKV header.
Video track 2 contains 173544 frames.
Subtitle track 7 contains 1385 captions.
eac3to processing took 17 minutes, 57 seconds.
Done.

Laurent

29th June 2010, 07:26

Midzuki

29th June 2010, 07:38

@ jasonwc:

if neither MPC-HC's decoder nor madFlac complained, but ffdshow stumbled, then the problem is in ffdshow, not in eac3to, IMHO.
BTW, you didn't mention what version of ffdshow you're using.

{
Anyway, and JMNSHO again, ffdshow became even-buggier with the advent of the "DXVA-era". :devil: And I don't intend to talk again with people who believe certain *design flaws* "must be preserved" for "historical reasons". :rolleyes:
}

jasonwc

29th June 2010, 08:09

tebasuna51

29th June 2010, 08:53

something like this ?

eac3to.exe input.ts 2:output.stdout | aften.exe -readtoeof 1 - aften.ac3

if yes, this doesn't seem to work...
Try something like this instead:
eac3to input.ts 2: stdout.wav | aften -b 320 - output.ac3
Don´t forget the -readtoeof 1 because a wav 5.0 24 bits can be > 4GB.
eac3to.exe input.ts 2: stdout.wav | Aften.exe -readtoeof 1 - aften.ac3
The default aften bitrate for 5.0 is 448 Kb/s.

Midzuki

29th June 2010, 08:57

The default aften bitrate for 5.0 is 448 Kb/s.

Probably he was thinking of NeroAacEnc when he wrote the Aften command-line... :D

tebasuna51

29th June 2010, 13:58

No, the parameter -b 320 is correct. And maybe recommended because:
2: E-AC3, French, 5.0 channels, 256kbps, 48kHz, dialnorm: -27dB, -2140ms
We can select between ...,192,224,256,320,384,448,512,576,640 Kb/s.

asarian

29th June 2010, 14:11

Using eac3to, any way I can downconvert a ~25GB 192 Khz (!) LPCM track (Akira Blu-Ray) to something more human, like, say, 48 Khz?

nurbs

29th June 2010, 14:41

Yes you can do that. See the manual or the first post as to how.

Thunderbolt8

29th June 2010, 16:00

whats the best solution again to deal with 7.1 DTS-HD MA tracks which are described as "strange setup" ? I want to convert the track to flac, so what are my options here?

is this information here still correct? http://forum.doom9.org/showthread.php?p=1361493#post1361493 using sonic to convert it to 5.1 makes most sense atm?

Laurent

29th June 2010, 18:56

I noticed yesterday a similar problem with many macroblocks in video if I demux video in MKV file rather than a H.264 file. As my process is a 2 steps process using first eac3to and then TsMuxer to produce a M2TS file, I cannot be sure if the problem with video in MKV container is due to eac3to or TsMuxer.

I come back to this problem with more explanations and tests. In fact, the problem is not new, I just verified and I have the same strange behaviour with eac3to v3.21 and eac3to v3.18.

Here is my scenario:
1 - eac3to to demux video in MKV container and convert audio to AC3
2 - TsMuxer to mux audio and video in a M2TS container
3 - playback of the M2TS file with the PS3

Video is H.264. When the file is decoded by the PS3, I have many macroblocks, and the result is the same whatever the version of eac3to (3.18, 3.21 and 3.22).
If at step 1, instead of demuxing video in MKV container, I demux the video in a H.264 file, then the playback with the PS3 is fine without macroblocks.

Note that all these files (M2TS and MKV) are played correctly on the PC.

So the problem is relative to the usage of MKV container, either when produced by eac3to, or when used as input by TsMuxer. And the problem is noticeable only when the final file is played with the PS3.

Now that I discovered this problem, I will of course avoid using MKV container.

b66pak

29th June 2010, 19:19

eac3to use haali media splitter (http://haali.su/mkv/) for demuxing...try other version (11-1-2009 (http://www.videohelp.com/download/MatroskaSplitter110109.exe))

Snowknight26

29th June 2010, 19:39

No it doesnt.

Laurent

29th June 2010, 20:24

eac3to use haali media splitter (http://haali.su/mkv/) for demuxing...try other version (11-1-2009 (http://www.videohelp.com/download/MatroskaSplitter110109.exe))

This is the version I am using.

asarian

29th June 2010, 20:26

I come back to this problem with more explanations and tests. In fact, the problem is not new, I just verified and I have the same strange behaviour with eac3to v3.21 and eac3to v3.18.

Here is my scenario:
1 - eac3to to demux video in MKV container and convert audio to AC3
2 - TsMuxer to mux audio and video in a M2TS container
3 - playback of the M2TS file with the PS3

Video is H.264. When the file is decoded by the PS3, I have many macroblocks, and the result is the same whatever the version of eac3to (3.18, 3.21 and 3.22).
If at step 1, instead of demuxing video in MKV container, I demux the video in a H.264 file, then the playback with the PS3 is fine without macroblocks.

Note that all these files (M2TS and MKV) are played correctly on the PC.

So the problem is relative to the usage of MKV container, either when produced by eac3to, or when used as input by TsMuxer. And the problem is noticeable only when the final file is played with the PS3.

Now that I discovered this problem, I will of course avoid using MKV container.
Hmm, you wouldn't be using an older version of CoreAVC, would ya? What you describe is a known issue with it. Just make sure you let ffdshow handle H264, and you should be fine. I stopped using CoreAVC for this precise reason.

Laurent

29th June 2010, 22:38

Hmm, you wouldn't be using an older version of CoreAVC, would ya? What you describe is a known issue with it. Just make sure you let ffdshow handle H264, and you should be fine. I stopped using CoreAVC for this precise reason.

But the playback problem is with the PlayStation, not the PC. And when demuxing with eac3to, I don't think a decoder is involved.
By the way, CoreAVC is not installed on my PC.

GoodzMastaJ

29th June 2010, 23:15

I've extracted the TrueHD track from Evangelion 1.11 blu-ray using eac3to. ffdshow reports it as having channels as shown in the screenshot below.

http://stuff.damagedgoodz.net/forumposts/7chtruehd.png

Note the presence of the side channels and absence of an LFE channel.
In actuality, there are no side channels but there is an LFE and a rear center channel which I suppose would make this a 6.1 track (rather than the 7.0 I'm getting from ffdshow). Using the mixer, I found the LFE channel is playing from what ffdshow thinks is Back Left (all that comes out of BackL is rumbling).

I'm not sure where the bug is, if it's in eac3to or ffdshow. Any ideas on how to pinpoint it? I made a similar post on the ffdshow-tryouts forum to stir up some suggestions there as well.

Here is a short sample that has a pretty good demonstration.
http://www.mediafire.com/?mmzydemji5m

Killroy™

29th June 2010, 23:40

Using eac3to, any way I can downconvert a ~25GB 192 Khz (!) LPCM track (Akira Blu-Ray) to something more human, like, say, 48 Khz?

OK... I'll be the first to ask... Why would you want to take such a beautiful audio track and throw away 2/3 of the data?

Atak_Snajpera

30th June 2010, 00:12

the same reason why 24bit is converted to 16bit. Human hearing has limitations.

dansrfe

30th June 2010, 01:28

Currently, are there any speaker systems/receiver's that actually support 192Khz? Now THAT would be the 9th Wonder of the World.

Killroy™

30th June 2010, 01:44

Currently, are there any speaker systems/receiver's that actually support 192Khz? Now THAT would be the 9th Wonder of the World.

Most modern AVR's support 192Khz with no problems.

Midzuki

30th June 2010, 02:01

Currently, are there any speaker systems/receiver's that actually support 192Khz?

Digital: In the world of today, 192kHz audio should already be something quite trivial. :)

Analog: In the world of today, speakers and audio amplifiers that support up to 96kHz still are very rare, and (IMHO) somewhat pointless...

I'm not sure where the bug is, if it's in eac3to or ffdshow.

Both eac3to and ffdshow use libavcodec for TrueHD decoding,
however I daresay eac3to "is less unreliable than" ffdshow. :devil:
I'd recommend converting the original 6.1 audio to a 6.1 .WAV
(assuming disk space is not an issue...)

Evangelion 1.11

I didn't see it, and I didn't like it. :D

(Yes man, quite often the remakes are quite inferior to the original versions.)

dansrfe

30th June 2010, 03:03

Midzuki

30th June 2010, 03:17

Thanks to some "oddities" of the English language, and also thanks to the marketing people :devil:, the word "speakers" became somewhat... undefined. :D From what I've read so far, it's never very clear when the word loudspeaker refers to the (so-called) "driver" itself, or to the "box" (enclosure) which contains one or more "drivers"... :confused: Anyway, usually the expression "digital speakers" applies to a set of "powered" drivers which includes a Digital2Analog converter. ACTUAL digital speakers are an entirely-different beast, and there is an article on Wikipedia (http://en.wikipedia.org/wiki/Digital_speakers) about them. HTH.

Killroy™

30th June 2010, 03:35

You know this might be a n00b question but I still don't understand exactly what a digital speaker is and what connectors are used with it. In a normal home theater setup where, for example, a htpc is outputting digital S/PDIF audio via HDMI (coming from the sound card or mobo sound module built-in to the graphics card), then is transmitted to the receiver all through a single HDMI cable, then then the receiver processes and decodes the audio if bitstreaming. But after that doesn't the receiver send a analog signal to the actual speakers? I'm totally confused about this. What exactly is a digital speaker and what is an analog speaker? What sort of connectors does a digital speaker have?

By "digital" we usually mean HDMI connectors. LPCM or bitstream of the audio source. "Analog" usually refers to the old multi-channel RCA-plugs used in older AVR's. Audio from the HTPC or STB comes from the analog output (HTPC) or the multi-channel analog outputs of the STB's. Since only very expensive HTPC analog cards could output 192Khz (properly) it is unsure if you will get a un-molested 192Khz output.

No such issues with the HDMI outputs.

The signal from the AVR to the speakers is always the same (analog) but can be degraded with lower quality wire. But even modest 24GA wires should have no problems with 192Khz at very short throws. Always use 12GA wire for optimum quality.

dansrfe

30th June 2010, 04:55

ah. I had this idea that digital speakers were some obscure, advanced, audio professional type stuff.

Xorp

30th June 2010, 06:36

Decoding this 6.1 DTS-MA track with eac3to 3.22 and Arcsoft 1.1.0.7 results in distortion: http://stfcc.org/misc/7channels.dtshd

tebasuna51

30th June 2010, 11:22

Decoding this 6.1 DTS-MA track with eac3to 3.22 and Arcsoft 1.1.0.7 results in distortion: http://stfcc.org/misc/7channels.dtshd
Read this post (http://forum.doom9.org/showthread.php?p=1409645#post1409645) and next (the channelmapping is solved with eac3to v3.21)

Test also my sample in the post using Arcsoft 1.1.0.0.

tebasuna51

30th June 2010, 12:10

I've extracted the TrueHD track from Evangelion 1.11 blu-ray using eac3to. ffdshow reports it as having channels as shown in the screenshot below.

Note the presence of the side channels and absence of an LFE channel.
In actuality, there are no side channels but there is an LFE and a rear center channel which I suppose would make this a 6.1 track (rather than the 7.0 I'm getting from ffdshow). Using the mixer, I found the LFE channel is playing from what ffdshow thinks is Back Left (all that comes out of BackL is rumbling).

I'm not sure where the bug is, if it's in eac3to or ffdshow. Any ideas on how to pinpoint it? I made a similar post on the ffdshow-tryouts forum to stir up some suggestions there as well.
Seems a ffdshow problem because eac3to decode the TrueHD fine to a wav 6.1 with MaskChannel 0x070F (FL FR FC LFE BC SL SR).
Try play the wav file.

b66pak

1st July 2010, 20:31

WARNING...the new mkvtoolnix 4.1.0 "header removal compression" (http://www.bunkus.org/videotools/mkvtoolnix/faq.html#header_removal_compression) make impossible decoding/demuxing for eac3to from mkv...

eac3to v3.22
command line: eac3to "mkvtoolnix.4.1.0.mkv" -demux
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 1 subtitle track, 0:01:16, 24p /1.001
1: h264/AVC, 1280x536 23.975p (160:67)
2: DTS, 5.1 channels, 48kHz
3: Subtitle (SRT)
[v01] The video bitstream is encoded in a non-standard framerate. <WARNING>
[v01] The video bitstream framerate field doesn't match the container framerate. <WARNING>
Bitstream parsing for track 2 failed. <WARNING>
Demuxing this track may still produce correct results - or not. <WARNING>
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Creating file "mkvtoolnix.4.1.0 - 2 - DTS, 5.1 channels, 48kHz.dts"...
[v01] Creating file "mkvtoolnix.4.1.0 - 1 - h264, 1280x536 23.975p.h264"...
[s03] Extracting subtitle track number 3...
[s03] Creating file "mkvtoolnix.4.1.0 - 3 - Subtitle (SRT).srt"...
Video track 1 contains 1819 frames.
eac3to processing took 2 seconds.
Done.

eac3to v3.22
command line: eac3to "mkvtoolnix.4.1.0 - 2 - DTS, 5.1 channels, 48kHz.dts" mkvtoolnix.4.1.0.wav
The format of the source file could not be detected.

_

dansrfe

1st July 2010, 20:34

XhmikosR

1st July 2010, 22:19

You can use Nero Lite/Micro. http://forum.doom9.org/showthread.php?p=1398214&highlight=lite#post1398214

If you want to experiment, you can manually find the needed files to get it to work.

ACrowley

2nd July 2010, 07:43

Can anyone please post a simple and effective guide to getting Nero 7 to work with eac3to. I have the software and have bought it with the Nero Blu-ray / HD DVD plugin but I do not want to install it just for a few plugins. I prefer extracting/using only what is necessary for eac3to to utilize nero 7 for AC3 decoding. Another thing I do not understand about AC3 encoding with eac3to and libav is that why does it say "Reducing bit depth from 64bits to 24 bits" while processing?

I wouldnt use NERO AC3/EAC3 Decoder
http://forum.doom9.org/showthread.php?p=1404212#post1404212
Looks like DRC isnt disabled properly

Better use libav..works without Problems. And i dont think you hear big a Quality Difference generally between Nero and libavcodec on AC3/EAC3

"Reducing bit depth from 64bits to 24 bits" while processing means that the libanvcodec internal processing runs at 64Bit and eac3to reduces the output to 24bit. Its not a Problem ,its just the Process

shogo_kawada

2nd July 2010, 08:36

Hi, I tried to post these questions in the newbies forum but I wasn't lucky, so I'll try again here, which maybe would have been the right place to begin with.

I'm willing to rip my HD-DVDs to multi-audio MKVs. I have an AV-receiver which can decode HD-audio, and the files will be played on a Dune Base HD, which, AFAIK, CAN bitstream HD audio from MKVs, so I don't need any kind of encoding nor transcoding of the tracks. I just need to demux them. So here are my doubts:

1. About dialog normalization. Should I keep it or not? If not, why is it there (in the disc) in the first place? I'd like the MKVs to sound the closest to the actual disc playback I can get.

2. About delay. I read in a guide that eac3to doesn't fix delay when simply demuxing. Is that true? I tried some demuxing and the log said delay WAS removed. Did I do anything wrong or is it supposed to happen even with simple demux?

3. Just to be sure: to simple demux I just have to put *.eac3, *.dtshd, *.thd as name for the target track, am I right?

Ok, that's it, thanks for your help :)

nibus

2nd July 2010, 08:51

Can eac3to demux DTS Express audio tracks?

TinTime

2nd July 2010, 12:11

1. About dialog normalization. Should I keep it or not? If not, why is it there (in the disc) in the first place? I'd like the MKVs to sound the closest to the actual disc playback I can get.

I'd keep it but this is just a matter of personal preference. It only affects the volume of the decoded audio.

2. About delay. I read in a guide that eac3to doesn't fix delay when simply demuxing. Is that true? I tried some demuxing and the log said delay WAS removed. Did I do anything wrong or is it supposed to happen even with simple demux?

It will fix the delay for everything except for TrueHD tracks where the delay required will be added to the output filename.

3. Just to be sure: to simple demux I just have to put *.eac3, *.dtshd, *.thd as name for the target track, am I right?

Yes, or you can just use the -demux switch.

mini-moose

2nd July 2010, 12:43

sorry if this was discussed before but I can't find it.

I'm trying to convert a 24.000fps DTS to 23.976fps.

when using -slowdown eac3to assumes my source is 25.000 thus the slowdown is by 4% which is not what I'm aiming for.

I tried to define the fps of the DTS by creating a copy using the -24.000 switch. then ran the slowdown again and it still treated it as 25.000.

-changeto didn't seem to work either:

"C:\eac3to\eac3to.exe" C:\one.dts two.dts -changeTo24.000
Was asked to modify source to 24.000, but the original FPS value is unknown.
Please specify the original FPS value (e.g. option "-23.976").

"C:\eac3to\eac3to.exe" C:\one.dts two.dts -24.000 -changeTo24.000
"C:\eac3to\eac3to.exe" C:\one.dts two.dts -24.000 -changeTo23.976
both got me a "Please specify the source and dest files first and then the options."

I guess I'm doing something wrong. Hopefully someone can help :)

TinTime

2nd July 2010, 13:36

"C:\eac3to\eac3to.exe" C:\one.dts two.dts -24.000 -changeTo23.976

I can't see anything wrong with this. Does it work if you put quotes around the filenames?

Or how about two steps?
"C:\eac3to\eac3to.exe" C:\one.dts two.wav -24.000 -changeTo23.976
"C:\eac3to\eac3to.exe" two.wav two.dts
Which step fails?

mini-moose

2nd July 2010, 15:01

[QUOTE] TinTime
I can't see anything wrong with this. Does it work if you put quotes around the filenames?

Or how about two steps?
Code:

"C:\eac3to\eac3to.exe" C:\one.dts two.wav -24.000 -changeTo23.976
"C:\eac3to\eac3to.exe" two.wav two.dts

thanks. I will give it a try. but would that even do the slowdown I need if it works ? I figured I might have to "tag" the original dts somehow as 24fps so -slowdown will perform a slowdown from 24 tp 23.976 and from 25.

mini-moose

2nd July 2010, 15:02

TinTime:
I can't see anything wrong with this. Does it work if you put quotes around the filenames?

Or how about two steps?
Code:

"C:\eac3to\eac3to.exe" C:\one.dts two.wav -24.000 -changeTo23.976
"C:\eac3to\eac3to.exe" two.wav two.dts

thanks. I will give it a try. but would that even do the slowdown I need if it works ? I figured I might have to "tag" the original dts somehow as 24fps so -slowdown will perform a slowdown from 24 tp 23.976 and from 25.

tebasuna51

2nd July 2010, 19:11

.... but would that even do the slowdown I need if it works ? I figured I might have to "tag" the original dts somehow as 24fps so -slowdown will perform a slowdown from 24 tp 23.976 and from 25.
An audio file (dts, ac3, ...) don't have tags about the video fps, only have a duration. And aply a -24.000 -changeTo23.974 means only modify the duration with a coefficient of 1.001

With audio files -slowdown means modify the duration with a coefficient of (1001/960) always.

Use:
"C:\eac3to\eac3to.exe" "C:\one.dts" "C:\two.ac3" -24.000 -changeTo23.976
or
"C:\eac3to\eac3to.exe" "C:\one.dts" "C:\two.wavs" -24.000 -changeTo23.976
and use a external dts encoder.

shogo_kawada

3rd July 2010, 08:24

I'd keep it but this is just a matter of personal preference. It only affects the volume of the decoded audio.
Ok, that's what I was thinking too, but just for the sake of knowledge: is there a reason for dialog normalization to be there (in the disc) in the first place? Why do most of people remove it? Why do studios put it there?

It will fix the delay for everything except for TrueHD tracks where the delay required will be added to the output filename.
Ok, I just tried to demux some DD+ tracks and indeed the delays were fixed, but the log informed me that a remaning delay of -8 ms couldn't be fixed. To fix the last unfixed gap, can I just add that exact delay in the track properties in MKVMerge?

Yes, or you can just use the -demux switch.
It's surely much simpler this way since I can avoid the very long command line. Just a doubt: will the video track be handled properly this way? I read somewhere that it would be better to demux video tracks to mkv for eac3to to properly deal with framerate, pulldown flags, and stuff like this (which I actually don't really get, but still...).

Thanks a lot for your help ;)

TinTime

3rd July 2010, 10:40

Ok, that's what I was thinking too, but just for the sake of knowledge: is there a reason for dialog normalization to be there (in the disc) in the first place? Why do most of people remove it? Why do studios put it there?

The reason it's there is to make all audio from different sources sound roughly the same volume. The problem is that it's not universally used. When you ask "why do studios put it there" it might be more reasonable to ask "why do some studios not use it".

Ok, I just tried to demux some DD+ tracks and indeed the delays were fixed, but the log informed me that a remaning delay of -8 ms couldn't be fixed. To fix the last unfixed gap, can I just add that exact delay in the track properties in MKVMerge?

You can if you think you can hear an 8ms delay! Different audio types come with different frame sizes which are the minimum chunks you can edit with. EAC3 is (I think) the same as AC3 in that it has a 32ms frame size. So if the original audio is 40ms out of sync then the closest you can get to fixing the delay is to take 32ms off, leaving a delay of 8ms that can't be fixed.

It's surely much simpler this way since I can avoid the very long command line. Just a doubt: will the video track be handled properly this way? I read somewhere that it would be better to demux video tracks to mkv for eac3to to properly deal with framerate, pulldown flags, and stuff like this (which I actually don't really get, but still...).

I've got a feeling that there used to be a problem with mkvmerge and raw AVC streams, or something like that, but there isn't any more. I always demux my video with eac3to and then mux it to mkv with mkvmerge and I don't have any problems though.

shogo_kawada

3rd July 2010, 20:03

The reason it's there is to make all audio from different sources sound roughly the same volume. The problem is that it's not universally used. When you ask "why do studios put it there" it might be more reasonable to ask "why do some studios not use it".

Ok so it's something which DO have some reason to be there and for us to keep when demuxing. That's the answer I was looking for, -keepdialnorm it is!

You can if you think you can hear an 8ms delay! Different audio types come with different frame sizes which are the minimum chunks you can edit with. EAC3 is (I think) the same as AC3 in that it has a 32ms frame size. So if the original audio is 40ms out of sync then the closest you can get to fixing the delay is to take 32ms off, leaving a delay of 8ms that can't be fixed.

Eheh you're right, I don't think I can hear an 8ms delay. BTW thanks again for the explanation, and if I get it right that means that I'll never end with a delay over 31 ms, which I think won't be an issue either.

I've got a feeling that there used to be a problem with mkvmerge and raw AVC streams, or something like that, but there isn't any more. I always demux my video with eac3to and then mux it to mkv with mkvmerge and I don't have any problems though.
Ok, so I got nothing left to worry :). It's time to start some conversion :thanks:

SomeJoe

3rd July 2010, 22:38

Eheh you're right, I don't think I can hear an 8ms delay. BTW thanks again for the explanation, and if I get it right that means that I'll never end with a delay over 31 ms, which I think won't be an issue either.

Actually, maximum possible unfixable offset is 16 msec because if the offset in one direction is over 16 msec, then remove/add one frame will give 32-(original offset) msec offset in the other direction.

So if audio is 20 msec behind video, then remove one audio frame and now audio is 12 msec ahead of video.

shogo_kawada

4th July 2010, 09:47

Actually, maximum possible unfixable offset is 16 msec because if the offset in one direction is over 16 msec, then remove/add one frame will give 32-(original offset) msec offset in the other direction.

So if audio is 20 msec behind video, then remove one audio frame and now audio is 12 msec ahead of video.
That's even better, awesome! Definetely no need to worry about the audio offset then.

Thunderbolt8

4th July 2010, 12:18

tebasuna51

4th July 2010, 12:53

is this still a problem with eac3to (or for players)?
Is a problem for eac3to (at least DTS tracks, AC3 seems ok).

PC players tested without problems: mpc-hc(Haali), vlc and KMP

Standalone player tested without problems: Xtreamer

BTW, I don't know how can compress AC3 headers. There are different values in headers: CRC's, RF protection, DRC values, ...

dbone1026

5th July 2010, 00:02

Madshi,

I have a Blu Ray disc of a tv show (Supernatural Season 1, Disc 1). The disc has the first 6 episodes of the tv show. When I run the disc through eac3to (using Clown_BD and/or Ripbot) eac3to only picks up 3 playlists (the 1st playlist is of all episodes, the 2nd playlist is of episode 1, and the 3rd playlist is of episode 2). Episode 3-6 don't show anywhere. If I use MakeMKV I can see all episodes listed out individually to select. Is there anything you can think of that would be causing this issue? I can just copy the individual episodes which are each m2ts files from the disc to my harddrive and then from there run through ripbot so no big deal, but thought it worth asking. Below is my eac3to log, let me know if I can provide anything else.

eac3to v3.22
command line: "C:\Program Files\Clown_BD_v0.76\eac3to\eac3to.exe" "F:" -progressnumbers -LOG="C:\Users\DAMIAN\Documents\Clown_BD\LOGS\eac3to_PASS1_LOG.LOG"
------------------------------------------------------------------------------
1) 00100.mpls, 4:20:33
[34+35+36+37+38+39].m2ts
- Chapters, 41 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48kHz
- AC3, French, stereo, 48kHz
- AC3, German, stereo, 48kHz
- AC3, Spanish, stereo, 48kHz
- AC3, Portuguese, stereo, 48kHz
- AC3, Japanese, stereo, 48kHz
- AC3, Japanese, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz

2) 00401.mpls, 00034.m2ts, 0:46:25
- Chapters, 6 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48kHz
- AC3, French, stereo, 48kHz
- AC3, German, stereo, 48kHz
- AC3, Spanish, stereo, 48kHz
- AC3, Portuguese, stereo, 48kHz
- AC3, Japanese, stereo, 48kHz
- AC3, Japanese, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz

3) 00402.mpls, 00037.m2ts, 0:42:11
- Chapters, 7 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48kHz
- AC3, French, stereo, 48kHz
- AC3, German, stereo, 48kHz
- AC3, Spanish, stereo, 48kHz
- AC3, Portuguese, stereo, 48kHz
- AC3, Japanese, stereo, 48kHz
- AC3, Japanese, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz

AnryV

6th July 2010, 21:31

TrueHD/AC3, 5.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
Extracting TrueHD stream...
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Writing WAVs...
[libav] Substream 0 parity check failed <WARNING>
[libav] Substream 0 checksum failed <WARNING>
[libav] Substream 0 length mismatch. <WARNING>
The libav decoder reported error -1 while decoding. <ERROR>
Aborted at file position 262144. <ERROR>

eac3to v3.22
command line: eac3to.exe "F:\T4\fre.thd" "F:\T4\fre.wavs" -nero
------------------------------------------------------------------------------
TrueHD, 5.1 channels, 48kHz
Disabling DRC for Nero (E-)AC3 decoding...
Decoding with DirectShow (Nero Audio Decoder 2)...
The DirectShow audio decoder didn't accept the input stream. <ERROR>
Aborted at file position 262144. <ERROR>

It was under w7...

Under Windows XP SP3:
TrueHD, 5.1 channels, 48kHz
Disabling DRC for Nero (E-)AC3 decoding...
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 5.1 channels, 24 bits, 48kHz
Writing WAV...
Creating file "fre.wav"...
The original audio track has a constant bit depth of 16 bits.
Superfluous zero bytes detected, will be stripped in 2nd pass.
Starting 2nd pass...
Reading WAV...
Stripping zero bytes...
Writing WAV...
Creating file "fre.wav"...
eac3to processing took 9 seconds.
Done.

The decoding by libav under XP failed too.

What this can be?

Sample - http://multi-up.com/301756

mr.duck

9th July 2010, 20:29

eac3to cannot fully demux my TS files (from DVB-T, UK).

Unlike other programs like tsMuxeR, eac3to at least recognizes that there are subtitles there. But -demux only extracts the video and audio.

Also when I -demux some <path>\video.ts, it extracts the audio and video to the same folder as eac3to.exe, not the video file folder and no other way of setting the output to another folder. Pretty annoying.

Thunderbolt8

9th July 2010, 23:54

Is a problem for eac3to (at least DTS tracks, AC3 seems ok).

PC players tested without problems: mpc-hc(Haali), vlc and KMP

Standalone player tested without problems: Xtreamer

BTW, I don't know how can compress AC3 headers. There are different values in headers: CRC's, RF protection, DRC values, ...
hm can eac3to please be fixed that way that demuxing these know problematic filetypes from .mkv files works normally again? (video streams (avc/h264, vc-1, mpeg2, flac, dts, ac3, .ass subs)

otherwise there could be problems when muxing e.g. hdtv stuff and then deleting the .ts source afterwards only to find out that you then cannot demux one of those streams from the .mkv when you need to.
thanks!

tebasuna51

10th July 2010, 01:08

hm can eac3to please be fixed that way that demuxing these know problematic filetypes from .mkv files works normally again? (video streams (avc/h264, vc-1, mpeg2, flac, dts, ac3, .ass subs)
+1
But the the change only is for:
* mkvmerge: enhancement: mkvmerge uses header removal compression
by default for AC3, DTS, MP3, Dirac and MPEG-4 part 2 tracks.

otherwise there could be problems when muxing e.g. hdtv stuff and then deleting the .ts source afterwards only to find out that you then cannot demux one of those streams from the .mkv when you need to.
thanks!
We always can use MkvExtract.

Thunderbolt8

10th July 2010, 12:17

Ive had it in the past that flac tracks extracted by mkvextract could not be recognised by eac3to afterwards. so if this problem has been fixed, I wouldnt mind to use mkvextract.

tebasuna51

10th July 2010, 15:28

With old versions of MkvExtract the flac tracks was extracted in ogg container by default. Now is not extract in ogg container.

2010-05-28 Moritz Bunkus

* mkvextract: feature removal: Removed support for extracting FLAC
tracks into Ogg FLAC files. Instead they're always written into
raw FLAC files. The option "--no-ogg" has been removed as well.

Midzuki

10th July 2010, 15:37

MKVmerge can be used for copying a FLAC track to a .mka file.

73ChargerFan

10th July 2010, 19:27

I have a Blu Ray disc of a tv show (Supernatural Season 1, Disc 1). The disc has the first 6 episodes of the tv show. When I run the disc through eac3to (using Clown_BD and/or Ripbot) eac3to only picks up 3 playlists (the 1st playlist is of all episodes, the 2nd playlist is of episode 1, and the 3rd playlist is of episode
Cool show, I just watched all of season 5 last week.

Use BDInfo to identify the playlists for each episode (it can sort by time.) Then you can use eac3to on each.

Or, if you don't care about chapter times, you can demux the m2ts directly (sort by file size).

IIRC, eac3to is supposed to show playlists longer than 15 minutes, so this is probably a bug.

deathlord

11th July 2010, 12:05

Cool show, I just watched all of season 5 last week.

Use BDInfo to identify the playlists for each episode (it can sort by time.) Then you can use eac3to on each.

Or, if you don't care about chapter times, you can demux the m2ts directly (sort by file size).

IIRC, eac3to is supposed to show playlists longer than 15 minutes, so this is probably a bug.

You can also run eac3to on the playlist files, like
eac3to xyz.mpls
This way you see, which playlist corresponds to which .m2ts file.
The bigger Problem is, you don't know the right order for the episodes, i.e. you can't tell which ist the first.
There is no guarantee that either the .m2ts file name of the .mpls file name helps you there. I have not found a satisfying solution to this problem.

SamuriHL

11th July 2010, 19:39

You can also run eac3to on the playlist files, like
eac3to xyz.mpls
This way you see, which playlist corresponds to which .m2ts file.
The bigger Problem is, you don't know the right order for the episodes, i.e. you can't tell which ist the first.
There is no guarantee that either the .m2ts file name of the .mpls file name helps you there. I have not found a satisfying solution to this problem.

If you want a manual way to do it, install MPC-HC and open each MPLS in that. You should quickly see which episode corresponds to which MPLS. It's a pain, yes, but, it should work.

73ChargerFan

11th July 2010, 22:18

There are dozens and dozens of playlist MPLS files on some blu-rays. BDInfo will load the disk structure show immediately which mpls files are of interest.

Snowknight26

12th July 2010, 00:27

How does eac3to know how much gain to apply after clipping's been detected?

a05 Applying -0.27dB gain...

Seems a bit strange to apply only a .27dB gain.

tebasuna51

12th July 2010, 11:23

How does eac3to know how much gain to apply after clipping's been detected?

Seems a bit strange to apply only a .27dB gain.
Internally the samples value are managed with float values and can overflow the max value (lossy encoder/decoder imperfections, samplerate/mix operations, ...).
The second pass restore the max peak found to the max value allowed.

When only decode (only lossy encoder/decoder imperfections) I use -no2ndpass because this overflows are always errors (can't be present in original source) and are imperceptibles. But is your choice.

deathlord

12th July 2010, 15:56

If you want a manual way to do it, install MPC-HC and open each MPLS in that. You should quickly see which episode corresponds to which MPLS. It's a pain, yes, but, it should work.

How do I see which episode is which? Almost no show has a text encoded in the image: "Episode 1: ...", in particular not at the very beginning of the episode. The only way I see is to start each episode from the BD structure in TMT/PDVD, start the .m2ts files in MPC and compare visually. I find this extremely unsatisfying.

Sometimes, the .mpls files are in the same order as the episodes (the .m2ts files hardly ever are). However, in some shows, you can apparently not even play each episode seperately. Example: Terminator: TSCC Season 1 in TMT, there is not a playlist for each episode, only one for all three in a row. (However, there *is* a .m2ts file for each episode.

SamuriHL

12th July 2010, 15:59

How do I see which episode is which? Almost no show has a text encoded in the image: "Episode 1: ...", in particular not at the very beginning of the episode. The only way I see is to start each episode from the BD structure in TMT/PDVD, start the .m2ts files in MPC and compare visually. I find this extremely unsatisfying.

Sometimes, the .mpls files are in the same order as the episodes (the .m2ts files hardly ever are). However, in some shows, you can apparently not even play each episode seperately. Example: Terminator: TSCC Season 1 in TMT, there is not a playlist for each episode, only one for all three in a row. (However, there *is* a .m2ts file for each episode.

For a show you've not seen before, this would be incredibly difficult, yes. I'm not sure if there's any easy way to determine what order to use "programmatically" as it were. The idea of loading each MPLS into MPC-HC to see what episode it is only works if you know the order they should be in and can recognize them by the first few seconds of watching it. Admittedly, this does suck. I don't know of a better way.

deathlord

12th July 2010, 16:31

For a show you've not seen before, this would be incredibly difficult, yes.

I usually wait for the shows to be released on blu-ray...

I don't know of a better way.

Does anyone?

TinTime

12th July 2010, 17:27

I don't know of a better way.

Does anyone?

No, I'm in the same boat as you. I just stick the blu-ray in my PS3 and note down the run times of anything I want to rip. Not ideal but I can't think of any other way to do it. I then run a batch script which runs eac3to against all the playlists on the disk and outputs a text file with all the playlists and run times on it so I can select the right ones easily enough.

I say bring back HD DVD. They have nice xpl files with title and chapter names in them :)

dbone1026

12th July 2010, 17:43

No, I'm in the same boat as you. I just stick the blu-ray in my PS3 and note down the run times of anything I want to rip. Not ideal but I can't think of any other way to do it. I then run a batch script which runs eac3to against all the playlists on the disk and outputs a text file with all the playlists and run times on it so I can select the right ones easily enough.

I say bring back HD DVD. They have nice xpl files with title and chapter names in them :)

The easiest method I had when eac3to doesn't correctly display each episode is to use MakeMKV which so far has always picked up each episode individually. Once the mkvs are created I then play the beginning of the mkv and cross reference it against TVDB to figure out the actual episode. A little manual but gets the job done.

Another option, I think a program like DVDFab you can use to rip individual episodes to mkv. Each episode in DVDFab will have a Title# which you should then be able to match off using a player like PDVD to figure out which title# is what episode (this is what I do for SD DVD tv shows)

rica

13th July 2010, 01:28

Guys, dunno if it's a known issue but i can not use Nero audio decoder with eac3to anymore. (with my Seven 64)
I have hd patch of Nero 7 but i can't see Nero audio decoder2; instead nero audible decoder appears.

And when i try to decode any thd with eac3to and i force it to decode with nero, here is what madshi says:

eac3to v3.22
command line: eac3to\eac3to C:\pcm\try.m2ts 3: C:\pcm\audionerothd.pcm -nero
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 0:05:00, 24p /1.001
1: VC-1, 1080p24 /1.001 (16:9)
2: RAW/PCM, English, 2.0 channels, 24 bits, 48kHz
3: TrueHD/AC3, English, 5.1 channels, 96kHz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB)
[a03] I'd like to use the Nero decoder, but it doesn't seem to work. <ERROR>

What's up?

EDIT: I forgot libav is the default decoder for thd because of the nero's channel limitation. But anyway eac3to had to work with nero when it's forced? I'll give it a go with ac3 and eac3 which nero supposed to be default for them.

EDIT: Yes i gave it a go with an ac3 file and eac3to couldn't decode it with its default decoder Nero and forcing it to use nero gave the same caution: "I'd like to use the Nero decoder, but it doesn't seem to work."
eac3to can not recognize Ne.Audible.ax- fake nero audio decoder2 on Seven.

Shall we forget nero anymore?

Or is there any hack?

_ _ _ _ _

rica

13th July 2010, 03:09

BTW, this is an "how to" guide for arcsoft decoders on Seven or Vista SP2.
Somebody may be interested:

http://www.avsforum.com/avs-vb/showpost.php?p=18884347&postcount=3065

_ _ _ _ _ _

crl2007

13th July 2010, 17:59

Sometimes, when I join multiple m2ts files and demux the streams, the end result shows 13 hours as duration.

http://i32.tinypic.com/141nsd2.jpg I join multiple m2ts files from branched discs, for backup as a single m2ts. So it should have 2 hours, not 13.

hubblec4

16th July 2010, 15:39

hello

can everybody help me with the second pass. i have a trueHD track. i want convert it to flac. when the first pass ready the second pass starts, with the message superfluous bits detected: reduce bitdepth from 24 to 16bit. but the second-pass-file sounds damaged.
But the first-pass-flie sounds good(perfect)

is there an option to disable the second pass?

and how i get a full-list of the option by eac3to?

Inspector.Gadget

16th July 2010, 15:46

What do you mean "sounds damaged"? What's the difference in sound between the original and the encoded FLAC file? eac3to is pretty good about throwing out only padding in lossless audio tracks.

Midzuki

16th July 2010, 15:55

is there an option to disable the second pass?

and how i get a full-list of the option by eac3to?

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

but I'm not sure if dropping the 2nd pass will be a solution

hubblec4

16th July 2010, 15:57

you cant understand speech in this file. the musik is distorted. icant explain that exactly, but you hear nothing right.

the bit depth reducing from 24 to 16 bit is not necessary. you know the option to disable the second pass?

i can manually abort the cmd after finished the first pass and then i have the firstpass-file.

hubblec4

16th July 2010, 15:57

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

but I'm not sure if dropping the 2nd pass will be a solution

ok. i've found it. thanks.

i think is a good solution, because the firstpass-audio sounds perfect with a bit depth of 24.
and the secondpass-file with 16bit sounds very bad.

deathlord

16th July 2010, 16:11

ok. i've found it. thanks.

i think is a good solution, because the firstpass-audio sounds perfect with a bitdeth of 24.
and the secondpass-file with 16bit sounds very bad.

The only reason eac3to uses only 16 bits and not 24 in this track is that only those bits contain audio information. The 8 least significant bits are empty. Eac3to does not need to do a second pass in order to strip these. So the second pass must have another reason, e.g. because eac3to detects clipping.
I suspect there is something (slightly) wrong with the file and that's why the second pass produces garbage (this happend to me once).
In any event, if the result of the first pass is ok, i.e. audio is in sync throughout the movie, you can use that of course.

hubblec4

16th July 2010, 17:44

here the log with second-pass

eac3to v3.21
command line: eac3to.exe "00000 - 6 - TrueHD+AC3, German, 5.1 channels, 48kHz DELAY 125ms.thd+ac3" "deu.flac" -50mb
------------------------------------------------------------------------------
TrueHD/AC3, 5.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
Extracting TrueHD stream...
Decoding with libav/ffmpeg...
Encoding FLAC with libFlac...
Creating file "deu.flac"...
The original audio track has a constant bit depth of 16 bits.
Superfluous zero bytes detected, will be stripped in 2nd pass.
Starting 2nd pass...
Decoding FLAC...
Reducing depth from 24 to 16 bits...
Encoding FLAC with libFlac...
Creating file "deu.flac"...
The processed audio track has a constant bit depth of 16 bits.
eac3to processing took 25 seconds.
Done.

Snowknight26

16th July 2010, 19:48

eac3to v3.21
Enough said.

Atak_Snajpera

16th July 2010, 19:54

hubblec4

16th July 2010, 20:57

Enough said.

ok you have right. with eac3to3.22 it works. eac3to make the second-pass and reduce the bitdepth to 16bit. but the file size is the same like 24bit. before the 24bit file was 1,4gb and the 16bit-file 0,9gb.

why two passes? the truehd+ac3-source is 16bit and eac3to convert it to 24bit first and then to 16bit??

so i can use the switch -down16 then start eac3to one pass only.

yesgrey

17th July 2010, 09:34

why two passes? the truehd+ac3-source is 16bit and eac3to convert it to 24bit first and then to 16bit??
It starts converting to 24bit because it's the indicated bit depth. After processing the entire file eac3to realizes that the effective bit depth is 16 bit, hence the bit depth reduction.

If you start using always the -down16, the files which would be effectively 24 bit will be reduced to 16 bit, and you might not want that.

hubblec4

17th July 2010, 15:13

If you start using always the -down16, the files which would be effectively 24 bit will be reduced to 16 bit, and you might not want that.

no, thats for this conversion only.

other 16bit sources will directly encoded with 16bit.

deathlord

17th July 2010, 17:54

no, thats for this conversion only.

other 16bit sources will directly encoded with 16bit.

TrueHD is always 24 bit, but sometimes it only contains 16 bit of audio data, which can only be seen after reading the whole track. Some time ago, eac3to wrote two tracks, a 16 and a 24 bit one. Apparently, now eac3to starts a second pass to write a 16 bit track in case it was only 16 effectively. So if you always use -down16 on thd tracks, you will lose the least significant 8 bit of audio data in case of a "real" 24 bit track.
However, if this is an audible loss of audio quality is highly questionable. I only use 16 bit for all formats.

ron spencer

17th July 2010, 18:21

Just a quick question google has not helped with.

I would like to downmix dts to 2.0, which is easy. But I cannot seem to find what flac compression options eac3to uses? Can I choose 8? Or does it do that by default?

thanks!!!

deathlord

17th July 2010, 19:15

eac3to uses level 8:
http://forum.doom9.org/showthread.php?p=1222771&highlight=flac+level#post1222771

hubblec4

18th July 2010, 14:06

TrueHD is always 24 bit, but sometimes it only contains 16 bit of audio data, which can only be seen after reading the whole track. Some time ago, eac3to wrote two tracks, a 16 and a 24 bit one. Apparently, now eac3to starts a second pass to write a 16 bit track in case it was only 16 effectively. So if you always use -down16 on thd tracks, you will lose the least significant 8 bit of audio data in case of a "real" 24 bit track.
However, if this is an audible loss of audio quality is highly questionable. I only use 16 bit for all formats.

ok i understand this stuff now.
i dont use the switch -down16 for other convertions. only in this case.
with the v3.22 create eac3to a 24bit firstpass file and then the secondpass file with 16bit, which works fine. the original truhd.file is defintiv 16bit.

mastrandrea

20th July 2010, 17:57

Hi, is there a way to stretching an audio file to a given time?
In my case I would like to stretch a track by adding 400ms to it, is it possible with eac3to?

(Thanks for this really useful tool!)

hubblec4

20th July 2010, 22:06

Hi, is there a way to stretching an audio file to a given time?
In my case I would like to stretch a track by adding 400ms to it, is it possible with eac3to?

(Thanks for this really useful tool!)

stretch by 400ms?? i think you mean a delay of 400ms.

you can use this switch +400ms
(eac3to input.xxx output.xxx +/-400ms

mastrandrea

20th July 2010, 23:21

No, I was meaning I need to sync audio and video, and the audio is 400ms shorter.

rica

21st July 2010, 00:58

Guys,

Haven't got any response to this post, pls:

http://forum.doom9.org/showpost.php?p=1417047&postcount=10321

_ _ _ _ _

rica

21st July 2010, 01:07

And another question even i'm not sure being in the right place?

What code should be used in converting any lossless HD audio to PCM before remuxing to BD with TSMuxer? (if i want to make a PCM BD?)

eac3to input.thd(dtshd) output.pcm
and later
pcm2Tsmu input.pcm output.pcm (to be recognized by TSMuxer)

OR,

eac3to input.thd(dtshd) output.wav

Or, doesn't matter?
_ _ _ _ _

dansrfe

21st July 2010, 02:14

@ rica post 1

forget nero. libav works fine if not better.

@ rica post 2

I don't think it matters.

rica

21st July 2010, 02:21

Thanks.

1) OK i forgot nero but i still wonder where is neroaudiodecoder2?

2) Yes, thanks.

SamuriHL

21st July 2010, 02:23

Rica, are you just thinking about Nero for the DTS-HD MA decoding? TMT3's DTS-HD MA codec works fine if you register it properly, but, I think you know that? Am I missing something?

EDIT: Oh, sorry, you meant to use Nero for TrueHD decoding. Yea, libav does a fine job of that. Nero's not needed.

rica

21st July 2010, 02:32

EDIT: Oh, sorry, you meant to use Nero for TrueHD decoding. Yea, libav does a fine job of that. Nero's not needed.

Samuri,
the question was not nero needed or not; it was why i can not use it even it's worse than libav or even i force it to be used in eac3to?
And where is this f***g nero decoder?

SamuriHL

21st July 2010, 02:34

Samuri,
the question was not nero needed or not; it was why i can not use it even it's worse than libav or even i force it to be used in eac3to?

Yea, I don't know the answer to that one. I haven't installed Nero in a very long time.

rica

21st July 2010, 02:37

Pls reinstall it on your seven and tell me where it is?
You will see Ne.Audible.ax- fake nero audio decoder2 on Seven.
And i need a hack. But i haven't been able to do.

SamuriHL

21st July 2010, 02:38

Oh god no. :) I don't have a recent version of Nero. I don't even know what the last version I bought was. 8 maybe? I'm not spoiling a good 7 install with that! :D

rica

21st July 2010, 02:42

Nero 7; what i've been speaking about which eac3to supposed to be used to use. With its HD patch.

SamuriHL

21st July 2010, 02:45

Is Nero 7 even compatible with W7 64? I've read a lot of reports that it no workie. I've never tried. I don't even know if I still have the email with my key kicking around. Is there a reason you're wanting to get it to work so badly?

rica

21st July 2010, 02:48

Normally all 32 bit SW should work on 64 bit OS.

Is there a reason you're wanting to get it to work so badly?

No reason; i just wonder :)

SamuriHL

21st July 2010, 02:52

Yea, I don't have my license for 7. I only have a license for 8. I never bothered with anything after that cause it became ridiculously bloated. I have no intention of installing Nero again. As for eac3to and TrueHD, I've had no problem at all with the built in libav code. Works perfectly.

rica

21st July 2010, 02:53

OK, thanks Samuri.

So, i think nautilus7 should revize the wiki.

ACrowley

21st July 2010, 07:20

Nero 7; what i've been speaking about which eac3to supposed to be used to use. With its HD patch.

Remember that theres a bigger Problem with the NeroAudio Decoder in eac3to! The DRC isnt removed properly so the Decoder is not recommended

hubblec4

21st July 2010, 09:25

No, I was meaning I need to sync audio and video, and the audio is 400ms shorter.

streching is possible in MKVmergeGUI.

you need the time of the video and the audio( audio= 100sek ; video 100,400sek) then you divide the videotime by the audiotime and you get the stretch-value.

deado

21st July 2010, 12:06

rebkell

21st July 2010, 12:56

No, I was meaning I need to sync audio and video, and the audio is 400ms shorter.

I assume the audio is out of sync? If it drifts over the entire clip, you could just add some audio loops spaced equally over the entire audio track.

-edit=0:00:00,0ms loops or removes audio data at the specified runtime
-silence/-loop forces usage of silence (or looping) for audio edits

If it's 48Khz AC3 you could add 12 or 13 32ms edits spaced equally and get with 16ms.

SamuriHL

21st July 2010, 14:05

Using Nero 7 with Win 7 64 here no problems.

Well, I guess that's good. :) I could only find my key for Nero 8. Not that I care in either case. I'm perfectly content with the libav support for TrueHD in eac3to. And I use ImgBurn for all my burning needs. That's all I ever really used Nero for anyway...stable burning. Then they started bloating it up with all kinds of other useless stuff. Meh.

stax76

21st July 2010, 20:47

There is a improved dialog for Blu-ray demuxing in StaxRip:

http://i.imagehost.org/t/0562/Unbenannt_3.jpg (http://i.imagehost.org/view/0562/Unbenannt_3)

http://forum.doom9.org/showthread.php?p=1419618#post1419618

rica

23rd July 2010, 00:35

Using Nero 7 with Win 7 64 here no problems.

Do you mean eac3to can use nero's forced decoder?

_ _ _ _ __

Snowknight26

23rd July 2010, 06:37

How can I reproduce this? I've just tried "eac3to some.wav 1.aac 2.aac" and both AAC files are created just fine.

Was able to find a small enough sample to be able to reproduce the issue:

http://stfcc.org/misc/eac3to.aac.issue.zip

Run the bat, wait for eac3to to stop outputting. Might be a while, but it will. Either put the two files in the folder eac3to is in or modify the bat.

tvjunky

24th July 2010, 15:10

Hi Madshi!

Maybe this is a bug in eac3to:

I always rip my BRs directly with eac3to like this:
eac3to d: 1) -demux

This was functioning for 100% of my BRs the last years. With the new version 3.22 i get the following error on ~50% of my discs (example):

D:\_hd>eac3to y: 1)
Skipping 66240 bytes to work around damaged source file...
Hint: You can use the option "-fix" to clean up the damage.
The format of the source file could not be detected.

The parameter "-fix" does not help. With eac3to version 3.21 all BRs are ok and can be copied to hd.

jdawn1

27th July 2010, 18:18

the same reason why 24bit is converted to 16bit. Human hearing has limitations.

Word. Personally, I stick with as high as possible source sample rate and bit depth, and usually use lossy encoding. Much more efficient use of space, IMO. Especially when you can use something like AAC. I'd rather have a 5.1 96kHz/24bit (sourced) 96kHz AAC @ 650kbit/s then a 5.1 48Khz/16bit FLAC @ 3000kbit/sec. All that extra superfluous audio data is much better spent on the video. By superfluous, I would personally guarantee that no one here could spot the difference between a "high bitrate" AAC and a FLAC from the same source, on your choice of audio equipment. I am talking an AAC file 1/5 to 1/10th the size of the FLAC.

I am still amazed when I see movie encodes people make with 10GB file size: 5GB of video and 5GB of a DTS-HD-MA audio track. :rolleyes: Makes no sense whatsoever. Make it 9.2GB of video and 800MB of audio (AAC for PC playback or DTS/AC3 core if you need to "bitstream") and it's overall much better quality.

Digital: In the world of today, 192kHz audio should already be something quite trivial. :)

Analog: In the world of today, speakers and audio amplifiers that support up to 96kHz still are very rare, and (IMHO) somewhat pointless. :rolleyes:ss......

Yeah. If you mean by frequency response, 96KHz could represent 48Khz frequency audio. Given I can't hear much above 16KHz, yeah, speaker reproduction of 48kHz would be pretty stupid.

Currently, are there any speaker systems/receiver's that actually support 192Khz? Now THAT would be the 9th Wonder of the World.

Well, speaker support is irrelevant, they're analog. My cheapie receiver supported it just fine, even the original bitstream (DTS-HD-MA 5.1 192 / 24)
Funnily enough, it actually sounds better if I playback a version of it downsampled to 48KHz as my receiver supports Audyssey but can't process 5.1 streams over 48Khz. The audio benefit of running with Audyssey enabled far exceeded listening to the track at 96 or 192 KHz...
My digital copy of the movie is 96/24 audio encoded to AAC, which I have Windows downsample to 48khz (LCPM over HDMI) so the receiver can enable Audyssey.

OK... I'll be the first to ask... Why would you want to take such a beautiful audio track and throw away 2/3 of the data?

Because it is a pointless waste of space if it's to be stored on a PC. In my case, resampling from 192 to 96 and encoding to AAC (Nero, -q 0.5) resulted in "throwing away" something like 94% of the data, lol. 13.4 GB to less then 1 GB. Yet, you can't hear a damn bit of difference on a nice audio setup; sounds great. I'd dare you to even try.

It's still a beautiful audio track even if you remove a few samples or use a transparent lossy encode. How it SOUNDS is what's important, and what makes a great score. File size has, and should not have, in anyone's mind, anything to do with it.

Using eac3to, any way I can downconvert a ~25GB 192 Khz (!) LPCM track (Akira Blu-Ray) to something more human, like, say, 48 Khz?

Of course! Just look at the help. It's really easy. I own the blu-ray, and I make .MKV's for my media server. This is the only movie I had to re-sample. I re-sampled to 96KHz / 24bit and encoded to AAC with Nero, file went from 13+ GB (DTS-HD MA) to less then 1 GB and sounds awesome. :)

The movie went from taking most of a BD-50 to only about 13.4 GB using the original video and AAC audio. Nothing lost quality wise, lol.

I did keep a copy of that sucker on my server for testing purposes, though. My receiver seems to work just fine with it... a few years old and a cheapie, too. Onkyo 606. Kinda neat to see it say 5.1ch DTS-HD MA 192 khz (bitstreamed from a PC with ATI 5xxx HDMI card). My old PS3 plays it back without a hitch either. 5.1 ch 192 kHz PCM shows up on the receiver, not that I would be able to hear a difference even if it was not fully decoding or down/upsampling, lol.

lintran

28th July 2010, 12:11

Hello
I just tried extract DTS-ES Discrete 6.1 to wav(input.dts output.wavs) with Arcsoft Decoder (version 3.0.160) and it produced 7 individual WAV files, but there was something wrong with them. Too much high-frequency and noise.
Same occurs when I try to create an AC3 file.
So please tell me where do i wrong? Or how to do it proper?
Thank you so much.

Atak_Snajpera

28th July 2010, 12:18

Arcsoft decoder cannot properly decode 6.1 dts. I had the same problem.

deathlord

28th July 2010, 17:26

Arcsoft decoder cannot properly decode 6.1 dts. I had the same problem.

Sonic works well on these!

I re-sampled to 96KHz / 24bit and encoded to AAC with Nero

Why not 48KHz, 96 is a waste of diskspace, too!

Atak_Snajpera

28th July 2010, 17:34

Why not 48KHz, 96 is a waste of diskspace, too!
because people like in placebo effect ;)

deathlord

28th July 2010, 20:40

@ madshi,

Could you allow for longer -edits, in particular silence? At the moment the maximum possible (positive) edit seems to be 4999ms.

longtom

29th July 2010, 06:28

I want to extract from a BluRay the 2.0 PCM soundtrack to wav.
The output should be outputchapter1.wav outputchapter2.wav etc...
So that i have all songs in a extra wav files and not in one LARGE output.wav file!

tebasuna51

29th July 2010, 10:23

You can use the BeSplit command line, or with CUE_Splitter GUI, to split the full wav file.
For CUE_Splitter you need a .cue file, but maybe you can convert the Chapters file to a cue file easily with a text editor.

longtom

29th July 2010, 11:27

tebasuna51

29th July 2010, 11:35

WARNING:
- When the language field of a track (in mkv) is empty this mean the language is the default: eng (from MkvMerge v4.0.0)
- eac3to don't manage audio tracks (MP2/MP3/DTS/AC3?) from mkv's generated with MkvMerge v4.1.0 defaults (header compression).
- eac3to crash (without messages) with video track H264 from mkv's generated with MkvMerge v4.2.0 defaults (header compression).

@madshi

Do you have any plan to support these new defaults?

frank

31st July 2010, 10:41

Test of mkvtoolnix 4.2.0 header removal compression
(deletes repeating header infos)

Remuxing of a ts Stream to mkv:
8.292.939 kB without header removal compression, like before
8.292.174 kB with Compression

= 0,00923 % difference!!

Wow, what a great saving! :devil: Forget that new default.
The mkvtoolnix developer bunkus should remove that annoying standard setup!!

First Aid:
Disable for every track the option Compression: to None.
Not empty.

setarip_old

31st July 2010, 20:08

@frank

Hi!Wow, what a great saving! Forget that new default.
The mkvtoolnix developer bunkus should remove that annoying standard setup!!If you read the MKVToolnix thread, you'll find that "bunkus" is adamant that since this is technically correct, it will remain as the far-from-obvious default setting...

dansrfe

1st August 2010, 20:02

Does eac3to do encodings to .mp3?

Inspector.Gadget

1st August 2010, 21:51

dansrfe - Not directly. But you can pipe output to the LAME executable. tebasuna51 has posted a command template several times around here, and I imagine you'll find it pretty easily searching for eac3to lame.exe or something similar,

mr.duck

4th August 2010, 15:27

Is it possible for eac3to demux a TS file and pipe it directly into mkvtoolnix?

If that's too hard I can just demux the files and then run mkvtoolnix when eac3to is done. Any recomendations?

longtom

5th August 2010, 12:34

Great, Thanks!
I'll try with Cue_splitter!

I've splitted it now with foobar!

LeXXuz

6th August 2010, 09:17

How do you convert AC3 2.0 Surround to another 2.0 Surround file and keep the surround information?

I'd like to make a speedup on an AC3 2.0 Surround f.e.:
eac3to input.ac3 output.ac3 -224 -speedup

However the output file is only AC3 2.0 (when reopening the output file, eac3to reports "AC3 2.0" and not "AC3 2.0 Surround" as for the input file) and my AVR does not show the Surround channel any more.

Is there any switch I need to use to keep the output file as 2.0 surround?

tebasuna51

6th August 2010, 12:26

How do you convert AC3 2.0 Surround to another 2.0 Surround file and keep the surround information?
You need use the external encoder Aften.exe.

Suggested command line:
eac3to input.ac3 stdout.wav -speedup | Aften -b 224 -m 0 -dsur 2 -pad 0 -readtoeof 1 -exps 32 -s 1 - output.ac3

tebasuna51

6th August 2010, 12:43

Is it possible for eac3to demux a TS file and pipe it directly into mkvtoolnix?
Not now, eac3to can only write the demuxed track files in hard disk.
If that's too hard I can just demux the files and then run mkvtoolnix when eac3to is done. Any recomendations?
You can try the GUI UsEac3to (http://forum.doom9.org/showthread.php?t=145574).

You can order extract the video to mkv, one or two audios to the desired format, and then push 'Run and MkvMux'.
After eac3to finish, the output files are sended to MkvMerge automatically.

deathlord

6th August 2010, 17:38

Is it possible for eac3to demux a TS file and pipe it directly into mkvtoolnix?

If that's too hard I can just demux the files and then run mkvtoolnix when eac3to is done. Any recomendations?

You can also let mkvmerge show you the command line. Then you can make a batch file with the eac3to command followed by the mkvmerge command.
This is very efficient for tv shows.

LeXXuz

6th August 2010, 18:16

You need use the external encoder Aften.exe.

Suggested command line:
eac3to input.ac3 stdout.wav -speedup | Aften -b 224 -m 0 -dsur 2 -pad 0 -readtoeof 1 -exps 32 -s 1 - output.ac3

Worked like a charm. Thanks a lot tebasuna51! :thanks:

robpdotcom

7th August 2010, 07:10

Hi,

I was searching this thread for problems with TrueHD in seamless branching discs, and read this:

It's not a good idea to extract THD tracks from seamless branching Blu-Rays. Audio and video will drift out of sync, because eac3to can't fix the audio overlaps with THD tracks, when demuxing them. You should convert to THD/WAVs directly from the original Blu-Ray structure. Then those "lossless check failed" warnings should go away and audio and video will stay in sync.

Does this mean that you can convert to THD? In other words, can eac3to fix overlaps if you extract as THD as opposed to THD+AC3? Or, must you convert to Flac or WAV?

TinTime

7th August 2010, 08:52

No, you can't convert to THD. I think it's just a typo. Possibly madshi meant to say "You should convert to FLAC/WAVs directly from the..."

robpdotcom

7th August 2010, 09:47

OK, thanks for clarifying.

What exactly is the difference between THD and THD+AC3, and why would you want to use one over the other?

mr.duck

7th August 2010, 15:59

You can also let mkvmerge show you the command line. Then you can make a batch file with the eac3to command followed by the mkvmerge command.
This is very efficient for tv shows.

If I understand you correctly this is how I have it set up at the moment. For TV shows too. Saves 10s of gigabytes with no transcoding! Performance and reliability are much better too, so pretty excellent IMO.

One recording I ran through eac3to and mkvmerge had chunk of audio or video missing because of some big spike of interference corrupting the data. So 1/2 way into the show, the audio and video went way out of sync by around a whole second.

So I notice eac3to says something like "... it may work, or not" as part of a warning message. For these files I want the batch file to abort and leave the original file as is rather than try and remux it to mkv with audio sync problems. So how can this be done? Maybe...
IF ERRORLEVEL = 1 (after eac3to is done) or...
eac3to.exe ..... | FIND "it may work, or not" <-- will that work doing it that way?

TinTime

7th August 2010, 17:18

Yes, try...

eac3to.exe commands -log="log.txt"
if %errorlevel% neq 0...
find /c /i "warning" "log.txt"
if %errorlevel%==0...

Shamus76

7th August 2010, 20:25

Hi guys,

I am having big problems encoding DTS. Example:

I:\KICKASS>eac3to bdmv 2) 3: Kick.Ass2.dts
M2TS, 1 video track, 1 audio track, 1:57:45, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: AC3, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
a03 The Nero decoder doesn't seem to work, will use libav instead.
a03 Extracting audio track number 3...
a03 Removing AC3 dialog normalization...
a03 Decoding with libav/ffmpeg...
a03 Reducing depth from 64 to 32 bits...
a03 Writing WAVs...
a03 Creating file "Kick.Ass2.L.wav"...
a03 Creating file "Kick.Ass2.R.wav"...
a03 Clipping detected, a 2nd pass will be necessary.
a03 Starting 2nd pass...
a03 Extracting audio track number 3...
a03 Removing AC3 dialog normalization...
a03 Decoding with libav/ffmpeg...
a03 Reducing depth from 64 to 32 bits...
a03 Writing WAVs...
a03 Applying -1,34dB gain...
a03 Creating file "Kick.Ass2.R.wav"...
a03 Creating file "Kick.Ass2.L.wav"...
Encoding DTS <768kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.29.0.
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

I have tried encoding DTS like above on THD and AC3 tracks, same problem. Also tried to convert a THD-track to DTS, same problem.

I am running eac3to v3.22 and i have the exact same setup on another computer and there it works fine.

I have re-installed Surcode several times, no difference.

I read that surcode sometimes have path-length issues but i have run it on the root folder with same result.

setarip_old

7th August 2010, 20:36

@Shamus76

Hi! I am running eac3to v3.22 and i have the exact same setup on another computer and there it works fine. I have re-installed Surcode several times, no difference.

Does your purchase license of "Surcode" allow for it to run on more than one machine? If not, that could be your problem...

Shamus76

7th August 2010, 20:39

@Shamus76

Hi!

Does your purchase license of "Surcode" allow for it to run on more than one machine? If not, that could be your problem...

Yes, it should be allowed, any other ideas?

Shamus76

7th August 2010, 20:51

@Shamus76

Hi!

Does your purchase license of "Surcode" allow for it to run on more than one machine? If not, that could be your problem...

OMG, almost don't wanna say this but the problem is solved. I had forgotten to activate surcode :rolleyes:

:thanks:

Lyle_JP

7th August 2010, 21:28

How do you convert AC3 2.0 Surround to another 2.0 Surround file and keep the surround information

2 channel is 2 channel. The "surround information" is always there, out of phase between the left and right. If you mean the cute little surround light that pops on in your receiver's display, that's because Dolby Digital allows a surround "flag" to be added to a 2.0 stream. If you transcode to another format (AAC, MP3, etc.) you lose the flag, and those formats don't support such a surround flag AFAIK.

As long as you instruct your receiver to do surround processing on any stereo source, you won't notice a difference.

mr.duck

9th August 2010, 04:10

Yes, try...

eac3to.exe commands -log="log.txt"
if %errorlevel% neq 0...
find /c /i "warning" "log.txt"
if %errorlevel%==0...

Unfortunately I have lots of errors and warnings. Typically I see something like this 10-20 times per recording...
[v01] [0:01:25] The source file seems to be damaged (transport error indicated). <WARNING>
But the video still seems to be able to remux to mkv ok and should not be aborted.

I found out what was causing audio sync issues (I said it was missing data due to interference). For TV recordings, it turns out...

eac3to.exe ...... -demux <-- Catastrophically bad audio sync. Was mkvmerge's fault for throwing away video (some complaint about GOPs).
eac3to.exe ...... 1: blah_blah.mkv ...... <-- Perfect audio sync with same file. mkvmerge has no complaints. eac3to is great :D

Obviously I'm now going to switch to using mkv output in eac3to instead of -demux. But just to improve my error handling for problem files, does anyone know about a warning message in eac3to that reads something like "... it may work, or not"? I don't have any files that produce this warning and I'm not sure as I may be making all this up having remembered it wrong. I had a look on the wiki but didn't find anything. Anyone know what I'm on about?

Shamus76

9th August 2010, 18:12

Hi again guys,

Got another problem. I am trying to extract an LCPM track from a Bluray and then merge it with MKVmerge.

I have tried several times extracting the LCPM track to a PCM file using no manual settings and once using "-16 -48000" setting. Result is this:

I:\KILLBILL1_GBR>eac3to bdmv 1) 3: Kill.Bill.1.pcm
M2TS, 1 video track, 8 audio tracks, 9 subtitle tracks, 1:50:44, 24p /1.001
1: Chapters, 19 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 5.1 channels, 16 bits, 48kHz
4: AC3, English, 5.1 channels, 640kbps, 48kHz
5: DTS, Italian, 5.1 channels, 16 bits, 1509kbps, 48kHz
6: AC3, Italian, 5.1 channels, 640kbps, 48kHz
7: DTS, Spanish, 5.1 channels, 16 bits, 1509kbps, 48kHz
8: AC3, Spanish, 5.1 channels, 640kbps, 48kHz
9: DTS, German, 5.1 channels, 24 bits, 1509kbps, 48kHz
10: AC3, German, 5.1 channels, 640kbps, 48kHz
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), Italian
14: Subtitle (PGS), Spanish
15: Subtitle (PGS), German
16: Subtitle (PGS), English
17: Subtitle (PGS), Italian
18: Subtitle (PGS), Spanish
19: Subtitle (PGS), German
a03 Extracting audio track number 3...
a03 Reading RAW/PCM...
a03 Swapping endian...
a03 Remapping channels...
a03 Swapping endian...
a03 Remapping channels...
a03 Creating file "Kill.Bill.1.pcm"...
a03 The original audio track has a constant bit depth of 16 bits.
Video track 2 contains 159287 frames.
eac3to processing took 9 minutes, 56 seconds.
Done.

Then when i drag'n'drop the file to MKVmerge 4.20, it takes a while and then i get a File Identification-error saying:

"C:\Program Files\MKVtoolnix\mkvmerge.exe" --output-charset UTF-8 --identify-for-mmg "I:\KILLBILL1_GBR\Kill.Bill.1.pcm"

Output:

Error end-of-file

Error: File I:\KILLBILL1_GBR\Kill.Bill.1.pcm has unknown type.

I cannot see any errors in the extraction, anyone got any ideas?

:thanks:

b66pak

9th August 2010, 19:20

@Shamus76 why don't you extract to .flac, .w64 or .wav?
_

Shamus76

9th August 2010, 19:24

@Shamus76 why don't you extract to .flac, .w64 or .wav?
_

Well i thought the uncompressed pcm track loose information when converting to another format, perhaps that isn't so?

Btw, .w64, what is that format?

b66pak

9th August 2010, 19:31

Shamus76

9th August 2010, 19:59

you don't lose anything...flac is lossless (you will save space!) and .w64 or .wav is as .pcm but with headers...

.w64 is short for .wav with 64bits...valid .wav files can't be bigger than 4gb so .w64 is the solution...
_

Oh i see, didn't know about .w64. Thnx for the tip.

Unfortunately .w64 isn't recognized by MKVmerge either.

I'll try standard .wav or .flac.

Thnx for your help.

mr.duck

9th August 2010, 22:46

Definitely use FLAC. There's nothing else that's better while being open source and widely supported. For stereo music, file size is around 60% using FLAC compared to WAV.

ajamils

11th August 2010, 21:57

TinTime

11th August 2010, 22:51

It's nothing to do with the audio. eac3to (or specifically the haali muxer) doesn't handle some AVC streams when muxing to mkv. The Apocalypto blu-ray is one of these types.

Demux to raw AVC and then mux with mkvmerge.

Shamus76

12th August 2010, 05:39

I'm trying to rip a Blu-ray using HdBrStreamExtractor (which is a GUI for eac3to) and I'm getting the following error. I also tried ripping the same blu-ray with anotherEac3to GUI and Blu-rip and all failed with the same error. It seems like eac3to does not like extracting the audio which is in Mayan language. Any suggestion ?

Earlier post by madshi explains the error:

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

I used the "-seekToIFrames" option and it works fine and i haven't yet had any problems with artifacts.

ramicio

12th August 2010, 15:35

I have a mono (1.0) DTS-HD Master audio file I'm trying to decode and it won't work. The output is:
"The ArcSoft DTS Decoder reported an error while decoding. Aborted file at position 262144."

Tried it directly from the m2ts file and it gave the same error, just a different position number. Eac3to shows the correct info about the file, but the decoder doesn't like it. Anything else to try here? I want lossless, I don't wan't just the core!

deathlord

12th August 2010, 18:48

ramicio

12th August 2010, 18:59

Hehe I am ripping my hair out trying to find the sonic decoder. I don't even know what to look for. I am trying to get Cineplayer HD-DVD Player 4.2 with no luck so far. And I'm sure once I do, it probably won't work.

Well I found it, and I am ripping my hair out once again trying to figure out why it won't work. It's giving a "can't find mfc71.dll" error when I run "eac3to -test" and it gets to the part where it checks the sonic decoder.

The mfc71.dll error is gone, and I found 4.3.0.169. This version is showing as installed but not working in eac3to as well. Would this have anything to do with having an x64 version of windows? I can use the sonic filter in graphedit but it makes a 16-bit file, not 24 :( The sonic decoder is only decoding the core of the dts file. HELP!!!

LeXXuz

13th August 2010, 12:36

I must be blind. Again. Please tell me how do I create six WAV files decoded from a 6ch dts source?

I always end up with one (probably 6-ch) wav file when I use "eac3to input.dts output.wav". I only get 6 intermediate wave files when I set .dts as output format to feed the dts encoder directly afterwards.
But I dont want that, because my DTS encoder is on another machine where I'd like to import those WAV's.

nurbs

13th August 2010, 12:53

eac3to input.dts output.wavs

LeXXuz

13th August 2010, 13:08

eac3to input.dts output.wavs

Ah no way! It can't be that obvious. :D
:thanks:

[)370|\|470!2

13th August 2010, 14:03

eac3to v3.22
command line: "D:\TRANS\eac3to3.22\eac3to.exe" "E:\Batman.The.Dark.Knight\odds\Batman.track_4354.ac3" "D:\Batman\Batman.track_4354.flac"
------------------------------------------------------------------------------
TrueHD/AC3, 5.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
Extracting TrueHD stream...
Decoding with libav/ffmpeg...
Encoding FLAC with libFlac...
Creating file "D:\Batman\Batman.322.track_4354.flac"...
[libav] Substream 0 parity check failed <WARNING>
[libav] Substream 0 checksum failed <WARNING>
[libav] Lossless check failed - expected 87, calculated 92 <WARNING>
[libav] Substream 1 parity check failed <WARNING>
[libav] Substream 1 checksum failed <WARNING>
[libav] Lossless check failed - expected 85, calculated 33 <WARNING>
[libav] Substream 0 parity check failed <WARNING>
[libav] Substream 0 checksum failed <WARNING>
[libav] Lossless check failed - expected fd, calculated c9 <WARNING>
[libav] End of stream indicated <WARNING>
The original audio track has a constant bit depth of 16 bits.
Superfluous zero bytes detected, will be stripped in 2nd pass.
Starting 2nd pass...
Decoding FLAC...
Reducing depth from 24 to 16 bits...
Encoding FLAC with libFlac...
Creating file "D:\Batman\Batman.track_4354.flac"...
The processed audio track has a constant bit depth of 16 bits.
eac3to processing took 24 minutes, 28 seconds.
Done.

Any ideas what could be wrong with the track? Oh, and inb4, I own the original disc.

BatKnight

13th August 2010, 17:49

Any ideas what could be wrong with the track? Oh, and inb4, I own the original disc.
Looks like a corrupted or broken track. You could try to re-rip the track, since you own the original disc. Or, better than that, you could rip the TrueHD track directly to FLAC, using eac3to.

[)370|\|470!2

13th August 2010, 20:20

You could try to re-rip the track, since you own the original disc.
But of course, it was the first thing I've tried to no avail.
Or, better than that, you could rip the TrueHD track directly to FLAC, using eac3to.
Already done. The question is how am I to be sure if it's b0rked or not.

setarip_old

13th August 2010, 21:19

@[)370|\|470!2

Already done. The question is how am I to be sure if it's b0rked or not.Post the log (Just as you did for the ripped-to-harddrive version)...

[)370|\|470!2

13th August 2010, 22:23

@[)370|\|470!2

Post the log (Just as you did for the ripped-to-harddrive version)...

The log posted above actually is.

nurbs

13th August 2010, 23:09

What happens if you convert to flac without demuxing the track first?

Also, since the source file in the command line is a .ac3 I find it strange that it says the content is TrueHD. I thought the had a .thd extnsion.

[)370|\|470!2

14th August 2010, 01:26

What happens if you convert to flac without demuxing the track first?
That's brilliant. Just tell me what command line to use to achieve that.
since the source file in the command line is a .ac3 I find it strange that it says the content is TrueHD. I thought the had a .thd extnsion.
And I thought file's being recognized by its headers, not the extension.

setarip_old

14th August 2010, 03:36

@[)370|\|470!2
"E:\Batman.The.Dark.Knight\odds\Batman.track_4354.ac3" "D:\Batman\Batman.track_4354.flac"
This line from the log certainly doesn't indicate that the source is your original disc (as I suggested you use)...

nurbs

14th August 2010, 08:27

That's brilliant. Just tell me what command line to use to achieve that.
eac3to DiscFolder shows the playlists
eac3to DiscFolder 1) shows the tracks in the playlists
eac3to DiscFolder 1) 3:audio.flac encodes the third track in the first playlist to flac

And I thought file's being recognized by its headers, not the extension.
Well that depends, doesn't it? I suspect eac3to does, but if you double click on a file with a .doc extension I wouldn't expect a media player to pop up even if the file is actually an .mp3, at least not on windows.

Anyway what software did you use to extract the audio? Did that software put the wrong extension on the file or did you do that?

[)370|\|470!2

14th August 2010, 15:17

if you double click on a file with a .doc extension I wouldn't expect a media player to pop up even if the file is actually an .mp3, at least not on windows.
Obviously you should learn a bit about the file assotiation per se and how it works.

Anyway what software did you use to extract the audio? Did that software put the wrong extension on the file or did you do that?
Let me get this straight — you're claiming that extension is a problem there?

nurbs

14th August 2010, 15:39

Let me get this straight — you're claiming that extension is a problem there?
No, I'm thinking something went wrong during demuxing the .m2ts files, which is why I asked what software you used to demux the audio. If it was eac3to you'd end up with a .thd extension, so you either renamed it or you used something else. I'd be interested in how you ended up with the file so I'll know how to avoid such problems.

So, have you found time to try if reencoding the audio straight from the .m2ts without demuxing it first works?

xkodi

15th August 2010, 07:43

Will be anything made with the decoding of "strange setup" 7.1?

solution for that problem was finally found:

http://forum.doom9.org/showthread.php?t=156168

actually, that approach could be used for every DTS-HD (MA) file that Arcsoft can't decode correctly and respectively eac3to can't.

i wish eac3to be able to write such 140 bytes headers for easier decoding of DTS-HD (MA) file outside eac3to in case that is necessary.

sub24ox7

15th August 2010, 07:47

yes PLEASE!!! that would be so awesome I can't decode a DTS-HD mono track at this very moment :(

madshi

15th August 2010, 11:12

Maybe this is a bug in eac3to:

I always rip my BRs directly with eac3to like this:
eac3to d: 1) -demux

This was functioning for 100% of my BRs the last years. With the new version 3.22 i get the following error on ~50% of my discs
That's weird. I have a hard time believing that the new eac3to is responsible for the problems. Maybe something else changed on your PC? It seems nobody else has this problem.

I just tried extract DTS-ES Discrete 6.1 to wav(input.dts output.wavs) with Arcsoft Decoder (version 3.0.160) and it produced 7 individual WAV files, but there was something wrong with them. Too much high-frequency and noise.
Arcsoft decoder cannot properly decode 6.1 dts. I had the same problem.
Newer versions (e.g. 1.1.0.7) of the dtsdecoder.dll seem to have this problem. Older versions (e.g. 1.1.0.0) work just fine.

Could you allow for longer -edits, in particular silence? At the moment the maximum possible (positive) edit seems to be 4999ms.
Yes, will change that.

WARNING:
- When the language field of a track (in mkv) is empty this mean the language is the default: eng (from MkvMerge v4.0.0)
- eac3to don't manage audio tracks (MP2/MP3/DTS/AC3?) from mkv's generated with MkvMerge v4.1.0 defaults (header compression).
- eac3to crash (without messages) with video track H264 from mkv's generated with MkvMerge v4.2.0 defaults (header compression).

@madshi

Do you have any plan to support these new defaults?
Yes.

Hi, is there a way to stretching an audio file to a given time?
In my case I would like to stretch a track by adding 400ms to it, is it possible with eac3to?
It is not possible with eac3to because the resampling library is limited to specific stretching factors.

An audio file (dts, ac3, ...) don't have tags about the video fps, only have a duration. And aply a -24.000 -changeTo23.974 means only modify the duration with a coefficient of 1.001

With audio files -slowdown means modify the duration with a coefficient of (1001/960) always.
FWIW, "-slowdown" always defines a target FPS value of 23.976. You can do "-24.000 -slowdown" and it will do a 24.000 -> 23.976 slowdown.

I have a Blu Ray disc of a tv show (Supernatural Season 1, Disc 1). The disc has the first 6 episodes of the tv show. When I run the disc through eac3to (using Clown_BD and/or Ripbot) eac3to only picks up 3 playlists (the 1st playlist is of all episodes, the 2nd playlist is of episode 1, and the 3rd playlist is of episode 2). Episode 3-6 don't show anywhere. If I use MakeMKV I can see all episodes listed out individually to select. Is there anything you can think of that would be causing this issue?
I've seen this before and just thought that there were not any playlists for the other episodes. But maybe there are and there's a bug in eac3to. Can you please zip up the playlist and clipinf folders for me?

The decoding by libav under XP failed too.

What this can be?
Broken TrueHD track.

eac3to cannot fully demux my TS files (from DVB-T, UK).

Unlike other programs like tsMuxeR, eac3to at least recognizes that there are subtitles there. But -demux only extracts the video and audio.
I don't know how to demux DVB subtitles in any format that can be used by other applications.

Also when I -demux some <path>\video.ts, it extracts the audio and video to the same folder as eac3to.exe, not the video file folder and no other way of setting the output to another folder. Pretty annoying.
Just use "eac3to source c:\whatever\movie.*" that will demux all tracks to any desired target path.

How do I see which episode is which?
Often there's a "play all" playlist. It usually lists the episodes in the right order.

Sometimes, when I join multiple m2ts files and demux the streams, the end result shows 13 hours as duration.
Simply joining multiple m2ts files makes problems because after joining it is not possible for eac3to (or any other program) to quickly see that the file was joined from multiple smaller files and that the timestamps do not match and where each smaller file starts and ends in the joined file. This is the way it is. If you manually join m2ts files, you'll have to live with this effect.

Can eac3to demux DTS Express audio tracks?
Yes.

I come back to this problem with more explanations and tests. In fact, the problem is not new, I just verified and I have the same strange behaviour with eac3to v3.21 and eac3to v3.18.

Here is my scenario:
1 - eac3to to demux video in MKV container and convert audio to AC3
2 - TsMuxer to mux audio and video in a M2TS container
3 - playback of the M2TS file with the PS3

Video is H.264. When the file is decoded by the PS3, I have many macroblocks, and the result is the same whatever the version of eac3to (3.18, 3.21 and 3.22).
If at step 1, instead of demuxing video in MKV container, I demux the video in a H.264 file, then the playback with the PS3 is fine without macroblocks.

Note that all these files (M2TS and MKV) are played correctly on the PC.

So the problem is relative to the usage of MKV container, either when produced by eac3to, or when used as input by TsMuxer. And the problem is noticeable only when the final file is played with the PS3.
If you do this:

(1) demux blu-ray directly, by using eac3to
(2) mux video to MKV by using eac3to, then demux again by using eac3to

The demuxed raw h264 video file produced by both methods is identical. So the problem must obviously be caused by tsMuxeR not handling MKV input correctly.

madshi

15th August 2010, 13:58

eac3to don't remove the dialog normalization with Arcsoft decoder (1.1.0.1)

Test .dtshd

http://multi-up.com/295013

DNorm.rar - center before and after decoding.
I've checked this and all decoders output the same volume - even libav which definitely doesn't do dialnorm processing for DTS when eac3to removes it from the headers. I believe that the volume was already changed (hard coded) by the DTS encoder. And I believe that ArcSoft totally ignores the dialnorm when decoding.

One key information is that decoding the track with ArcSoft and Sonic decoders output bit perfect identically results. Which means that decoding should be lossless which would not be the case if dialnorm processing was active.

madshi

15th August 2010, 14:04

I just noticed that a bug which I had reported months ago is still present.
The mapping of 3.0 soundtracks converted to flac is incorrect. The center channel ends up in the left surround.
This time the source was DTS MA 3.0 (last time: PCM 3.0), as found on the Criterion BluRays of Yojimbo and Sanjuro, so I don't think it's a decoder problem.
When creating Mono-Wavs instead of flac, the three tracks are correctly titled L, R and C.
The FLAC is correct. The latest madFlac v1.9 now also outputs the correct channelmask (0x7). The previous version used channelmask 0xffffffff. If you still get incorrect channel assignments then it's the fault of some other filter, maybe even the audio renderer. eac3to and madFlac definitely handle 3.0 correctly now.

madshi

15th August 2010, 14:22

Could you allow for longer -edits, in particular silence? At the moment the maximum possible (positive) edit seems to be 4999ms.
I've checked. There's no specific limitation for the "-edit" command. You cannot edit in such a way, though, that looping wouldn't work. E.g. "-edit=0:00:01,+2000ms" is not allowed, but "-edit=0:00:05,+2000ms" works just fine.

madshi

15th August 2010, 19:19

Was able to find a small enough sample to be able to reproduce the issue:

http://stfcc.org/misc/eac3to.aac.issue.zip

Run the bat, wait for eac3to to stop outputting. Might be a while, but it will. Either put the two files in the folder eac3to is in or modify the bat.
I think I got a fix for this one. Could you please try with the next eac3to build and report back?

madshi

15th August 2010, 19:58

I'm seeing odd behavior with Battleship Potemkin, a newly restored BluRay release from Kino.
Thanks for the upload, the problem will be fixed in the next build.

madshi

15th August 2010, 20:06

@madshi
can you add option to automatically ignore .m2ts in playlist if number of audio channels is not constant. In this example also frame rate is different in first file.

Movie Downfall
eac3to v3.22
------------------------------------------------------------------------------
1) 00014.mpls, 00034.m2ts+00014.m2ts, 2:35:21
- Chapters, 35 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, German, stereo, 48kHz

00034.m2ts
M2TS, 1 video track, 1 audio track, 0:00:37, 29.766p
1: Chapters, 35 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, German, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB

00014.m2ts
M2TS, 1 video track, 1 audio track, 0:00:29, 24p /1.001
1: VC-1, 1080p24 /1.001 (16:9)
2: AC3, German, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB

Hmmmmm... This is a difficult one. Sometimes there might be a small intro with different channels before the real movie which I could easily ignore without doing any damage. But then there might be movies where actually different parts have different audio tracks (e.g. seamless branching theatrical cut vs. director's cut). How can eac3to know? E.g. if the first m2ts file is 10 minutes long, shall I still cut it away? For an intelligent human it's not so difficult to find out whether an m2ts file can be dropped or not, but for a computer program it's much more difficult. I fear that if I simply drop the first m2ts file if it has different audio track properties, then I might damage some real movies.

Thoughts?

(BTW, the frame rate is not different, eac3to is more or less "guessing" the m2ts framerate, it's sometimes wrong. The video track framerate is reliable and it's identical.)

madshi

15th August 2010, 20:11

eac3to v3.23 released

http://madshi.net/eac3to.zip

* fixed: handling of MKV files with header compression didn't work
* fixed: track listing of some Blu-Rays failed silently
* added support for decoding mono tracks with ArcSoft DTS decoder
* added reading support for DTS tracks with additional DTS-HD header
* improved m2ts framerate detection
* video bitdepth is displayed now, if different than 8 bits
* chroma format is displayed now, if different than 4:2:0

b66pak

15th August 2010, 20:17

thanks a lot...

Thoughts?

keep it as it is...or you will open a pandora box!..
_

Thunderbolt8

15th August 2010, 20:20

thanks for the new version again :)

a few questions: what would be the problem if you left all those small .m2ts files listed? a overly long amount of different playlists? is there maybe any way to say that eac3to should list, and not cut away, all small .m2ts files of certain playlists only, e.g. such playlists which contain the 2-3 biggest .m2ts files? (sometimes, there are extras on a BD which are bigger than the actual movie)

regarding audio, so dtsma mono tracks should work now. so whats still left that doesnt work properly? what about those 6.1 dtsma/truehd/lpcm tracks? 2.0 dtsma? anything with "strange setup"? dts express afaik was a problem of non-existing mkvmerge support.

what are those dts-hd tracks with additional header, what does it mean? again, do we have to do any re-demuxing of our dtsma track collection to have compability with hardware receivers when just bitstreaming the file?

thanks!

madshi

15th August 2010, 20:29

a few questions: what would be the problem if you left all those small .m2ts files listed? a overly long amount of different playlists?
There are many Blu-Rays with literally hundreds of playlists. You'd be burried by useless information.

is there maybe any way to say that eac3to should list, and not cut away, all small .m2ts files of certain playlists only
No.

regarding audio, so dtsma mono tracks should work now. so whats still left that doesnt work properly? what about those 6.1 dtsma/truehd/lpcm tracks? 2.0 dtsma? anything with "strange setup"?
The only problem I'm aware of are those 7.1 tracks listed as "strange setup". Everything else, including mono, stereo and 6.1 should work perfectly, if you use the right ArcSoft decoder version (e.g. 1.1.0.0 works fine).

what are those dts-hd tracks with additional header, what does it mean?
If you don't know what it means that you don't need it. No need to re-demux anything.

Thunderbolt8

15th August 2010, 20:48

If you don't know what it means that you don't need to do it. No need to re-demux anything.well, I dont have a HD receiver yet, so I wouldnt know if I needed it ;) but if I had to redo something then I'd like to have done that by the time I have that kind of receiver, thats why I ask.

There are many Blu-Rays with literally hundreds of playlists. You'd be burried by useless information.

No.would it be possible that eac3to leaves a message when it cuts away some small parts of those playlists it displays? then you dont have to check the playlist of the main movie manually each time for those files, but only if they are really included in the list for that movie.

mrr19121970

15th August 2010, 20:48

Thoughts?

(BTW, the frame rate is not different, eac3to is more or less "guessing" the m2ts framerate, it's sometimes wrong. The video track framerate is reliable and it's identical.)

The answer would be 'depends' on what you want to do ultimately (have all streams separetly demuxed, have 1 .ts/.m2ts/.mkv file, or the full BD structure).

Are you (Atak_Snajpera) looking to simply demux the playlist, or recode the audio? There are a few options available currently:

Use 00014.m2ts instead of 00014.mpls for example
Use another tool (eg tsMuxeR) on each .m2ts and then correct the playlist (http://clownbd.com/Clown_BD%20MPLS%20Builder.exe).

Snowknight26

15th August 2010, 20:52

I think I got a fix for this one. Could you please try with the next eac3to build and report back?

Seems to be working beautifully, thanks.

madshi

15th August 2010, 21:19

eac3to v3.24 released

http://madshi.net/eac3to.zip

* fixed: m2ts framerate detection for h264 tracks was half of what it should be
Oooooops.

jj666

16th August 2010, 02:39

Hello Madshi,

I tested the new processing .DTSHD functionality, it seems to be working with all the files I threw at it so far.

However, I found that demuxing a .DTSHD file introduced a 21ms delay to the file.

I verified by demuxing the original WAV, demuxing the encoded WAV with DTS Stream Player (both matched perfectly) and demuxing with EAC3TO (+21ms added). Waveforms were checked with Adobe Audition.

FYI, before EAC3TO added this functionality, I was demuxing the .DTSHD files with TSMUXER, TSMUXER mentioned "bad frame found, resycning stream" - this also introduced the same +21ms delay.

Cheers,

-jj-

Thunderbolt8

16th August 2010, 04:17

However, I found that demuxing a .DTSHD file introduced a 21ms delay to the file.when you demux it from what? you also get a delay now when demuxing it from the original blu-ray source?

madshi

16th August 2010, 08:21

However, I found that demuxing a .DTSHD file introduced a 21ms delay to the file.

I verified by demuxing the original WAV, demuxing the encoded WAV with DTS Stream Player (both matched perfectly) and demuxing with EAC3TO (+21ms added). Waveforms were checked with Adobe Audition.

FYI, before EAC3TO added this functionality, I was demuxing the .DTSHD files with TSMUXER, TSMUXER mentioned "bad frame found, resycning stream" - this also introduced the same +21ms delay.
I think the delay is probably caused by the encoder. The "demuxing" (= removal of added DTS-HD header structure) should not have any effect on the audio data or the delay.

Thunderbolt8

16th August 2010, 11:36

yes, seems like it. did a waveform comparison of the center channel of

blu-ray source -> flac -> wavs and
blu-ray source -> demuxed dtsma -> flac -> wavs

and both are exactly the same with no delay.

tebasuna51

16th August 2010, 11:45

Thanks madshi.

Very good fixes.

madshi

16th August 2010, 11:46

Pleasure... :)

ACrowley

16th August 2010, 13:14

* added reading support for DTS tracks with additional DTS-HD header

Thank you :) A restore/add additional DTSHD Header Function would complete it :)

madshi

16th August 2010, 14:20

A restore/add additional DTSHD Header Function would complete it :)
Not going to happen, at least not anytime soon. There's no specification about how the headers need to look like and there are lot of fields in there that need to be filled (e.g. sampling rate etc). I'm not going to write guessed headers. That would only open up a can of worms. A lot of follow up bugs would probably show up, and I neither have the fun nor the time to spend week after week trying to do all the necessary bugfixes... :(

AnryV

16th August 2010, 15:54

I've checked this and all decoders output the same volume - even libav which definitely doesn't do dialnorm processing for DTS when eac3to removes it from the headers. I believe that the volume was already changed (hard coded) by the DTS encoder. And I believe that ArcSoft totally ignores the dialnorm when decoding.

One key information is that decoding the track with ArcSoft and Sonic decoders output bit perfect identically results. Which means that decoding should be lossless which would not be the case if dialnorm processing was active.

http://thumbnails30.imagebam.com/9335/2de43b93341946.jpg (http://www.imagebam.com/image/2de43b93341946)

eac3to -test
eac3to (v3.22) is installed
Nero Audio Decoder (Nero 7) works fine
ArcSoft DTS Decoder (1.1.0.1) works fine
Sonic Audio Decoder (4.3.0.169) works fine
Haali Matroska Muxer (2009-01-11) is installed
Nero AAC Encoder (1.3.3.0) is installed
Surcode DTS Encoder (1.0.29.0) is installed
MkvToolnix (2.9.9.0, release version) is installed

jj666

16th August 2010, 16:01

jj666

16th August 2010, 16:06

I think the delay is probably caused by the encoder. The "demuxing" (= removal of added DTS-HD header structure) should not have any effect on the audio data or the delay.

Process:
- I demuxed a two channel .WAV and encoded to DTS-HD MA.
- Demuxing (or rather splitting the outputted .DTSHD) to individual .WAVs with DTS StreamPlayer results in an identical to
original source file when compared to original in Adobe Audition.
- Demuxing (or rather splitting the outputted .DTSHD) to individual .WAVs with EAC3TO results in an file with +21ms delay when compared to original in Adobe Audition.

Cheers,

-jj-

madshi

16th August 2010, 16:11

http://thumbnails30.imagebam.com/9335/2de43b93341946.jpg
You're right, my fault, must have made a mistake when testing. I've now done some more checks. The problem is that whenever I try to manipulate the DTS-HD headers, the ArcSoft DTS decoder seems to ignore the DTS-HD information. I guess that there's a CRC somewhere in the headers which I'd need to update, but I don't know where it is nor how to update it. So it seems that my hands are tied here.

Ergo: If you have a DTS-HD track with dialnorm in it, ArcSoft will honor the dialnorm, while Sonic ignores it. Nothing I can do about it, it seems. Fortunately DTS tracks with dialnorm in it seem to be pretty rare...

madshi

16th August 2010, 16:15

Process:
- I demuxed a two channel .WAV and encoded to DTS-HD MA.
- Demuxing (or rather splitting the outputted .DTSHD) to individual .WAVs with DTS StreamPlayer results in an identical to
original source file when compared to original in Adobe Audition.
- Demuxing (or rather splitting the outputted .DTSHD) to individual .WAVs with EAC3TO results in an file with +21ms delay when compared to original in Adobe Audition.
Then I can only guess that there's additional information in the DTSHD header which tells the decoder to cut off 21ms from the beginning of the track. 21ms should be exactly 2 DTS frames. Or maybe the DTS StreamPlayer always cuts off 2 DTS frames? I've no idea. What I can say is that eac3to is not doing anything fancy, it just strips of the DTSHD header and feeds the DTS frames to the ArcSoft decoder. As far as I can see, eac3to is doing exactly the right thing.

Of course I could add extra processing to eac3to to always ignore the first 2 DTS frames when there's a DTSHD header, but then who guarantees us that it's always 2 frames? Maybe sometimes it's only 1 frame or 3 frames?

jj666

16th August 2010, 16:21

In my experience encoding LPCM (2.0 and 5.1) to DTS-HD MA, it's always 21ms (as mentioned in the past, I demuxed first with TSMUXER to remove header, then processed with EAC3TO). That's probably around 100 encodings and all waveform checked with Adobe Audition (and 21ms removed after with EAC3TO).

That's my experience, but like you said there may be certain circumstances where this is different. I would humbly request that the 21ms be automatically removed to save me a step in my process, but that's up to you :-) What I can say is that I will report back if I ever see sync issues different to 21ms :-)

Cheers,

-jj-

madshi

16th August 2010, 16:33

jj666

16th August 2010, 19:01

Hello Madshi,

Thanks for your quick reply. Yes, its less than a millisecond out. I took the liberty of making some samples for you:-

DTS-HD (60mb): http://www.mediafire.com/?49x9zs9up9uhvh9
DTS StreamPlayer demuxed left channel (30mb): http://www.mediafire.com/?gfmwkaa4pqhmea4
EAC3TO -21ms demuxed left channel (30mb): http://www.mediafire.com/?88484ugdeiehbzz
EAC3TO demuxed left channel (30mb): http://www.mediafire.com/?36y5labf5uqw1vf

Audition (same file order):

http://thumbnails24.imagebam.com/9336/53bd9f93358609.jpg (http://www.imagebam.com/image/53bd9f93358609)

Let me know in case you need any more info. As mentioned, this sync issue does not apply to a straightforward demux from a Blu-ray, it's only happening where the DTS-HD contains the header information.

Cheers,

-jj-

Thunderbolt8

16th August 2010, 19:28

is there possibly a very small bug with the red colour of "the format of the source file could not be detected"?

doesnt seem to appear normally, but I did "eac3to movie.htm movie.srt > moviefinal.srt" (basically I accidently used eac3to command instead of srtmaker.exe, that small file to combine OCR'ed subs (timestamps + spoken text)) and then after that sentence above, the path "X:\" also had that red background colour and every line after as well.

srtmaker.exe in order to reproduce: http://www.mediafire.com/?zzqihdc5p11bkjn

rapscallion

16th August 2010, 22:20

Here (http://rghost.net/1659797) they are.
SNR:
out.10N.nero.L.wav 27.386 dB
out.10N.libav.L.wav 27.411 dB
out.10S.nero.L.wav 5.5173 dB
out.10S.libav.L.wav 27.398 dB
tartak (http://forum.doom9.org/member.php?u=169863) wrote on another forum, that with nero decoder DRC is only partially ignored: volume increasing still applies, while attenuation is ignored.

Goal is to decode Dolby/TrueHd 5.1 to wavs.

From several previous posts, it was stated that Nero decoder doesn't handle drc removal properly. An example was given, as a comparison to libav, that the snr was substantially different. Is this really true and, if so, why was their no problem all the time that this was considered a "fine" decoder.
(still is in madshi's first post of the thread)

No other user has mentioned problems with either the audio quality or the snr of Nero since the beginning of this thread 3 years ago.
A google search doesn't reveal the CompAudio program used to measure the outputs. Anyone know where to find it or does it go by a different name ?

When using Nero (in eac3to_more gui) the dialog box states that drc is being disabled, however when using libav no such msg. Is libav removing drc or not ?

Separate curiousity question :

Blue Thunder contains both a ac3 and THD track. Why is it that the THD track is 16bit (24 bit padded) and yet the ac3 track is 24 bit (eac3to reduces from 64 to 24)? Logic would indicate that it should be just the opposite.

When decoding these tracks to wavs, the HD tracks result in ~600mb files and the ac3 tracks to ~900mb files. Again, seems like it should be just the opposite or is the size the resullt of the different bit depths.

This is starting to make my head hurt !

Thunderbolt8

16th August 2010, 23:35

there was some talk that lossy tracks dont have a real bitdepth or something like that, so just ignore the 24-bit.

rik1138

17th August 2010, 00:07

well, I dont have a HD receiver yet, so I wouldnt know if I needed it ;) but if I had to redo something then I'd like to have done that by the time I have that kind of receiver, thats why I ask.

It only affects DTS-HD files created by the DTS Encoder (before they've been used on a Blu-ray disc). If you are only demuxing from existing discs, it has no affect on you.

Madshi- Thanks so much for adding this! Saves me the hassle of having to remove the headers when I need to convert files! :cool:

Abradoks

17th August 2010, 00:47

No other user has mentioned problems with either the audio quality or the snr of Nero since the beginning of this thread 3 years ago.
madshi has already tried (http://forum.doom9.org/showthread.php?p=1404212#post1404212) to fix it.
A google search doesn't reveal the CompAudio program used to measure the outputs. Anyone know where to find it or does it go by a different name ?
Actually first link (http://www-mmsp.ece.mcgill.ca/documents/Software/Packages/AFsp/CompAudio.html) in google. It's part of AFsp (http://www-mmsp.ece.mcgill.ca/Documents/Downloads/AFsp/) package.
BTW, you can use any other tool to compute SNR.
Is libav removing drc or not ?
Yes, it doesn't apply DRC by default.

rapscallion

17th August 2010, 21:16

madshi has already tried (http://forum.doom9.org/showthread.php?p=1404212#post1404212) to fix it.
Yes, I saw his post but didn't think it was an eac3to issue, but a Nero issue.
Isn't it the Nero decoder that's removing/disabling drc?

Or am I misunderstanding how the decoder processes the stream to wavs ?

Actually first link (http://www-mmsp.ece.mcgill.ca/documents/Software/Packages/AFsp/CompAudio.html) in google. It's part of AFsp (http://www-mmsp.ece.mcgill.ca/Documents/Downloads/AFsp/) package.
BTW, you can use any other tool to compute SNR.
Thanks, however I actually saw that link but didn't see anywhere to download the file, just an explanation of what it does. Am I missing something ?

Yes, it doesn't apply DRC by default.
By "apply" do you mean it removes it when decoding an ac3 track? Because that's what I'm looking for the decoder to do.

nurbs

17th August 2010, 21:47

By "apply" do you mean it removes it when decoding an ac3 track? Because that's what I'm looking for the decoder to do.
It removes it on demuxing and doesn't apply it on decoding which is essentially the same as removing it.

rapscallion

17th August 2010, 21:49

Gotcha, thanks !

Abradoks

17th August 2010, 21:58

Or am I misunderstanding how the decoder processes the stream to wavs ?
DRC is metainformation. By default Nero decoder always applies DRC when such information is presented. eac3to uses some hacks to avoid it, but as you see it doesn't work always. On the contrary, with libav you have full control over this and by default it doesn't apply DRC. So you don't need to "remove" anything.

Thanks, however I actually saw that link but didn't see anywhere to download the file, just an explanation of what it does. Am I missing something ?
Here is direct link: AFsp-v8r2.tar.gz (http://www-mmsp.ece.mcgill.ca/Documents/Downloads/AFsp/AFsp-v8r2.tar.gz). win32 binaries are in MSVC/bin folder.

rapscallion

17th August 2010, 22:29

DRC is metainformation. By default Nero decoder always applies DRC when such information is presented. eac3to uses some hacks to avoid it, but as you see it doesn't work always. On the contrary, with libav you have full control over this and by default it doesn't apply DRC. So you don't need to "remove" anything.
Ok, I really did misunderstand that process. And now I see why it was so long before the problem was discovered.
Reminds me of years ago when I worked for the Norwegian audio co Tandberg. The running joke was that the name translated to "intermittent".

Here is direct link: AFsp-v8r2.tar.gz (http://www-mmsp.ece.mcgill.ca/Documents/Downloads/AFsp/AFsp-v8r2.tar.gz). win32 binaries are in MSVC/bin folder.

Thank you again and for the link..... had to go back to my command prompt days to do this one :)

Atak_Snajpera

18th August 2010, 20:06

Newer versions (e.g. 1.1.0.7) of the dtsdecoder.dll seem to have this problem. Older versions (e.g. 1.1.0.0) work just fine.
Does older version contain any serious bugs which were fixed in newer? In other words. Is it safe to downgrade decoder to older version?

tebasuna51

18th August 2010, 20:11

Does older version contain any serious bugs which were fixed in newer? In other words. Is it safe to downgrade decoder to older version?
v1.1.0.0 works fine for me, without know bugs, with XP SP3.

MikeEby

20th August 2010, 03:43

Hmmmmm... This is a difficult one. Sometimes there might be a small intro with different channels before the real movie which I could easily ignore without doing any damage. But then there might be movies where actually different parts have different audio tracks (e.g. seamless branching theatrical cut vs. director's cut). How can eac3to know? E.g. if the first m2ts file is 10 minutes long, shall I still cut it away? For an intelligent human it's not so difficult to find out whether an m2ts file can be dropped or not, but for a computer program it's much more difficult. I fear that if I simply drop the first m2ts file if it has different audio track properties, then I might damage some real movies.

Thoughts?

(BTW, the frame rate is not different, eac3to is more or less "guessing" the m2ts framerate, it's sometimes wrong. The video track framerate is reliable and it's identical.)

I've run into a couple of disks with playlists such as this too. The latest one is called “Dead Man Running”.

Perhaps if eac3to could look at the first 2 files in the playlist and then it notices they do not have matching tracks it could then use only the file with the higher track count and longest time. If the track counts don’t match eac3to will fail anyway, so there would be nothing to loose. The chapter times would also have to be offset by missing file length. Maybe if the first file is less than 30 seconds long just drop it, or perhaps a switch?

Mike

Atak_Snajpera

20th August 2010, 10:17

Hmmmmm... This is a difficult one. Sometimes there might be a small intro with different channels before the real movie which I could easily ignore without doing any damage. But then there might be movies where actually different parts have different audio tracks (e.g. seamless branching theatrical cut vs. director's cut). How can eac3to know? E.g. if the first m2ts file is 10 minutes long, shall I still cut it away? For an intelligent human it's not so difficult to find out whether an m2ts file can be dropped or not, but for a computer program it's much more difficult. I fear that if I simply drop the first m2ts file if it has different audio track properties, then I might damage some real movies.

Thoughts?

(BTW, the frame rate is not different, eac3to is more or less "guessing" the m2ts framerate, it's sometimes wrong. The video track framerate is reliable and it's identical.)

I agree with MikeEby. eac3to should compare two first files in playlist. File with higher number of channels should be kept. File with lower number of channels should be automatically discarded.

I've run into a couple of disks with playlists such as this too.
MikeEby can you provide samples or at least logs like mine
eac3to v3.22
------------------------------------------------------------------------------
1) 00014.mpls, 00034.m2ts+00014.m2ts, 2:35:21
- Chapters, 35 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, German, stereo, 48kHz

00034.m2ts
M2TS, 1 video track, 1 audio track, 0:00:37, 29.766p
1: Chapters, 35 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, German, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB

00014.m2ts
M2TS, 1 video track, 1 audio track, 0:00:29, 24p /1.001
1: VC-1, 1080p24 /1.001 (16:9)
2: AC3, German, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB

mrr19121970

20th August 2010, 10:31

Or change the way the info is presented, currently:

M2TS, 1 video track, 1 audio track, 0:20:43, 50i
1: Chapters, 4 chapters
2: MPEG2, 576i50 (16:9)
3: AC3, German, 2.0 channels, 192kbps, 48kHz

if streams differ in underlying .m2ts files

00001.M2TS, 1 video track, 1 audio track, 0:00:43, 50i
1: Chapters, 4 chapters
2: MPEG2, 576i50 (16:9)
3: AC3, German, 2.0 channels, 192kbps, 48kHz

00002.M2TS, 1 video track, 1 audio track, 0:20:00, 24p
4: VC-1,1080p24 (16:9)
5: DTS, German, 5.1 channels

nautilus7

20th August 2010, 12:07

madshi,

"-mono" switch to output only center channel (in mono wavs) doesn't seem to work correctly when a second pass is needed.

eac3to v3.24
command line: eac3to audio.thd audio.bd.wavs -mono
------------------------------------------------------------------------------
TrueHD, 5.1 channels, 48kHz
Decoding with libav/ffmpeg...
Writing WAVs...
Creating file "audio.bd.C.wav"...
[libav] Lossless check failed - expected 0, calculated c <WARNING>
The original audio track has a constant bit depth of 16 bits.
Superfluous zero bytes detected, will be stripped in 2nd pass.
Starting 2nd pass...
Decoding with libav/ffmpeg...
Reducing depth from 24 to 16 bits...
Writing WAVs...
Creating file "audio.bd.L.wav"...
Creating file "audio.bd.C.wav"...
Creating file "audio.bd.SR.wav"...
Creating file "audio.bd.LFE.wav"...
Creating file "audio.bd.R.wav"...
Creating file "audio.bd.SL.wav"...
The processed audio track has a constant bit depth of 16 bits.
eac3to processing took 21 minutes, 21 seconds.
Done.

Snowknight26

20th August 2010, 18:05

Hate to nag, but any hopes of getting -check working with SRT subs in an upcoming build? ASS subs work fine, but SRT..

chompy

20th August 2010, 19:10

I don't know if this is a bug, or if that's what's meant to be, but demuxed H264 videos cannot be opened with Adobe Premiere, while the same videos demuxed with tsMuxeR can be opened and edited perfectly.

Greetings

EDIT: The files I'm demuxing are m2ts ripped from bluray

b66pak

20th August 2010, 19:36

I agree with MikeEby. eac3to should compare two first files in playlist. File with higher number of channels should be kept. File with lower number of channels should be automatically discarded.

what about movies like "Fireflies In The Garden"?

Logo 1, Channels: 2.0, 10.400s
Logo 2, Channels: 5.1, 4.992s
Logo 3, Channels: 2.0, 15.680s
Movie, Channels: 5.1

it should stay as it is...
_

Momber

21st August 2010, 03:48

The FLAC is correct. ... If you still get incorrect channel assignments then it's the fault of some other filter
I'm not in the least surprised to hear that.

eac3to and madFlac definitely handle 3.0 correctly now.
Great! Thank you very much.

tormento

21st August 2010, 13:21

I am trying to convert "American Gangster" extended edition from BD to MKV.

Firstly I have to correctly demux every track but a problem exist.

The standard lenght is composed by the following M2TS:

2:36:50

235+222
236+223
237+224
238+225
239+226
240+227
241+228
242+229
243+230
244+231
245+232
246+233
247+234
248

The extended lenght from the following:

2:55:46

235+249
236+250
237+251
238+252
239+253
240+254
241+255
242+256
243+257
244+258
245+259
246+260
247+261
248

The standard lenght has every language and subs, the extended only the english in the alternative M2TS.

I need the italian+english(when italian missing) extended version. How is possible to tell eac3to to select a particular stream from a M2TS and changing it while joining?

MikeEby

22nd August 2010, 00:53

MikeEby can you provide samples or at least logs like mine

Here you go.
The Girl With The Dragon Tatoo (US Version)

1) 00001.mpls, 2:33:02
[1+0+17+18].m2ts
- Chapters, 17 chapters
- h264/AVC, 1080p24 /1.001 (16:9)

00001.m2ts
M2TS, 1 video track, 0:00:10, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)

00000.m2ts
M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 2:32:44, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: AC3, Swedish, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
3: AC3, English, 5.0 channels, 448kbps, 48kHz, dialnorm: -27dB
4: Subtitle (PGS), English

00017.m2ts
M2TS, 1 video track, 0:00:04, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)

00018.m2ts
M2TS, 1 video track, 0:00:04, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)

Mary and Max (US Version) This one has matching track counts just the track properties don't match.

1) 00000.mpls, 00023.m2ts+00000.m2ts, 1:32:26
- Chapters, 17 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz

2) 00007.mpls, 00007.m2ts, 0:22:04
- MPEG2, 480i60 /1.001 (16:9)
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz

3) 00002.mpls, 00005.m2ts, 0:15:48
- Chapters, 6 chapters
- MPEG2, 480i60 /1.001 (4:3)
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz
- RAW/PCM, English, stereo, 48kHz

00023.m2ts
M2TS, 1 video track, 4 audio tracks, 2 subtitle tracks, 0:00:10, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
3: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
4: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
5: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
6: Subtitle (PGS), Spanish
7: Subtitle (PGS), English

00000.m2ts
M2TS, 1 video track, 4 audio tracks, 2 subtitle tracks, 1:32:16, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: RAW/PCM, English, 2.0 channels, 24 bits, 48kHz
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
4: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
5: RAW/PCM, English, 2.0 channels, 16 bits, 48kHz
6: Subtitle (PGS), Spanish
7: Subtitle (PGS), English

Dead Man Running This one has matching track counts too...Different audio types in the two files. My original thought won't work.

1) 00001.mpls, 00009.m2ts+00003.m2ts, 1:31:48
- Chapters, 12 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS, English, stereo, 48kHz
- DTS, English, stereo, 48kHz

2) 00005.mpls, 00011.m2ts, 0:23:55
- h264/AVC, 1080i60 /1.001 (16:9)
- DTS, English, stereo, 48kHz

3) 00004.mpls, 00010.m2ts, 0:23:12
- h264/AVC, 1080i60 /1.001 (16:9)
- DTS, English, stereo, 48kHz

00009.m2ts
M2TS, 1 video track, 2 audio tracks, 0:00:15, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: DTS, English, 2.0 channels, 16 bits, 510kbps, 48kHz
3: DTS, English, 2.0 channels, 16 bits, 510kbps, 48kHz

00003.m2ts
M2TS, 1 video track, 2 audio tracks, 1:31:32, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: DTS Master Audio, English, 5.1 channels, 16 bits, 48kHz core: DTS, 5.1 channels, 16 bits, 1509kbps, 48kHz)
3: DTS, English, 2.0 channels, 16 bits, 768kbps, 48kHz

Thanks!
Mike

setarip_old

22nd August 2010, 20:15

@tormento

Hi! I am trying to convert "American Gangster" extended edition from BD to MKV.That's a simple task for MakeMKV (including stream selection)...

krosswindz

23rd August 2010, 01:05

I am using the eac3to v3.24 to extract wave files from a DTS MA source. I noticed that the wave files have a bitrate of 1152kbps though the core is 1509 kbps. I am using the arcsoft DTS decoder v1.1.0.0. I extracted the wave files using the following command line

c:\path\to\eac3to.exe e:\path\to\blu-ray\source 1) 3: output.wavs

I am wondering is there something that I am doing wrong. If I extract the core as wave files the bitrate is again at 1152 kbps.

TinTime

23rd August 2010, 01:25

No, nothing's wrong. The bitrate of a PCM WAV file is always going to be (bitdepth * sample rate * channels / 1000) kbps. More specifically bitdepth will be rounded up to the nearest byte.

So in your example it's 24 * 48000 * 1 / 1000 = 1152kbps

It doesn't matter if the source is lossy or lossless. If they both decode to the same bitdepth then the resultant wavs will be the same size.

tormento

23rd August 2010, 02:21

That's a simple task for MakeMKV (including stream selection)... [/Color]
Sorry, I have tried but Make MKV sees only the eng audio for the extended version.. My will is to use the eng audio where the ita audio is missing... I could try to encode every M2TS separately and join them but doing this would not correct the audio gaps and so.

krosswindz

23rd August 2010, 04:16

krosswindz

23rd August 2010, 05:11

TinTime

23rd August 2010, 11:44

I am not sure if this feature exists already but it would be nice if one is able to use some form of a queue for processing in eac3to. For example if one wants to create both 768Kbps DTS and 1536 Kbps DTS eac3to could only extract the wave files once. Currently, I do the extraction myself and then use surcode encoder manually.

If your source is DTS MA with a full rate core as mentioned in your earlier post then then just extract the core and add the half rate file separately.
c:\path\to\eac3to.exe e:\path\to\blu-ray\source 1) 3: output1.dts -core 3: output2.dts -768
More generally you might be able do the following:
c:\path\to\eac3to.exe e:\path\to\blu-ray\source 1) 3: output1.dts -1536 3: output2.dts -768
I've never tried it but give it a go :)

If the problem is that it takes a long time to decode your source audio then another option is to decode to wav first and then encode from that wav - it won't take so long to split to wavs:
c:\path\to\eac3to.exe e:\path\to\blu-ray\source 1) 3: inter.wav
c:\path\to\eac3to.exe inter.wav output1.dts -1536
c:\path\to\eac3to.exe inter.wav output2.dts -768

jasonwc

23rd August 2010, 16:08

Ok, so should I only ignore the first 2 frames when there's a DTSHD header? When there's no such header, I should not cut off 2 frames?

Can you please try "eac3to source.dts dest.wavs -21ms" to check whether the WAV files are then perfectly identical to the DTS StreamPlayer output?

I did some testing and found that I got a 24 ms delay. I took a TrueHD track from Dexter (16/48 Khz) and converted to FLAC. I then converted the TrueHD to mono WAVS and used the Master Audio Suite to create a DTS-HD MA. Here's what I found:

Decoded FLAC - 54:09.288 (155965840 samples)
Decoded TrueHD- 54:09.288 (155965840 samples)
Decoded DTS-HD MA- 54:09.312 (155966976 samples)

The DTS-HD MA has an extra 24 ms of samples (1,136 samples). I used the foobar2000 bit-comparison tool and found that the TrueHD and FLAC tracks were bit-identical.

I used eac3to remove 24 ms, but I was still a few samples short. Is there a way to remove an exact number of frames? In any case, I'm not sure why I'm getting a 24 ms rather than a 21 ms delay.

Like the others, when I decode to WAV with the DTS HD Streamplayer, there are no extra samples, and the output matches the source in foobar's bit-comparison tool.

Thunderbolt8

23rd August 2010, 17:37

so far ive only had delays of 21ms as well.

sub24ox7

23rd August 2010, 19:15

BUG report:
I have a disc that has a 24 bit 96 khz 5.1 LPCM track when extracting the LPCM (eac3to movie.m2ts 4: audio.wavs) as pcm or wav(s) the runtime is twice as long as it should be with the new eac3to 3.24 so I reverted to the only other copy I had on my computer which was 3.17 I think and the LPCM track extracted fine. Also the M2ts had a 1080i60 video track in it.

ramicio

23rd August 2010, 19:35

I wonder about the order of processing, and this is why I haven't ventured to use eac3to for certain tasks. Anything with a lossless 5.1 track I find I demux to wavs with eac3to. Then with goldwave I mix these into a stereo 24/192 wav, resample to 24/96, and then use eac3to to flac it. If I could just do a 1 step process to do this that would be nice.

Lossless 5.1 -> 6 x 24/192 -> downmix -> 24/96 -> flac

I also do downmixing like:

Left = Center + LFE + FL + RL
Right = Center + LFE + FR + RR

So I don't know how anything else does it. After that of course it is clipped, but Goldwave thankfully stores everything in floating point that is being worked on, so I just find the highest peak and reduce the volume by that MINUS 0.01 dB.

Alcohor

23rd August 2010, 20:09

madshi, first of all thank you for your wonderful tool :D

i have a "little" request for you... can you add an option that can split .m2ts files by chapters (like dvddecrypter)? i've got a lot of bluray of tv series that contain all episodes merged in one .m2ts and i'd like to make them separated

nurbs

23rd August 2010, 20:36

@ramicio:

You can't get that downmix with eac3to.

-down2 uses a DPLII downmix with
L' = 0.5 x L + 0.3535 x C + 0.433 x SL + 0.250 x SR
R' = 0.5 x R + 0.3535 x C - 0.250 x SL - 0.433 x SR
with -mixlfe you also add that to it.

eac3to will use 64 bit precision for processing. Output precision will be same as input if you encode lossless, highest supported by the encoder if you encode lossy. Clipping will be detected and automatically dealt with. You can change the sampling rate, apply gain or normalize the audio if you like. It's in the manual.

By the way, the downmix you use is not standard AFAIK. A standard normalized stereo downmix uses
L' = 0.3694 x L + 0.2612 x C + 0.3694 x SL
R' = 0.3694 x R + 0.2612 x C + 0.3694 x SR

ramicio

23rd August 2010, 20:53

Yes, I know what eac3to can do. It's the order of it I was asking about. Is that downmix mapping really optimal for lossless? My downmixes sound just fine to me. Just the center will be louder than standard. And also, what level would the LFE get mixed in at? I wish I wouldn't have gotten rid of a lot of things I mixed already. Just to try out those levels.

Hmm. I just took 0.2612/0.3694 and it is ~ 0.707... the sin of 45°. I think the validity of that downmix is coming to play making its point and I will try that in the future.

nurbs

23rd August 2010, 21:26

It decodes, increases precision to 64 bit, resamples, downmixes and decreases the bitdepth. FYI the order of most of the steps is listed in the log. No idea if the second downmix is "optimal" for normal stereo, but it is what is usually being used around here. No idea about the LFE either.

Boulder

24th August 2010, 03:53

I've been having a problem with eac3to and NeroAACEnc with later versions of eac3to. After encoding the decoded WAV, the log says that NeroAACEnc seems to be stuck (but I've no clue as to why) and it starts to encode the file again. I don't remember at which point the problem started to occur but it hasn't been there for long.

eac3to v3.24
command line: c:\utils\eac3to\eac3to.exe "e:\temp\dvd-rip\star trek tng\4x17 T81 3_2ch 384Kbps DELAY 0ms.ac3" "f:\temp\captures\tng_4x17_audio.m4a" -quality=0.42 -normalize -down2
------------------------------------------------------------------------------
AC3, 5.1 channels, 0:43:41, 384kbps, 48kHz, dialnorm: -27dB
Disabling DRC for Nero (E-)AC3 decoding...
Removing AC3 dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 5.1 channels, 24 bits, 48kHz
Downmixing multi channel audio to stereo...
Writing WAV...
Creating file "f:\temp\captures\tng_4x17_audio.m4a.pass1.wav"...
Starting 2nd pass...
Reading WAV...
Reducing depth from 64 to 32 bits...
Encoding AAC <0.42> with NeroAacEnc...
Applying 3,6dB gain...
The original audio track has a constant bit depth of 64 bits.
The processed audio track has a constant bit depth of 32 bits.
The Nero AAC encoder seems to be stuck... <ERROR>
[NeroAacEnc] Processed 0 seconds...
[NeroAacEnc] Processed 1 seconds...
[NeroAacEnc] Processed 2 seconds...
[NeroAacEnc] Processed 3 seconds...
[NeroAacEnc] Processed 4 seconds...
[NeroAacEnc] Processed 5 seconds...
[NeroAacEnc] Processed 6 seconds...
[NeroAacEnc] Processed 7 seconds...
[NeroAacEnc] Processed 8 seconds...
[NeroAacEnc] Processed 9 seconds...
[NeroAacEnc] Processed 10 seconds...
[NeroAacEnc] Processed 11 seconds...
[NeroAacEnc] Processed 12 seconds...
[NeroAacEnc] Processed 13 seconds...
[NeroAacEnc] Processed 14 seconds...
[NeroAacEnc] Processed 15 seconds...
[NeroAacEnc] Processed 16 seconds...
[NeroAacEnc] Processed 17 seconds...
[NeroAacEnc] Processed 18 seconds...
[NeroAacEnc] Processed 19 seconds...
[NeroAacEnc] Processed 20 seconds...
[NeroAacEnc] Processed 21 seconds...
.
.
.
.
[NeroAacEnc] Processed 2531 seconds...
[NeroAacEnc] Processed 2532 seconds...
[NeroAacEnc] Processed 2533 seconds...
[NeroAacEnc] Processed 2534 se
Aborted at file position 2012946500. <ERROR>

ramicio

24th August 2010, 16:51

I have been getting the same thing lately, and randomly. I cannot say for what version. I only started using the Nero encoder on the newest version.

jasonwc

24th August 2010, 18:16

Is there any plan to extract chapter information from MKVs? Currently, eac3to can extract video, audio, and subtitle formats, but it doesn't show or extract the chapter data.

jasonwc

24th August 2010, 23:59

madshi,

I did some more testing and I figured out why I thought the delay was 24 ms while others said it was 21ms. We're both right. We were just counting differently. I assumed that if you simply calculated the difference in length between the DTS-HD MA --> WAV output and the Source PCM output, you would find the delay, since I assumed it was just added that number of samples of digital silence to the beginning of the track.

In fact, it seems more complicated but the bottom line is that the delay information is contained in the "Codec delay" tag which can be found in the DTS-HD MA header, readable by DTS-HD Streamtools. All BD sourced DTS-HD MA tracks lack this tag, and therefore decode properly with no delay. "Raw" DTS-HD MA files created with MAS have a Codec Delay of 1024 samples. A DTS frame is 512 samples, so the delay is always 2 DTS frames, or 21.33ms (assuming 48 Khz source). eac3to need merely read the DTS-HD MA header to properly decode the track. Unfortunately the Arcsoft DTS decoder does not strip the first 2 DTS frames as instructed by the header. In contrast, DTS Streamplayer does. Thus, eac3to ought to either read the DTS-HD MA header to detect the codec delay, or, if that is not possible, simply strip the first 2 DTS frames from any raw DTS-HD MA track.

I tested this way using eac3to 3.24 - Source (TrueHD) ---> WAVS ---> DTS-HD MA ---> WAVS
I compared the Center channel mono-wavs.

Original Source: 54:09.288 (155965840 samples)
DTS-HD MA --> WAV: 54:09.312 (155966976 samples)

Difference: 1136 Samples / 48,000 Samples/sec = 23.666 ms

I checked the length/# of samples in foobar2000 and Adobe Audition, and they matched. The DTS-HD MA --> WAV track is 23.666 ms longer than the source WAV.

However, when I checked the actual audio delay in Adobe Audition, I found the delay was actually only 21.35 ms.

Attached I have posted an Adobe Audition screenshot with 4 WAVs compared side-by-side. See
http://img823.imageshack.us/img823/1902/audition5.th.png (http://img823.imageshack.us/i/audition5.png/)

Wave 1: Source (TrueHD --> WAVS)
Wave 2: DTS-HD MA --> WAVS with -20ms delay
Wave 3: DTS-HD MA --> WAVS with -21ms delay
Wave 4: DTS-HD MA --> WAVS with -22 ms delay

The closest match is Wave3 (-21ms). However, the -21ms sample still appears to be behind by 0.35ms. I calculated this in two ways. First, I calculated the delay visually and saw that the difference was approximately 10:41.777 seconds - 10:41.77665 seconds, which is 0.00035 seconds, or 3.5 ms. I also counted the number of audio samples within the selection and found 17 (18 appear highlighted, but that's because Audition doesn't allow you to precisely select, and it went a bit too far to the right).

17 samples / 48,000 samples/sec = 0.354 ms
Total Delay = 21.35 ms

Thus, it appears that raw DTS-HD MA tracks encoded with the DTS-HD Master Audio Suite (Version 2.50.20) when decoded in eac3to using the Arcsoft DTS decoder, have an additional 23.66 ms in length, but only have a positive delay of 21.35 ms. I have no idea what is going on with the additional 2.3 ms of samples.

Both jj666 and I were able to replicate the same results using a source THD from the Dexter S04 BDs.

Source ---> WAVS ---> DTS-HD MA (MAS Encoder) ---> WAVS

DTS-HD MA WAVS v. Source WAVS:
Additional 23.66 ms in Length
Positive delay of 21.35 ms (appx)

As a DTS frame is 10.666 ms, it appears we were right that the delay is exactly 2 DTS frames, or 21.33 ms.

But it turns out all this testing was unnecessary. Here's the relevant portion of the DTS-HD MA header:

...

Source Samples : 155965840
Sample Rate : 48000Hz
Samples Per Frame : 512
Codec Delay : 1024
...

In other words, the delay can be calculated simply by reading the DTS-HD MA header, and from what we've seen it should always be exactly 2 DTS frames.

1024 samples / 48,000 samples/sec = 21.33 ms delay

So, all eac3to has to do is to read the DTS-HD MA header. DTS-HD MA tracks sourced from Blu-Rays do not have a "Codec Delay" tag in the header, and thus are decoded properly. In contrast, raw DTS-HD MA files produced with MAS do have such a header, and appear to always add a delay of 1024 PCM samples/2 DTS frames. The DTS Streamplayer reads the header and outputs accordingly. Arcsoft does not head the delay info, explaining the 21.33 ms delay in eac3to.

jj666

25th August 2010, 00:09

Madshi,

In addition to the above post, I also saw that a manual delay is ignored when doing a second pass to strip zero bytes (ie, eac3to test.thd test.lpcm -21ms will actually not apply the delay if a second pass is needed).

Cheers,

-jj-

jasonwc

25th August 2010, 02:42

It seems MAS always uses a codec delay of 2 DTS frames (21.33ms). At 48 Khz, the delay is 1024 samples, 2048 at 96 Khz, and 4096 at 192 Khz.

marcio

25th August 2010, 22:44

http://img833.imageshack.us/img833/6417/eac3to.jpg

Bard2

26th August 2010, 18:39

Just to clarify: is "demuxing" multichannel WAV to mono WAVs trough eac3to using commandline like this "eac3to source.wav demuxed.wavs" lossless?

ramicio

26th August 2010, 18:48

Just to clarify: is "demuxing" multichannel WAV to mono WAVs lossless?

Yes it is.

Chouonsoku

27th August 2010, 04:28

eac3to v3.24
command line: eac3to2 english.thd+ac3 english.wavs
------------------------------------------------------------------------------
TrueHD/AC3, 5.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
Extracting TrueHD stream...
Decoding with libav/ffmpeg...
Writing WAVs...
[libav] Substream 0 parity check failed <WARNING>
[libav] Substream 0 checksum failed <WARNING>
[libav] Substream 0 length mismatch. <WARNING>
The libav decoder reported error -1 while decoding. <ERROR>
Aborted at file position 262144. <ERROR>

I've been getting this error when trying to split the TrueHD track into wavs using the latest eac3to, whether it's in the M2TS file or demuxed. It's from the Afro Samurai: Resurrection Blu-Ray. This happens when I try to do any conversion with the files, .wavs, .flac, but I can still extract the core AC3 file just fine.

DreckSoft

28th August 2010, 14:17

Short notice:
There are some kind of MLP files which libav cannot handle: Those with mixed sampling rates (i.e. 96khz for Lf-Rf-C and 48khz for LFE-Ls-Rs). Seems Nero is the only option there. Unfortunately getting the Nero decoder working is ...
It now works after removing Nero and installing the latest Nero Lite 7 version.

Maybe eac3to should check for the mixed sampling rates and select the decoder approriately.

An example for this type of file can be found on the DVD Audio "Nena feat. Nena Live".

If you have any questions about this please write a PM as I'm not reading this thread all the time.

Rumbah

29th August 2010, 02:51

TinTime

29th August 2010, 09:24

One thing I just noticed when extracting audio of an avc hd disc to stereo flac (for mp3 encoding) is that even if eac3to notices that a second pass is needed it just continues writing the compressed flac file till the end. After that it gets overwritten with the second pass flac file.
Wouldn't it be faster to abort the encoding and file writing after it discovers that it needs a second pass (especially if the source and target disc are the same)?

I agree. If I think there's a chance of a second pass then I generally convert to wav as an intermediate file first. I tend to go TrueHD -> wav -> FLAC for example. I find it's the encoding (certainly of FLAC) rather than file writing that takes the time.

tebasuna51

29th August 2010, 09:25

...
Wouldn't it be faster to abort the encoding and file writing after it discovers that it needs a second pass (especially if the source and target disc are the same)?
We need continue until the end to know all gaps/ovelaps/overflows before begin the second pass

TinTime

29th August 2010, 09:37

We need continue ultil the end to know all gaps/ovelaps/overflows before begin the second pass

He's not talking about aborting all processing though - just "the encoding and file writing".

I admit that I initially read the post exactly the same way as you did :)
I'd typed out a similar reply to yours before the penny dropped and I realised what Rumbah meant.

Rumbah

29th August 2010, 12:17

He's not talking about aborting all processing though - just "the encoding and file writing".

Exactly. I'm aware that the whole file has to be scanned but if you know that you have to do a second pass there is no need to actually encode and write the first pass results to disc.

video_magic

30th August 2010, 00:12

1st pass could specify -o NUL , and only write the info to a log file which it needs for the 2nd pass.

2nd pass uses the info from the log file generated from the 1st pass, and this time writes the actual outfile which we want.

It would save some diskdrive wear, be a bit faster, & use less CPU power and generated heat, than writing an output file on the 1st pass for nothing.

Jeff Flowerday

30th August 2010, 02:14

Any one else seeing this with the Cars Blu-Ray? eac3to 3.24

v02 0:01:15 The source file seems to be damaged (discontinuity).
a04 0:01:15 The source file seems to be damaged (discontinuity).
a03 0:01:15 The source file seems to be damaged (discontinuity).
v02 0:01:15 Detected PTS break, increasing PTS by 41.7ms...
a03 0:01:15 Detected PTS break, increasing PTS by 8ms...
a03 libav Lossless check failed - expected b, calculated 3c
a04 0:01:15 Detected PTS break, increasing PTS by 32.0ms...
a03 0:01:50 The source file seems to be damaged (sync byte missing).
a04 0:01:50 The source file seems to be damaged (sync byte missing).
v02 0:01:51 The source file seems to be damaged (sync byte missing).
v02 0:01:56 The source file seems to be damaged (discontinuity).
a03 0:01:56 The source file seems to be damaged (discontinuity).
a04 0:01:56 The source file seems to be damaged (discontinuity).
a03 libav Lossless check failed - expected ef, calculated d0
v02 0:01:29 The source file seems to be damaged (discontinuity).
a04 0:01:30 The source file seems to be damaged (discontinuity).
a03 0:01:30 The source file seems to be damaged (discontinuity).

Thunderbolt8

30th August 2010, 04:02

is it OK if that line "swapping endian..." appears twice?

eac3to v3.24
command line: eac3to movie 1) 3: movie.pcm
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 1:40:40, 24p /1.001
1: Chapters, 17 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 1.0 channels, 24 bits, 48kHz
4: AC3, English, 1.0 channels, 192kbps, 48kHz
5: Subtitle (PGS), English
[a03] Extracting audio track number 3...
[a03] Reading RAW/PCM...
[a03] Swapping endian...
[a03] Swapping endian...
[a03] Creating file "movie.pcm"...
[a03] The original audio track has a constant bit depth of 24 bits.
Video track 2 contains 144816 frames.
eac3to processing took 17 minutes, 51 seconds.
Done.

SC05

3rd September 2010, 22:36

I love eac3to, thanks for all the work put into it.

I have one issue which I have never been able to solve, despite numerous attempts and searches.

Two DTS-MA 6.1 soundtracks will not decode properly in eac3to no matter what I have tried (Top Gun and X-Men the Last Stand). I am using eac3to v3.24, Arcsoft DTS decoder v1.1.0.7. It doesn't matter whether I decode straight from the disc, or from the disc to a .dtsma file, then decode. The resultant FLAC file always sounds like a garbled mess (though you can hear the underlying dialog). I have tried doubling the back channel, to no avail.

I though the Arcsoft decoder was supposed to work great with 6.1 and 7.1, as stated in the original post of this thread. What gives? Am i using the wrong version, or do I need to use a different decoder?

Thanks, and I hope I can find some guidance on this issue.

P.S. I have also tried decoding to mono WAVs, and I end up with seven files that all sound identical and garbled (even the LFE, which contains plenty of non-low frequency sound in this case).

tebasuna51

4th September 2010, 00:21

It's a know bug of v1.1.0.7, use v1.1.0.0 for DTS-MA 6.1

SC05

4th September 2010, 00:37

Thanks for the quick reply. I had no idea about that. I have been pulling my hair out over this one. I'll track down that version and give it a go.

SC05

4th September 2010, 02:08

Wow, it works perfectly now, thanks again!

yesgrey

4th September 2010, 11:18

madshi,
Recently I noticed that when demuxing all files to the same disk the video file is created with hundreds of fragments, even though the disk has been defragged and lots of empty space available. Since I have 3 hard disks, and I only extract a few streams (original audio, video, chapters and 1 or 2 subs), I thought that I might be able to speed up the processing by writing the demuxed files to separate disks. I've created a 256 MB ram disk for putting the smaller files, and sent the video and audio for different hard disks. It didn't seem to have resulted, because the processing time increased by 50% (different BR), but on the other hand the fragmentation was smaller. Instead of 137 fragments I only got 29. The first case was a demuxed mpeg2 stream, and the second was a H264 stream to mkv file. The curious thing is that only the video file is fragmented, so would it be possible for you to avoid the fragmentation? For example, by first allocating the full size of the file, or any other method? I think this would be a nice feature because besides the speedup it would also help reducing the stress on the hard disks...

laserfan

4th September 2010, 15:40

It's a know bug of v1.1.0.7, use v1.1.0.0 for DTS-MA 6.1Do you (or anyone) know:

1. Was 1.1.0.0 the version with v2.0 of TMT? I have v2 but am using 3.0 now.

2. Is it as simple as overwriting the 1.1.0.7 dll (mine's in my eac3to dir)

3. Was there a "known bug" in 1.1.0.0 that we exchange for when we revert from .7 to .0

TIA

:confused:

tebasuna51

4th September 2010, 18:08

1) ArcSoft.TotalMedia.Theatre.v2.1.6.129

2) If you have the full TMT 3.0 installed maybe you can't use v1.0.0.0.
If you have only some files in eac3to folder:
- unregister ASAudioHD.ax from v1.1.0.7 (maybe not needed but is a good method)
- replace the files
- register the ASAudioHD.ax from v1.1.0.0

3) I don't know bugs of v1.1.0.0 working with eac3to

laserfan

4th September 2010, 18:16

7ekno

5th September 2010, 11:21

1. Was 1.1.0.0 the version with v2.0 of TMT? I have v2 but am using 3.0 now.
Version 1.1.0.0 was the absolute final version for the v2.x series (the "Update" files are still in the TMT archive) ..

2. Is it as simple as overwriting the 1.1.0.7 dll (mine's in my eac3to dir)
Yes, don't even bother with any other files, just overwrite "dtsdecoderdll.dll" with the older version (or rename 1.1.0.7 and copy 1.1.0.0 into same directory) ...

3. Was there a "known bug" in 1.1.0.0 that we exchange for when we revert from .7 to .0
Doesn't seem to be, but changelogs don't appear for these files, so very difficult to assess what the actual differences are ...

7ek

laserfan

5th September 2010, 16:19

Version 1.1.0.0 was the absolute final version for the v2.x series... just overwrite "dtsdecoderdll.dll" with the older version (or rename 1.1.0.7 and copy 1.1.0.0 into same directory) ...Thanks 7ekno. I found it easily and just copied it over and it works. :)

Frogger13

7th September 2010, 18:22

I want to UP this Issue as I'm getting the same since I moved from version 3.22 to 3.24. Happens especially when under heavy load (e.g. multiple encodings at the same time) but also spuriously when encoding one file at a time :(

I've been having a problem with eac3to and NeroAACEnc with later versions of eac3to. After encoding the decoded WAV, the log says that NeroAACEnc seems to be stuck (but I've no clue as to why) and it starts to encode the file again. I don't remember at which point the problem started to occur but it hasn't been there for long.

eac3to v3.24
command line: c:\utils\eac3to\eac3to.exe "e:\temp\dvd-rip\star trek tng\4x17 T81 3_2ch 384Kbps DELAY 0ms.ac3" "f:\temp\captures\tng_4x17_audio.m4a" -quality=0.42 -normalize -down2
------------------------------------------------------------------------------
AC3, 5.1 channels, 0:43:41, 384kbps, 48kHz, dialnorm: -27dB
Disabling DRC for Nero (E-)AC3 decoding...
Removing AC3 dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 5.1 channels, 24 bits, 48kHz
Downmixing multi channel audio to stereo...
Writing WAV...
Creating file "f:\temp\captures\tng_4x17_audio.m4a.pass1.wav"...
Starting 2nd pass...
Reading WAV...
Reducing depth from 64 to 32 bits...
Encoding AAC <0.42> with NeroAacEnc...
Applying 3,6dB gain...
The original audio track has a constant bit depth of 64 bits.
The processed audio track has a constant bit depth of 32 bits.
The Nero AAC encoder seems to be stuck... <ERROR>
[NeroAacEnc] Processed 0 seconds...
[NeroAacEnc] Processed 1 seconds...
[NeroAacEnc] Processed 2 seconds...
[NeroAacEnc] Processed 3 seconds...
[NeroAacEnc] Processed 4 seconds...
[NeroAacEnc] Processed 5 seconds...
[NeroAacEnc] Processed 6 seconds...
[NeroAacEnc] Processed 7 seconds...
[NeroAacEnc] Processed 8 seconds...
[NeroAacEnc] Processed 9 seconds...
[NeroAacEnc] Processed 10 seconds...
[NeroAacEnc] Processed 11 seconds...
[NeroAacEnc] Processed 12 seconds...
[NeroAacEnc] Processed 13 seconds...
[NeroAacEnc] Processed 14 seconds...
[NeroAacEnc] Processed 15 seconds...
[NeroAacEnc] Processed 16 seconds...
[NeroAacEnc] Processed 17 seconds...
[NeroAacEnc] Processed 18 seconds...
[NeroAacEnc] Processed 19 seconds...
[NeroAacEnc] Processed 20 seconds...
[NeroAacEnc] Processed 21 seconds...
.
.
.
.
[NeroAacEnc] Processed 2531 seconds...
[NeroAacEnc] Processed 2532 seconds...
[NeroAacEnc] Processed 2533 seconds...
[NeroAacEnc] Processed 2534 se
Aborted at file position 2012946500. <ERROR>

Thunderbolt8

9th September 2010, 12:46

whats the status on the framerate recognition thing in case of movies which consist of more than 1 .m2ts file? is it considered a problem if that value is estimated wrong at the stage of parsing? got a case here in which both .m2ts files have the same framerate, but the complete playlist is nevertheless estimated wrongly.

edit: for the same movie I got problems with the subtitles of those combined .m2ts files. its ok with another program, but this problem does not occur at all when just using the subs from the main .m2ts file alone.

mastrandrea

11th September 2010, 18:06

I'm trying to slow down a 4 channel ac3, but eac3to return this error:
The AC3 encoder received a non-supported data format (float, 4, 64, -).
Aborted at file position 262144.
Am I doing somethig wrong?

Here's the mediainfo log for this particular track:
Audio
Format : AC-3
Format/Info : Audio Coding 3
Mode extension : CM (complete main)
Duration : 1h 29mn
Bit rate mode : Constant
Bit rate : 384 Kbps
Channel(s) : 4 channels
Channel positions : Front: L C R, Side: C
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Stream size : 245 MiB (100%)
Thanks in advance!

Snowknight26

11th September 2010, 20:16

Am I doing somethig wrong?

Apart from not searching beforehand, no.

http://forum.doom9.org/showthread.php?p=1412746#post1412746

tebasuna51

11th September 2010, 20:39

Use eac3to and a 'pipe' to Aften.exe

eac3to input stdout.wav -slowdown | Aften -b 384 -readtoeof 1 - output.ac3

Thunderbolt8

13th September 2010, 18:50

when using -slowdown, to which fps rate does eac3to actually slow down, to 23.976 or precisely to 24/1.001? and when having normal source blu-rays with 23.976 audio and video, is it 24/1.001 here as well or just 23.976?

(basically Im asking when trying to combine track from different releases, if there could be a very slight sync problem in cases there are difference between source and slowed tracks due to that)

and in that respect I got a 7.1 blu-ray track here:

eac3to v3.24
command line: eac3to H:\express.flac H:\expressslowed.flac -24.000 -slowdown
------------------------------------------------------------------------------
FLAC, 7.1 channels, 1:42:14, 16 bits, 977kbps, 48kHz
Decoding FLAC...
Changing FPS from 24.000 to 23.976...
Reducing depth from 64 to 24 bits...
Encoding FLAC with libFlac...
Creating file "H:\expressslowed.flac"...
Clipping detected, a 2nd pass will be necessary. <WARNING>
The original audio track has a constant bit depth of 16 bits.
The processed audio track has a constant bit depth of 24 bits.
Starting 2nd pass...
Decoding FLAC...
Changing FPS from 24.000 to 23.976...
Reducing depth from 64 to 24 bits...
Encoding FLAC with libFlac...
Applying -0,38dB gain...
Creating file "H:\expressslowed.flac"...
The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 50 minutes, 35 seconds.
Done.

doesnt clipping usually only occur with hdtv caps? and has that anything to do with those -0,5dB gain? or why else must gain have been applied (the 7.1 source was a DTS-HD MA track with strange setup btw.)

regarding that 7.1 strange setup, I only get that note in combination with the audio of the complete blu-ray structure. when I demux that track, that information is gone. so is some kind of information now lost regarding channel setup when I want to transform that demuxed track into another audio format, would audio channels or the converted content be different compared to when using the source audio within its blu-ray structure?
if so, is it maybe to fix this with a kind of -switch which assumes the source uses such a kind of setup or maybe adding that information to the demuxed dtsma track header?

dansrfe

13th September 2010, 19:49

Does eac3to do timestreching or just a simple slowdown or speedup neglecting pitch etc.?

TinTime

13th September 2010, 21:37

doesnt clipping usually only occur with hdtv caps?

Clipping can occur when re-sampling, as is happening here.

tebasuna51

14th September 2010, 00:35

when using -slowdown, to which fps rate does eac3to actually slow down, to 23.976 or precisely to 24/1.001?
24/1.001
and when having normal source blu-rays with 23.976 audio and video, is it 24/1.001 here as well or just 23.976?
24/1.001
doesnt clipping usually only occur with hdtv caps?
Can occur with any lossy conversion, like here changing audio duration.
and has that anything to do with those -0,5dB gain?
Is only a info (that I request to madshi).

Thunderbolt8

14th September 2010, 02:51

24/1.001
Can occur with any lossy conversion, like here changing audio duration.I used it on a flac file, so after a slowdown (or speedup) the audio quality, aside from the lost or gained pitch, is not lossless any more?

TinTime

14th September 2010, 04:34

I used it on a flac file, so after a slowdown (or speedup) the audio quality, aside from the lost or gained pitch, is not lossless any more?

No, it's been re-sampled - not (usually) a lossless procedure. The only way to keep it lossless in this case would be to play the original track slower and have a DAC that handles a (48000/1.001)kHz sample rate.

Thunderbolt8

15th September 2010, 05:00

has anyone already made experiences with that low volume problem of those dtsma strange setup 7.1 channels? is the volume still evenly distributed over the channels like normal, just lower now, so that cranking up the volume solves the problem? are there any other problems aside the volume thing which could come along when trying to transform to 7.1 flac? (only have a 5.1 system, so I cannot really test whether all the channels are correct)

xkodi

15th September 2010, 13:13

has anyone already made experiences with that low volume problem of those dtsma strange setup 7.1 channels? ...

Sonic DTS decoder correctly and bit-perfectly decodes DTS-HD MA 7.1 with "strange setup" to 5.1 channels.

so, it's your choice: get bit-perfect and correct 5.1 channels from your DTS-HD MA 7.1 with "strange setup" using Sonic DTS decoder or get 7.1 channels with using Arcsoft decoder, but all of those 7.1 channels are not-bit-perfectly decoded and since at least to me it's not clear exactly what processing of the audio data Arcsoft decoder is doing in case of "strange setup" we can even assume they are wrong.

so, at least in my opinion the safest way to go is using Sonic DTS decoder in such case of "strange setup" DTS-HD MA 7.1.

for the sake of completeness i can elaborate a little more with listing the facts:

1. "normal" (not "strange setup") DTS-HD MA 7.1 has the following channel layout:

http://thumbnails31.imagebam.com/9780/43bb3497794528.jpg (http://www.imagebam.com/image/43bb3497794528)

which in the way how Microsoft (and thus eac3to) named the channels is the same as:

http://thumbnails30.imagebam.com/9780/160e7497795034.jpg (http://www.imagebam.com/image/160e7497795034)

or DTS channel names mapped to Microsoft channel names are as follows:

DTS channel name <---> Microsoft channel name

L <---> L
R <---> R
C <---> C
LFE <---> LFE
Lsr <---> BL
Rsr <---> BR
Lss <---> SL
Rss <---> SR

2. "strange setup" DTS-HD MA 7.1 has the following channel layout:

http://thumbnails28.imagebam.com/9780/da2a3097795712.jpg (http://www.imagebam.com/image/da2a3097795712)

or using the way how Microsoft (and thus eac3to) named the channels then DTS channel names mapped to Microsoft channel names are as follows:

L <---> L
R <---> R
C <---> C
LFE <---> LFE
Lsr <---> BL
Rsr <---> BR
Ls <---> i don't know how it's called in Microsoft terms
Rs <---> i don't know how it's called in Microsoft terms

3. DTS-HD MA 5.1 has the following channel layout:

http://thumbnails30.imagebam.com/9780/addcf997797266.jpg (http://www.imagebam.com/image/addcf997797266)

which is the same as "strange setup" DTS-HD MA 7.1 layout with missing "Lsr" (or "BL" in MS terms) and "Rsr" (or "BL" in MS terms)

4. with using Sonic DTS decoder for "normal" (not "strange setup") DTS-HD MA 7.1 you get the following 5.1 channels:

L <---> L
R <---> R
C <---> C
LFE <---> LFE
Lsr <---> not decoded
Rsr <---> not decoded
Lss <---> SL
Rss <---> SR

which even that are bit-perfectly decoded are not the correct 5.1 channel configuration according to 3., because you're missing "Ls" and "Rs" channels that are used in DTS-HD MA 5.1 channel configuration as it's shown in 3., i.e. you can't convert correctly "normal" (not "strange setup") DTS-HD MA 7.1 to 5.1 channels with just using Sonic DTS decoder and actually you don't even need that, because Arcsoft gives 7.1 channels bit-perfectly decoded in case of "normal" (not "strange setup") DTS-HD MA 7.1.

5. with using Sonic DTS decoder for "strange setup" DTS-HD MA 7.1 you get the following 5.1 channels:

L <---> L
R <---> R
C <---> C
LFE <---> LFE
Lsr <---> not decoded
Rsr <---> not decoded
Ls <---> decoded, but i don't know how it's called in Microsoft terms
Rs <---> decoded, but i don't know how it's called in Microsoft terms

or you have 5.1 channels bit-perfectly decoded and also they are:

L, R, C, LFE, Ls, Rs

which is the correct configuration for DTS-HD MA 5.1 channels according to 3.

so, in very short: DTS-HD MA 7.1 with "strange setup" can be decoded bit-perfectly to correct 5.1 channels and i'm not sure why "eac3to" doesn't inform the user about it and even do that by default, because at least in my opinion it's much proper way than using Arcsoft. also, because of those facts it's correct to say that you can think in case of DTS-HD MA 7.1 with "strange setup" that you actually have DTS-HD MA 5.1, because that is what information you can extract correctly and bit-perfectly from it.

nautilus7

15th September 2010, 15:13

Thanks, xkodi. So you can use sonic to get (correct) 5.1 ch and arcsoft to get the (not correct) extra 2 ch and combine them in an audio editor to reach 7.1 ch.

Midzuki

15th September 2010, 16:26

Sonic DTS decoder correctly and bit-perfectly decodes DTS-HD MA 7.1 with "strange setup" to 5.1 channels.

:thanks: 4 sharing that info.

...
or get 7.1 channels with using Arcsoft decoder, but all of those 7.1 channels are not-bit-perfectly decoded and since at least to me it's not clear exactly what processing of the audio data Arcsoft decoder is doing in case of "strange setup" we can even assume they are wrong.

FWIW, and for the time being, I have confirmed that ArcSoft doesn't decode stereo lossless-DTS correctly. :(

Ls <---> i don't know how it's called in Microsoft terms
Rs <---> i don't know how it's called in Microsoft terms

Ls == Back-Left
Rs == Back-Right

Besides, Lsr and Rsr "do not exist" in the Wave-Format-Extensible "universe"
( and if they existed, they would be called
"Back Left of Center" && "Back Right of Center" ).

Thunderbolt8

15th September 2010, 18:42

is it really bit-perfect, meaning that the information from those 2 other channels is then added somewhere to the front or side channels of 5.1? or is that information from those 2 channels dropped (because you said "not decoded", which would not be bit-perfect then compared to the source information)?

and in case its added to the 5.1 channels, is there any way to measure how good it would sound compared to the same sound recorded and distributed to 5.1 only in the first place (so how accurate is the distribution of those 2 non decoded channel to the 5.1 channels)?

xkodi

15th September 2010, 19:05

So you can use sonic to get (correct) 5.1 ch and arcsoft to get the (not correct) extra 2 ch and combine them in an audio editor to reach 7.1 ch.

yes, if you want 7.1 channels that's one way to lower the error, but still you won't get completely correct 7.1 channels as in case if you go for just 5.1 channels.

one other possibility to lower error maybe is to use "Ls" and "Rs" to calculate from them new "Lsr" and "Rsr", because according to:

http://thumbnails28.imagebam.com/9780/da2a3097795712.jpg (http://www.imagebam.com/image/da2a3097795712)

they seem on very close positions to each other and "tebasuna51" is very good in making such calculation. however, it's just an idea, because i don't know if new "Lsr" and "Rsr" made in such way will be better than what Arcsoft is decoding for "Lsr" and "Rsr".

Ls == Back-Left
Rs == Back-Right

Besides, Lsr and Rsr "do not exist" in the Wave-Format-Extensible "universe"
( and if they existed, they would be called
"Back Left of Center" && "Back Right of Center" ).

thanks, it's very confusing, because it seems to me in 5.1 channels configuration:

DTS Ls == MS BL
DTS Rs == MS BR

but in 7.1 channel configuration:

DTS Lsr == MS BL
DTS Rsr == MS BR

or maybe i'm wrong again - at least that's what i can conclude from the picture in my initial post - i took all of those pictures from DTS and MS.

is it really bit-perfect, meaning that the information from those 2 other channels is then added somewhere to the front or side channels of 5.1? or is that information from those 2 channels dropped (because you said "not decoded", which would not be bit-perfect then compared to the source information)?

i'm not sure that i understand what you're asking, but let me try to explain it again in different way:

* "strange setup" DTS-HD MA 7.1 decoded with Arcsoft: all 7.1 channels are decoded, but not any of those 7.1 channels is decoded bit-perfectly, which means not lossless and thus it's not correct, because DTS-HD MA is lossless. how wrong Arcsoft decodes them is not clear at least to my knowledge, because it's not clear what kind of processing of the audio data Arcsoft does in the case of "strange setup" DTS-HD MA 7.1.

* "strange setup" DTS-HD MA 7.1 decoded with Sonic: only 5.1 channels are decoded from the initial 7.1 channels, but those 5.1 channels are decoded:

- in bit-perfectly way, i.e. lossless
- as correct set of channels for 5.1 configuration, because what you get of decoding with Sonic is: (L, R, C, LFE, Lsr, Rsr, Ls, Rs) are reduced to (L, R, C, LFE, Ls, Rs) and so there is no any loss of information for 5.1 channel setup

or as i tried to summarize it my initial post "strange setup" DTS-HD MA 7.1 decoded with Sonic is way to convert that to 5.1 channel configuration without any loss of audio data for 5.1 channel setup. so, if you have movie with "strange setup" DTS-HD MA 7.1 better think of it the audio is DTS-HD MA 5.1, because there is no way to get more of it on computer.

i don't if this way of explaining is more clear or not.

and in case its added to the 5.1 channels, is there any way to measure how good it would sound compared to the same sound recorded and distributed to 5.1 only in the first place (so how accurate is the distribution of those 2 non decoded channel to the 5.1 channels)?

if you're talking about what "nautilus7" there is no way to know, because we don't know what Arcsoft is doing wrong, that's why the safest way is just get 5.1 which are at least bit-perfect/lossless and correct set of channels for 5.1 - for example even if assume we can get all channels of "strange setup" DTS-HD MA 7.1 decoded bit-perfectly then on 5.1 channel system "Lsr" and "Rsr" will be stripped and that's clear form DTS channel layout in 2. and 3. in my initial post.

Thunderbolt8

15th September 2010, 19:18

so its basically like theres 2 kind of channel setup information stored for the same audio content, one for 7.1 strange setup and one for normal 5.1, and sonic chooses the 5.1 setup then?

my 2nd question basically was whether that XX channel information was added arbitrarily to those 5.1 channels. but if theres another mask available which also takes all the information into consideration from the same source sound, only this time distribution is 5.1 instead of 7.1, then I guess it should sound as if that source sound was recorded to 5.1 channels only in the first place.

can I use the same procedure also for normal 7.1 dtsma tracks, those without strange setup? or is there nothing like 5.1 channel distribution information available for sonic then?

btw. ive had 24-bit 7.1 tracks which then got transformed to 16-bit 5.1 with sonic, is it still lossless?

FWIW, and for the time being, I have confirmed that ArcSoft doesn't decode stereo lossless-DTS correctly. :(hm where? does at least sonic do it correctly?

Midzuki

15th September 2010, 19:44

Just for the record, these are the only "7.1 DTS channel layouts" which are compatible with the Wave-Format-Extensible definitions:

L, R, C, LFE, Ls, Rs, Lw, Rw

L, R, C, LFE, Ls, Rs, Cs, Ch
(Cs == Back Center, Ch == Top Front Center)

L, R, C, LFE, Ls, Rs, Cs, Oh
(Oh == Top Center)

L, R, C, LFE, Ls, Rs, Lh, Rh
(Lh == Top Front Left, Rh == Top Front Right)

Argh!

Anacletus

16th September 2010, 10:14

I want to UP this Issue as I'm getting the same since I moved from version 3.22 to 3.24. Happens especially when under heavy load (e.g. multiple encodings at the same time) but also spuriously when encoding one file at a time :(

I'm also experiencing this problem lately on many of my audio encodes, sources varies from flac to dts/dtshd.. neroaacenc abort the encoding after a while.. any clues?

flebber

18th September 2010, 14:51

Hi. Having an issue with subs being extracted by eac3to. Seems to have an error passing bitstream. I inputted a Mkv file made with Makemkv into Megui HD Streams Extracto.

Initial post in Megui Thread here, referred here as error appeared in eac3to process http://forum.doom9.org/showthread.php?p=1444148#post1444148

I have uploaded a small sample which casues the error.

http://www.mediafire.com/file/87sp8re9edx8te0/Snow%20White%20%26%20The%20Seven%20Dwarves%20%281%29-001.mkv

buzzqw

18th September 2010, 18:43

eac3to is unable to extract properly vobsub from mkv
(that why in HDC now i use mkvextract...)

BHH

xkodi

19th September 2010, 16:00

can I use the same procedure also for normal 7.1 dtsma tracks, those without strange setup? or is there nothing like 5.1 channel distribution information available for sonic then?

in very short - you can't use what i explained for "normal 7.1 dtsma tracks" (i.e. the one that are not "strange setup"). the reason for that is simple:

- channels inside "strange setup" 7.1 dtsma track are the set (L, R, C, LFE, Lsr, Rsr, Ls, Rs) and sub-set of (L, R, C, LFE, Ls, Rs) is what you need for proper 5.1 channels and exactly that set (L, R, C, LFE, Ls, Rs) is what Sonic decodes bit-perfectly from such track and that's why using Sonic you end up with bit-perfect and correct set of channels for 5.1

- channels inside "normal 7.1 dtsma tracks" are the set (L, R, C, LFE, Lsr, Rsr, Lss, Rss) and so there is no "Ls" and "Rs" that you need for 5.1 and using Sonic you will end up with 5.1 channels that are simple not the set of channels you need for proper 5.1.

btw. ive had 24-bit 7.1 tracks which then got transformed to 16-bit 5.1 with sonic, is it still lossless?

if the track is 24-bit for sure (and not 16-bit padded with "0" to 24-bit) then of course 24->16 is not lossless operation.

hm where? does at least sonic do it correctly?

i don't know, but i will check that when have some spare time.

Thunderbolt8

19th September 2010, 16:10

alright, thanks!

SLKabaker

21st September 2010, 02:26

I am relatively new to Video Conversion. I have only been doing it for a couple months now. I purchased ArcSoft MediaConverter 7. It includes the same DTS Decoder that comes with ArcSoft TMT. Is there a way to get eac3to to recognize that? I registered ASAudioHD.ax (using regsvr32) in the directory it was installed to. However, when I run eac3to -test, it still does not detect it. I ran GraphEdit, and it sees that the ArcSoft Audio Decoder is a DirectShow CODEC. I would appreciate any help in how to get eac3to to use the DTS Decoder. Do I need to copy the files and put them in another directory? I am running Windows 7 64Bit.

Also, for those of us unable to purchase Nero 7, as they are up to Nero 10 now, is it possible for eac3to to be updated to use ArcSoft's AC-3/E-AC3 decoder. It does not support Dolby TrueHD but it does support 7.1 Dolby Digital Plus. As well as, for AAC, Nero gives away free its Nero AAC Encoder and Decoder. eac3to already uses the NeroAACEncoder.exe, is there a way to have it use NeroAACDecoder.exe instead of Nero 7? Also, Microsoft put a Full E-AC3 Decoder into Windows 7...

I am sorry if any of this is already explained in the 500+ pages of this forum. Please just point me in the right direction and I will learn quickly. Thank you for all your help.

Midzuki

21st September 2010, 02:55

@ SLKabaker:

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

IMPORTANT: don't forget the file checkactivate.dll.

Thunderbolt8

21st September 2010, 03:15

afaik madshi wont bother to implement 7.1 DD+, because DD+ died when HD DVD died and there wasnt really any movie which used 7.1 DD+.

mr.duck

23rd September 2010, 02:03

Mark_A_W

23rd September 2010, 12:23

I'm having terrible trouble with the last 4 Seamless Branching BD discs I've done.

The lipsync is all over the place - the audio does not match the video.

I've tried with 3.22 and 3.24. It's running the second pass, but the result is wrong. I'm converting the audio to Flac.

Has anyone else had this problem?

Thunderbolt8

23rd September 2010, 13:19

afaik never had these problems with BDs since it was working correctly.

flac's framelength is 1ms, so there shouldnt be any delay at all.

Mark_A_W

23rd September 2010, 13:53

No, not a constant delay.

The audio varies from scene to scene, which must have been different m2ts file chunks. It's like the overlap correction is incorrect.

It used to work fine, but I haven't been able to figure which version it went wrong.

Mark_A_W

23rd September 2010, 13:58

I get a correct result if I use MakeMKV tp make a MKV, then use eac3to to convert convert the audio from that to flac (then mux with mkvmerge).

Inspector.Gadget

23rd September 2010, 14:46

Mark_A_W: Are you getting the message seen in mr. duck's post immediately above?

DJ_Phatic

23rd September 2010, 21:06

SamuriHL

23rd September 2010, 21:08

I would like to start using eac3to again but I find that when I install Haali, it decides to take over my computer even though i untick everything when installing.

I don't Haali to be used for playback of mkv and ts/m2ts. Any suggestions how I can install haali without it taking priority? Filemerit doesn't seem to make a difference.

I use batch files to unregister all the dlls/ax files for both 32 and 64 bit environments. Then I just register the files needed for mkv when I need it in the 32 bit environment.

Midzuki

23rd September 2010, 22:54

I would like to start using eac3to again but I find that when I install Haali, it decides to take over my computer even though i untick everything when installing

I don't Haali to be used for playback of mkv and ts/m2ts. Any suggestions how I can install haali without it taking priority? Filemerit doesn't seem to make a difference.

It looks like the "dictator complex" of Haali has been fixed in version 1.10.262.12 (2010/Aug/14). Except for the "generic" (Matroska) splitter, all its other components remain happily set at merit=0x00200000.

As for the playback of transport streams, I highly recommend the ArcSoft splitter and decoders. :)

Mark_A_W

23rd September 2010, 22:57

Mark_A_W: Are you getting the message seen in mr. duck's post immediately above?

Yes.
And lots of discontinuity errors. Then Video overlap errors.

But it then runs the second pass, which I thought fixed that.

Do I need to manually run a 3rd pass? (I'll have to talk to MikeEby about including that automatically in his Gui, if that is the case).

Thanks

Mark

sub24ox7

23rd September 2010, 23:38

yea I have done a few 2 in one encodes lately last was Anchoman: The Legend Of Ron Burgundy which had an authoring error where there was two of the same dreamwork clip intro in both of the .mpls files for the two runtimes. Anyway the disc uses branching obviously and when demuxing the video and audio and after encoding the video and adding the audio and muxing with mkvmerge there are no audio gaps via that method. I have noticed
that playback with mpc-hc of blu-ray discs with branching via the mpls file the audio is getting out of sync everytime a new m2ts file is played in the mpls.

mr.duck

23rd September 2010, 23:46

Do I need to manually run a 3rd pass? (I'll have to talk to MikeEby about including that automatically in his Gui, if that is the case).

I've already tried another pass with eac3to and the results are less than perfect.

Something else I have found out, if I simply demux the file with another program like TSmuxer, and then play the output files (which is badly out of sync at this point) but then manually adjust the audio sync in MPC HC to the exact number reported by eac3to when it scans the file (-650ms in my example file), then I get better results than if eac3to did the muxing to mkv which is supposed to take care of this offset! Even after all these multiple passes.

eac3to is nearly there, but I think I worked out what the problem is. eac3to needs to insert missing frames in the audio or video to maintain the correct timecodes to keep it in sync. For me it happens when the source file is not perfect.

SLKabaker

24th September 2010, 05:46

@Midzuki

Thank you for the link. I appreciate the help. I will try these options out.

hoju3508

24th September 2010, 06:06

BDInfo shows these "Angles" at the beginning of the movie.

DISC INFO:

Disc Title: MEET_THE_ROBINSONS
Disc Size: 41,463,082,132 bytes
Protection: AACS
BD-Java: Yes
BDInfo: 0.5.6

PLAYLIST REPORT:

Name: 00035.MPLS
Length: 1:34:30 (h:m:s)
Size: 26,194,931,712 bytes
Total Bitrate: 36.96 Mbps

Angle 1 Length: 0:08:39 (h:m:s) / 1:34:30 (h:m:s)
Angle 1 Size: 2,380,879,872 bytes / 26,189,082,624 bytes
Angle 1 Total Bitrate: 36.63 Mbps / 36.95 Mbps

Angle 2 Length: 0:08:39 (h:m:s) / 1:34:30 (h:m:s)
Angle 2 Size: 2,382,815,232 bytes / 26,191,017,984 bytes
Angle 2 Total Bitrate: 36.66 Mbps / 36.95 Mbps

All Angles Length: 1:51:50 (h:m:s)
All Angles Size: 30,958,626,816 bytes
All Angles Bitrate: 36.91 Mbps

VIDEO:

Codec Bitrate Description
----- ------- -----------
MPEG-4 AVC Video 24057 kbps 1080p / 23.976 fps / 16:9 / High Profile 4.1
MPEG-4 AVC Video (1) 2176 (23726) kbps 1080p / 23.976 fps / 16:9 / High Profile 4.1
MPEG-4 AVC Video (2) 2177 (23745) kbps 1080p / 23.976 fps / 16:9 / High Profile 4.1

AUDIO:

Codec Language Bitrate Description
----- -------- ------- -----------
LPCM Audio English 4608 kbps 5.1 / 48 kHz / 4608 kbps / 16-bit
DTS Audio Dutch 1509 kbps 5.1 / 48 kHz / 1509 kbps / 24-bit / DN -4dB
DTS Audio French 1509 kbps 5.1 / 48 kHz / 1509 kbps / 24-bit / DN -4dB
Dolby Digital Audio English 640 kbps 5.1 / 48 kHz / 640 kbps / DN -4dB
Dolby Digital Audio Dutch 640 kbps 5.1 / 48 kHz / 640 kbps / DN -4dB
Dolby Digital Audio Dutch 640 kbps 5.1 / 48 kHz / 640 kbps / DN -4dB
Dolby Digital Audio French 640 kbps 5.1 / 48 kHz / 640 kbps / DN -4dB
Dolby Digital Audio English 192 kbps 2.0 / 48 kHz / 192 kbps / DN -4dB / Dolby Surround

SUBTITLES:

Codec Language Bitrate Description
----- -------- ------- -----------
Presentation Graphics English 37.935 kbps
Presentation Graphics English 41.886 kbps
Presentation Graphics English 0.086 kbps
Presentation Graphics Dutch 25.859 kbps
Presentation Graphics Dutch 49.223 kbps
Presentation Graphics Dutch 0.146 kbps
Presentation Graphics French 29.440 kbps
Presentation Graphics French 59.777 kbps
Presentation Graphics French 1.373 kbps

FILES:

Name Time In Length Size Total Bitrate
---- ------- ------ ---- -------------
00120.M2TS 0:00:00.000 0:00:49.883 225,810,432 36,214
00000.M2TS 0:00:49.883 0:00:56.765 264,382,464 37,260
00034.M2TS (1) 0:00:49.883 0:00:56.765 261,402,624 36,840
00317.M2TS (2) 0:00:49.883 0:00:56.765 261,414,912 36,842
00001.M2TS 0:01:46.648 0:06:24.258 1,782,933,504 37,119
00002.M2TS 0:08:10.907 0:00:35.076 162,091,008 36,968
00036.M2TS (1) 0:08:10.907 0:00:35.076 162,109,440 36,973
00318.M2TS (2) 0:08:10.907 0:00:35.076 162,109,440 36,973
00003.M2TS 0:08:45.983 0:02:30.066 694,622,208 37,030
00004.M2TS 0:11:16.050 0:00:30.030 139,524,096 37,169
00102.M2TS (1) 0:11:16.050 0:00:30.030 139,499,520 37,163
00319.M2TS (2) 0:11:16.050 0:00:30.030 139,505,664 37,164
00005.M2TS 0:11:46.080 0:14:22.987 4,076,199,936 37,787
00006.M2TS 0:26:09.067 0:00:27.277 126,222,336 37,019
00046.M2TS (1) 0:26:09.067 0:00:27.277 126,160,896 37,001
00320.M2TS (2) 0:26:09.067 0:00:27.277 126,259,200 37,030
00309.M2TS 0:26:36.344 0:00:28.028 129,921,024 37,083
00306.M2TS 0:27:04.372 0:00:56.306 261,163,008 37,106
00307.M2TS (1) 0:27:04.372 0:00:56.306 261,163,008 37,106
00321.M2TS (2) 0:27:04.372 0:00:56.306 261,322,752 37,129
00009.M2TS 0:28:00.678 0:01:25.376 395,784,192 37,086
00010.M2TS 0:29:26.055 0:00:43.668 202,156,032 37,035
00054.M2TS (1) 0:29:26.055 0:00:43.668 202,100,736 37,024
00055.M2TS (2) 0:29:26.055 0:00:43.668 202,192,896 37,041
00011.M2TS 0:30:09.724 0:08:27.340 2,389,186,560 37,674
00012.M2TS 0:38:37.064 0:00:20.729 95,852,544 36,993
00058.M2TS (1) 0:38:37.064 0:00:20.729 95,877,120 37,002
00059.M2TS (2) 0:38:37.064 0:00:20.729 95,784,960 36,966
00013.M2TS 0:38:57.793 0:03:02.140 844,523,520 37,093
00014.M2TS 0:41:59.933 0:00:56.181 260,536,320 37,099
00062.M2TS (1) 0:41:59.933 0:00:56.181 260,560,896 37,103
00063.M2TS (2) 0:41:59.933 0:00:56.181 260,640,768 37,114
00015.M2TS 0:42:56.115 0:09:09.882 2,628,986,880 38,248
00016.M2TS 0:52:05.997 0:00:23.815 109,897,728 36,916
00066.M2TS (1) 0:52:05.997 0:00:23.815 109,928,448 36,927
00067.M2TS (2) 0:52:05.997 0:00:23.815 109,928,448 36,927
00017.M2TS 0:52:29.813 0:03:46.184 1,043,951,616 36,924
00018.M2TS 0:56:15.997 0:00:20.520 95,102,976 37,076
00070.M2TS (1) 0:56:15.997 0:00:20.520 94,955,520 37,019
00071.M2TS (2) 0:56:15.997 0:00:20.520 94,955,520 37,019
00019.M2TS 0:56:36.517 0:05:38.379 1,576,636,416 37,275
00020.M2TS 1:02:14.897 0:00:21.146 98,248,704 37,169
00104.M2TS (1) 1:02:14.897 0:00:21.146 98,260,992 37,174
00105.M2TS (2) 1:02:14.897 0:00:21.146 98,242,560 37,167
00021.M2TS 1:02:36.043 0:03:47.060 1,105,901,568 38,964
00022.M2TS 1:06:23.103 0:00:18.852 87,263,232 37,031
00078.M2TS (1) 1:06:23.103 0:00:18.852 87,250,944 37,025
00079.M2TS (2) 1:06:23.103 0:00:18.852 87,281,664 37,038
00023.M2TS 1:06:41.956 0:06:06.074 2,046,320,640 44,719
00024.M2TS 1:12:48.030 0:00:20.603 100,663,296 39,085
00082.M2TS (1) 1:12:48.030 0:00:20.603 100,700,160 39,099
00083.M2TS (2) 1:12:48.030 0:00:20.603 100,663,296 39,085
00025.M2TS 1:13:08.634 0:01:58.576 547,350,528 36,928
00026.M2TS 1:15:07.210 0:00:21.187 97,579,008 36,843
00086.M2TS (1) 1:15:07.210 0:00:21.187 97,634,304 36,864
00087.M2TS (2) 1:15:07.210 0:00:21.187 97,640,448 36,867
00027.M2TS 1:15:28.398 0:04:37.151 1,269,331,968 36,639
00028.M2TS 1:20:05.550 0:00:20.186 93,007,872 36,859
00090.M2TS (1) 1:20:05.550 0:00:20.186 92,971,008 36,844
00091.M2TS (2) 1:20:05.550 0:00:20.186 93,001,728 36,856
00029.M2TS 1:20:25.737 0:02:30.066 695,531,520 37,079
00030.M2TS 1:22:55.803 0:00:19.686 90,765,312 36,885
00094.M2TS (1) 1:22:55.803 0:00:19.686 90,746,880 36,877
00095.M2TS (2) 1:22:55.803 0:00:19.686 90,734,592 36,872
00106.M2TS 1:23:15.490 0:02:00.578 556,707,840 36,936
00107.M2TS 1:25:16.068 0:02:05.458 579,508,224 36,953
00032.M2TS 1:27:21.527 0:00:27.944 102,273,024 29,279
00098.M2TS (1) 1:27:21.527 0:00:27.944 99,557,376 28,501
00099.M2TS (2) 1:27:21.527 0:00:27.944 101,136,384 28,953
00033.M2TS 1:27:49.472 0:06:41.192 1,218,994,176 24,307

After remuxing the movie with Clown_BD, the fps doesn't look right and BDInfo wasn't able to scan it.

eac3to v3.24
command line: "C:\Tools\Clown_BD_v0.76\eac3to\eac3to.exe" "F:" 1) -progressnumbers -LOG="C:\Clown_BD_WORK\Robinsons\LOGS\eac3to_PASS2_LOG.LOG"
------------------------------------------------------------------------------
M2TS, 1 video track, 8 audio tracks, 9 subtitle tracks, 1:34:29, 79.657p
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 5.1 channels, 16 bits, 48kHz
4: AC3, English, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
5: DTS, French, 5.1 channels, 24 bits, 1509kbps, 48kHz, dialnorm: -4dB
6: AC3, French, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
7: DTS, Dutch, 5.1 channels, 24 bits, 1509kbps, 48kHz, dialnorm: -4dB
8: AC3, Dutch, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
9: AC3, Dutch, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
10: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), French
14: Subtitle (PGS), Dutch
15: Subtitle (PGS), French
16: Subtitle (PGS), Dutch
17: Subtitle (PGS), English
18: Subtitle (PGS), French
19: Subtitle (PGS), Dutch

Any ideas?

Thunderbolt8

24th September 2010, 11:32

if the movie is in sync afterwards, dont care whats written there.

if not, try to add the .m2ts files (1+2+3...) manually instead of using the playlist.

hoju3508

24th September 2010, 14:55

Yes, it's in sync but TMT3 thinks it's a 13 hr movie for some reason. It's kind of odd.

if not, try to add the .m2ts files (1+2+3...) manually instead of using the playlist.

I might give this a try. Thanks.

Dogway

24th September 2010, 20:36

Hello, my source audio file is 25fps ac3 with 2 front channels and 2 rear channels. As for now Im using this code:
eac3to "input.ac3" "output.aac" -quality=0.30 -changeTo24.000 -down2 -phaseShift -mixlfe -resampleTo44100 -25.000
I want to enlength the audio to fit into the 24fps film, and downmixing all channels to Stereo 2. Does -phaseShift fix the pitch problem when changing audio lenght? and second, with this code I still have the 4 channels, how can I downmix them to 2?

tebasuna51

25th September 2010, 01:17

Hello, my source audio file is 25fps ac3 with 2 front channels and 2 rear channels.

I want to enlength the audio to fit into the 24fps film, and downmixing all channels to Stereo 2. Does -phaseShift fix the pitch problem when changing audio lenght?
Nope. BTW, don't use -phaseShift never with ac3 source.
And -mixlfe do nothing because don't exist LFE channel in the source
and second, with this code I still have the 4 channels, how can I downmix them to 2?
The -down2 parameter only work with 5.1 source.

You can use an .avs file like this:
a = NicAc3Source("D:\path\input.ac3")

# 4 Channels Quadro L,R,SL,SR, downmix to simple stereo
# flr = GetChannel(a, 1, 2)
# blr = GetChannel(a, 3, 4)
# MixAudio(flr, blr, 0.5, 0.5)

# 4 Channels Quadro L,R,SL,SR, downmix to DPL II
flr = GetChannel(a, 1, 2)
bl = GetChannel(a, 3)
br = GetChannel(a, 4)
sl = MixAudio(bl, br, 0.3714, 0.2144)
sr = MixAudio(bl, br, -0.2144, -0.3714)
blr = MergeChannels(sl, sr)
MixAudio(flr, blr, 0.4142, 1.0)

# [DSP: TimeStretch - 25 -> 24] pitch preserved
TimeStretch(tempo=96)

# [DSP: Resample SSRC - 44100 Hz]
SSRC(44100)

# [DSP: Normalize to 100%]
Normalize()
and encode with NeroAacEnc to .m4a with MeGUI, BeHappy or Wavi.

Dogway

25th September 2010, 03:45

Just WOW! it worked perfect. I didnt know I could load an avs into the audio input of MeGUI. Aside solving my issue, I just learned a little more, I hope this enlighting post awakes an interest on me on the audio part of the digital encoding.gracias:D

BTW what bitrate do you recommend me for medium quality towards good in m4a/AAC? something like 128,160 of mp3.

Also, in your script you are downmixing, normalizing,etc but in Megui the audio encoder also does its stuff in the Config options. Do I need to uncheck, keep originals everything from the options?

tebasuna51

25th September 2010, 10:26

...
BTW what bitrate do you recommend me for medium quality towards good in m4a/AAC? something like 128,160 of mp3.
To transcode from lossy sources (AC3,DTS,...) I always recommend at least -quality=0.35 (AAC-LC), with 0.30 you obtain AAC-HE, and is not always compatible with some players.
Also, in your script you are downmixing, normalizing,etc but in Megui the audio encoder also does its stuff in the Config options. Do I need to uncheck, keep originals everything from the options?
A little offtopic but ...
Yeep, you need uncheck all, or don't put Resample/Normalize in avs and use the Config options.
(These two functions are equivalent to eac3to, resample with SSRC lib).

The predefined downmix with MeGUI/BeHappy work only with 5.1 sources, like with eac3to. Then you need include the special downmix avs method for sources with 3,4 or 5 channels.

The change audio duration (for different fps video) in MeGUI only work for 23.976 <-> 25 using SSRC method without preserving pitch (like eac3to, but eac3to have more options).
To preserve pitch you need use the AviSynth function TimeStretch(tempo) included in BeHappy.

deathlord

26th September 2010, 21:52

I have just found something weird on "Fringe Season 2" BD. On disc three, the playlist for episode 14 "Jacksonville" is not listed by eac3to. It is relatively obvious that this must be 00072.mpls. Running eac3to on this file shows the right length (about 40 min), but instead of showing 00027.m2ts it shows 00027.m2ts+00027.m2ts and consequently demuxes files of double length.
With this length, eac3to should actually list this playlist aswell, but maybe there is an error in the playlist?
In any event, I just thought I would report this, perhaps somebody can make sense of it. Of course the problem can be solved easily by avoiding playlists for this episode and using the .m2ts file directly.

cavediver

26th September 2010, 23:18

setarip_old

27th September 2010, 00:05

@deathlord, @cavediver

Hi!

Does the "Backup>>Playlist" sub-folder contain .MPLS files IDENTICAL to those of the primary "Playlist" sub-folder?

Mark_A_W

27th September 2010, 04:07

Is there anything that can be done to reduce audio sync issues with eac3to? It usually happens when I get a message like this...

v01 The MKV file was created without making use of the gap/overlap information.
v01 Please check whether audio is in sync. If it is in sync everything is fine.
v01 Otherwise ask eac3to to repeat the muxing. It will then automatically make
v01 use of the detailed gap/overlap information.

If I don't repeat the muxing then sometimes the audio can be really badly out of sync. It is not predictable how much. If I repeat the muxing as it says, it successfully makes the new MKV file without this error message. But then the audio goes out of sync by a small but very noticeable amount.

So repeating the muxing does do something but maybe eac3to needs extra help by having the file cleaned up by another program that can take care of the gaps etc before passing it on to eac3to? Any other ideas how I can stop the audio going out of sync?

Anyone having success (or failure..) with seamless branching titles?

Somewhere along the way they stop working for me - major audio sync issues, changing from scene to scene.

It's not clear if the automatic second pass is supposed to correct this, or if we need to make a manual third pass (and possibly manually compensate for an audio delay).

Umm..a little guidance? Please.

Here is an edited log (sorry for the long post..)

eac3to v3.24

[v02] [0:07:08] The source file seems to be damaged (sync byte missing). <WARNING>
[s31] [0:07:05] The source file seems to be damaged (sync byte missing). <WARNING>
[s13] [0:06:34] The source file seems to be damaged (sync byte missing). <WARNING>
[a04] [0:07:07] The source file seems to be damaged (sync byte missing). <WARNING>
[s31] Creating file "F:\MKV temp files\Moofie - 31 English Subtitle.sup"...
[s31] [0:53:20] The source file seems to be damaged (discontinuity). <WARNING>
[a04] [0:53:21] The source file seems to be damaged (discontinuity). <WARNING>
[v02] [0:53:20] The source file seems to be damaged (discontinuity). <WARNING>
[s13] [0:52:47] The source file seems to be damaged (discontinuity). <WARNING>
[v02] [0:57:31] The source file seems to be damaged (discontinuity). <WARNING>
[a04] [0:57:31] The source file seems to be damaged (discontinuity). <WARNING>
[s31] [0:53:21] The source file seems to be damaged (discontinuity). <WARNING>
[s13] [0:56:55] The source file seems to be damaged (discontinuity). <WARNING>
[s31] [1:03:24] The source file seems to be damaged (discontinuity). <WARNING>
[a04] [1:03:52] The source file seems to be damaged (discontinuity). <WARNING>
[v02] [1:03:52] The source file seems to be damaged (discontinuity). <WARNING>
[s13] [1:02:51] The source file seems to be damaged (discontinuity). <WARNING>
[v02] [1:23:10] The source file seems to be damaged (discontinuity). <WARNING>
[a04] [1:23:10] The source file seems to be damaged (discontinuity). <WARNING>
[s13] [1:22:27] The source file seems to be damaged (discontinuity). <WARNING>
[s31] [1:03:52] The source file seems to be damaged (discontinuity). <WARNING>
[s31] [1:25:53] The source file seems to be damaged (discontinuity). <WARNING>
[v02] [1:25:55] The source file seems to be damaged (discontinuity). <WARNING>
[a04] [1:25:55] The source file seems to be damaged (discontinuity). <WARNING>
[s13] [1:25:19] The source file seems to be damaged (discontinuity). <WARNING>
[v02] [1:30:06] The source file seems to be damaged (discontinuity). <WARNING>
[a04] [1:30:06] The source file seems to be damaged (discontinuity). <WARNING>
[s31] [1:25:56] The source file seems to be damaged (discontinuity). <WARNING>
[s13] [1:28:43] The source file seems to be damaged (discontinuity). <WARNING>
[s31] [1:33:10] The source file seems to be damaged (discontinuity). <WARNING>
[v02] [1:33:24] The source file seems to be damaged (discontinuity). <WARNING>
[a04] [1:33:24] The source file seems to be damaged (discontinuity). <WARNING>
[s13] [1:32:29] The source file seems to be damaged (discontinuity). <WARNING>
[v02] [1:35:41] The source file seems to be damaged (discontinuity). <WARNING>
[a04] [1:35:41] The source file seems to be damaged (discontinuity). <WARNING>
[s13] [1:34:26] The source file seems to be damaged (discontinuity). <WARNING>
[s31] [1:33:25] The source file seems to be damaged (discontinuity). <WARNING>
[v02] Video overlaps for 20 frames at playtime 0:53:21. <WARNING>
[v02] Video overlaps for 7 frames at playtime 0:57:30. <WARNING>
[v02] Video overlaps for 14 frames at playtime 0:57:31. <WARNING>
[v02] Video overlaps for 21 frames at playtime 0:57:32. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:03:52. <WARNING>
[v02] Video overlaps for 14 frames at playtime 1:03:52. <WARNING>
[v02] Video overlaps for 43 frames at playtime 1:03:52. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:23:10. <WARNING>
[v02] Video overlaps for 14 frames at playtime 1:23:11. <WARNING>
[v02] Video overlaps for 175 frames at playtime 1:23:11. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:25:53. <WARNING>
[v02] Video overlaps for 14 frames at playtime 1:25:57. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:25:57. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:30:07. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:30:08. <WARNING>
[v02] Video overlaps for 28 frames at playtime 1:30:08. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:33:20. <WARNING>
[v02] Video overlaps for 15 frames at playtime 1:33:27. <WARNING>
[v02] Video overlaps for 12 frames at playtime 1:33:27. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:35:43. <WARNING>
[v02] Video overlaps for 14 frames at playtime 1:35:44. <WARNING>
[v02] Video overlaps for 2 frames at playtime 1:35:44. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:37:46. <WARNING>
[v02] The MKV file was created without making use of the gap/overlap information. <WARNING>
[v02] Please check whether audio is in sync. If it is in sync everything is fine. <WARNING>
[v02] Otherwise ask eac3to to repeat the muxing. It will then automatically make <WARNING>
[v02] use of the detailed gap/overlap information. <WARNING>
[a04] The original audio track has a constant bit depth of 24 bits.
[a04] Audio overlaps for 12ms at playtime 0:17:27. <WARNING>
[a04] Audio overlaps for 6ms at playtime 0:49:19. <WARNING>
[a04] Audio overlaps for 247ms at playtime 0:53:20. <WARNING>
[a04] Audio overlaps for 258ms at playtime 0:57:31. <WARNING>
[a04] Audio overlaps for 382ms at playtime 1:03:51. <WARNING>
[a04] Audio overlaps for 1161ms at playtime 1:23:09. <WARNING>
[a04] Audio overlaps for 173ms at playtime 1:25:55. <WARNING>
[a04] Audio overlaps for 261ms at playtime 1:30:06. <WARNING>
[a04] Audio overlaps for 204ms at playtime 1:33:24. <WARNING>
[a04] Audio overlaps for 143ms at playtime 1:35:41. <WARNING>
[a04] Starting 2nd pass...
[a04] Decoding FLAC...
[a04] Realizing RAW/PCM gaps...
[a04] Encoding FLAC with libFlac...
[a04] Creating file "F:\MKV temp files\Moofie - 4 DTS Master.flac"...
[a04] The processed audio track has a constant bit depth of 24 bits.
Added fps value (24 /1.001) to MKV header.

hoju3508

27th September 2010, 05:02

eac3to v3.24

[v02] [0:07:08] The source file seems to be damaged (sync byte missing). <WARNING>

I've gotten this before due to a bad rip (bad source).

Mark_A_W

27th September 2010, 05:14

I've gotten this before due to a bad rip (bad source).

I don't think that's it.

Every seamless-branching title has this issue.

They all play fine in TMT, and if I process with MakeMKV, they work fine.

madshi

27th September 2010, 07:37

Just a quick answer: "The source file seems to be damaged" is *NOT* supposed to appear for seamless branching titles. I've never in my whole life seen it on my PC with any seamless branching title. Try ripping to harddisk with AnyDVD HD first and then run eac3to on the harddisk folder.

Jeff Flowerday

27th September 2010, 14:50

Bad Source: I had the same problem recently with my Cars Blu-Ray. I waited for a new AnyDVD HD, just installed the release from a few days ago, and the problem went away.

deathlord

30th September 2010, 20:46

@deathlord, @cavediver

Hi!

Does the "Backup>>Playlist" sub-folder contain .MPLS files IDENTICAL to those of the primary "Playlist" sub-folder?

They are not identical,
00072.mpls in the Playlist folder has md5
46612f7c8c248b74dde7c57ccad25a67

00072.mpls in the Backup\Playlist folder has md5
e9d5b291218c5409228547b9ebfcb0d2

However, the playlists are identical as far as eac3to is concerned:

eac3to Fringe_Season_2\FRINGE_SEASON_2_DISC_3\BDMV\PLAYLIST\00072.mpls
1) 00072.mpls, 00027.m2ts+00027.m2ts, 0:43:58
- Chapters, 9 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48kHz
- AC3, Portuguese, stereo, 48kHz

eac3to Fringe_Season_2\FRINGE_SEASON_2_DISC_3\BDMV\BACKUP\PLAYLIST\00072.
mpls
1) 00072.mpls, 00027.m2ts+00027.m2ts, 0:43:58
- Chapters, 9 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48kHz
- AC3, Portuguese, stereo, 48kHz

BDInfo shows the same. Also, 00075.mpls shows the same double file for 00023.m2ts. I think it's an error in the mpls files.

For my 00075.mpls it does not show a double file, at least not on disc 3.

sfcav

1st October 2010, 21:38

Problem with ac3 conversion

Hi All:

I have a 25 fps PAL ac3 file I'm trying to slowdown to 23.976 fps so I can burn an AVCHD disc and play in my Bluray player. I do this regularly and have never had a problem. This file, although it plays fine in Media Player Classic, eac3to conversion hasn't worked.

When I looked at it more closely, it's characteristics differ depending on what I use to examine it:

MediaInfo for the mkv file (tsMuxeR does NOT recognize the audio track)
ID : 2
Format : AC-3
Format/Info : Audio Coding 3
Codec ID : A_AC3
Duration : 1h 27mn
Bit rate mode : Constant
Bit rate : 384 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Surround: L R, LFE
Sampling rate : 48.0 KHz
Stream size : 242 MiB (11%)

MediaInfo for the extracted audio track (using mkvextract)
Format : AC-3
Format/Info : Audio Coding 3
Duration : 1h 27mn
Bit rate mode : Constant
Bit rate : 192 Kbps
Channel(s) : 2 channels
Channel positions : L R
Sampling rate : 48.0 KHz
Stream size : 121 MiB (100%)

And eac3to sees the same extracted audio track as:
AC3, 5.1 channels, 0:44:00, 384 kbps, 48kHz, dialnorm: -22dB

Any ideas on what's happening here (or how to fix it)??

sfcav

tebasuna51

2nd October 2010, 00:26

Problem with ac3 conversion
...
Channel(s) : 6 channels
...
Channel(s) : 2 channels

Any ideas on what's happening here (or how to fix it)??
Maybe you have an AC3 with 2 kind of frames (a TV capture?, movie and commercials).

Try run:

SplitAc3 audio.ac3

and put the log file created.

[EDIT] SplitAc3.exe and SplitAc3m.exe included in AC3Util.7z here (https://forum.doom9.org/showthread.php?p=1485920#post1485920)

sfcav

2nd October 2010, 01:16

Maybe you have an AC3 with 2 kind of frames (a TV capture?, movie and commercials).

Try run:

SplitAc3 audio.ac3

and put the log file created.

Thanks tebasuna51!

Here is the output:

FileSize : 126680064 bytes
---------- First valid Header
Time eq. : 2639168 ms.
SamplCod : 0 (0:48, 1:44.1, 2:32 KHz.)
BitRate : 384 Kb/s
ChanMode : 7 (1:1/0, 2:2/0, 3:3/0, 4:2/1, 5:3/1, 6:2/2, 7:3/2)
FrameSize: 1536 bytes
---------- Process ( 25.000000 fps is used for Trim)
Time: 0 ms. Written: 1 frames 5.1 ( 32 ms.) Trim(0, 0)
Time: 32 ms. Written: 164946 frames 2.0 ( 5278272 ms.)
---------- End of File
Total time: 5278304 ms. at EOF
T. written: 1 frames 5.1.
T. written: 164946 frames 2.0.
___________________________________________--

It yielded two ac3 files, one of which fits what I expected (1hr 27 min) and looks the same with both MediaInfo and eac3to!

Thanks : )

hubblec4

2nd October 2010, 13:44

neroaacencoder work not more correctly. have anyone an idee whats the problem is?

eac3to v3.24
command line: eac3to.exe "00003 - 3 - DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz.dtsma" "eng.mp4" -288
------------------------------------------------------------------------------
DTS Master Audio, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
Decoding with ArcSoft DTS Decoder...
Encoding AAC <288kbps> with NeroAacEnc...
The original audio track has a constant bit depth of 24 bits.
The Nero AAC encoder seems to be stuck... <ERROR>
[NeroAacEnc] Processed 0 seconds...
[NeroAacEnc] Processed 1 seconds...
[NeroAacEnc] Processed 2 seconds...
[NeroAacEnc] Processed 3 seconds...
[NeroAacEnc] Processed 4 seconds...
[NeroAacEnc] Processed 5 seconds...
[NeroAacEnc] Processed 6 seconds...
[NeroAacEnc] Processed 7 seconds...
[NeroAacEnc] Processed 8 seconds...
[NeroAacEnc] Processed 9 seconds...
[NeroAacEnc] Processed 10 seconds...
[NeroAacEnc] Processed 11 seconds...
.........
[NeroAacEnc] Processed 2848 seconds...
[NeroAacEnc] Processed 2849 seco
Aborted at file position 1506659076. <ERROR>

after abort the file is deleted by eac3to. when i cancel the neroaac-process i keep the mp4 and its playable. sounds good.

hubblec4

2nd October 2010, 14:53

i have testet the same file with eac3to 3.22 and it works fine.

the same problem happened with ac3 files but with 3.22 all fine.

i have winxp32bit sp3. arcsoftdecoder 1.1.0.0

SoleBastard

2nd October 2010, 20:53

Hi Madshi,
Thanks again for your great program!

As eac3to doesn't support merging FLAC files (yet) I tried doing it using Foobar2000 and flac.exe 1.2.1. However, when the merged file reaches 2.00GB the encoding stops with an error. I asked about this in the Foobar2000 forum (see this thread (http://www.hydrogenaudio.org/forums/index.php?showtopic=84014)) and it seems to be a specific limitation of the Windows FLAC build. However you manage to create FLAC files with eac3to in Windows that are >2GB just fine. How is that possible?

TinTime

2nd October 2010, 21:47

You could always convert to a .w64 file with foobar and then convert that to FLAC with eac3to. Not completely ideal but it's a workaround.

SoleBastard

2nd October 2010, 22:06

Yep, I figured that out and doing the conversion at the moment ;)

So yeah, this work-around is okay for me because I don't need to do these kind of merges that often. I was just curious how Madshi was able to implement encoding >2GB flac files :)

TinTime

2nd October 2010, 22:57

Yep, I figured that out and doing the conversion at the moment ;)

Sorry, I see now that I replied to something you didn't ask and ignored what you did ask :)

Anyway, from your link it looks like an i/o limitation in flac.exe, so that's why eac3to is ok - it handles i/o itself and uses libflac to encode.

deathlord

3rd October 2010, 20:40

Often there's a "play all" playlist. It usually lists the episodes in the right order.

This is a very helpful hint, thanks a lot!
I have found it to work on Terminator: The Sarah Connor Chronicles and Dexter.

Groucho2004

4th October 2010, 17:37

SoleBastard

4th October 2010, 19:24

Anyway, from your link it looks like an i/o limitation in flac.exe, so that's why eac3to is ok - it handles i/o itself and uses libflac to encode.

Ah right, that makes sense. :thanks:

A kind HA.org user modified the flac.exe to work with large input files and I find it to work perfectly with my FLACs! See the thread I linked to earlier.

deathlord

4th October 2010, 19:59

Would it be possible to add an option to normalize to a certain level?
As far as I know, eac3to only normalizes to 0 dB. I always like to leave a little headroom for further conversions so that would be nice.

Possibly something like "-normalize -2dB".

You can certainly do "-2dB" (see first post). I don't know what happens if eac3to has to normalize, too. You can try it out or you can simply do two passes.

b66pak

4th October 2010, 20:26

"-normalize -2dB" this don't work (the -2dB part)

you must do:
eac3to audio.xxx normalized_audio.wav -normalize
eac3to normalized_audio.wav final-2dB_audio.wav -2dB
_

shh

4th October 2010, 21:22

Is there any way to prevent the massive file fragmentation in conjunction with eac3to?

I understand, that this is an issue of the (sometimes) unknown size of handled streams, but the count of segments of some of eac3to's files is just insane. :eek:

Suggestions:
- reserving file space for separate streams what can safely cover the size of a (sub)stream, later crop filesize to the real size of the stream.
- command line option: "just reserve +100MB per stream"
- ...?

Groucho2004

4th October 2010, 21:28

this don't work (the -2dB part)

you must do:
eac3to audio.xxx normalized_audio.wav -normalize
eac3to normalized_audio.wav final-2dB_audio.wav -2dB
_

Sigh.

Yes, that's what I've been doing so far. However, I was asking if this feature could added so I don't have to do 2 passes (actually 3 passes, because the "normalize" function already needs 2 passes).

LeXXuz

5th October 2010, 16:19

I'm getting this error message when encoding an audio stream:
eac3to v3.24
command line: eac3to eng.ac3 eng_new.ac3 -384 -normalize
------------------------------------------------------------------------------
AC3, 5.0 channels, 1:33:59, 384kbps, 48kHz
The Nero decoder doesn't seem to work, will use libav instead.
Decoding with libav/ffmpeg...
Remapping channels...
Writing WAV...
Creating file "eng_25p.ac3.pass1.wav"...
Caution: The WAV file is bigger than 4GB. <WARNING>
Some WAV readers might not be able to handle this file correctly. <WARNING>
Starting 2nd pass...
Reading WAV...
The AC3 encoder received a non-supported data format (float, 5, 64, -). <ERROR>
Aborted at file position 262144. <ERROR>

Creating the wav file works, but as soon as I start the ac3 encoder it aborts with a similar error message:
eac3to v3.24
command line: eac3to eng_new.wav eng_new.ac3 -384
------------------------------------------------------------------------------
WAV, 5.0 channels, 1:33:59, 24 bits, 5760kbps, 48kHz
Reading WAV...
The AC3 encoder received a non-supported data format (pcm, 5, 24, -). <ERROR>
Aborted at file position 262144. <ERROR>

Source is from an HDTV recording.

Snowknight26

5th October 2010, 20:59

Search the thread for the 2nd to last line's error. It's been answered many times.

Think Aften.

icon

6th October 2010, 16:36

rack04

6th October 2010, 16:43

Does anyone have a .bat/.reg file that would allow me to right click on a .mkv file and have it call a .bat file to automatically convert dts to ac3? I don't want to use a gui.

If not can anyone help me with creating a bat file that I run to convert dts to ac3?

Thanks.

Why not just use the cmd? At bat file wouldn't work for all instances since the track number of the dts is a variable. Well you could use mediainfo cmd to determine the track number for the dts and use that in the bat file but that would be more work than using eac3to cmd.

eac3to input.mkv track: output.ac3

If you must use a bat file here is a quick and dirty. It only works with mkv, single audio track, and doesn't allow for additional options. Hence the reason why using the cmd is a better approach.

@SET MediaInfo_PATH=C:\Program Files\MediaInfo\CLI\MediaInfo.exe
@SET eac3to_PATH=C:\Program Files\eac3to\eac3to.exe
@SET Working_Directory=C:\Personal\Videos

@Echo off
Echo.
Echo.
Echo\*************** Input ****************
Echo.
Echo.
set /p INPUT_VIDEO="Path to Source: "
Echo.
Echo.
Echo\******************************************
Echo.

Echo General;%%FileName%%>"%Working_Directory%\temporary-generalfilename.txt"
"%MediaInfo_PATH%" --Inform=file://"%Working_Directory%\temporary-generalfilename.txt" "%INPUT_VIDEO%">"%Working_Directory%\temporary-analyzegeneralfilename.txt"
Echo General;%%FolderName%%>"%Working_Directory%\temporary-generalfoldername.txt"
"%MediaInfo_PATH%" --Inform=file://"%Working_Directory%\temporary-generalfoldername.txt" "%INPUT_VIDEO%">"%Working_Directory%\temporary-analyzegeneralfoldername.txt"
Echo General;%%FileExtension%%>"%Working_Directory%\temporary-generalfileextension.txt"
"%MediaInfo_PATH%" --Inform=file://"%Working_Directory%\temporary-generalfileextension.txt" "%INPUT_VIDEO%">"%Working_Directory%\temporary-analyzegeneralfileextension.txt"
type "%Working_Directory%\temporary-analyzegeneralfilename.txt">"%Working_Directory%\temporary-analyzegeneralfilename.tmp"
type "%Working_Directory%\temporary-analyzegeneralfoldername.txt">"%Working_Directory%\temporary-analyzegeneralfoldername.tmp"
type "%Working_Directory%\temporary-analyzegeneralfileextension.txt">"%Working_Directory%\temporary-analyzegeneralfileextension.tmp"
set /p SOURCE_FILENAME=<"%Working_Directory%\temporary-analyzegeneralfilename.tmp"
set /p SOURCE_FOLDER=<"%Working_Directory%\temporary-analyzegeneralfoldername.tmp"
set /p SOURCE_EXTENSION=<"%Working_Directory%\temporary-analyzegeneralfileextension.tmp"

DEL "%Working_Directory%\temporary-generalfilename.txt"
DEL "%Working_Directory%\temporary-analyzegeneralfilename.txt"
DEL "%Working_Directory%\temporary-analyzegeneralfilename.tmp"
DEL "%Working_Directory%\temporary-generalfoldername.txt"
DEL "%Working_Directory%\temporary-analyzegeneralfoldername.txt"
DEL "%Working_Directory%\temporary-analyzegeneralfoldername.tmp"
DEL "%Working_Directory%\temporary-generalfileextension.txt"
DEL "%Working_Directory%\temporary-analyzegeneralfileextension.txt"
DEL "%Working_Directory%\temporary-analyzegeneralfileextension.tmp"

if "%SOURCE_EXTENSION%" EQU "mkv" goto process

Echo.
Echo.Source file is not MKV !
Echo.
Echo.
goto end

:process
Echo Audio;%%ID%%>"%SOURCE_FOLDER%\%SOURCE_FILENAME%-audid.txt"
"%MediaInfo_PATH%" --Inform=file://"%SOURCE_FOLDER%\%SOURCE_FILENAME%-audid.txt" "%INPUT_VIDEO%">"%SOURCE_FOLDER%\%SOURCE_FILENAME%-analyzeaudid.txt"
type "%SOURCE_FOLDER%\%SOURCE_FILENAME%-analyzeaudid.txt">"%SOURCE_FOLDER%\%SOURCE_FILENAME%-analyzeaudid.tmp"
set /p ID=<"%SOURCE_FOLDER%\%SOURCE_FILENAME%-analyzeaudid.tmp"

DEL "%SOURCE_FOLDER%\%SOURCE_FILENAME%-audid.txt"
DEL "%SOURCE_FOLDER%\%SOURCE_FILENAME%-analyzeaudid.txt"
DEL "%SOURCE_FOLDER%\%SOURCE_FILENAME%-analyzeaudid.tmp"

@Echo on
"%eac3to_PATH%" "%INPUT_VIDEO%" %ID%: "%SOURCE_FOLDER%\%SOURCE_FILENAME%.ac3" >"%SOURCE_FOLDER%\%SOURCE_FILENAME%-eac3to.txt"
@Echo off

DEL "%SOURCE_FOLDER%\%SOURCE_FILENAME%-eac3to.txt"
DEL "%SOURCE_FOLDER%\%SOURCE_FILENAME% - Log.txt"

Echo.
Echo.
Echo.Finished !
Echo.
Echo.
:end
Pause

icon

6th October 2010, 18:01

Thanks for the reply. I should have mentioned that my mkv files only have (1) dts track and I don't use any options. I use cmd: eac3to input.mkv input.ac3
I'll take a look at the script you posted to see if I can make sense of it. Thanks again.

Ideally, I wanted to right click on a mkv file from the context menu, pass that file to a bat script to automagically create the ac3 output. I do a lot of ripping and converting. This cmd part is getting a little old. Just must be getting lazy.

Edit: I agree, that if I have to type the source input in everytime, then it doesn't make much since to use a bat file. I was hoping I could run it and it would locate all the 'mkv files I have and spit out ac3 files with the same input name and the mkv. Maybe I am asking for too much. I can't believe no one else has a need for this though.

If only I could use wildcards: eac3to *.mkv *.ac3
something like that

rack04

6th October 2010, 18:25

icon

6th October 2010, 18:53

I see. I use handbrake to do a lot of my encodings. It keeps the dts but does not convert to ac3. It seems that I need to run my encodes through mkmerge anyhow for proper playback on my video boxes. I did find an old script in this thread from Snowkight26, that I was able to modify for my needs. Thanks again for the help.

icon

7th October 2010, 00:23

Assuming there is only 1 video and 1 DTS track:

for /f "delims=;" %%a in ('dir *.mkv /b /s') do {
eac3to %%a 1: %%a.video.mkv 2: %%a.ac3
mkvmerge -o %%a.DTS.mkv --language 1:eng --default-track 1:yes -d 1 -A -S %%a.video.mkv -a 0 -D -S %%a.ac3 --track-order 0:1,1:0
del %%a.video.mkv
del %%a.ac3
}
pause

Something like that at least.

I modified this script to just convert dts from a mkv to ac3. I had to use quotes and change { } to ( ). Now I am trying to figure out how the mkvmerge portion of the script works. I seem to be having a problem with it. Can anyone help me out. It's a very powerful script at least for my needs.

tebasuna51

7th October 2010, 02:27

...
Now I am trying to figure out how the mkvmerge portion of the script works. I seem to be having a problem with it. Can anyone help me out. It's a very powerful script at least for my needs.
What is your problem with mkvmerge?

Try this:
for %%a in (*.mkv) do (
"C:\Program Files\eac3to\eac3to.exe" "%%a" 2: "%%a.ac3"
"C:\Program Files\MKVtoolnix\mkvmerge.exe" -o "%%a.AC3.mkv" -d 1 --compression 1:none -A "%%a" -a 0 --compression 0:none "%%a.ac3"
del %%a.ac3
)
pause

icon

7th October 2010, 03:02

Try this:
for %%a in (*.mkv) do (
"C:\Program Files\eac3to\eac3to.exe" "%%a" 2: "%%a.ac3"
"C:\Program Files\MKVtoolnix\mkvmerge.exe" -o "%%a.AC3.mkv" -d 1 --compression 1:none -A "%%a" -a 0 --compression 0:none "%%a.ac3"
del %%a.ac3
)
pause

Thanks. The code works, but I need help modifing it. I am new to mkvmerge cli, I ususally use the gui.

Your script removes the dts track. I need the following:
Keep the dts track, mark for english, set as default and make sure no compression is set.
Add the ac3 track, mark for english, no default and make sure no compression is set.
(anything else I am missing)

I think that is it. Thanks for your help.

tebasuna51

7th October 2010, 10:26

...
Your script removes the dts track. I need the following:
Keep the dts track, mark for english, set as default and make sure no compression is set.
Using the original mkv like source the defaults and language are preserved. If you want preserve also the the DTS track use:
-d 1 --compression 1:none -a 2 --compression 2:none "%%a"
(also subs and chapters are preserved if any)
Add the ac3 track, mark for english, no default and make sure no compression is set...
If you add a new track isn't marked as default, then you only need:
-a 0 --language 0:eng --compression 0:none "%%a.ac3"

icon

7th October 2010, 15:44

Using the original mkv like source the defaults and language are preserved. If you want preserve also the the DTS track use:
-d 1 --compression 1:none -a 2 --compression 2:none "%%a"
(also subs and chapters are preserved if any)

If you add a new track isn't marked as default, then you only need:
-a 0 --language 0:eng --compression 0:none "%%a.ac3"

Thanks again. I was able to figure out the rest of the things I needed based upon your examples and explanations.

dansrfe

8th October 2010, 04:14

I get this "Audio overlaps for 157ms at playtime 1:25:08. <WARNING>" and after checking I have found out that eac3to did not fix this delay. What do I do?

dansrfe

8th October 2010, 21:57

so I have found out that eac3to only shows a warning for the overlap/delay and does NOT fix it automatically. I ended up extracting both dts-hdma's from their m2ts's and delaying the second one manually then rejoining in eac3to and outputting wavs.

elmarikon

9th October 2010, 15:31

Hi to all!
Great tool! Stable, handy functional!

1. I just have some questions about the multichannel to mono wav conversion:
How are the channels mapped?
I guess, I found out, when the Input is 6 or 8 channels, but at the latest when it's about 10 channels, I am not sure anymore...

An idea would be to have a 'numbered output channel' option, so that channel 1 in the source would end up being xxx1.wav in the end?!

2. Would you consider adding 'multiple mono wavs' as input as well? So one could easily re/mux again to multi channel wav / ac3 / ...

:thanks: for all your work!

All the best from Berlin!

dansrfe

9th October 2010, 21:15

i dont know about numerical channel mapping however it does say L for left, R for right, LFE for LFE, SR for surround right, and SL for surround left on a 5.1 stream.

Richard1485

10th October 2010, 13:59

I am having difficulty downmixing a 5.1 full DTS track (i.e. not HD) to a two-channel PCM wav that I can import to Womble MPEG DVD Wizard. I am using yr_eac3to_more_gui and have been playing with the settings for ages. Eac3to generates files succesfully, but Womble cannot import them, and when I view the files in Mediainfo the details are not as I want them to be. This is what I want the final file to be like:

Bit rate : 1536 Kbps
Codec : PCM
Family : PCM
Info : Microsoft PCM
Bit rate : 1536 Kbps
Channel(s) :2 channels
Sampling rate : 48 KHz
Resolution : 16

Midzuki

10th October 2010, 15:25

^ @ Jeff B: try something like this:

eac3to input.dts output.wav -down2 -simple

Richard1485

10th October 2010, 23:53

try something like this:
eac3to input.dts output.wav -down2 -simple

Thanks! :) I really appreciate the help. I had to use...

eac3to input.dts output.wav -down16 -down2 -simple

...as it as defaulted to 24bit, but other than it worked perfectly. Womble accepts and plays the file without any problems.

I notice that this defaults to Arcsoft. Would there be any benefit to using Sonic or Nero?

krosswindz

12th October 2010, 18:15

Lyle_JP

12th October 2010, 18:27

I hope I'm not asking something that's been asked before, but I didn't see anything on the search (but after 500 pages, who knows). Can eac3to encode DTS at 1,234 kbps (44.1 khz sampling) and wrap it in a PCM stream? I am trying to create a DTS CD.

Shirtfull

12th October 2010, 18:56

Copied from the first post on page one.

For DTS encoding you need:
(1) SurCode DVD DTS - version 1.0.21 or newer

hoju3508

13th October 2010, 05:31

I think the op wants to create a DTS core track inside a PCM track like the way that DTS-HD MA has a DTS core track.

Midzuki

13th October 2010, 06:17

This is a n00b question. Eac3to uses flags of -1536/-768 for DTS encoding. DTS.com says legal bitrate is 1509.75 Kbps, are the various representations the same, i.e, does DTS 1536 Kbps = DTS 1510 Kbps = DTS 1509.75 Kbps?

1509.75 == exact value
1510 == "convenient" approximation
1536 == DTS data plus zero-padding (for SPDIFing as stereo PCM @ 48kHz)

Lyle_JP

13th October 2010, 06:48

I think the op wants to create a DTS core track inside a PCM track like the way that DTS-HD MA has a DTS core track.

Not really. I'm trying to create a DTS signal wrapped in a redbook compliant PCM stream exactly like they used to do on laserdiscs and on DTS CDs. All DTS receivers are capable of recognizing these streams, but the DTS stream has to be 1,234 kbps with a 44.1 khz sampling rate or it doesn't work. I'm assuming now that this is probably beyond eac3to's capabilities.

Midzuki

13th October 2010, 09:12

I'm trying to create a DTS signal wrapped in a redbook compliant PCM stream exactly like they used to do on laserdiscs and on DTS CDs.

Both Surcode and the DTSHDMAS Encoder can do that.
Why do you want or need eac3to? :confused:

tebasuna51

13th October 2010, 10:33

Not really. I'm trying to create a DTS signal wrapped in a redbook compliant PCM stream exactly like they used to do on laserdiscs and on DTS CDs. All DTS receivers are capable of recognizing these streams, but the DTS stream has to be 1,234 kbps with a 44.1 khz sampling rate or it doesn't work. I'm assuming now that this is probably beyond eac3to's capabilities.

Eac3to only can automate the job with SurCode DVD DTS, then only work with 48 KHz and bitrates 1536/768.

With eac3to you can create the 6 monowavs 44.1 KHz and after encode manually with DTS HD Encoder Suite or SurCode CD-DTS (or the new SurCode 5.1 DTS Encoder for DVD & CD)

tebasuna51

13th October 2010, 13:27

This is a n00b question. Eac3to uses flags of -1536/-768 for DTS encoding. DTS.com says legal bitrate is 1509.75 Kbps, are the various representations the same, i.e, does DTS 1536 Kbps = DTS 1510 Kbps = DTS 1509.75 Kbps?
Please read the new #8 DTS FAQ (http://forum.doom9.org/showthread.php?p=424069#post424069)

krosswindz

13th October 2010, 14:41

Please read the new #8 DTS FAQ (http://forum.doom9.org/showthread.php?p=424069#post424069)

Thanks for the detailed explanation.

_hawk_

14th October 2010, 09:00

I have a problem to grab the eac3to error output with C# - normal output works fine. Does any1 have an idea how to do this? Might it be possible to add a switch to eac3to to print errors on standard output?

Btw thanks for your great tool!

iSeries

14th October 2010, 10:12

Hi,

When downmixing 5.1 FLAC soundtracks to AC3 2.0, what is the optimal bitrate to go for? 192kbps? Or maybe higher?

tebasuna51

14th October 2010, 11:13

When downmixing 5.1 FLAC soundtracks to AC3 2.0, what is the optimal bitrate to go for? 192kbps? Or maybe higher?
Is your choice, at least 192 like DVD tracks 2.0, but you can use 224 or 256 also.

iSeries

14th October 2010, 11:16

Is it ok to use the -normalize switch when converting to AC3, or is that not recommended?

tebasuna51

14th October 2010, 12:00

Is it ok to use the -normalize switch when converting to AC3, or is that not recommended?
Is recommended when downmix.

iSeries

14th October 2010, 12:42

Thank you!

Is 256kbps the maximum for 2 channel AC3?

TinTime

14th October 2010, 13:57

Is 256kbps the maximum for 2 channel AC3?

No, it's 640 (!).

iSeries

14th October 2010, 14:29

I realised that 640 is the max for 5.1 AC3, but did not know that was also the max for 2.0 AC3.

TinTime

14th October 2010, 14:42

It's probably complete overkill. I think eac3to defaults to 448 for two channel AC3.

I've seen 640 2.0 AC3 occasionally on Blu-rays though.

iSeries

14th October 2010, 14:56

As 640kbps equals 106.66kbps per channel for 5.1 AC3 and is considered transparent enough for most people, then for 2.0 I guess 224kbps should offer similar quality?

TinTime

14th October 2010, 15:13

As 640kbps equals 106.66kbps per channel for 5.1 AC3 and is considered transparent enough for most people, then for 2.0 I guess 224kbps should offer similar quality?

Only if each channel is treated equally. I don't know how the bits are distributed so I'm not sure what the 2.0 equivalent would be. I suspect the front centre gets a good chunk of the allocation in a 5.1 stream, and the LFE gets very little.

Midzuki

14th October 2010, 16:49

OverSimplification:

640kbps / 5 = 128kbps per channel

iSeries

14th October 2010, 18:30

So do not include lfe when working out? So with your method 2.0 AC3 256kbps is equivalent to 5.1 AC3 640kbps in terms of quality?

Midzuki

14th October 2010, 18:53

^

So with your method 2.0 AC3 256kbps is equivalent to 5.1 AC3 640kbps in terms of quality?

YES ( or at least the theory says so :devil: :) ).
As TinTime said, the LFE channel consumes a very-very-low bitrate, and this is the reason why we say 5.1 and not simply 6-channel audio.

flyingernst

14th October 2010, 21:40

Hi guys...I tink i am not the first how asks, but i didn´t find the answer: is it possible to demux only the forced captures of the PGS Subtitles of "Avatar"?! forced and normal are in the same pgs stream in this movie

Thanks, greetings, Michael

TinTime

14th October 2010, 21:59

eac3to tells you about forced subs but no more. You can then use BDSup2Sub to extract the forced subs.

dansrfe

16th October 2010, 02:54

eac3to does not fix the delay when I extract wavs from 2 joined m2ts's. It says realizing raw/pcm gaps but that doesn't fix it. Any way to have it auto-fix the delay?

dansrfe

17th October 2010, 08:37

Can someone please help me?^

Thunderbolt8

17th October 2010, 15:09

in dts master audio suite you can tick the option to display all 7.1 channel layouts. doing so, theres another setup similar to that of "strange setup" 7.1 tracks which has the 2 additional speakers at the front and not at the back (Lw, Rw instead of Lsr and Rsr). how can we actually be sure that these layouts named as "strange setup" by eac3to are that one with the additional 2 channels at the back and not at the front?

Snowknight26

17th October 2010, 17:24

2: h264/AVC (left eye), 1080p24 /1.001 (16:9)
3: h264/AVC (right eye), 1080p24 /1.001 (16:9)
Is there supposed to be an extra space before '(left eye)?'

dansrfe

17th October 2010, 17:38

So I'm still having the same problem with audio overlaps and eac3to not fixing them automatically. Any switch or option I have to select to have it auto-fix the overlaps? Thanks.

liquidskin76

17th October 2010, 23:15

Hi,

Can i clarify how the audio delay is handle when demuxing Blu-ray audio.

Am i right in thinking that the audio delay on DTS-HD is included when demuxing, however on TrueHD it isn't. For TrueHD, the delay required is shown in the filename, and needs to be added when for example you mux using mkvmerge?

Many thanks

dansrfe

18th October 2010, 00:52

That's exactly the problem. The delay is not fixed for DTS-HDMA when demuxing or extracting wavs. I always see a list of "Audio overlaps" but eac3to never actually fixes them. Therefore the only way for me to possibly fix them is to extract each and every single dts-hdma file from each of the m2ts files with different delays then join them all together and extract the wavs. I hope there is some switch or something which fixes the delay or it might just be a bug.

ACrowley

18th October 2010, 10:54

Hi

Is there any standalone TrueHD+AC3 Joiner besides the eac3to thd+ac3 Function ? Imho it would be nice to have a small App/Gui to join THD and AC3 ?

Frogger13

19th October 2010, 00:26

Hi,

just wanted to know if there is a solution to this issue yet cause it's prety annoying :(

Edit: This error happens more often when more that one encoding is in progess (didn't happen in previous versions), CPU load *may* enhance the issue.

neroaacencoder work not more correctly. have anyone an idee whats the problem is?

eac3to v3.24
command line: eac3to.exe "00003 - 3 - DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz.dtsma" "eng.mp4" -288
------------------------------------------------------------------------------
DTS Master Audio, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
Decoding with ArcSoft DTS Decoder...
Encoding AAC <288kbps> with NeroAacEnc...
The original audio track has a constant bit depth of 24 bits.
The Nero AAC encoder seems to be stuck... <ERROR>
[NeroAacEnc] Processed 0 seconds...
[NeroAacEnc] Processed 1 seconds...
[NeroAacEnc] Processed 2 seconds...
[NeroAacEnc] Processed 3 seconds...
[NeroAacEnc] Processed 4 seconds...
[NeroAacEnc] Processed 5 seconds...
[NeroAacEnc] Processed 6 seconds...
[NeroAacEnc] Processed 7 seconds...
[NeroAacEnc] Processed 8 seconds...
[NeroAacEnc] Processed 9 seconds...
[NeroAacEnc] Processed 10 seconds...
[NeroAacEnc] Processed 11 seconds...
.........
[NeroAacEnc] Processed 2848 seconds...
[NeroAacEnc] Processed 2849 seco
Aborted at file position 1506659076. <ERROR>

after abort the file is deleted by eac3to. when i cancel the neroaac-process i keep the mp4 and its playable. sounds good.

dansrfe

19th October 2010, 05:14

Calling madshi urgently...

jd213

19th October 2010, 16:21

Sorry if this has been asked already...

Is it possible or will it ever be possible for eac3to to make BD compatible Dolby Digital Plus and TrueHD streams from HD DVD sources?

I converted the Apollo 13 HD DVD to Blu-ray the other day, but I noticed some static coming out of the surround speakers so I assume the conversion to LPCM using eac3to and Pcm2Tsmu13 wasn't quite perfect.

Even if it's just a placebo effect, I'd rather not have to convert the DD+ and THD streams before re-authoring. Is a parallel ac3 stream all that's needed? If so, could a new version of eac3to allow for encoding this stream?

I'd be willing to make a donation if it will help. Actually, if donations are being accepted I'd like to make one regardless, since I've already gotten plenty of use out of eac3to. I don't see any mention of donations on the first post or on madshi.net, though.

TinTime

19th October 2010, 17:27

TrueHD is fine. Just use the following to generate the AC3:

eac3to.exe input.thd output.thd+ac3

I'm not sure why you've got a problem with LPCM though.

tebasuna51

19th October 2010, 22:05

...
I converted the Apollo 13 HD DVD to Blu-ray the other day, but I noticed some static coming out of the surround speakers so I assume the conversion to LPCM using eac3to and Pcm2Tsmu13 wasn't quite perfect.
...
Pcm2Tsmu work perfect or don't work at all (if used with incorrect parameters). BTW use .w64 instead LPCM to be load after in tsMuxeR, to avoid the use of incorrect parameters.

Or use the TinTime option.

rica

20th October 2010, 01:30

madshi,
PDVD10 audio decoder decodes DTSHD streams perfectly.
Maybe you can give it a go?

jd213

20th October 2010, 11:17

TrueHD is fine. Just use the following to generate the AC3:

eac3to.exe input.thd output.thd+ac3

Oh, I see. So it's just DD+ that can't be converted to a BD-compatible DD+ stream. It would need to be eac3+ac3 I assume, and eac3to doesn't seem to support that format apparently.

I'm not sure why you've got a problem with LPCM though.

I don't have a problem with LPCM itself, but it seems the libavcodec in eac3to isn't decoding DD+ 100% accurately since I got some occasional static out of the surround speakers when playing the decoded LCPM but not when playing the HD DVD itself. Also, I'd prefer to just keep the original DD+ stream, since decoding to LPCM makes Apollo 13 too big for a BD25.

TinTime

20th October 2010, 11:36

There's a difference between eac3 on HD DVDs and eac3 on BD so you're going to have to convert it to something - you can't just interleave ac3 to make it compatible like TrueHD.

If 24bit LPCM is too big how about 16bit? Alternatively just convert it to AC3.

jd213

21st October 2010, 02:59

Thanks for the information. 16-bit LPCM would normally be fine, but I suspect that it will still have the static problem.

If there's no way to convert HD DVD eac3 to BD eac3, then I guess I'll just use my Popcorn Hour (although I'd rather have a burned disc). Just found out that HD DVD eac3 will work on it when remuxed into a BD m2ts file. I previously didn't think it would since I once tried playing a evo file and got video but no sound.

dansrfe

21st October 2010, 06:56

Thunderbolt8

22nd October 2010, 22:30

for the time being, to get around that problem just encode to .flac instead of .wav and then convert to .wav afterwards. other than that, you need to wait for madshi to fix that problem.

dansrfe

23rd October 2010, 00:07

k thx^ :)

dansrfe

23rd October 2010, 00:55

To Thunderbolt8:

Going back to your earlier question about 7.1 config in DTS HD Encoder. Did you find out how to figure out which channels from eac3to works with which config for 7.1 in DTS HD Encoder?

xxx666yyy777

23rd October 2010, 01:56

dansrfe

23rd October 2010, 04:08

Excellent^ Thanks!

Sharc

23rd October 2010, 09:17

Can someone PLEASE address the issue of eac3to NOT solving delay/overlap issues when extracting wavs or the audio files from a set of stringed-together/joined m2ts files from an mpls? It gives all the warnings which is great but it doesn't actually "do" anything about it.
IIRC joining the multiple .m2ts from the .mpls into one big .mkv (as intermediate file for further processing) using MakeMKV solved the problem as well.

Added:
If you start from a BD structure you should also be successful with jdobb's BD Rebuilder.

Thunderbolt8

23rd October 2010, 11:07

Try this link:

http://forum.doom9.org/showthread.php?p=1443085#post1443085
I saw that link, but since both setups are so similar, how can you really be sure that its the one and not the other?

xkodi

23rd October 2010, 11:25

Thunderbolt8

23rd October 2010, 12:32

just tick "display all 7.1 channel layouts" under preferences ;)

Thunderbolt8

23rd October 2010, 17:43

when having a 32-bit float wavepack file and I want to convert it losslessly to .flac, how to do that best?

when converting what 32-bit float file to .wav and then to flac (because wavepack input is not supported by eac3to), then I get the message "reducing bithdepth from 64 to 24-bits" and the final flac file will have 24-bits which is not lossless any more, right?

same when I convert the 32-bit float file to 32-bit non floating .wav (is this step still lossless btw?) and then to flac, then I get "reducing bitdepth from 32 to 24 bits" and the final flac file isnt lossless any more either, correct?

if so, why is the creation of 32-bit flac files not supported? according to wikipedia, flac supports up to 32-bit: "It can handle any PCM bit resolution from 4 to 32 bits per sample"

flyingernst

25th October 2010, 09:23

Hi guys. I have a problem
System Win 7 x64 + newest eac3to + Arcsoft tmt3 3.0.1.180

I cant encode a *.dtsma to flac because eac3to didn´t find the arcsoft decoder.
What did I wrong?

Thank, greetings, flyingernst

Edit: I tryed this: http://forum.doom9.org/showpost.php?p=1302586&postcount=9076 but I get an error

edit edit: i installed Version 2, now it works

yaous

27th October 2010, 11:22

Sorry for the newbie questions.

1: h264/AVC 1080p24 /1.001 (16:9)
2: AC3, German, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
3: TrueHD/AC3, English, 7.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)

'eac3to 1) 3: d:\audio.ac3'

this command line extracts embedded ac3 track?
or re-encode TrueHD to ac3?

mrr19121970

27th October 2010, 11:30

It will extract the core, to be 100% sure of this add -core.

eac3to 1) 3: d:\audio.ac3 -core

xxx666yyy777

1st November 2010, 14:39

IIRC joining the multiple .m2ts from the .mpls into one big .mkv (as intermediate file for further processing) using MakeMKV solved the problem as well.

Added:
If you start from a BD structure you should also be successful with jdobb's BD Rebuilder.

So, does anyone have a definite description of this problem and when/how it occurs? Are these audio delays occuring, when converting to .wav or also, when simply demuxing (to .dts/.dtsma)?

Thanks.

xxx666yyy777

1st November 2010, 19:38

So, does anyone have a definite description of this problem and when/how it occurs? Are these audio delays occuring, when converting to .wav or also, when simply demuxing (to .dts/.dtsma)?

Thanks.

I did some small tests and its getting really interesting. First, here is the "make-up" of the BD:

C:\Windows\System32>eac3to y:\ 3)
M2TS, 1 video track, 5 audio tracks, 4 subtitle tracks, 1:43:01, 100.758p
1: Chapters, 34 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48kHz)
5: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
6: AC3 EX, French, 5.1 channels, 640kbps, 48kHz
7: AC3 EX, Spanish, 5.1 channels, 640kbps, 48kHz
8: Subtitle (PGS), English
9: Subtitle (PGS), English
10: Subtitle (PGS), French
11: Subtitle (PGS), Spanish

Now, I performed several "actions" to obtain the 4th stream - the DTS-HD with the embedded DTS-ES and I compared the durations of the different streams. The audiogaps, as reported by eac3to are:

a04 Audio overlaps for 5ms at playtime 0:00:44.
a04 Audio overlaps for 12ms at playtime 0:03:12.
a04 Audio overlaps for 9ms at playtime 0:04:16.
a04 Audio overlaps for 11ms at playtime 0:11:46.
a04 Audio overlaps for 6ms at playtime 0:13:37.
a04 Audio overlaps for 7ms at playtime 0:15:39.
a04 Audio overlaps for 11ms at playtime 0:16:00.
a04 Audio overlaps for 6ms at playtime 0:16:43.
a04 Audio overlaps for 10ms at playtime 0:19:24.
a04 Audio overlaps for 9ms at playtime 0:21:04.
a04 Audio overlaps for 8ms at playtime 0:43:06.
a04 Audio overlaps for 12ms at playtime 0:48:20.
a04 Audio overlaps for 10ms at playtime 0:48:53.
a04 Audio overlaps for 10ms at playtime 1:05:27.
a04 Audio overlaps for 10ms at playtime 1:05:56.
a04 Audio overlaps for 11ms at playtime 1:08:40.
a04 Audio overlaps for 10ms at playtime 1:10:41.
a04 Audio overlaps for 10ms at playtime 1:12:10.
a04 Audio overlaps for 6ms at playtime 1:15:25.
a04 Audio overlaps for 11ms at playtime 1:17:36.
a04 Audio overlaps for 11ms at playtime 1:20:47.
a04 Audio overlaps for 10ms at playtime 1:25:00.
a04 Audio overlaps for 7ms at playtime 1:35:54.
a04 Audio overlaps for 6ms at playtime 1:36:42.
a04 Audio overlaps for 10ms at playtime 1:38:39.

The different actions I used to extract the audio are listed below. The streams have the same length in hours/mins/seconds, except for the miliseconds. I have listed the miliseconds and the ectraction method used in the list below:

363 - eac3to straight to w64
363 - eac3to straight to flac
363 - eac3to straight to wavs
363 - eac3to straight to flac then eac3to to wavs
355 - tsmux the playlist, then eac3to to w64
364 - MakeMKV to mkv, then eac3to to w64
309 - eac3to to dtsma, then eac3to to dts (core)
309 - eac3to straight to dts (core)

When I extract the video part by itself via eac3to (.264) and then mux with tsmuxer to .m2ts, the miliseconds are 272. Extracting the video with eac3to to .mkv results in the miliseconds being 314.

Does anyone know, which one is actually right??? :scared:

dansrfe

1st November 2010, 19:50

This has already been discussed before. When extracting DTS-HDMA to wavs eac3to does NOT fix the delays as reported. Hopefully madshi will come out with an update soon *fingers crossed*. For now you can encode to FLAC then demux the wavs because according to an earlier poster, eac3to DOES fix the delays when encoding from DTS-HDMA to FLAC however I'm still not very keen with the idea and want to see a real fix.

mindbomb

2nd November 2010, 15:49

In my experience, when converting truehd to .wav, it always reduces bitdepth from 24 to 16 with an additional pass.
So are truehd encoders required to output 24 bit regardless of original or something of that nature?
there isn't a bug with decoding, is there?

dansrfe

2nd November 2010, 15:54

The truehd streams are padded to 24 bit because, as you said, "truehd encoders [are] required to output 24 bit regardless of original" is true. eac3to doesn't seem the need to leave the padding so it removes it and leaves the job to the decoder to pad it back and output it.

setarip_old

7th November 2010, 00:59

@Murkurial

Hi!

Are you talking about the "Machete" DVD that came out in 2007, or the one that won't be released until January 2011?

Murkurial

7th November 2010, 01:30

Alright, I'll give it another shot.

I would simply like to know if anyone has discovered a solution to the problem discussed here (http://forum.doom9.org/showthread.php?t=149695)? The inability of eac3to to properly identify and convert DTS-MA audio.

jj666

7th November 2010, 15:48

Murkurial, that problem was fixed in the last version of EAC3TO, you can process .DTSHD created with DTS Master Audio Suite directly. Recommended is to remove -21ms from the start of the file to stay in sync with the original.

Setarip, Machete has been out in Russia for a while on Blu-ray, perhaps that's what he's working with. I don't see his original post to see what he was asking.

Cheers,

-jj-

setarip_old

7th November 2010, 23:29

@jj666

Hi! Setarip, Machete has been out in Russia for a while on Blu-ray, perhaps that's what he's working with.

Snce you've specifically addressed me, if your source of information for your statement is the following remark in Wikipedia: Home video release
Director Robert Rodriguez has expressed in an interview that a more violent director's cut will be released on home media.

The R-rated theatrical version of Machete was released in Russia on DVD and Blu-Ray on September 30, 2010. Because of licensing reasons, these releases only contain Russian and Ukrainian audio and no English-language track. I'd suggest you read between the lines regarding "Because of licensing reasons." Keep in mind that the first official showing of this movie anywhere was on September 3rd.

Additionally, since you addressed this to me, I did a rather thorough Googlesearch and could find no legitimate sources selling the DVD or Blu-ray discs. If you are aware of any legitimate sources, please be good enough to post them...

Murkurial

8th November 2010, 06:25

jj666

8th November 2010, 08:15

Thanks for that. Can eac3to be used to cut that 21ms from the file or would I need another program to do that? And if you could, how would I do that?

EAC3TO input.dtshd output.dts -21ms should do what you need.

Cheers,

-jj-

follz20

8th November 2010, 08:17

Can someone help me out with a query I have?

When I rip an lossless audio track (LPCM or TrueHD) to individual channel .wavs, why is the output bit rate always constant? I've done it multiple times for different movie tracks, but the bit rate is always at 768 Kbps and the output file size is exactly the same for each channel - which I find highly confusing. Shouldn't the bit rate be variable and the output file size for each channel differ greatly?

I'm probably just confused on this, so any help/info would be very much appreciated.

I just want to make sure no information is discarded in the process - I want know for sure that the .wavs are lossless (ie identical to the source).

jj666

8th November 2010, 08:27

@jj666

Hi!

Snce you've specifically addressed me, if your source of information for your statement is the following remark in Wikipedia: I'd suggest you read between the lines regarding "Because of licensing reasons." Keep in mind that the first official showing of this movie anywhere was on September 3rd.

Additionally, since you addressed this to me, I did a rather thorough Googlesearch and could find no legitimate sources selling the DVD or Blu-ray discs. If you are aware of any legitimate sources, please be good enough to post them...

Hi Setarip,

Here you go:

bolero (http://www.bolero.ru/product-82456918.html?terms=machete)
ozon (http://www.ozon.ru/context/detail/id/5537482/)

I'm not sure what the original, now removed, question was, but just wanted to point out this disk was already available (same also goes for the Expendables), with language caveats you already mentioned.

Cheers,

-jj-

tebasuna51

8th November 2010, 13:54

@follz20
.wav, .w64 and .pcm outputs contain the same uncompressed audio samples and are identical to lossless sources (TrueHD, DTS-MA, FLAC) and LPCM uncompressed of course.

The bitrate of uncompressed audio is always the same:
BitRate = BitDepth x SampleRate x Num_Channels

Then for a monochannel wav 16 bits 48 KHz
Bitrate = 16 x 48000 x 1 = 768 Kb/s

of course all monochannel wavs must have the same size.

rapscallion

8th November 2010, 16:25

@tebasuna51
Further to that, I decoded (via eacto_more) a True HD track, 3.8gb, to wavs ( I tried both nero and libav decoders) and result was 6 wavs ~992,600 KB ea.

Then I downconverted the same HD track, via tsmuxer, to extract the core ( ~551,300 KB). I then repeated the process, as above, and the resulting wavs were still all the same size.

So, in the first step, are the wavs truly lossless ? And why, in the 2nd step are the wav files still the same size? Logically, I would think that the first set of wavs should be substantialy larger.

nurbs

8th November 2010, 16:37

If the sources have the same bitdepth, samplerate, duration and number of channels the resulting wavs will have the same size. It's uncompressed so the content doesn't matter for the final size.

shogo_kawada

8th November 2010, 17:34

Hi, I'm trying to demux my Blade Runner collection HD-DVD, disc 3. It's the disc with the 3 versions (international, american and 1992 dir cut) put together via seamless branching. I'm having trouble to correctly demux these 3 versions of the movie.

I run this command (disc is already ripped to hard drive, playlist 1 is the international version):
eac3to c:\bladerunner_branching 1) -demux -keepdialnorm

eac3to gives me the following log (notice the red part), and when I try to mux the streams back together with MkvMerge audio is out of sync starting from 1:39:58 onwards.

Another strange thing: there should only be 3 audio streams, while eac3to detects (and demux) 6. None of them works in sync with the video. Can anybody help me? Here's the log

eac3to v3.24
command line: eac3to c:\ripped\bladerunner_branching 1) -demux -keepdialnorm
------------------------------------------------------------------------------
EVO, 1 video track, 6 audio tracks, 6 subtitle tracks, 1:57:29
1: Joined EVO file
2: Chapters, 35 chapters
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: E-AC3, English, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB, -21ms
5: E-AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB, -21ms
6: E-AC3 Surround, French, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB, -21ms
7: E-AC3, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB, -18ms
8: E-AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
9: E-AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB, -7ms
10: Subtitle (DVD), English
11: Subtitle (DVD), French
12: Subtitle (DVD), Spanish
13: Subtitle (DVD)
14: Subtitle (DVD)
15: Subtitle (DVD)
Creating file "SEG01_DC - Chapters.txt"...
[v03] Extracting video track number 3...
[a04] Extracting audio track number 4...
[a06] Extracting audio track number 6...
[a05] Extracting audio track number 5...
[a08] Extracting audio track number 8...
[a07] Extracting audio track number 7...
[a09] Extracting audio track number 9...
[v03] Writing new framerate "24fps /1.001" to bitstream.
[a04] Applying (E-)AC3 delay...
[a04] A remaining delay of -5ms could not be fixed.
[a05] Applying (E-)AC3 delay...
[a07] Applying (E-)AC3 delay...
[a07] A remaining delay of -2ms could not be fixed.
[v03] Removing VC-1 pulldown...
[a06] Applying (E-)AC3 delay...
[a06] A remaining delay of +11ms could not be fixed.
[a09] A remaining delay of -7ms could not be fixed.
[a05] A remaining delay of +11ms could not be fixed.
[v03] Creating file "SEG01_DC - 3 - VC-1, 1080p24.vc1"...
[a04] Creating file "SEG01_DC - 4 - E-AC3, English, 5.1 channels, 640kbps, 48kHz.eac3"...
[a07] Creating file "SEG01_DC - 7 - E-AC3, 5.1 channels, 640kbps, 48kHz.eac3"...
[a05] Creating file "SEG01_DC - 5 - E-AC3 Surround, English, 2.0 channels, 192kbps, 48kHz.eac3"...
[a08] Creating file "SEG01_DC - 8 - E-AC3 Surround, 2.0 channels, 192kbps, 48kHz.eac3"...
[a09] Creating file "SEG01_DC - 9 - E-AC3 Surround, 2.0 channels, 192kbps, 48kHz.eac3"...
[a06] Creating file "SEG01_DC - 6 - E-AC3 Surround, French, 2.0 channels, 192kbps, 48kHz.eac3"...
[s15] Extracting subtitle track number 15...
[s12] Extracting subtitle track number 12...
[s14] Extracting subtitle track number 14...
[s11] Extracting subtitle track number 11...
[s10] Extracting subtitle track number 10...
[s13] Extracting subtitle track number 13...
[s15] Creating file "SEG01_DC - 15 - Subtitle (DVD).sup"...
[s14] Creating file "SEG01_DC - 14 - Subtitle (DVD).sup"...
[s12] Creating file "SEG01_DC - 12 - Subtitle (DVD), Spanish.sup"...
[s13] Creating file "SEG01_DC - 13 - Subtitle (DVD).sup"...
[s11] Creating file "SEG01_DC - 11 - Subtitle (DVD), French.sup"...
[s10] Creating file "SEG01_DC - 10 - Subtitle (DVD), English.sup"...
[v03] Video overlaps for 705 frames at playtime 1:40:02. <WARNING>
[v03] Video overlaps for 1 frames at playtime 1:40:32. <WARNING>
[v03] Video overlaps for 7 frames at playtime 1:52:24. <WARNING>
[v03] Video overlaps for 6 frames at playtime 1:52:25. <WARNING>
[a04] Audio overlaps for 4192ms at playtime 1:39:58. <WARNING>
[a05] Audio overlaps for 4192ms at playtime 1:39:58. <WARNING>
[a06] Audio overlaps for 4192ms at playtime 1:39:58. <WARNING>
[a07] Audio overlaps for 4192ms at playtime 1:39:58. <WARNING>
[a08] Audio overlaps for 4192ms at playtime 1:39:58. <WARNING>
[a09] Audio overlaps for 4192ms at playtime 1:39:58. <WARNING>
[a04] Starting 2nd pass...
[a04] Realizing (E-)AC3 gaps...
[a04] Creating file "SEG01_DC - 4 - E-AC3, English, 5.1 channels, 640kbps, 48kHz.eac3"...
[a05] Starting 2nd pass...
[a05] Realizing (E-)AC3 gaps...
[a05] Creating file "SEG01_DC - 5 - E-AC3 Surround, English, 2.0 channels, 192kbps, 48kHz.eac3"...
[a06] Starting 2nd pass...
[a06] Realizing (E-)AC3 gaps...
[a06] Creating file "SEG01_DC - 6 - E-AC3 Surround, French, 2.0 channels, 192kbps, 48kHz.eac3"...
[a07] Starting 2nd pass...
[a07] Realizing (E-)AC3 gaps...
[a07] Creating file "SEG01_DC - 7 - E-AC3, 5.1 channels, 640kbps, 48kHz.eac3"...
[a08] Starting 2nd pass...
[a08] Realizing (E-)AC3 gaps...
[a08] Creating file "SEG01_DC - 8 - E-AC3 Surround, 2.0 channels, 192kbps, 48kHz.eac3"...
[a09] Starting 2nd pass...
[a09] Realizing (E-)AC3 gaps...
[a09] Creating file "SEG01_DC - 9 - E-AC3 Surround, 2.0 channels, 192kbps, 48kHz.eac3"...
Video track 3 contains 169079 frames.
Subtitle track 10 contains 875 captions.
Subtitle track 11 contains 741 captions.
Subtitle track 12 contains 741 captions.
Subtitle track 13 contains 919 captions.
Subtitle track 14 contains 799 captions.
Subtitle track 15 contains 798 captions.
eac3to processing took 17 minutes, 26 seconds.
Done.

rapscallion

8th November 2010, 18:42

If the sources have the same bitdepth, samplerate, duration and number of channels the resulting wavs will have the same size. It's uncompressed so the content doesn't matter for the final size.

Well, the True HD has a variable bitrate (peak 5200 kbps) while the core is constant @ 640.

So are you saying that it's just a matter of compression, and that the extracted wavs from both sources will be lossless and the same as the originals?

If that's the case, you should be able to take a 640 ac-3 track, extract the wavs, and then process them to DTS-MA (DTS MA Audio Suite) ?

Edit : Btw, here are the log files from both decodes: Note that it took twice as long to decode the HD stream.

eac3to v3.24
------------------------------------------------------------------------------
TrueHD/AC3, 5.1 channels, 48kHz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB)
Disabling DRC for Nero (E-)AC3 decoding...
Extracting TrueHD stream...
Removing TrueHD dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 5.1 channels, 24 bits, 48kHz
Writing WAVs...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 10 minutes, 56 seconds.
Done.

eac3to v3.24
------------------------------------------------------------------------------
AC3, 5.1 channels, 1:57:37, 640kbps, 48kHz, dialnorm: -27dB
Disabling DRC for Nero (E-)AC3 decoding...
Removing AC3 dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 5.1 channels, 24 bits, 48kHz
Writing WAVs...
The last (E-)AC3 frame is incomplete and thus gets skipped.
<WARNING>
eac3to processing took 4 minutes, 17 seconds.
Done.

TinTime

8th November 2010, 21:59

...here are the log files from both demuxes...

The thing is you're not demuxing, you're decoding to PCM. This will result in a fixed size as nurbs and tebasuna51 said. The compression of the source is irrelevant, assuming that you decode all sources to the same sample rate / bit depth / number of channels.

rapscallion

8th November 2010, 22:19

The thing is you're not demuxing, you're decoding to PCM. This will result in a fixed size as nurbs and tebasuna51 said. The compression of the source is irrelevant, assuming that you decode all sources to the same sample rate / bit depth / number of channels.
Yes, wrong phraseology on my part. I do realize that the wav file sizes will be the same.

What I'm asking is why the same results from both the True HD source and the 640 source of the same track? And as asked in the post, are the resulting wavs indeed lossless? and can either set be used to create a DTS MA track ?

nurbs

8th November 2010, 22:48

You don't get the same results from TrueHD and AC3. The content of the wavs will be different, only the size will be the same. They are still lossless representations of the source. You can use either to create a DTS-HD MA track, but using the wavs created from the TrueHD files would be better.

rapscallion

8th November 2010, 23:18

Ah, ha....that was the definitive answer I was looking for. Thanks nurbs !!

follz20

9th November 2010, 04:56

Thanks for the responses tebasuna51, rapscallion & nurbs... you've all been a great help.

kieranrk

9th November 2010, 14:13

@Murkurial

Hi!

Are you talking about the "Machete" DVD that came out in 2007, or the one that won't be released until January 2011?

And the accosting is back...

saint-francis

9th November 2010, 15:05

I can't seem to use eac3to over a network. Is there a reason for this? Do I have to map a drive to get this to work?

Thunderbolt8

9th November 2010, 21:33

nurbs

9th November 2010, 21:40

You could try mediainfo or decode it with ffdshow to get a third and fourth opinion.

rapscallion

9th November 2010, 21:55

how precise is the 3.1 channel detection of eac3to? eac3to reports the sound of music blu-ray a DD track with 3.1 channel mask, but review sites state its 4.0. they could be wrong of course, but how can I know which is supposed to be correct?
Actually, eac3to reports it as 3/1 , not 3.1.

Mediainfo reports 4 - L C R, Surround : C

So eac3to is reporting correctly just stating it differently. Strange track BTW, especially when DTS MA 7.1 is the main audio track

Thunderbolt8

10th November 2010, 08:38

so what about the last channel, is it back centrered? because back surround only at 1 side doesnt make much sense

shogo_kawada

10th November 2010, 09:56

A little help here? :helpful:

To sum up: Blade Runner HD-DVD, seamless branching with 3 different versions of the movie. eac3to gives me the list of playlist, listing all of them with 3 audio. When I demux the first playlist, i get this "video and audio overlaps" problem, and I get 6 audio streams instead of 3. More importantly, none of the demuxed audio track is correctly in sync with the video, starting from 1:39:58.

This is the log (look at the red part please):

eac3to v3.24
command line: eac3to c:\ripped\bladerunner_branching 1) -demux -keepdialnorm
------------------------------------------------------------------------------
EVO, 1 video track, 6 audio tracks, 6 subtitle tracks, 1:57:29
1: Joined EVO file
2: Chapters, 35 chapters
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: E-AC3, English, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB, -21ms
5: E-AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB, -21ms
6: E-AC3 Surround, French, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB, -21ms
7: E-AC3, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB, -18ms
8: E-AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
9: E-AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB, -7ms
10: Subtitle (DVD), English
11: Subtitle (DVD), French
12: Subtitle (DVD), Spanish
13: Subtitle (DVD)
14: Subtitle (DVD)
15: Subtitle (DVD)
Creating file "SEG01_DC - Chapters.txt"...
[v03] Extracting video track number 3...
[a04] Extracting audio track number 4...
[a06] Extracting audio track number 6...
[a05] Extracting audio track number 5...
[a08] Extracting audio track number 8...
[a07] Extracting audio track number 7...
[a09] Extracting audio track number 9...
[v03] Writing new framerate "24fps /1.001" to bitstream.
[a04] Applying (E-)AC3 delay...
[a04] A remaining delay of -5ms could not be fixed.
[a05] Applying (E-)AC3 delay...
[a07] Applying (E-)AC3 delay...
[a07] A remaining delay of -2ms could not be fixed.
[v03] Removing VC-1 pulldown...
[a06] Applying (E-)AC3 delay...
[a06] A remaining delay of +11ms could not be fixed.
[a09] A remaining delay of -7ms could not be fixed.
[a05] A remaining delay of +11ms could not be fixed.
[v03] Creating file "SEG01_DC - 3 - VC-1, 1080p24.vc1"...
[a04] Creating file "SEG01_DC - 4 - E-AC3, English, 5.1 channels, 640kbps, 48kHz.eac3"...
[a07] Creating file "SEG01_DC - 7 - E-AC3, 5.1 channels, 640kbps, 48kHz.eac3"...
[a05] Creating file "SEG01_DC - 5 - E-AC3 Surround, English, 2.0 channels, 192kbps, 48kHz.eac3"...
[a08] Creating file "SEG01_DC - 8 - E-AC3 Surround, 2.0 channels, 192kbps, 48kHz.eac3"...
[a09] Creating file "SEG01_DC - 9 - E-AC3 Surround, 2.0 channels, 192kbps, 48kHz.eac3"...
[a06] Creating file "SEG01_DC - 6 - E-AC3 Surround, French, 2.0 channels, 192kbps, 48kHz.eac3"...
[s15] Extracting subtitle track number 15...
[s12] Extracting subtitle track number 12...
[s14] Extracting subtitle track number 14...
[s11] Extracting subtitle track number 11...
[s10] Extracting subtitle track number 10...
[s13] Extracting subtitle track number 13...
[s15] Creating file "SEG01_DC - 15 - Subtitle (DVD).sup"...
[s14] Creating file "SEG01_DC - 14 - Subtitle (DVD).sup"...
[s12] Creating file "SEG01_DC - 12 - Subtitle (DVD), Spanish.sup"...
[s13] Creating file "SEG01_DC - 13 - Subtitle (DVD).sup"...
[s11] Creating file "SEG01_DC - 11 - Subtitle (DVD), French.sup"...
[s10] Creating file "SEG01_DC - 10 - Subtitle (DVD), English.sup"...
[v03] Video overlaps for 705 frames at playtime 1:40:02. <WARNING>
[v03] Video overlaps for 1 frames at playtime 1:40:32. <WARNING>
[v03] Video overlaps for 7 frames at playtime 1:52:24. <WARNING>
[v03] Video overlaps for 6 frames at playtime 1:52:25. <WARNING>
[a04] Audio overlaps for 4192ms at playtime 1:39:58. <WARNING>
[a05] Audio overlaps for 4192ms at playtime 1:39:58. <WARNING>
[a06] Audio overlaps for 4192ms at playtime 1:39:58. <WARNING>
[a07] Audio overlaps for 4192ms at playtime 1:39:58. <WARNING>
[a08] Audio overlaps for 4192ms at playtime 1:39:58. <WARNING>
[a09] Audio overlaps for 4192ms at playtime 1:39:58. <WARNING>
[a04] Starting 2nd pass...
[a04] Realizing (E-)AC3 gaps...
[a04] Creating file "SEG01_DC - 4 - E-AC3, English, 5.1 channels, 640kbps, 48kHz.eac3"...
[a05] Starting 2nd pass...
[a05] Realizing (E-)AC3 gaps...
[a05] Creating file "SEG01_DC - 5 - E-AC3 Surround, English, 2.0 channels, 192kbps, 48kHz.eac3"...
[a06] Starting 2nd pass...
[a06] Realizing (E-)AC3 gaps...
[a06] Creating file "SEG01_DC - 6 - E-AC3 Surround, French, 2.0 channels, 192kbps, 48kHz.eac3"...
[a07] Starting 2nd pass...
[a07] Realizing (E-)AC3 gaps...
[a07] Creating file "SEG01_DC - 7 - E-AC3, 5.1 channels, 640kbps, 48kHz.eac3"...
[a08] Starting 2nd pass...
[a08] Realizing (E-)AC3 gaps...
[a08] Creating file "SEG01_DC - 8 - E-AC3 Surround, 2.0 channels, 192kbps, 48kHz.eac3"...
[a09] Starting 2nd pass...
[a09] Realizing (E-)AC3 gaps...
[a09] Creating file "SEG01_DC - 9 - E-AC3 Surround, 2.0 channels, 192kbps, 48kHz.eac3"...
Video track 3 contains 169079 frames.
Subtitle track 10 contains 875 captions.
Subtitle track 11 contains 741 captions.
Subtitle track 12 contains 741 captions.
Subtitle track 13 contains 919 captions.
Subtitle track 14 contains 799 captions.
Subtitle track 15 contains 798 captions.
eac3to processing took 17 minutes, 26 seconds.
Done.

setarip_old

10th November 2010, 16:33

@shogo_kawada

Hi!

I believe you should be able to use tsMuxeR instead (in conjunction with AnyDVDHD) ...

follz20

10th November 2010, 17:06

OK, another potentially embarrassing question coming up again.. here goes anyway.

I have several FLAC files that have been converted by eac3to from raw LPCM streams - so obviously lossless -> lossless = lossless. So far so good!

Now, if I load up those certain converted FLAC files in eac3to and rip to .wavs, is the resulting output the exact same as the original LPCM stream (ie. if I ripped the original LPCM stream to .wavs, would the result exactly match the current .wavs from the FLAC file - bit-for-bit)?

Apologies for the question, as I'm sure the answer is yes... but just would like the comfort of confirmation from the experts.

Cheers

dansrfe

10th November 2010, 18:03

yes^ It should be bit-for-bit identical.

shogo_kawada

10th November 2010, 18:20

@shogo_kawada

Hi!

I believe you should be able to use tsMuxeR instead (in conjunction with AnyDVDHD) ...
Hi :)

So TsMuxeR can also demux hd-dvds? I'll try that as soon as I go home from work

rapscallion

10th November 2010, 19:31

so what about the last channel, is it back centrered? because back surround only at 1 side doesnt make much sense
Yes, it's rear, center. Again, strange track, can't imagine why they included it as a 2nd Eng track.

shogo_kawada

12th November 2010, 15:45

No luck with my Blade Runner HD-DVD yet :(. Anybody here has it? Could it be a damaged disc?

setarip_old

12th November 2010, 18:03

@shogo_awada No luck with my Blade Runner HD-DVD yetAnd EXACTLY what happened when you tried tsMuxeR?

Thunderbolt8

13th November 2010, 09:58

for me, the HD DVD has already worked back then when it came out, so the disc itself & eac3to working with it shouldnt be the problem.

shogo_kawada

13th November 2010, 17:03

@shogo_awada And EXACTLY what happened when you tried tsMuxeR?

I just can't find a way to select the correct playlist. I mean, with tsmuxer (I use the official GUI) I don't have a way to select a disc drive or a hd-dvd folder structure. Do I have to select every single segment evo file and join them?

Thunderbolt8

13th November 2010, 17:45

only the two biggest ones, like feature_1 & feature_2 or pevob_1, pevob_2

shogo_kawada

13th November 2010, 18:04

only the two biggest ones, like feature_1 & feature_2 or pevob_1, pevob_2
Normally that's right, but this is a seamless branching disc. It has lots of segments (called DC_01, DC_02, INT_01, INT_02 and so on). Each of the three version of the movie is a different playlist which combines in its own way the existing segments (for exemple: DC_01+DC_02+...+DC_XX is the dir cut, while DC_01+INT_01+INT02+DC_02 etc is the domestic).

setarip_old

13th November 2010, 19:18

@shogo_kawada Do I have to select every single segment evo file and join them?Every one that's part of your selected playlist/version, yes (IIRC).

On the other hand (It's been a while since I last dealt with my HD-DVDs), I believe an earlier program, "EVODemux" can use the "VTI" file, which I believe contains the playlists, to join the requisite video files...

shogo_kawada

14th November 2010, 10:50

Ok I'll try these two methods (evodemux and tsmuxer selecting all the segments), even though eac3to ALREADY selects the right playlist, joining all the segments.

I think one of the problems is that this particular seamless branching isn't just a matter of different playlists of fixed segments, since there's a version of the movie (the director's cut) which uses a different soundtrack. I think this is the reason for eac3to listing 6 soundtrack for each playlist instead of the 3 listed on the back cover.

BTW since this is an eac3to thread, maybe I should post the results on another thread. The issue with this program stands though: is there anyone (madshi maybe) who knows how to make eac3to work with this particular disc?

UPDATE: trying tsmuxer (manually adding every segment) seems to have worked! I can't understand why it doesn't work in eac3to though

setarip_old

14th November 2010, 16:55

@shogo_kawada UPDATE: trying tsmuxer (manually adding every segment) seems to have worked! I can't understand why it doesn't work in eac3to thoughGlad to hear my first suggestion worked for you...

I can't understand why it doesn't work in eac3to thoughThat's why there are programs that perform similar functions ;>}

asarian

14th November 2010, 17:52

I'm getting some real fu-barred result extracting the English .sup track (nr. 15) from Avatar, the Extended Blu-Ray edition; here it is: sups (http://www.mediafire.com/file/9b1b6tb2uc8ouz3/avatar_e2.sup).

The timings are extremely off. It starts with a suptitle at 15 mins, then at 4 mins, etc. And I can't get any of the subs to actually show with the AviSynth SupTitle plugin.

Can anyone please look into this?

Thanks.

setarip_old

14th November 2010, 18:09

@asarian

Hi!

If it's not resolved using EAC3to, then I believe you too should be able to use tsMuxeR instead (in conjunction with AnyDVDHD) ...

asarian

14th November 2010, 18:40

@asarian

Hi!

If it's not resolved using EAC3to, then I believe you too should be able to use tsMuxeR instead (in conjunction with AnyDVDHD) ...

Thanks! That actually worked (I had never used tsMuxeR to extract subs from a playlist before). The resultant .sup file is perfect. :)

Still, for future reference, would be nice if eac3to could get it right, too, of course.

shogo_kawada

14th November 2010, 18:43

@shogo_kawada Glad to hear my first suggestion worked for you...

That's why there are programs that perform similar functions ;>}

Thanks again :). But I've got another one for you: I still got issues with the subtitle streams. TsMuxer doesn't seem to be able to demux those (am I missing something or TsMuxer only sees video and audio tracks? No chapter nor subtitles), and the ones demuxed by eac3to got sync issues (like audio tracks demuxed by eac3to did). How can I properly demux the subs track?

Thanks! That actually worked (I had never used tsMuxeR to extract subs from a playlist before). The resultant .sup file is perfect. :)

Did you use the GUI? When I load an evo file from that tsmuxer only shows video and audio streams

asarian

14th November 2010, 18:54

Did you use the GUI? When I load an evo file from that tsmuxer only shows video and audio streams

The Avatar Extended disc I used is a Blu-Ray, it doesn't have EVO files. :) It has a PLAYLIST dir with .mpls files (which I needed here, as the main feature is scattered over 20+ individual segments).

setarip_old

15th November 2010, 10:03

@asarian Thanks! That actually workedYou're quite welcome ;>}

hoju3508

16th November 2010, 10:36

@shogo_kawada, have you tried using Clown_BD (http://forum.slysoft.com/showthread.php?t=25818)? I've used this many times to extract from HD-DVD.

setarip_old

16th November 2010, 14:54

@shogo_kawada

If your desired end result is an .MKV file, have you tried using MakeMKV (NO other programs needed) to, with one click, decrypt and convert to an .MKV file - and then, if necessary, use MKVMerge?

shogo_kawada

17th November 2010, 08:58

@shogo_kawada, have you tried using Clown_BD (http://forum.slysoft.com/showthread.php?t=25818)? I've used this many times to extract from HD-DVD.
I tried it, but it had the same problems eac3to had. I guess it's because it uses eac3to for the demuxing process.
@shogo_kawada

If your desired end result is an .MKV file, have you tried using MakeMKV (NO other programs needed) to, with one click, decrypt and convert to an .MKV file - and then, if necessary, use MKVMerge?
I didn't know Makemkv worked with hd-dvd too! I'm trying right now. I read that it doesn't strip the pulldown flag, so I guess I'll just have to do a second demux/remux with either eac3to or TsMuxer + mkvmerge. I'll post the results by the way, thanks again ;)

shogo_kawada

17th November 2010, 14:13

Ok, I tried and here's some partial results (I had to go to work so I won't be able to continue until tomorrow): Makemkv seems to have correctly identified the 3 playlists, and audio tracks are well in sync throughout the whole movies. I can't tell about the subs though, since vlc didn't display them, even though mediainfo detects them. When I come back home I got to try to demux the mkvs created by makemkv (stripping the pulldown flag in the process), open the idx/sub files with suprip and see if if there's anything wrong with them, and remux with mkvmerge.

Thunderbolt8

17th November 2010, 15:18

tiny bit off topic (but not entirely, since this tool/frontend uses eac3to), maybe some people with programming skills like to participate (not my project btw)

http://code.google.com/p/xin1generator/

oddwunn

17th November 2010, 16:10

Is this the right thread to report problems with eac3to? If so, I would like to report 3 new issues:

1. On the German release of Lethal Weapon 4, the audio is out of sync with the video by about 15-20 seconds using eac3to. If demuxed with TSMuxer everything lines up correctly.

2. The subtitle streams in Avatar Extended Collector's Edition are not in sync with the Extended version or the Director's Cut. Once again, if you open the extended playlist file in TSMuxer and demux subs from there, they line up correctly, so it is just eac3to misunderstanding how to demux the subs for the different playlists.

3. I don't know if this is a DVDFab Passkey issue or an issue with eac3to, but on ALL titles ripped with DVDFab Passkey 7.0.5.7 eac3to reports:
Skipping 467712 bytes to work around damaged source file...
Hint: You can use the option "-fix" to clean up the damage.
The format of the source file could not be detected.
If I run the -fix switch, the message gets reduced to:
The format of the source file could not be detected.
And eac3to will no longer do anything to the "fixed" files.

Inspector.Gadget

18th November 2010, 02:53

Arcsoft has just released TotalMedia Theatre 5 (http://www.arcsoft.com/EN-US/software_title.asp?ProductCode=TMT5P). I and I'm sure others would appreciate it if anyone purchasing this software could find out whether the latest revision of Arcsoft's DTS decoder will work with eac3to.

mrr19121970

18th November 2010, 08:32

jeremy33

20th November 2010, 16:27

madshi could you please put a link to my GUI on the first page. Thank you.

Hi,

I made a little GUI for eac3to. I made it because I use only some eac3to functions and I wanted a GUI with only the function I use.
So I kept only the "essentials" functions of eac3to.

With eac3to µGUI you can :

* Change the framerate of AC3/TrueHD/DTS/DTS HD MA/Flac (ex: 25.000 -> 23.976)
* Encode an AC3/TrueHD/DTS/DTS HD MA/Flac (also in mkv) into AC3/DTS/Flac
* Extract the DTS-Core contained in DTS HD MA

I'm french, but eac3to µGUI is available in english. On my site there is also a tuto about mpchc, ffdshow, reclock, ac3filter, madVR.

Download eac3to µGUI (http://www.homecinema-hd.com/eac3tougui_en.html)

Uxorious

20th November 2010, 20:15

I'm seeing a problem when using eac3to/ArcSoft to transcode DTS 6.1 to anything.
The sound terrible ... like a constant ringing distortion.

From comparisons with files trascoded using -libav, I think the problem is that eac3to is mistaking the ArcSoft data as 7.1 instead of 6.1.
If I compare the start time of peaks in the sound of files created with -libav with files using ArcSoft, the ArcSoft files are increasingly delayed by a factor if 1/7.

Anybody else seeing this?
Is this the proper forum for bug reports?

Boulder

20th November 2010, 20:30

Is there the same problem(s) reported somewhere in this thread regarding other versions than 1.1.0.0? I think they had something to do with 6.1 or 7.1 audio decoding.

Uxorious

20th November 2010, 20:32

Boulder

20th November 2010, 20:34

Maybe I wasn't clear.
I already have ArcSoft installed, and eac3to reports that its working: ArcSoft DTS Decoder (1.1.0.7) works fine

Decoding DTS 5.1 works fine, and DTS-MA works too.
However, DTS 6.1 produces incorrect files.

Have you verified that 6.1 works with the older version you listed, or did you think I had nothing working at all?

I was actually answering to a different post which was removed while I was writing..

But still, your problem might be that one. There are reports in this thread regarding decoding tracks with channel configuration 6.1 (or 7.1, unfortunately I don't remember) with newer versions than 1.1.0.0.

dansrfe

20th November 2010, 20:35

ArcSoft HD Decoder 1.1.0.8 is working great with eac3to from the TotalMedia Theatre 5 trial version ;)

Extracted from the installer of course. no way I was going to install it lol

Thunderbolt8

20th November 2010, 21:15

does the version from tmt5 still have that problem with 6.1 dtsma tracks?

Uxorious

20th November 2010, 21:56

does the version from tmt5 still have that problem with 6.1 dtsma tracks?

Interestingly, that package seems to be generally newer than the one I had.
eac3to reports the same version (1.1.0.8), but looking at the individual files, they generally have a (slightly) higher version number than the files I had.

The problem persists with this version. DTS 6.1 files (they are not DTS-MA actually) still don't produce correct output.

b66pak

20th November 2010, 22:03

downgrade to 1.1.0.0 to fix it...
_

dansrfe

20th November 2010, 22:10

ArcSoft HD Decoder eac3to pack v1.1.0.8 (http://www.sendspace.com/file/unwnh5)

Instructions are inside.

Note: Any pack over 1.1.0.0 does not decode 6.1 DTS files accurately.

Uxorious

20th November 2010, 22:35

downgrade to 1.1.0.0 to fix it...
_

Awesome, that one works.
(To save people some time, search for TMT3, or the TMT2 codec pack).

Still, it would be nice if eac3to detected the version and worked around the issue ... if possible.

Does the author read the forum (good luck keeping up!), or is there some other way to do bugreports?

Inspector.Gadget

20th November 2010, 23:38

ArcSoft HD Decoder 1.1.0.8 is working great with eac3to from the TotalMedia Theatre 5 trial version ;)

Extracted from the installer of course. no way I was going to install it lol

Excellent news. Did you merely need to replace the old registered dll with the newer? Did you have to replace any of the dependencies (checkactivate.dll, etc.)?

Inspector.Gadget

21st November 2010, 00:15

If anyone else wants to test the 6.1 DTS issue

And yes, the DTS ES 6.1 issue is still present with 1.1.0.8. What installer is 1.1.0.0 from anyway?

dansrfe

21st November 2010, 00:25

1.1.0.0 is TMT2 is believe. I just replaced all the dlls and ASAudioHD.ax from TMT5 and used the same checkactivate.dll which I believe is still trailing from TMT2.

Uxorious

21st November 2010, 00:27

And yes, the DTS ES 6.1 issue is still present with 1.1.0.8. What installer is 1.1.0.0 from anyway?

In case URL's are deleted, Google for "TMT3 ArcSoft s Codec Pack".
I used the filestube link.
Password is URL in the "File Description" field.

dansrfe

21st November 2010, 00:40

If someone can point me to the exact installer for 1.1.0.0 then I can make a collective pack out of it which everyone on the forum can easily find.

Inspector.Gadget

21st November 2010, 02:05

I can confirm that eac3to downmixes and decodes the DTS-ES 6.1 track from Ghost in the Shell 2: Innocence to stereo FLAC correctly when using 1.1.0.0 (TMT2's final update, rev. 129) and incorrectly (ringing artifacts) when using 1.1.0.7 (my working copy of TMT3, but I've been lazy about updates) and 1.1.0.8 (from TMT5 installer).

7ekno

21st November 2010, 02:09

v1.1.0.0 comes from the update to v2 (http://www.arcsoft.com/downloads/digitaltheatre/arcsoft/totalmediatheatre_2.1.6.105_2.1.6.125_Update_ALL.exe) that is still on the Arcsoft file servers.

Incase the link didn't work, the full URL is here:
http://www.arcsoft.com/downloads/digitaltheatre/arcsoft/totalmediatheatre_2.1.6.105_2.1.6.125_Update_ALL.exe

7ek

shogo_kawada

21st November 2010, 10:08

Ok, I tried and here's some partial results (I had to go to work so I won't be able to continue until tomorrow)
Ok, here's the last update :). I finally succeeded doing what I was trying to, here's how:

. First used eac3to to identify how every playlist was chained together using the various evo segments

. Then demuxed video and audio with TsMuxer, manually inputting every single segment for the chosen playlist (no subs since TsMuxer didn't detect them)

. Came back to eac3to to extract subs and chapters

. Converted subs to srt via SupRip. I guess I could have converted them in idx/sub much faster, but since I needed to resync them (yes, on the next step) I went with srt, I wasn't sure if Subtitles Workshop works with idx/sub too

. Subs needed to be resynced with Subtitles Workshop after every "break point" (I don't know how it is called) in the former seamless branching playlist. I guess this makes sense, since they were extracted with eac3to which didn't correctly demux them in the first place.

. Finally, remuxed everything with MkvMerge

A final thought about what could have bothered eac3to so much to prevent it to correctly treat this disc. Forgive if I'm too verbose, but english isn't my language and I have some trouble expressing everything in a better form :p.

First, I understood why eac3to and every other program I tried to demux the playlists, listed twice as tracks as they should have (subs and audio). It's just that this particular seamless branching is a little more complicated than the usual one, where you have one of the following:

. A shorter and a longer version of the same movie. The short playlist just has to skip the segment of added content. Both use the same soundtrack, so every segment has it and plays it if part of the chosen playlist.

. Two version of the movie featuring alternative versions of some scenes. Robocop comes to my mind for the scene of the killing of Murphy. It's quite close to the first option, there will just be alternative segments for those scenes and the playlist will chose the right one to play. Again, every segment has just one version of the soundtrack.

. Here we have 3 versions. International and domestic fall in the first case scenario: domestic is just shorter than international, so the playlist just chose which segments must be removed from the former. But there's a third version, the 1992 director's cut, which is also shorter than the other two video-wise (first case scenario), but more importantly has a different soundtrack too, since it doesn't feature Harrison's Ford voice over. This of course reflects on the subs as well. So every segment in the disc has to have a soundtrack to be used in international/domestic playlists, and another one for the director's cut playlist.

BTW thanks to everyone who gave me advice, very much appreciated :thanks:

dalex_257

21st November 2010, 15:35

@madshi: Just wanted to report that eac3to has a problem demuxing subtitles from recent Avatar Extended edition bluray. That disk uses seamless branching if I extract subtitles for the extended edition they still have all the timings from the theatrical (shorter) version.

dansrfe

21st November 2010, 15:42

Thunderbolt8

21st November 2010, 16:28

is there actually anything 1.1.0.8 can do better than 1.1.0.0? otherwise, no other version than 1.1.0.0 would be needed.

Inspector.Gadget

21st November 2010, 17:06

is there actually anything 1.1.0.8 can do better than 1.1.0.0? otherwise, no other version than 1.1.0.0 would be needed.

I too wonder about this.

dansrfe

21st November 2010, 18:23

I too wonder about this.

I second that wonder.

Inspector.Gadget

21st November 2010, 18:48

Well, I guess the thing to do is put together a test suite and run it against all known versions of the Arcsoft decoder. Unfortunately, I don't have a Blu-ray collection large enough to do that.

Known versions of the Arcsoft decoder: 1.1.0.0, 1.1.0.7, 1.1.0.8, ???

DTS Variants: "Regular" DTS in up to 5.1 channels, DTS-ES (Matrix and Discrete) in up to 6.1 channels, DTS 96/24 in up to 5.1 channels, DTS-HD HR in up to 8 channels, DTS-HD MA in up to 8 channels, ???

Uxorious

21st November 2010, 18:53

Well, I guess the thing to do is put together a test suite and run it against all known versions of the Arcsoft decoder.

Or preferably, ask madshi to fix eac3to to work with the new versions.

If my suspicion from looking at the produced WAV file is correct, the new ArcSoft decoder is outputting a record with 8 channels per sample - but of course only 7 of them have correct content.
When eac3to converts it, it treats it as a 7-channel stream, thus cycling data and padding through all the channels ...

dansrfe

21st November 2010, 19:10

^that could very well be a possibility, but seeing how madshi has left this and his madVR thread in the dark, I don't know if there will ever be a next version to eac3to.

robpdotcom

21st November 2010, 19:56

Instructions in zip:

ArcSoft HD Decoder pack eac3to v1.1.0.0 (http://www.sendspace.com/file/ief0t0)

Latest version: ArcSoft HD Decoder pack eac3to v1.1.0.8 (http://www.sendspace.com/file/unwnh5)

Note: All versions above v1.1.0.0 do not accurately decode 6.1 DTS tracks with eac3to.

First, thanks for the easy decoder installation. Every other method I've seen was much more complicated.

Second, when using v1.1.0.8, I only get 6 channel WAV's from a DTS-ES 6.1 track.:confused:

Third, how do I go from v1.1.0.8 to v1.1.0.0? I tried just replacing the files, but I get an error "MSVCP71.dll" is missing. Anyone mind helping?

Uxorious

21st November 2010, 20:30

How do I go from v1.1.0.8 to v1.1.0.0? I tried just replacing the files, but I get an error "MSVCP71.dll" is missing. Anyone mind helping?

I'm running Windows7 64-bit and needed that file too.
Just google for the filename ... there are a couple zip files out there with the file.

Uxorious

21st November 2010, 20:37

ArcSoft HD Decoder pack eac3to v1.1.0.0 (http://www.sendspace.com/file/ief0t0)

The files I found are slightly newer and still work (but probably have no differences).
All your Mag* files report v1.0.0.148, but there's also a set of 1.0.0.156 out there.
As mentioned earlier, Google for TMT3 codec pack, and you will find it.

As somebody else mentioned, the 1.1.0.0 version needs MSVCP71.DLL whereas the new version doesn't.

Also, I don't think DtsDec.dll is used at all....

b66pak

21st November 2010, 20:37

Just google for the filename ... there are a couple zip files out there with the file.

or better get the real deal (and not some virus) form ms: Microsoft Visual C++ 2008 SP1 Redistributable Package (x86) (http://www.microsoft.com/downloads/en/details.aspx?familyid=A5C84275-3B97-4AB7-A40D-3802B2AF5FC2&displaylang=en)
_

Uxorious

21st November 2010, 20:47

or better get the real deal (and not some virus) form ms: Microsoft Visual C++ 2008 SP1 Redistributable Package (x86) (http://www.microsoft.com/downloads/en/details.aspx?familyid=A5C84275-3B97-4AB7-A40D-3802B2AF5FC2&displaylang=en)
_

True. I think it's from the 2005 redistributable though.

dansrfe

21st November 2010, 22:40

yea it uses 2005 redistributable.

robpdotcom

22nd November 2010, 00:04

I'm running Windows7 64-bit and needed that file too.
Just google for the filename ... there are a couple zip files out there with the file.

Thanks, that helped.

Still wondering though, are you guys getting 7 channel WAV's from DTS-ES Matrix? Or from DTS-ES Discrete? I'm not sure I've ever actually seen a 6.1 Discrete DTS-ES stream.

Uxorious

22nd November 2010, 00:34

Still wondering though, are you guys getting 7 channel WAV's from DTS-ES Matrix? Or from DTS-ES Discrete? I'm not sure I've ever actually seen a 6.1 Discrete DTS-ES stream.

I assume it's discrete since otherwise it would just show as 5.1 ... right?

So far I have seen DTS 6.1 content on 2 BluRay's:
Lord of War
Disturbia

robpdotcom

22nd November 2010, 01:09

I assume it's discrete since otherwise it would just show as 5.1 ... right?

Probably so. I was trying with matrixed content, which shows as DTS-ES 5.1 in eac3to. Too bad the Arcsoft decoder can't pull the matrixed back channel out to create 6.1 WAV's.

So far I have seen DTS 6.1 content on 2 BluRay's:
Lord of War
Disturbia

I'm seeing more 6.1 and 7.1 DTS-HD MA recently: Alice in Wonderland; 6.1 Matrix, and Toy Story 3; 7.1 Discrete, come to mind. But all of the 6.1 and 7.1 DTS-HD discs I've seen, whether it was matrix or discrete, has a DTS-ES 6.1 Matrix core.

dansrfe

23rd November 2010, 02:05

So audio overlaps are still not being fixed in the second pass. This is depressing.

asarian

23rd November 2010, 02:48

So, dumb question maybe, but how can I tell which version of the ArcSoft decoder I have?

Inspector.Gadget

23rd November 2010, 02:50

asarian: Open a command prompt in the folder where eac3to resides and run eac3to -test and wait a second as the output is printed to the console.

asarian

23rd November 2010, 03:04

asarian: Open a command prompt in the folder where eac3to resides and run eac3to -test and wait a second as the output is printed to the console.

Thanks. :)

Hmm, seems I have 1.1.0.7. So, do I need to downgrade if I want 7.1 DTS-MA (correctly) converted to 5.1 LPCM?

Uxorious

23rd November 2010, 03:31

Hmm, seems I have 1.1.0.7. So, do I need to downgrade if I want 7.1 DTS-MA (correctly) converted to 5.1 LPCM?

No. The only problem mode is 6.1

sreemv

23rd November 2010, 04:02

Could someone help me with the following error - i demuxed the dts track from dvd using eac3to but converting to 24fps appears to be a challenge.

N:\RipBot264v1.16.3\Tools\eac3to>eac3to dvd-demuxed.dts eac3to-24fps.dts -29.970 -changeTo24.000 -r8brain
DTS, 5.1 channels, 2:06:02, 24 bits, 1510kbps, 48kHz
Decoding with ArcSoft DTS Decoder...
Changing FPS from 29.970 to 24.000...
Reducing depth from 64 to 32 bits...
Writing WAVs...
Creating file "eac3to-24fps.R.wav"...
Creating file "eac3to-24fps.LFE.wav"...
Creating file "eac3to-24fps.L.wav"...
Creating file "eac3to-24fps.C.wav"...
Creating file "eac3to-24fps.SR.wav"...
Creating file "eac3to-24fps.SL.wav"...
Clipping detected, a 2nd pass will be necessary.
The ArcSoft DTS Decoder reported an error while decoding.
Aborted at file position 290717696.

N:\RipBot264v1.16.3\Tools\eac3to>

Thanks in advance.

asarian

23rd November 2010, 04:02

No. The only problem mode is 6.1

Thanks. I downgraded to 1.1.0.0 anyway, to be safe (I just have the pertinent .ax stuff installed; not the entire bloatware). If I read the reports in this thread properly, there's nothing 1.1.0.7 can do that 1.1.0.0 won't do as well.

tebasuna51

23rd November 2010, 04:41

Could someone help me with the following error - i demuxed the dts track from dvd using eac3to but converting to 24fps appears to be a challenge.
...
Maybe the DTS is corrupt, try Delaycut to fix it or other method to extract.
Try also with -libav

BTW, your parameters are incorrect, to convert a DTS from DVD NTSC you need:
eac3to dvd-demuxed.dts eac3to-24fps.dts -23.976 -changeTo24.000

sreemv

23rd November 2010, 04:48

Maybe the DTS is corrupt, try Delaycut to fix it or other method to extract.
Try also with -libav

BTW, your parameters are incorrect, to convert a DTS from DVD NTSC you need:
eac3to dvd-demuxed.dts eac3to-24fps.dts -23.976 -changeTo24.000

Thanks for the quick reply - you bring up my other dilemma. How do I find the source frame rate for sure? I got the 29.970 from doing "mediainfo" on the source VOBs. Please correct me if I am wrong. Thanks!

Inspector.Gadget

23rd November 2010, 06:06

i demuxed the dts track from dvd using eac3to but converting to 24fps appears to be a challenge

To add on to what was already said, if your target format is in fact 24000/1001 fps (~23.976fps commonly called 24p) and not 24000/1000 fps (24.000fps, perhaps instead should be called the "real" 24p), you may be doing yourself no favors.

Is your source an NTSC DVD (apparently so, given VOB files and 30000/1001 fps or ~29.97fps video) and your target either a file with inverse telecined (to ~23.976fps) video or a Blu-ray disc? If so, no audio framerate conversion is necessary!

In short, only do the contemplated operation (and flag the input as 23.976fps!) if you're sure that you need 24.000fps video.

sreemv

23rd November 2010, 06:33

To add on to what was already said, if your target format is in fact 24000/1001 fps (~23.976fps commonly called 24p) and not 24000/1000 fps (24.000fps, perhaps instead should be called the "real" 24p), you may be doing yourself no favors.

Is your source an NTSC DVD (apparently so, given VOB files and 30000/1001 fps or ~29.97fps video) and your target either a file with inverse telecined (to ~23.976fps) video or a Blu-ray disc? If so, no audio framerate conversion is necessary!

In short, only do the contemplated operation (and flag the input as 23.976fps!) if you're sure that you need 24.000fps video.

Sorry, not sure I understand everything you said - sounds too technical for me :)

Let me summarize what I am attempting to do in layman's [my] terms - I demuxed dts audio track from a dvd [supposedly NTSC] and now attempting to re-mux it with a mkv created from BluRay source.

When I muxed the dvd demuxed DTS to the mkv, I experienced gradual audio async - so more 'google' suggested I need to correct for the framerate diff?

If you have any advice helping me accomplish this task, I would highly appreciate it.

robpdotcom

23rd November 2010, 08:05

Is your source an NTSC DVD (apparently so, given VOB files and 30000/1001 fps or ~29.97fps video) and your target either a file with inverse telecined (to ~23.976fps) video or a Blu-ray disc? If so, no audio framerate conversion is necessary!

I didn't know it was possible to inverse telecine with eac3to. Is the result a progressive, 23.976fps file? And, would you mind giving me an example command line?

tebasuna51

23rd November 2010, 11:59

Let me summarize what I am attempting to do in layman's [my] terms - I demuxed dts audio track from a dvd [supposedly NTSC] and now attempting to re-mux it with a mkv created from BluRay source.

When I muxed the dvd demuxed DTS to the mkv, I experienced gradual audio async - so more 'google' suggested I need to correct for the framerate diff?
1) You can recode dts to ac3 (recommended). If you want recode to dts you need Surcode DVD (commercial) installed.
Check if you have Surcode with:
eac3to -test

2) Check with MediaInfo/eac3to if your mkv is exactly 24 fps and not 24000/1001 or 23.976

3) If your dts source is 768 Kb/s maybe you can add a explicit bitrate instead the default:
-448 to recode to ac3
-768 to recode to dts

4-1) If your DVD is PAL (check in the cover) you need:
eac3to input.dts output.ac3 -25.000 -changeTo24.000

4-2) If your DVD is NTSC you need:
eac3to input.dts output.ac3 -23.976 -changeTo24.000

You need correct the audio duration, forget the video fps.

Thunderbolt8

23rd November 2010, 15:42

^that could very well be a possibility, but seeing how madshi has left this and his madVR thread in the dark, I don't know if there will ever be a next version to eac3to.I guess its just a temporary thing. already happened I think twice before that there was a rather long break in between development. maybe he just needs some time off then and when. but so far, he has always come back.

Inspector.Gadget

23rd November 2010, 16:27

I didn't know it was possible to inverse telecine with eac3to. Is the result a progressive, 23.976fps file? And, would you mind giving me an example command line?

I don't believe you can do that except with soft-telecined VC-1 content, where eac3to discards the pulldown flags. I was thinking about the usual Avisynth route for the video, which the OP indicated is coming from an NTSC DVD.

Inspector.Gadget

23rd November 2010, 16:33

tempting to do in layman's [my] terms - I demuxed dts audio track from a dvd [supposedly NTSC] and now attempting to re-mux it with a mkv created from BluRay source.

When I muxed the dvd demuxed DTS to the mkv, I experienced gradual audio async - so more 'google' suggested I need to correct for the framerate diff?

If you experience gradual audio desync, then you are probably correct - that could reflect the effects of combining audio synced to a 23.976fps master with video that plays back at 24.000fps. Both 23.976fps and 24.000fps are legal for Blu-rays. However, there is a way to solve your desync problem without re-encoding. Simply flag the framerate of your video stream as 23.976fps in MKVMergeGUI when muxing your final file and if the DVD and the Blu-ray were made from the same master and have the same number of frames the audio sync should be perfect.

Caveat: If your DTS file had a delay value on the DVD, you will need to add that delay to the final Matroska file, again using MKVMergeGUI, to get the correct sync. Otherwise, your final file will have some small amount of constant desync as if you were watching the DVD without the correct audio delay added to the disc.

robpdotcom

23rd November 2010, 18:10

I don't believe you can do that except with soft-telecined VC-1 content, where eac3to discards the pulldown flags. I was thinking about the usual Avisynth route for the video, which the OP indicated is coming from an NTSC DVD.

:thanks: Thanks for the info.

sreemv

23rd November 2010, 21:44

1) You can recode dts to ac3 (recommended). If you want recode to dts you need Surcode DVD (commercial) installed.
Check if you have Surcode with:
eac3to -test

eac3to -test clearly shows surcode and it works by itself as well as I have used it at various times in the past.

N:\RipBot264v1.16.3\Tools\eac3to>eac3to -test
eac3to (v3.24) is up to date
Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed
http://www.nero.com/eng/store-blu-ray.html
CAUTION: You need Nero 7. Nero 8 won't work with eac3to.
ArcSoft DTS Decoder (1.1.0.0) works fine
Sonic Audio Decoder (3.24.0.0) doesn't seem to be installed
Haali Matroska Muxer (2010-05-19) is installed
There's a new version (2010-11-21) available
http://haali.net/mkv
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 (3.4.0.0, release version) is installed
There's a new release version (4.4.0.0) available
http://www.bunkus.org/videotools/mkvtoolnix
There's a new beta version (4.4.0.0, 2010-11-15) available
http://www.bunkus.org/videotools/mkvtoolnix/win32/pre

N:\RipBot264v1.16.3\Tools\eac3to>

But surcode with eac3to keeps failing and I am unable to decipher the reason:
N:\RipBot264v1.16.3\Tools\eac3to>eac3to "eac3to-demux.dts" N:\24fps.dts -23.976 -changeTo24.000 -libav
DTS, 5.1 channels, 2:06:02, 24 bits, 1510kbps, 48kHz
Decoding with libav/ffmpeg...
Changing FPS from 23.976 to 24.000...
Reducing depth from 64 to 32 bits...
Writing WAVs...
Creating file "N:\24fps.LFE.wav"...
Creating file "N:\24fps.SR.wav"...
Creating file "N:\24fps.SL.wav"...
Creating file "N:\24fps.C.wav"...
Creating file "N:\24fps.R.wav"...
Creating file "N:\24fps.L.wav"...
Clipping detected, a 2nd pass will be necessary.
libav ERROR: block code look-up failed
libav L
libav a
libav s
libav t
................................................
...............................................[truncated]
libav Didn't get subframe DSYNC
libav ERROR: block code look-up failed

[truncated]

libav Didn't get subframe DSYNC
Starting 2nd pass...
Decoding with libav/ffmpeg...
Changing FPS from 23.976 to 24.000...
Reducing depth from 64 to 32 bits...
Writing WAVs...
Applying -12.53dB gain...
Creating file "N:\24fps.R.wav"...
Creating file "N:\24fps.C.wav"...
Creating file "N:\24fps.SL.wav"...
Creating file "N:\24fps.L.wav"...
Creating file "N:\24fps.SR.wav"...
Creating file "N:\24fps.LFE.wav"...
libav ERROR: block code look-up failed
libav L
libav a
libav s
libav t
libav m

[truncated]

libav Didn't get subframe DSYNC
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.29.0.
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

N:\RipBot264v1.16.3\Tools\eac3to>

I checked the FAQ and amde sure there were no long path/names and yet surcode part keeps failing. As I mentioned above, surcode by itself works fine.

Is there a way to save the frame converted "wavs" and use the "surcode" independently? eac3to keeps deleting them after the above failure.

BTW, -libav has more success than the default - so, thanks for that!

2) Check with MediaInfo/eac3to if your mkv is exactly 24 fps and not 24000/1001 or 23.976

N:\RipBot264v1.16.3\Tools\eac3to>eac3to "original.mkv"
MKV, 1 video track, 1 subtitle track, 2:11:21, 24p /1.001
1: h264/AVC, 1920x818 24p /1.001 (960:409)
2: Subtitle (SRT)

N:\RipBot264v1.16.3\Tools\eac3to>

3) If your dts source is 768 Kb/s maybe you can add a explicit bitrate instead the default:
-448 to recode to ac3
-768 to recode to dts

N:\RipBot264v1.16.3\Tools\eac3to>eac3to "eac3to-demux.dts"
DTS, 5.1 channels, 2:06:02, 24 bits, 1510kbps, 48kHz

N:\RipBot264v1.16.3\Tools\eac3to>

4-1) If your DVD is PAL (check in the cover) you need:
eac3to input.dts output.ac3 -25.000 -changeTo24.000

4-2) If your DVD is NTSC you need:
eac3to input.dts output.ac3 -23.976 -changeTo24.000

You need correct the audio duration, forget the video fps.

N:\RipBot264v1.16.3\Tools\eac3to>eac3to "F:\Original\VIDEO_TS\VTS_01_1.VOB"
VOB, 1 video track, 2 audio tracks, 1 subtitle track, 0:01:53
1: MPEG2, 480i60 /1.001 (16:9)
2: AC3, 5.1 channels, 448kbps, 48kHz
3: DTS, 5.1 channels, 24 bits, 1510kbps, 48kHz
4: Subtitle (DVD)

N:\RipBot264v1.16.3\Tools\eac3to>

Thanks again for all the help!

tebasuna51

24th November 2010, 01:13

...
libav ERROR: block code look-up failed
...
Applying -12.53dB gain...
...

Your extracted dts seems damaged.
Try fix it with DelayCut (http://madshi.net/delaycut.rar).
Put here the log if there are errors.

Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.29.0.
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

Is there a way to save the frame converted "wavs" and use the "surcode" independently? eac3to keeps deleting them after the above failure.

Yes, use:
eac3to input.dts output.wavs -23.976 -changeTo24.000

eac3to "original.mkv"
MKV, 1 video track, 1 subtitle track, 2:11:21, 24p /1.001
1: h264/AVC, 1920x818 24p /1.001 (960:409)
2: Subtitle (SRT)
Your mkv is 23.976, in theory you don't need change the audio duration.

sreemv

24th November 2010, 07:54

Your extracted dts seems damaged.
Try fix it with DelayCut (http://madshi.net/delaycut.rar).
Put here the log if there are errors.

I dont see any option in "delaycut" [v1.2.12.] for verifying/fixing - the "crc errors" section is grayed out once I load the "dts" as the "input file" - I must be missing something?

Yes, use:
eac3to input.dts output.wavs -23.976 -changeTo24.000

Doh! I should have guessed that - very similar to what I was doing with LPCM-to-DTS - thanks :)

Your mkv is 23.976, in theory you don't need change the audio duration.

Hmmm....this is frustrating - at the beginning of the movie, audio appears lagging and then few minutes in the audio goes ahead without me changing anything...*scratch-head*

Thanks again for all your help!

sreemv

24th November 2010, 08:00

Your extracted dts seems damaged.
Try fix it with DelayCut (http://madshi.net/delaycut.rar).
Put here the log if there are errors.

Doh! Looks like delaycut appears to by default run a fixer algorithm? Didnt see any errors:

====== INPUT FILE INFO ========================
File is dts
Bitrate (kbit/s) 1536
Act rate (kbit/s) 1509.750
File size (bytes) 1427019726
Channels mode C+L+R+SL+SR
Sampling Frec 48000
Low Frec Effects LFE: Present
Duration 02:06:01.621
Frame length (ms) 10.666667
Frames/second 93.750000
Num of frames 708902
Bytes per Frame 2013.0000
Size % Framesize 0
CRC present: NO
=============================================
====== TARGET FILE INFO ======================
Start Frame 0
End Frame 708901
Num of Frames 708902
Duration 02:06:01.621
NotFixedDelay 0.0000
=============================================
====== PROCESSING LOG ======================
Number of written frames = 708902
Number of Errors= 0

acki

24th November 2010, 11:12

Quick question. If you want to convert audio from a 25fps sample to a 23.976 sample... does it make any difference if you use:

-slowdown convert 25.000 and 24.000 content to 23.976 fps

or either

-23.976/... define source fps to be "23.976", "24.000", "25.000", ...
-changeTo24.000 change source fps to "23.976", "24.000", "25.000", ...

Being in this last case -25.000 -changeTo23.976 ?

Which one should be more apropriate ?

:thanks:

tebasuna51

24th November 2010, 12:21

...
Hmmm....this is frustrating - at the beginning of the movie, audio appears lagging and then few minutes in the audio goes ahead without me changing anything...*scratch-head*
You have:
MKV, 1 video track, 1 subtitle track, 2:11:21, 24p /1.001
DTS, 5.1 channels, 2:06:02, 24 bits, 1510kbps, 48kHz

If you are sure than mkv and dvd movie is the same version (can be normal, extended, director's cut, ...), with same title credits also, the duration suggest you need a a more strong conversion like 25 -> 23.976 :

MKV: 7881 seconds
DTS: 7562 seconds
7562 x 25 / (24/1.001) = 7884

tebasuna51

24th November 2010, 12:25

...
-slowdown convert 25.000 and 24.000 content to 23.976 fps
...
Being in this last case -25.000 -changeTo23.976 ?

Absolutely equivalent.

acki

24th November 2010, 14:52

Absolutely equivalent.

OK, Gracias ;) Thanks for the info.:thanks:

sreemv

25th November 2010, 01:42

You have:
MKV, 1 video track, 1 subtitle track, 2:11:21, 24p /1.001
DTS, 5.1 channels, 2:06:02, 24 bits, 1510kbps, 48kHz

If you are sure than mkv and dvd movie is the same version (can be normal, extended, director's cut, ...), with same title credits also, the duration suggest you need a a more strong conversion like 25 -> 23.976 :

MKV: 7881 seconds
DTS: 7562 seconds
7562 x 25 / (24/1.001) = 7884

Absolutely positvely same movie - no special director cut or extended - just that they are sloppy during credits at the beginnig and end of the movies, there is no audio for few seconds.

Can you suggest a way to find the exact timecodes for where the audio starts and end on the dvd?

sreemv

25th November 2010, 03:35

I may have found the root of my problem - notice below different vob files appear to have different audio lags? Also - whats the duration printed at the end of the first line for each vob? what does that represent - please help! THANKS for all the expert advice!

N:\RipBot264v1.16.3\Tools\eac3to>eac3to N:\Original\VIDEO_TS\VTS_01_1.VOB
VOB, 1 video track, 1 audio track, 0:02:55
1: MPEG2, 480i60 /1.001 (16:9)
2: DTS, 5.1 channels, 24 bits, 1510kbps, 48kHz

N:\RipBot264v1.16.3\Tools\eac3to>eac3to N:\Original\VIDEO_TS\VTS_01_2.VOB
MPG, 1 video track, 1 audio track, 0:00:33
1: MPEG2, 480i60 /1.001 (16:9)
2: DTS, 5.1 channels, 24 bits, 1510kbps, 48kHz, -343ms

N:\RipBot264v1.16.3\Tools\eac3to>eac3to N:\Original\VIDEO_TS\VTS_01_3.VOB
MPG, 1 video track, 1 audio track, 0:01:55
1: MPEG2, 480i60 /1.001 (16:9)
2: DTS, 5.1 channels, 24 bits, 1510kbps, 48kHz, -376ms

N:\RipBot264v1.16.3\Tools\eac3to>eac3to N:\Original\VIDEO_TS\VTS_01_4.VOB
MPG, 1 video track, 1 audio track, 13:14:56
1: MPEG2, 480i60 /1.001 (16:9)
2: DTS, 5.1 channels, 24 bits, 1510kbps, 48kHz, -378ms

N:\RipBot264v1.16.3\Tools\eac3to>eac3to N:\Original\VIDEO_TS\VTS_01_5.VOB
MPG, 1 video track, 1 audio track, 13:13:51
1: MPEG2, 480i60 /1.001 (16:9)
2: DTS, 5.1 channels, 24 bits, 1510kbps, 48kHz, -482ms

N:\RipBot264v1.16.3\Tools\eac3to>eac3to N:\Original\VIDEO_TS\VTS_01_6.VOB
MPG, 1 video track, 1 audio track, 0:07:43
1: MPEG2, 480i60 /1.001 (16:9)
2: DTS, 5.1 channels, 24 bits, 1510kbps, 48kHz, -327ms

N:\RipBot264v1.16.3\Tools\eac3to>eac3to N:\Original\VIDEO_TS\VTS_01_7.VOB
MPG, 1 video track, 1 audio track, 13:13:10
1: MPEG2, 480i60 /1.001 (16:9)
2: DTS, 5.1 channels, 24 bits, 1510kbps, 48kHz, -236ms

N:\RipBot264v1.16.3\Tools\eac3to>

Do I need to account for different delays for different vobs? Or am I barking up the wrong tree?

tebasuna51

25th November 2010, 03:41

Use:
eac3to N:\Original\VIDEO_TS\VTS_01_1.VOB+...+N:\Original\VIDEO_TS\VTS_01_7.VOB

Inspector.Gadget

25th November 2010, 03:58

If things are still out of sync: Try demuxing the entire title with DGIndex to see if eac3to gives you the same output (e.g., load the VOBs 1-7 in order into DGIndex, make sure "demux all streams" is selected for audio, and run it. It'll print a delay value to the file name of the outputted DTS file.

There is another potential problem with audio from a DVD:

So far as I know, neither DGIndex nor eac3to can automatically correct a series of VOBs that either contains more than one title (a trailer at the beginning, etc.) or a blank cell in VOB 1 which consists of a few seconds of black screen and no audio. Either one will wreck delay values for the whole movie. The thing to do if you have a video-only cell near the beginning is to set DGIndex's first marker to the first frame of actual content (first frame that isn't totally black) and demux from there; that will correct the audio delay so it starts where that frame does, except for limitations built into the audio codec which may leave you with a remaining value of some milliseconds (again printed to the file name of the output audio).

The way to handle a multi-title VOB set is to run PGCDemux against a complete decrypted copy of the original DVD and make sure it runs in the PGC and Title domain (two select boxes). Your output will be only the movie, and you can pass that to DGIndex or eac3to without further problems.

tebasuna51

25th November 2010, 11:49

...
There is another potential problem with audio from a DVD:

So far as I know, neither DGIndex nor eac3to can automatically correct a series of VOBs that either contains more than one title (a trailer at the beginning, etc.)
I agree. Of course you need extract from DVD only the Title for main movie with DVDDecrypter or PGCDemux or other tool.

or a blank cell in VOB 1 which consists of a few seconds of black screen and no audio...
About this problem I always have a doubt.
The workaround explained for DGIndex is clear: the delay printed in audio extracted file is related to the .d2v video index created.
But I'm not sure how eac3to manage this problem.

Anyway you can't know if you need a different initial delay to match BD video and must be verified always.

But the problem with delays between VOB files (overlap) is managed correctly (I think) by both tools.

alex_t

26th November 2010, 12:39

Hi everybody and madshi.

First, BIG THANKS madshi for this software !

I would like to report here a bug (I think it is but I'm not sure). Revision is 3.24.

I wanted to convert a Blu-ray 23.976 in 25.000 FPS. The command was as below:

eact3to blu-ray_directory 1) 2: k:\movie.mkv 3: k:\audio.flac 13: k:\subtitle.sup -speedup

Conversion for audio was : DTS HD MA to FLAC with dts arcsoft decoder

During the 1st pass, clipping has been detected by eac3to and a 2nd pass has been done.

During the 1st pass: a message said 'convert 23.976 to 25.000'

But at the end of 2nd pass, a message said 'fps at 23.976 has been added in MKV header' (something like this)

Result was that the FPS remained at 23.976. I use MPC HC as player.

I would like to change 23.976 FPS into 25.000 FPS with eac3to. May be I did not use the good -option (I used -speedup)

What is the difference between -speedup and -changeTo25.000 ?

Thanks

Regards
(sorry for my english)

nurbs

26th November 2010, 15:21

I haven't used speedup myself, but normally you have to put the options next to the track you want to use them on, e.g.
eac3to blu-ray_directory 1) 2: k:\movie.mkv -speedup 3: k:\audio.flac -speedup 13: k:\subtitle.sup -speedup (I don't know if this actually works for subtitles)

May I ask why you want to convert the framerate?
It's usually more trouble than it's worth and converting from 23.976 to 25 is almost never necessary.

TinTime

26th November 2010, 16:17

But the problem with delays between VOB files (overlap) is managed correctly (I think) by both tools.

I've had occasional problems with eac3to and DVDs ("A Good Year" was one) so I always use DGIndex for demuxing DVDs.

Inspector.Gadget

26th November 2010, 18:11

Hi everybody and madshi.

First, BIG THANKS madshi for this software !

I would like to report here a bug (I think it is but I'm not sure). Revision is 3.24.

...

You have to do audio and video speed-up separately, because eac3to doesn't speed up video or subtitles. One way to accomplish this is to use eac3to as usual for audio speed-up.

For video, you have two methods: If you're re-encoding anyway, just add AssumeFPS(25, 1, false) to your Avisynth script. If you're not going to re-encode, mux your final MKV (original video + sped-up audio) with MKVMergeGUI, and make sure to set the video framerate to be 25.

For subtitles, you can OCR Blu-ray SUP to SRT with Suprip and then convert the framerate of the output SRT from 23.976 to 25.00fps using Subrip.

tebasuna51

26th November 2010, 20:53

You have to do audio and video speed-up separately, because eac3to doesn't speed up video ...

For video,... mux your final MKV (original video + sped-up audio) with MKVMergeGUI, and make sure to set the video framerate to be 25.
This work also:
eac3to blu-ray_directory 1) 2: k:\movie.mkv -speedup 3: k:\audio.flac -speedup

The movie.mkv is already 25 fps

Mtz

29th November 2010, 06:02

I know 3 modes for PAL > NTSC (25>23.976) audio conversion (slowdown) used in 2 programs. Can you recommend some audio conversion for AC3 > AC3?

1. eac3to > slowdown (or another setting?)

2. behappy > Timestretch with which options of these:
- Rate, tempo, and no pitch correction
- Pitch changed preserving tempo
- Tempo changed, pitch correction

3. behappy > Resample SSRC - Slowdown 25 -> 23.976
?

enjoy,
Mtz

sreemv

29th November 2010, 09:04

The way to handle a multi-title VOB set is to run PGCDemux against a complete decrypted copy of the original DVD and make sure it runs in the PGC and Title domain (two select boxes). Your output will be only the movie, and you can pass that to DGIndex or eac3to without further problems.

I did exactly what you suggested - used PGCDemux [by PGC and Titles domain] and got a clean DTS - tried it as is [23.976] and a sped up version [24fps] - still no sync. scratching head.

But, shouldnt I be setting the source to "29.970" for NTSC? Thats what it shows in mediainfo as frame reate for the video.

On an another note, any advice on how I might compare 2 audio [ac3, DTS] tracks side-by-side with timecodes?

tebasuna51

29th November 2010, 12:56

I know 3 modes for PAL > NTSC (25>23.976) audio conversion (slowdown) used in 2 programs. Can you recommend some audio conversion for AC3 > AC3?

A conversion implies 3 steps:

a) Decode audio:

a.1) eac3to can use Nero 7 decoder (certified but there are some issues reported about cancel DialNorm) or sonic (applies DRC always, not recommended) or free decoder libav.

a.2) BeHappy can use NicAudio or DirectShow decoder. I can't know your DirectShow decoder and how is configured.

After my test I prefer NicAudio but maybe is conditioned by parental love.

There are also other commercial certified Dolby Digital audio decoders, take care with DN and DRC.

b) Tempo conversion.

b.1) eac3to can use SSRC routines, by default, but also r8brain.

b.2) BeHappy can use SSRC routines or SoundTouch (TimeStretch).

SSRC, r8brain and TimeStretch-rate change the tempo and the pitch. If your audio is from a PAL-DVD maybe the original pitch is recovered, then for this source SSRC is recommended (maybe the SSRC routines in eac3to are more updated than included in AviSynth).

For other sources with correct pitch (PAL cameras for instance) maybe you want preserve the pitch and use TimeStrech-tempo.

There are also many Audio Editors (Audacity, Audition, SoundForge, ...) that can do the job with many options.
Maybe Prosoniq TimeFactory is a good option between the commercial soft.

c) Encode audio:
Both, eac3to and BeHappy, use Aften to encode AC3. The only difference can be the versión used. Maybe the best quality using Aften is with the parameters:
-b 640 -readtoeof 1 -exps 32 -s 1
You can use the last Aften.exe version with BeHappy and with eac3to with something like:

eac3to input stdout.wav -slowdown | Aften -b 640 -readtoeof 1 -exps 32 -s 1 - output.ac3

Of course you can use any commercial certified Dolby Digital audio encoder.

Make your choice.

tebasuna51

29th November 2010, 13:23

...
But, shouldnt I be setting the source to "29.970" for NTSC? Thats what it shows in mediainfo as frame reate for the video.

One more time, the video frame rate isn't related with audio.
Try yourself.

sreemv

29th November 2010, 17:40

One more time, the video frame rate isn't related with audio.
Try yourself.

Sorry for being a newbie but I was under the impression that I am having to compensate/stretch the audio to match the video frame rate difference[29.970 vs 23.976/24]? No?

TinTime

29th November 2010, 18:44

Sorry for being a newbie but I was under the impression that I am having to compensate/stretch the audio to match the video frame rate difference[29.970 vs 23.976/24]? No?

No. If the video framerate is changed by adding/dropping frames/fields then the duration is the same and the audio requires no adjustment.

Forteen88

30th November 2010, 10:21

How come, when I convert the audio from VOB-files (AC3) to AAC, I sometimes get this type of message: "a03 [NeroAacEnc] Processed 7260 second. Aborted at file position 4153661440.", but sometimes the ~third try it works OK?

mrr19121970

30th November 2010, 10:26

Forteen88

30th November 2010, 10:36

the 1st 2 times you get a read error, the 3rd you didn't.That never happens when I use MKVMerge, so I doubt that the harddrive is bad. Does eac3to got better errorcheck than MKVMerge, or what?
Also, when I convert to FLAC, I never got that problem.

EDIT: @MikeEby I run it on a copy (ripped) that is already on the harddrive. But as said, it seems to be an issue that has not been properly fixed yet.

MikeEby

30th November 2010, 16:50

That never happens when I use MKVMerge, so I doubt that the harddrive is bad. Does eac3to got better errorcheck than MKVMerge, or what?
Also, when I convert to FLAC, I never got that problem.

You get this error when your ripping directly off the Blu-ray or when the copy of the disk is already on the hard drive? I've never seen the error when AnyDVD has made a copy to hard drive. I have seen it when ripping directly off the Blu-Ray disk.

Mike

Snowknight26

30th November 2010, 18:37

It's an issue with the way NeroAacEnc and eac3to work together. Ever since I reported the issue, madshi tried to fix, but by doing so caused the same issue to appear under different circumstances. The only sure fire to not run into the issue is to encode individual demuxed tracks to AAC.

IanD

2nd December 2010, 15:39

shogo_kawada

2nd December 2010, 17:45

I strongly suggest, if working with HD-DVD, to become familiar with the basics of editing .XPL playlists as an additional tool: that way you have exact control over what is happening, instead of relying on software you have little control over.
Thanks for your reply, but I already solved my Blade Runner situation using the process described in this (http://forum.doom9.org/showthread.php?p=1459110#post1459110)post. I also managed to add the italian soundtrack (taken from a cheap dvd boxset), slowed down with eac3to

jd213

3rd December 2010, 07:55

Hmm, all of the HD DVD to BD-R conversions I've done with thd+ac3 tracks encoded by eac3to give me an "Unknown Disc" error with my Oppo BDP-80, although they work fine in my PS3 and Japanese Sony BD recorder. Discs without thd+ac3 work fine in the Oppo.

I'll be sure to ask in a forum dedicated to Oppo players since it seems to be more of a player problem, but any ideas what might be causing this?

edit: nevermind, it looks like all of the TrueHD titles I re-authored coincidentally were all burned to LTH discs, which won't play on the BDP-80.

TDiTP_

4th December 2010, 14:46

May be bug?

1). eac3to 3.24 can't decode any MLP encoded in SurCode Dolby MLP Encoder 1.0.29. (sample (http://www.mediafire.com/download.php?m26rmm80dk9feyk))

eac3to v3.24
command line: eac3to ID20.mlp ID20.wav -libav
------------------------------------------------------------------------------
MLP, 5.1 channels, 16 bits, 48kHz
Decoding with libav/ffmpeg...
The libav decoder output an unexpected bitdepth (-1). <ERROR>
Aborted at file position 262144. <ERROR>

eac3to v3.24
command line: eac3to ID20.mlp ID20.nero.wav -nero
------------------------------------------------------------------------------
MLP, 5.1 channels, 16 bits, 48kHz
Disabling DRC for Nero (E-)AC3 decoding...
Decoding with DirectShow (Nero Audio Decoder 2)...
The DirectShow audio decoder didn't accept the input stream. <ERROR>
Aborted at file position 262144. <ERROR>

At the same time older versions of eac3to (for example 3.17) work perfectly (bit-in-bit). As well as http://sourceforge.net/projects/dvdadecoder/ .

2). ArcSoft DTS Decoder 1.1.0.0 and 1.1.0.7 with eac3to can't normal decode DTS(-HD) 2.1. LFE-channel is clear.
In the same time Sonic (4.3.0.169) works well.
Samples: DTS 2.1 (http://www.mediafire.com/?g2n295cd847w9x1), DTS-96/24 2.1 (http://www.mediafire.com/?87a1nso5jho362v), DTS-HD MA 2.1 (http://www.mediafire.com/?0hz1kpe8ibyqffp).

Inspector.Gadget

4th December 2010, 16:42

ArcSoft DTS Decoder 1.1.0.0 and 1.1.0.7 with eac3to can't normal decode DTS(-HD) 2.1. LFE-channel is clear.

Arrrgh, can't Arcsoft just make something that ALWAYS WORKS?

Thunderbolt8

5th December 2010, 00:47

whats worse is they dont even seem to be able or willing to fix it in updates. those problems must have been known to them since quite a while.

xkodi

5th December 2010, 01:36

whats worse is they dont even seem to be able or willing to fix it in updates. those problems must have been known to them since quite a while.

since Sonic works well for those DTS(-HD) 2.1 tracks then i don't see why we should care about Arcsoft, but it would be very handy if we have a chart giving information which DTS(-HD) with which decoder (Arcsoft, Sonic, etc) works good.

dansrfe

5th December 2010, 04:30

Yep, I can confirm the bug. Well this just sucks.

TDiTP_

5th December 2010, 12:02

AFAIK ArcSoft bugs are:

1). Any version can't ignore DN in DTS-HD MA and DTS-HD Hi-Res.
Solition: Sonic, but we can't use it with > 6.1
BTW reference decoder of DTS-HD StreamPlayer can't ignore dialog normalization too (that is why file-header cutting won't help us)

2). DTS-ES Discrete 6.1/6.0 (in eac3to terms: DTS-ES 6.1/6.0) and DTS-HD with this core can be decoded with only ArcSoft 1.1.0.0. Other versions do not work.

3). DTS(-HD) 1.0 can be decoded with only ArcSoft 1.1.0.0 and 1.1.0.8.

4). Any version can't decode DTS-HD MA 7.1 ("strange setup").
Solution: if we have DTS-HD StreamPlayer we can cut file-header (first 140 byte) of similar stream obtained in DTS-HD Master Audio Suite and then merge it with our DTS-HD MA 7.1. This new track can be decoded in StreamPlayer.

5). Any version can't decode DTS(-HD) 2.1.
Solution: Sonic, Nero (libav gives too low level of LFE-channel)

Frogger13

5th December 2010, 17:43

Well beside being very time consuming this does not solve the problem at all as I had one time to restart single file encoding about 10 times until it finally succeded. In EAC3to v3.22 This error never happened to me and still doesn't (I reverted to that version since 3.24 is horrible)!

So if things where buggy then they where still way better than now.

Put simply: EAC3to v3.24 is useless for AC3 to AAC encoding and you must revert to v3.22!

It's an issue with the way NeroAacEnc and eac3to work together. Ever since I reported the issue, madshi tried to fix, but by doing so caused the same issue to appear under different circumstances. The only sure fire to not run into the issue is to encode individual demuxed tracks to AAC.

PS: I did never Encode from a muxed file. I always encode from extracted DTS or AC3 tracks. Both of them cause the Error, the only difference being that the error happens more often when more than 1 file is encoded at a time, but it still happens when encoding 1 file at a time.

Thunderbolt8

6th December 2010, 03:43

4). Any version can't decode DTS-HD MA 7.1 ("strange setup").
Solution: if we have DTS-HD StreamPlayer we can cut file-header (first 140 byte) of similar stream obtained in DTS-HD Master Audio Suite and then merge it with our DTS-HD MA 7.1. This new track can be decoded in StreamPlayer.
but whats the channel matrix then afterwards for that 7.1 track, is it that one of a strange setup track or of a normal 7.1 one? (they are slightly different)

tebasuna51

6th December 2010, 13:01

AFAIK ArcSoft bugs are:
...

4). Any version can't decode DTS-HD MA 7.1 ("strange setup").
Solution: if we have DTS-HD StreamPlayer we can cut file-header (first 140 byte) of similar stream obtained in DTS-HD Master Audio Suite and then merge it with our DTS-HD MA 7.1. This new track can be decoded in StreamPlayer.
I don't think so.
I think ArcSoft decode correctly DTS-HD MA 7.1 ("strange setup"), and search other "solution" is useless.

Let me explain the problem:

When decode standard 7.1 we can obtain a exact correlation between DTS speakers and WAV speakers:
Lss -> Side Left
Lsr -> Back Left
(the same for Right speakers)

But with "strange setup" we have:
Ls -> ?
Lsr -> Back Left
Don't exist, in WAV specs, a channel equivalent to Ls, we have only
SPEAKER_BACK_LEFT
SPEAKER_BACK_RIGHT
SPEAKER_BACK_CENTER
SPEAKER_SIDE_LEFT
SPEAKER_SIDE_RIGHT
(and others not related)

Then ArcSoft decode:
Side_Left = 0.6007 x Ls
Back_Left = 0.6804 x Lsr + 0.3196 x Ls
the Ls channel is separated in two components with this conditions:

0.6804 + 0.3196 = 1
to avoid overflow, and

(0.6007)^2 + (0.3196)^2 = (0.6804)^2
to preserve the audio power contribution of Ls.

All other channels must be attenuated also like:
Front_Left = 0.6804 x L
...
to preserve the volume balance.

And don't exist a better solution to the problem.
For what you want extract the exact Ls channel?
You can't convert it to other audio format, and you can't play it ,unless you change the speakers position physically in your audio room each time you want play a "strange setup".

When you play a DTS "strange setup" in your receiver it stop the play and send a message like "Please, change the Side speakers to Ls-Rs possitions and press Play to continue"?

I think not, or make a mix like ArcSoft do or you play incorrectly the audio.

TDiTP_

6th December 2010, 14:14

tebasuna51
Great explanation. Thanks.

For what you want extract the exact Ls channel?
Typical (but unusual) situation is decode "strange setup", then mix C-channel with translating voice (AFAIK it's popular in my country only :)) and then encode in the same scheme "strange setup".
Now I think that can use ArcSoft.

Joniii

6th December 2010, 20:41

I demuxed 2 BD's of The Aviator to use audio from import BD to Region A BD. German import has english DTS-HD Master Audio and I would like to use that with Region A BD with VC-1 because main track in it is AC3. Only problem is that the audio is not completely in synch.

I was thinking of converting DTS core and AC3 to wav with something like Wavelab and check the graph to see where the audio starts and add the delay when remuxing but that propably isn't the easiest way plus wavelab isn't freeware. Anyone have any ideas?

dansrfe

6th December 2010, 21:24

sshd

6th December 2010, 23:54

Small bug: When converting 24p mkv to 23.976

eac3to.exe file.mkv 1: vid.mkv 2: aud.flac -slowdown (works)
eac3to.exe file.mkv 1: vid.mkv 2: aud.flac 3: subs.srt -slowdown (-slowdown ignored)

tebasuna51

7th December 2010, 00:06

eac3to.exe file.mkv 1: vid.mkv 2: aud.flac -slowdown (works)
Only with flac, maybe you need:
eac3to.exe file.mkv 1: vid.mkv -slowdown 2: aud.flac -slowdown
eac3to.exe file.mkv 1: vid.mkv 2: aud.flac 3: subs.srt -slowdown (-slowdown ignored)
Yep, -slowdown don't work with subs.

Joniii

7th December 2010, 07:22

I don't quite understand why you you are bringing the Region A BD into the mix when your imported German disc has English DTS-HD Master Audio and a 1080p HD picture. Imported BD's, especially in AVC, are BETTER than VC-1 Region A BD's. Authoring/encoding houses in Germany actually know how to author/encode.

I thought of that too since it's AVC and has a high bitrate compared to VC-1 but I don't really understand the source of these german releases, I mean do they have the same quality masters? Is there a program that opens both AVC and VC-1 similar to VirtualDub so that I can go to specific frame and take a screenshots to compare?

dansrfe

7th December 2010, 09:33

I do believe The Aviator has the same master used for the Region A & B Discs. easiest way to compare is just open the file in vlc and take screenshots. another way is to index both .m2ts files with ffms2 and then do a stackhorizontal for side-by-side analysis of the same frame. But personally I would go with the AVC version with the higher bitrate.

Joniii

7th December 2010, 13:30

Joniii

7th December 2010, 13:36

Another question.

I'm trying to add AC3 track from Cars PAL DVD to Cars Region A BD. Everything goes fine except I think eac3to is remapping the channels incorrectly, I mean DVD ac3 should have the same channel order than in BD or am I wrong?

D:\eac3to>eac3to e:\c.ac3 d:\cf.ac3 -slowdown
AC3 EX, 5.1 channels, 1:51:44, 384kbps, 48kHz
The Nero decoder doesn't seem to work, will use libav instead.
Decoding with libav/ffmpeg...
Remapping channels...
Changing FPS from 25.000 to 23.976...
Encoding AC3 <640kbps> with libAften...
Creating file "d:\cf.ac3"...
Clipping detected, a 2nd pass will be necessary.
Starting 2nd pass...
Decoding with libav/ffmpeg...
Remapping channels...
Changing FPS from 25.000 to 23.976...
Encoding AC3 <640kbps> with libAften...
Applying -0,28dB gain...
Creating file "d:\cf.ac3"...
eac3to processing took 25 minutes, 40 seconds.
Done.

tebasuna51

7th December 2010, 14:59

...
Everything goes fine except I think eac3to is remapping the channels incorrectly, I mean DVD ac3 should have the same channel order than in BD or am I wrong?

Don't worry, the message:

Remapping channels...

is from ac3 internal order to uncompressed order, but the final ac3 have the same channel map than original.

Thunderbolt8

7th December 2010, 21:27

there has been a lot of checking done with different versions of the arcsoft decoder files. but are there maybe other versions of the sonic decoder which might give us more benefits?

xkodi

8th December 2010, 09:29

there has been a lot of checking done with different versions of the arcsoft decoder files. but are there maybe other versions of the sonic decoder which might give us more benefits?

i don't believe so, because i ran some tests before:

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

and also here (at the end of the post) is summary and conclusions from the tests:

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

in fact i tried recently even the latest Sonic version "4.3.0.238" and it can't work with eac3to at all.

xkodi

8th December 2010, 09:51

I think ArcSoft decode correctly DTS-HD MA 7.1 ("strange setup"), and search other "solution" is useless.

if that is the case then why we want to decode DTS-HD MA 5.1 in bit-perfect way (and both Sonic and Arcsoft do that), because if you look the channel layout diagrams here:

http://forum.doom9.org/showthread.php?p=1443085#post1443085

then it's obvious there is no match of the speaker positions between Mircosoft 5.1 WAV file and DTS-HD MA 5.1 - Microsoft 5.1 WAV files use only 0x3F as channel mask for 5.1 WAV files, which is FL, FR, FC, LFE, SL, SR.

so, maybe i'm wrong somehow, but please explain.

also, the same post gives information that DTS-HD MA 7.1 ("strange setup") tracks can be decoded with Sonic bit-perfectly to 5.1 channels that match DTS-HD MA 5.1 speaker positions, which is another thing that seems correct at least to me to be done.

P.S. and the same question(s) for DTS-HD MA 7.1 not-"strange setup" tracks based on DTS and Microsoft channel layout diagrams and respectively speaker positions.

[EDIT] BTW, in the mentioned old post, DTS channel layout diagrams are taken directly from DTS Labs software encoder software and Microsoft ones are from here:

http://download.microsoft.com/download/9/c/5/9c5b2167-8017-4bae-9fde-d599bac8184a/SpkrConfig5.docx

page 6 and 7, where is also explained how Microsoft 5.1 WAV is played back on 7.1 channel system in Windows.

TDiTP_

8th December 2010, 10:07

Then ArcSoft decode:
Side_Left = 0.6007 x Ls
Back_Left = 0.6804 x Lsr + 0.3196 x Ls

These coefficients must be function of the angle between C-speaker and Rs-speaker. May be anybody know angle? I can't find any information about it.
The angle between C-speaker and Rsr-speaker must be 150º, is not it?

tebasuna51

8th December 2010, 12:56

These coefficients must be function of the angle between C-speaker and Rs-speaker. May be anybody know angle? I can't find any information about it.
The angle between C-speaker and Rsr-speaker must be 150º, is not it?

Yes, must be function of the angle, but doesn't exist a common opinion about the exact angles between speakers:
http://en.wikipedia.org/wiki/File:Standard_7.1_surround_sound_speaker_placement.png

The same for 5.1 config, the Surround channels can be located between 110º and 135º.

tebasuna51

8th December 2010, 13:27

...
then it's obvious there is no match of the speaker positions between Mircosoft 5.1 WAV file and DTS-HD MA 5.1 - Microsoft 5.1 WAV files use only 0x3F as channel mask for 5.1 WAV files, which is FL, FR, FC, LFE, SL, SR.

so, maybe i'm wrong somehow, but please explain.
Please don't add more confussion.
Of course MS WAV channels don't have a exact correlation with Surround channels, for that admit two ChannelMask for 5.1 config:
0x003F channel mask for FL, FR, FC, LF, BL, BR
0x060F channel mask for FL, FR, FC, LF, SL, SR

Both must be played in the exact way:
- A 5.1 system must consider the last channels as Surround channels (120º more o less).
- A 7.1 system must split the Surround channels between Side and Back channels.

also, the same post gives information that DTS-HD MA 7.1 ("strange setup") tracks can be decoded with Sonic bit-perfectly to 5.1 channels that match DTS-HD MA 5.1 speaker positions, which is another thing that seems correct at least to me to be done.

P.S. and the same question(s) for DTS-HD MA 7.1 not-"strange setup" tracks based on DTS and Microsoft channel layout diagrams and respectively speaker positions.
I never tested the Sonic decoder and don't know what is exactly your questions, sorry.

For me the standard 7.1 have a exact correlation with MS channels (and standard location of speakers in a 7.1 audio system), but we need make some correction for "strange setup"

Also a receiver than accept 7.1 but have only 5.1 speakers must do some correction not only ignore the channels.

xkodi

8th December 2010, 18:09

Please don't add more confussion. sorry, but i'm confused and i believe i'm not the only one having the same questions regarding this subject.

Of course MS WAV channels don't have a exact correlation with Surround channels, for that admit two ChannelMask for 5.1 config:
0x003F channel mask for FL, FR, FC, LF, BL, BR
0x060F channel mask for FL, FR, FC, LF, SL, SR

the channel mask of 0x060F exists only in theory (and not even in theory if you ask me) and in the document i referred to:

http://download.microsoft.com/download/9/c/5/9c5b2167-8017-4bae-9fde-d599bac8184a/SpkrConfig5.docx

on page 9 Microsoft make it very clear that 5.1 WAV files no matter of the actual channel mask in newer versions of Windows are treatеd as with channel mask of 0x3F and effectively there is no channel mask of 0x060F, because even that channel mask is actually used it's the same for Windows as channel mask of 0x3F and thus in practice there is only one channel mask for 5.1 WAV files:

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. With this behavior, an audio driver never receives a 5.1-channel stream with a channel mask of 0x60F from the mixer.

Both must be played in the exact way:

i agree with this statement, because in practice Windows doesn't make difference between 0x3F and 0x60F channel mask as "SpkrConfig5.docx" document states

- A 5.1 system must consider the last channels as Surround channels (120º more o less).

i'm not sure about the exact angle and what "Surround channels" means in this case, but that is clearly defined in "SpkrConfig5.docx" document that in 5.1 setup Microsoft expect last two channel to be "Side Left" (SL) and "Side Right" (SR).

so, do you say that SL and SR in Microsoft world is what in DTS world is "Ls" ans "Rs"? because that is in contradiction with DTS 5.1 speaker positions:

http://www.imagebam.com/image/addcf997797266

and Microsoft 5.1 speaker positions as defined in page 6 "Playing a 5.1-Channel Stream on a 7.1 Speaker Configuration" of "SpkrConfig5.docx" document. that's the most confusing part at least for me.

- A 7.1 system must split the Surround channels between Side and Back channels.

that's not correct according page 6 "Playing a 5.1-Channel Stream on a 7.1 Speaker Configuration" of "SpkrConfig5.docx" document, where it's stated by Microsoft that the Back channels are just silence in such case. so, i can't agree here unless there is some document backing up your statement.

I never tested the Sonic decoder and don't know what is exactly your questions, sorry.

For me the standard 7.1 have a exact correlation with MS channels (and standard location of speakers in a 7.1 audio system)

how's that? at least from my point of view it's only in your opinion, because i don't think there is 1:1 match in speaker position between:

DTS-HD MA 7.1 (normal setup):

http://www.imagebam.com/image/43bb3497794528

and Microsoft 7.1 setup:

http://www.imagebam.com/image/160e7497795034

considering how DTS-HD MA 5.1 speaker position is:

http://www.imagebam.com/image/addcf997797266

and that there is no match in speaker position of Microsoft and DTS for 5.1 channels setup at least from what's defined in "SpkrConfig5.docx" document.

anyway, i don't want to argue about it, but there is no doubt it's very confusing.

tebasuna51

8th December 2010, 22:02

...
the channel mask of 0x060F exists only in theory (and not even in theory if you ask me)
Say you that to madshi and Flac developers, eac3to and flac decoder always output 0x060F.
and in the document i referred to:

http://download.microsoft.com/download/9/c/5/9c5b2167-8017-4bae-9fde-d599bac8184a/SpkrConfig5.docx
The document isn't clear, see in this extract of KsMedia.h:
// Speaker Positions:
#define SPEAKER_FRONT_LEFT 0x1
#define SPEAKER_FRONT_RIGHT 0x2
#define SPEAKER_FRONT_CENTER 0x4
#define SPEAKER_LOW_FREQUENCY 0x8
#define SPEAKER_BACK_LEFT 0x10
#define SPEAKER_BACK_RIGHT 0x20
#define SPEAKER_FRONT_LEFT_OF_CENTER 0x40
#define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80
#define SPEAKER_BACK_CENTER 0x100
#define SPEAKER_SIDE_LEFT 0x200
#define SPEAKER_SIDE_RIGHT 0x400
...
#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_5POINT1_SURROUND (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT)
Then both exist :
KSAUDIO_SPEAKER_5POINT1 = 0x3F with BL, BR
KSAUDIO_SPEAKER_5POINT1_SURROUND = 0x60F with SL, SR
but both must be played the same.

Other question is how the system mixer work:
- If the input is a 5.1 0x60F the output is 5.1 0x3F, no problem.
- If the input is 7.1 and need 5.1 output, discards channels 4 and 5 (BL,BR) and send a 5.1 0x3F with original channels SL, SR. This undesired behaviour can be modified using filters like ffdshow.

But all of this is irrelevant to your question:
so, do you say that SL and SR in Microsoft world is what in DTS world is "Ls" ans "Rs"?
No, I say than no matter how you call the last 2 channels (Back or Side) in a 5.1 audio they are always the "Ls, Rs" channels in DTS world (~120º), or Surround channels in AC3 world (~110º)

Forget also all the figures about exact speakers position in the MS docx, the audio is created like DTS (or DD) say and real audio systems must have the speakers located like DTS-DD say:
- For a 5.1 system: Surround channels at 110º-120º (not Side 90º nor Back 150º)
- For a 7.1 system: Side channels at ~90º, Back channels ~150º

TDiTP_

10th December 2010, 13:09

Yes, must be function of the angle, but doesn't exist a common opinion about the exact angles between speakers:
http://en.wikipedia.org/wiki/File:Standard_7.1_surround_sound_speaker_placement.png

I thought DTS Inc. published their angle recommendation.

Then another question please. How did you obtain that coefficients?

tebasuna51

10th December 2010, 17:28

Then another question please. How did you obtain that coefficients?
- I make 8 monowav test channel, each one with a max peak at 0 dB (normalized).

- I encode the 8 channels to 7.1 "strange setup" and after decoded with ArcSoft.

- The max peak of each channel in decoded wav say me the coefficients

Skinleech

10th December 2010, 20:48

I took delivery of a Blackgold 3620 DVB-T2 decoder yesterday so I can cap Freeview HD streams in the UK. I have got it set up with DVBViewer to record the incoming .ts streams - the broadcasts are 1440x1088/50i with AAC audio streams (AAC-LC afaik). Eac3to can correctly identify the streams in the source .ts but it can't proces the AAC tracks in anyway. For example, eac3to says the following when checking a recorded stream:

eac3to v3.24
command line: eac3to "L:\Recorded TV\12-10_19-18-52_BBC HD_Wild China.ts"
------------------------------------------------------------------------------
TS, 1 video track, 2 audio tracks, 1 subtitle track, 0:01:41, 50i
1: h264/AVC, 1440x1080 50i (16:9)
2: AAC, English, unknown parameters, -1718ms
3: AAC, English, unknown parameters, -1252ms
4: Subtitle (DVB), English
Bitstream parsing for tracks 2 and 3 failed. <WARNING>
Demuxing these tracks may still produce correct results - or not. <WARNING>

If I demux the tracks I get AAC files, but these still cannot be processed by eac3to. If I try playback in MPC, the audio is detected as a mpeg stream, and fails to play - no AAC filter is loaded. DVBViewer plays the recorded streams without issue. Is there any way to get eac3to to parse the AAC tracks, so I can convert them to ac3 - I can then edit the streams in VideoRedo.

Here's a sample:

http://www.freefilehosting.net/12-1019-18-52bbchdwildchina

tebasuna51

11th December 2010, 15:04

I took delivery of a Blackgold 3620 DVB-T2 decoder yesterday so I can cap Freeview HD streams in the UK. I have got it set up with DVBViewer to record the incoming .ts streams - the broadcasts are 1440x1088/50i with AAC audio streams (AAC-LC afaik).
...
Your Track 2 is a AAC LOAS-LATM.

After extract with eac3to you need convert this aac to a standard AAC with vlc:
"D:\your path\vlc.exe" -I dummy "China_2 DELAY -1718ms.aac" --sout #std{access=file,mux=mp4,dst="China_2_nolatm.mp4"} vlc://quit

Now you have a playable mp4:
Audio format : AAC-LC
Channels ....: 6
SampleRate ..: 48000 Hz.
Duration trak: 100.096 sec., (0h. 1m. 40.096s.)
BitRate .....: 315 Kb/s

You can use the mp4 to mux with video in a mkv/mp4 container or decode with NeroAacDec

Edit: your Track 3 (also LOAS-LATM) seems a:
Audio format : AAC-HE v2 (LC+SBR+PS)
Channels ....: 2
SampleRate ..: 48000 Hz.
Duration trak: 99.584 sec., (0h. 1m. 39.584s.)
BitRate .....: 63 Kb/s
but I get only silence.

Skinleech

11th December 2010, 16:32

Your Track 2 is a AAC LOAS-LATM.

After extract with eac3to you need convert this aac to a standard AAC with vlc:

Thank you very much - just tried these steps and I now have a playable audio file! I assume the conversion is not lossless, i.e. some kind of header removal? Is it possible to go direct to ac3 via VLC? I don't use it at all.

With regards to the 2nd audio track, it is likely supposed to be silence - the 2nd track is designed to carry an audio description service, but only 2 - 3 programmes a day actually use the track.

Thank you for your assistance.

tebasuna51

11th December 2010, 20:48

I assume the conversion is not lossless, i.e. some kind of header removal?...
Yes, the conversion is lossless, the audio data in LOAS-LATM stream is the same than audio data in .mp4.

vlc remove LOAS and LATM headers and put the header parameters in the mp4 container.

SeeMoreDigital

12th December 2010, 11:59

eac3to v3.24
command line: eac3to "L:\Recorded TV\12-10_19-18-52_BBC HD_Wild China.ts"So BBC have decided to broadcast using AAC audio. The same as ITV. This move could set a precedent for all futue FTA HDTV channels in the UK :eek:

I wonder.... Is there any chance some kind person on this forum can "whip-up" an dedicated application to strip these LOAS/LATM headers from AAC streams?

Cheers

Snowknight26

12th December 2010, 18:35

Any chance of implementing working outputting to NUL? Currently if you output a track to NUL:\file.ext, eac3to just stalls a few seconds in, and NUL doesn't work at all.

xkodi

12th December 2010, 21:25

I wonder.... Is there any chance some kind person on this forum can "whip-up" an dedicated application to strip these LOAS/LATM headers from AAC streams?

there is already such application(s) available:

LOAS_LATM_to_ADTS.exe: LOAS/LATM AAC to ADTS AAC

(ADTS_to_RAW.exe: ADTS AAC to RAW AAC
LOAS_LATM_to_RAW.exe: LOAS/LATM AAC to RAW AAC)

File name: AAC_Tools_NA_0.1.zip File size: 148.64 KB (http://www.fileserve.com/file/5wzwNpg)

as well as the zip file includes the source code in Delphi - the same programming language in which eac3to is written and thus such conversion(s) could be included easily in eac3to too

tebasuna51

13th December 2010, 02:59

there is already such application(s) available:

LOAS_LATM_to_ADTS.exe: LOAS/LATM AAC to ADTS AAC

(ADTS_to_RAW.exe: ADTS AAC to RAW AAC
LOAS_LATM_to_RAW.exe: LOAS/LATM AAC to RAW AAC)

File name: AAC_Tools_NA_0.1.zip File size: 148.64 KB (http://www.fileserve.com/file/5wzwNpg)

as well as the zip file includes the source code in Delphi - the same programming language in which eac3to is written and thus such conversion(s) could be included easily in eac3to too

Yes, but don't work for me.

See this post http://forum.doom9.org/showthread.php?t=157467

Is the same soft than proposed in http://www.dvbviewer.tv/forum/topic/35738-loaslatm-%26gt%3B-adts-conversion/

Skinleech

13th December 2010, 11:08

Yes, the conversion is lossless, the audio data in LOAS-LATM stream is the same than audio data in .mp4.

vlc remove LOAS and LATM headers and put the header parameters in the mp4 container.

Great, thanks. I'm currently messing about with a decent workflow so I can pull the files into VRD4 to edit ad breaks etc.

Is it possible to apply the associated audio delay to the .mp4 output by VRD during the header removal, rather than setting it in the container?

tebasuna51

13th December 2010, 11:54

...
Is it possible to apply the associated audio delay to the .mp4 output by VRD during the header removal, rather than setting it in the container?
Sorry, I don't know a tool to convert +delay to initial AAC silence like eac3to/DelayCut can do with AC3/DTS/MP3.

b66pak

13th December 2010, 14:27

demux the mp4 to aac and apply the delay then mux back to whatever container you need...
_

Skinleech

13th December 2010, 14:59

Thanks both.

I was hoping to avoid yet another step in the process, but it looks like I may just need to go with this. My testing today was:

Demux with eac3to
Convert audio with VLC
Mux video & AAC in .mp4m (with delay) to .ts with Tsmuxer
Edit .ts in VideoRedo4

Playback in TMT5 was OK on my dekstop, but it was way out of sync on my HTPC so I'll need to do a little more work.

Anyway, enough from me as these really are off topic posts for the eac3to thread.

gregt

14th December 2010, 07:24

Hi All,
I tried to extract the DTS track (#4) from this Blu-ray movie that I own and the resultant flac file had the wrong track or multiple tracks mixed together because the director's voice was heard over the regular movie audio. I looked at the streams and there was no single stream file for the movie so it looks like it is quite broken up.

Any ideas?

G

Skinleech

15th December 2010, 09:32

jd213

17th December 2010, 07:25

nurbs

17th December 2010, 11:31

:logfile: No way to tell if you are doing something wrong if we don't know what you are doing.

tebasuna51

17th December 2010, 12:02

When extracting a Blu-ray Dolby TrueHD track, how do I make eac3to just demux it rather than decoding/encoding? I tried inputting "-core" after "output.thd+ac3" but it's still decoding the TrueHD with libav and encoding the ac3 core with libAften.
Example:
eac3to v3.24
command line: "D:\Programa\eac3to\eac3to.exe"
"D:\Test\hd_dolby_truehd.m2ts"
2: "D:\Test\hd_dolby_truehd.m2ts_2.thd+ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 0:01:34, 60i /1.001
1: h264/AVC, 1080i60 /1.001 (16:9)
2: TrueHD/AC3, 7.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
[a02] Extracting audio track number 2...
[a02] Creating file "D:\Test\hd_dolby_truehd.m2ts_2.thd+ac3"...
Video track 1 contains 2832 frames.
eac3to processing took 2 seconds.
Done.
without decode or encode anything.

krosswindz

18th December 2010, 18:30

I was wondering if eac3to will ever support encoding to mp3 audio as well. Then eac3to would be the one stop shop for me for audio conversion.

Boulder

18th December 2010, 20:34

Just wondering, is it possible to pipe eac3to's output (WAV) to LAME?

tebasuna51

19th December 2010, 01:01

Just wondering, is it possible to pipe eac3to's output (WAV) to LAME?
Of course:
eac3to input stdout.wav [-down2 -normalize] | Lame -V 4 - output.mp3

Also to Oggenc, or Aften (for 3 or 4 channels).

jd213

20th December 2010, 02:12

Example:
eac3to v3.24
command line: "D:\Programa\eac3to\eac3to.exe"
"D:\Test\hd_dolby_truehd.m2ts"
2: "D:\Test\hd_dolby_truehd.m2ts_2.thd+ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 0:01:34, 60i /1.001
1: h264/AVC, 1080i60 /1.001 (16:9)
2: TrueHD/AC3, 7.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
[a02] Extracting audio track number 2...
[a02] Creating file "D:\Test\hd_dolby_truehd.m2ts_2.thd+ac3"...
Video track 1 contains 2832 frames.
eac3to processing took 2 seconds.
Done.
without decode or encode anything.

Thanks. It didn't do this when I tried demuxing from the playlist, but it worked for the m2ts file.

Kisa_AG

20th December 2010, 12:29

Hello!

I found small bug in eac3to v3.24. During ac3 decoding tt makes downmixing multi channel audio to stereo only in

case of 5.1 channel input (L,R,C,Ls,Rs,LFE), but doesn't do it with other multichannel configurations, like 5.0

(L,R,C,Ls,Rs), 4.1 (L,R,Ls,Rs,LFE), 4.0 (L,R,Ls,Rs) and 3.0 (L,R,C).

Here is two logs:
"Successful" downmixing of 5.1:
-----------------------------------------------------------------
command line: eac3to.exe 5.1.ac3 5.1.wav -down2 -mixlfe -down16
-----
AC3, 5.1 channels, 0:00:10, 448kbps, 48kHz
The Nero decoder doesn't seem to work, will use libav instead.
Decoding with libav/ffmpeg...
Remapping channels...
Downmixing multi channel audio to stereo...
Reducing depth from 64 to 16 bits...
Writing WAV...
Clipping detected, a 2nd pass will be necessary. <WARNING>
Creating file "5.1.wav"...
Starting 2nd pass...
Decoding with libav/ffmpeg...
Remapping channels...
Downmixing multi channel audio to stereo...
Reducing depth from 64 to 16 bits...
Writing WAV...
Applying -7.52dB gain...
Creating file "5.1.wav"...
eac3to processing took 2 seconds.
Done.
-----------------------------------------------------------------
After that I got Stereo WAV file.
There is a "Downmixing..." string in log file.

"Not Successful" downmixing of 3.0:
-----------------------------------------------------------------
command line: eac3to.exe 3.0.ac3 3.0.wav -down2 -down16
----
AC3, 3.0 channels, 0:00:10, 448kbps, 48kHz
The Nero decoder doesn't seem to work, will use libav instead.
Decoding with libav/ffmpeg...
Reducing depth from 64 to 16 bits...
Writing WAV...
Creating file "3.0.wav"...
eac3to processing took 1 second.
Done.
-----------------------------------------------------------------
After that I got 3.0 multichannel WAV file.
There is no "Downmixing..." string in log file. But I "asked" eac3to to downmix the file...

Just for information, here are Media infos of input 3.0 ac3 file and output 3.0 WAV file:
------------------------------------------
General
Complete name : 3.0.ac3
Format : AC-3
Format/Info : Audio Coding 3
File size : 548 KiB
Duration : 10s 16ms
Overall bit rate : 448 Kbps

Audio
Format : AC-3
Format/Info : Audio Coding 3
Mode extension : CM (complete main)
Duration : 10s 16ms
Bit rate mode : Constant
Bit rate : 448 Kbps
Channel(s) : 3 channels
Channel positions : Front: L C R
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Compression mode : Lossy
Stream size : 548 KiB (100%)
------------------------------------------
General
Complete name : 3.0.wav
Format : Wave
File size : 2.75 MiB
Duration : 10s 16ms
Overall bit rate : 2 304 Kbps

Audio
ID : 0
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Signed
Codec ID : 00001000-0000-0100-8000-00AA00389B71
Codec ID/Hint : Microsoft
Duration : 10s 16ms
Bit rate mode : Constant
Bit rate : 2 304 Kbps
Channel(s) : 3 channels
Channel positions : Front: L C R
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Stream size : 2.75 MiB (100%)
------------------------------------------

Joplin

20th December 2010, 17:42

Am I correct when I assume that when demuxing a .dtshd track the "DTS Core" is not included afterwards? If I want to have the DTS-HD and DTS, I have to demux them separately and then mux both tracks in my .mkv?

rapscallion

20th December 2010, 17:53

No, that's not correct. It will be demuxed to a MA track containing the core.

Select "dtsma" as the "save as" type.

_hawk_

21st December 2010, 08:43

Is there a commandline switch to disable color output?
Would it be possible to add something like this?

TDiTP_

21st December 2010, 10:08

mrr19121970

21st December 2010, 12:54

Is there a commandline switch to disable color output?
Would it be possible to add something like this?

yes it's like -simple or somthing like that. I'll try it.

Update

It was already discussed a few times (1 (http://forum.doom9.org/showthread.php?p=1333829#post1333829), 2 (http://forum.doom9.org/showthread.php?p=1348250#post1348250)).
Adding -simpleoutput switch into eac3to looks like the best solution.

But the answer to your question (that I assume you want to ask is):

AttachConsole Function (http://msdn.microsoft.com/en-us/library/ms681952%28VS.85%29.aspx)
GetConsoleScreenBufferInfo Function
(http://msdn.microsoft.com/en-us/library/ms683171%28VS.85%29.aspx)
as Clown_BD does.

_hawk_

21st December 2010, 13:44

yes it's like -simple or somthing like that. I'll try it.

AttachConsole Function (http://msdn.microsoft.com/en-us/library/ms681952%28VS.85%29.aspx)
GetConsoleScreenBufferInfo Function
(http://msdn.microsoft.com/en-us/library/ms683171%28VS.85%29.aspx)
as Clown_BD does.

Thank you for the info.
Will try that.

Edit: seems that -simpleoutput or something isn't implemented yet
AttachConsole seems not to work with C# :(

tebasuna51

21st December 2010, 16:10

When apply -double7 command.
Rs, Ls, Cs (DTS-HD Master Audio Suite) -> BL,BR,SL,SR (eac3to input.dtshd output.wavs -double7):

SL = Cs
SR = Ls
BL (and BR, BL=BR) ~Rs but level is lower

Bug. Anyone can confirm it?

I've checked DTS-HD MA 6.1 ES Discrete and DTS-ES 6.1 Discrete.

I think the right way is:
SL = Ls
SR = Rs
BL = BR = 0,707xCs

are not?

Yes, you are right, the output is:
BL = 0.7071xRs
BR = 0.7071xRs
SL = Cs
SR = Ls

Also the downmix 6.1 -> 5.1 dont work:
SurroundL = Cs + 0.7071xRs
SurroundR = Ls + 0.7071xRs

Seems there are still troubles with 6.1 channelmap.

jd213

22nd December 2010, 03:41

libav keeps giving me an error whenever I try to convert the thd track from the Japanese HD DVD release of Vexille to thd+ac3 or any other format:

eac3to v3.24
command line: eac3to G:\vexille\joined.evo 1: G:\vexille\vid.h264 2: G:\vexille\aud.thd+ac3 3: G:\vexille\aud3.wav 4: G:\vexille\aud4.wav
------------------------------------------------------------------------------
EVO, 1 video track, 3 audio tracks, 1:49:48
1: h264/AVC, 1080p24 /1.001 (16:9) with pulldown flags
2: TrueHD, 5.1 channels, 48kHz, dialnorm: -29dB, -83ms
3: RAW/PCM, 2.0 channels, 16 bits, 48kHz, -83ms
4: RAW/PCM, 2.0 channels, 16 bits, 48kHz, -83ms
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a03] Extracting audio track number 3...
[a04] Extracting audio track number 4...
[a02] Extracting audio track number 2...
[v01] Removing h264 pulldown...
[a02] Removing TrueHD dialog normalization...
[a03] Reading RAW/PCM...
[a04] Reading RAW/PCM...
[a02] Removing TrueHD dialog normalization...
[a03] Swapping endian...
[a04] Swapping endian...
[a02] Decoding with libav/ffmpeg...
[a03] Applying RAW/PCM delay...
[a04] Applying RAW/PCM delay...
[a02] Encoding AC3 <640kbps> with libAften...
[a03] Writing WAV...
[a04] Writing WAV...
[v01] Creating file "G:\vexille\vid.h264"...
[a02] Creating file "G:\vexille\aud.thd+ac3"...
[a04] Creating file "G:\vexille\aud4.wav"...
[a03] Creating file "G:\vexille\aud3.wav"...
[a02] [libav] No restart header present in substream 0. <WARNING>
[a02] The libav decoder reported error -1 while decoding. <ERROR>
Aborted at file position 12162433024. <ERROR>

I thought it might be due to a bad rip, but it does this both with the rip I made with dumpHD and when reading from the disc with the latest version of DVDFab Passkey. I tried first extracting the thd track and then converting to thd+ac3, then I tried joining the EVO files and then converting, but the same error still occurred.

Is there any way around this? I'd try the Nero decoder, but the HD DVD/BD plugin doesn't seem to be available any more (I can only find the BD plugin for Nero 10 on their site).

mrr19121970

22nd December 2010, 17:11

AttachConsole seems not to work with C# :(

See here (http://www.csharp411.com/console-output-from-winforms-application/). Maybe this helps?

rapscallion

28th December 2010, 05:44

In searching, I see that eac3to displays warnings for audio overlaps but does not fix them.

Extracting the DTS MA audio track from The A-Team extended cut ( 39 M2ts's !) I get the following 23 overlap warnings :

eac3to v3.24
command line: "C:\dvd_rw\eac3to\eac3to.exe" "D:\BD-Test\A Team\A_TEAM_FD\BDMV\" 1) 3: "D:\BD-Test\A Team\A_TEAM_FD\BDMV\team_dts.dts" -nero
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 3 subtitle tracks, 2:13:32, 24p /1.001
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
4: AC3, French, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
5: AC3, Spanish, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
6: AC3, English, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
7: Subtitle (PGS), English
8: Subtitle (PGS), Spanish
9: Subtitle (PGS), French
[a03] Extracting audio track number 3...
[a03] Creating file "D:\BD-Test\A Team\A_TEAM_FD\BDMV\team_dts.dts"...
[a03] Audio overlaps for 6ms at playtime 0:03:56. <WARNING>
[a03] Audio overlaps for 10ms at playtime 0:07:59. <WARNING>
[a03] Audio overlaps for 10ms at playtime 0:11:19. <WARNING>
[a03] Audio overlaps for 6ms at playtime 0:16:48. <WARNING>
[a03] Audio overlaps for 9ms at playtime 0:24:15. <WARNING>
[a03] Audio overlaps for 8ms at playtime 0:26:10. <WARNING>
[a03] Audio overlaps for 5ms at playtime 0:35:53. <WARNING>
[a03] Audio overlaps for 10ms at playtime 0:37:53. <WARNING>
[a03] Audio overlaps for 9ms at playtime 0:38:12. <WARNING>
[a03] Audio overlaps for 11ms at playtime 0:42:42. <WARNING>
[a03] Audio overlaps for 10ms at playtime 0:47:35. <WARNING>
[a03] Audio overlaps for 14ms at playtime 0:59:11. <WARNING>
[a03] Audio overlaps for 10ms at playtime 1:06:16. <WARNING>
[a03] Audio overlaps for 6ms at playtime 1:07:36. <WARNING>
[a03] Audio overlaps for 5ms at playtime 1:15:06. <WARNING>
[a03] Audio overlaps for 9ms at playtime 1:23:50. <WARNING>
[a03] Audio overlaps for 6ms at playtime 1:26:12. <WARNING>
[a03] Audio overlaps for 8ms at playtime 1:28:03. <WARNING>
[a03] Audio overlaps for 8ms at playtime 1:28:55. <WARNING>
[a03] Audio overlaps for 5ms at playtime 1:36:50. <WARNING>
[a03] Audio overlaps for 14ms at playtime 1:47:59. <WARNING>
[a03] Audio overlaps for 9ms at playtime 1:49:43. <WARNING>
[a03] Audio overlaps for 10ms at playtime 1:55:05. <WARNING>
[a03] Audio overlaps for 14ms at playtime 2:02:59. <WARNING>
[a03] Audio overlaps for 7ms at playtime 2:07:27. <WARNING>
[a03] Starting 2nd pass...
[a03] Realizing DTS gaps...
[a03] Creating file "D:\BD-Test\A Team\A_TEAM_FD\BDMV\team_dts.dts"...
Video track 2 contains 192139 frames.
eac3to processing took 10 minutes, 38 seconds.
Done.

Are these overlaps big enough to see a noticeable out of sync?

Secondly, what does "Realizing DTS gaps", in the second pass, mean ?

Thunderbolt8

28th December 2010, 11:12

1. yes (though it can vary depending on your position in the movie)
2. it means everything is fine.

rapscallion

28th December 2010, 16:03

1. yes (though it can vary depending on your position in the movie)
2. it means everything is fine.

2. Everything's fine as in the overlaps were fixed? ( I thought eac3to couldn't do that)

TinTime

28th December 2010, 16:28

2. Everything's fine as in the overlaps were fixed? ( I thought eac3to couldn't do that)

Only TrueHD overlaps can't be fixed if demuxing.

rapscallion

28th December 2010, 17:11

Only TrueHD overlaps can't be fixed if demuxing.
Ah Ha ! Thank you.

Kisa_AG

29th December 2010, 15:03

Hello!
Can anyone confirm the bug with multichannel downmixing I've stated above? I'm trying to understand, is this a bug of eac3to or is this due to other things?

TDiTP_

29th December 2010, 19:19

Can anyone confirm the bug with multichannel downmixing I've stated above?

It's not bug. eac3to never was programmed for such downmix (3.0->2.0). Only 5.1->2.0 and only DPLII (even 5.0->2.0 doesn't work).

mrr19121970

30th December 2010, 08:57

@madshi

do you know why the following messages are displayed on -demux?

[a06] Decoding with libav/ffmpeg...
[a06] Encoding AC3 <640kbps> with libAften...

I would have expected no encoding/decoding here. Is there a problem?

Here is the full log:

eac3to v3.24
command line: "D:\TVIX\Clown_BD\eac3to\eac3to.exe" M:\ 1) -demux
------------------------------------------------------------------------------
M2TS, 2 video tracks, 6 audio tracks, 19 subtitle tracks, 1:42:14, 24p /1.001
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 "00123 - Chapters.txt"...
[a06] Extracting audio track number 6...
[a06] Extracting TrueHD stream...
[a06] Decoding with libav/ffmpeg...
[a06] Encoding AC3 <640kbps> with libAften...
[s14] Extracting subtitle track number 14...
[s24] Extracting subtitle track number 24...
[s16] Extracting subtitle track number 16...
[a07] Extracting audio track number 7...
[s13] Extracting subtitle track number 13...
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Decoding with libav/ffmpeg...
[a04] Encoding AC3 <640kbps> with libAften...
[s20] Extracting subtitle track number 20...
[s10] Extracting subtitle track number 10...
[s22] Extracting subtitle track number 22...
[a05] Extracting audio track number 5...
[s23] Extracting subtitle track number 23...
[a08] Extracting audio track number 8...
[s18] Extracting subtitle track number 18...
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[s26] Extracting subtitle track number 26...
[s12] Extracting subtitle track number 12...
[s19] Extracting subtitle track number 19...
[s15] Extracting subtitle track number 15...
[s21] Extracting subtitle track number 21...
[a06] Extracting audio track number 6...
[a06] Extracting TrueHD stream...
[s28] Extracting subtitle track number 28...
[v02] Extracting video track number 2...
[s11] Extracting subtitle track number 11...
[s27] Extracting subtitle track number 27...
[s25] Extracting subtitle track number 25...
[s17] Extracting subtitle track number 17...
[a07] Removing AC3 dialog normalization...
[a08] Removing AC3 dialog normalization...
[a05] Removing AC3 dialog normalization...
[v02] Creating file "00123 - 2 - h264, 1080p24.h264"...
[a06] Creating file "00123 - 6 - TrueHD+AC3, German, 5.1 channels, 48kHz.thd+ac3"...
[a04] Creating file "00123 - 4 - TrueHD+AC3, English, 5.1 channels, 48kHz.thd+ac3"...
[a05] Creating file "00123 - 5 - AC3, English, 5.1 channels, 448kbps, 48kHz.ac3"...
[a08] Creating file "00123 - 8 - AC3, Ukrainian, 5.1 channels, 448kbps, 48kHz.ac3"...
[a07] Creating file "00123 - 7 - AC3, Russian, 5.1 channels, 448kbps, 48kHz.ac3"...
[s23] Creating file "00123 - 23 - Subtitle (PGS), English.sup"...
[s24] Creating file "00123 - 24 - Subtitle (PGS), German.sup"...
[s25] Creating file "00123 - 25 - Subtitle (PGS), Dutch.sup"...
[s10] Creating file "00123 - 10 - Subtitle (PGS), English.sup"...
[s11] Creating file "00123 - 11 - Subtitle (PGS), English.sup"...
[s15] Creating file "00123 - 15 - Subtitle (PGS), Estonian.sup"...
[s22] Creating file "00123 - 22 - Subtitle (PGS), Turkish.sup"...
[s18] Creating file "00123 - 18 - Subtitle (PGS), Hindi.sup"...
[s19] Creating file "00123 - 19 - Subtitle (PGS), Lithuanian.sup"...
[s13] Creating file "00123 - 13 - Subtitle (PGS), Dutch.sup"...
[s12] Creating file "00123 - 12 - Subtitle (PGS), German.sup"...
[s17] Creating file "00123 - 17 - Subtitle (PGS), Modern Greek.sup"...
[s21] Creating file "00123 - 21 - Subtitle (PGS), Swedish.sup"...
[s14] Creating file "00123 - 14 - Subtitle (PGS), Danish.sup"...
[s20] Creating file "00123 - 20 - Subtitle (PGS), Norwegian.sup"...
[s16] Creating file "00123 - 16 - Subtitle (PGS), Finnish.sup"...
[s27] Creating file "00123 - 27 - Subtitle (PGS), Ukrainian.sup"...
[a05] Audio overlaps for 27ms at playtime 0:05:51. <WARNING>
[a05] Audio overlaps for 23ms at playtime 0:27:55. <WARNING>
[a05] Audio overlaps for 16ms at playtime 0:32:31. <WARNING>
[a05] Audio overlaps for 9ms at playtime 0:34:13. <WARNING>
[a05] Audio overlaps for 20ms at playtime 0:48:54. <WARNING>
[a05] Audio overlaps for 18ms at playtime 0:49:56. <WARNING>
[a05] Audio overlaps for 17ms at playtime 0:52:28. <WARNING>
[a05] Audio overlaps for 29ms at playtime 0:53:10. <WARNING>
[a05] Audio overlaps for 31ms at playtime 0:57:34. <WARNING>
[a05] Audio overlaps for 8ms at playtime 1:04:46. <WARNING>
[a05] Audio overlaps for 25ms at playtime 1:07:09. <WARNING>
[a05] Audio overlaps for 7ms at playtime 1:10:47. <WARNING>
[a05] Audio overlaps for 23ms at playtime 1:13:47. <WARNING>
[a05] Audio overlaps for 24ms at playtime 1:15:20. <WARNING>
[a05] Audio overlaps for 18ms at playtime 1:25:56. <WARNING>
[a07] Audio overlaps for 27ms at playtime 0:05:51. <WARNING>
[a07] Audio overlaps for 23ms at playtime 0:27:55. <WARNING>
[a07] Audio overlaps for 16ms at playtime 0:32:31. <WARNING>
[a07] Audio overlaps for 9ms at playtime 0:34:13. <WARNING>
[a07] Audio overlaps for 20ms at playtime 0:48:54. <WARNING>
[a07] Audio overlaps for 18ms at playtime 0:49:56. <WARNING>
[a07] Audio overlaps for 17ms at playtime 0:52:28. <WARNING>
[a07] Audio overlaps for 29ms at playtime 0:53:10. <WARNING>
[a07] Audio overlaps for 31ms at playtime 0:57:34. <WARNING>
[a07] Audio overlaps for 8ms at playtime 1:04:46. <WARNING>
[a07] Audio overlaps for 25ms at playtime 1:07:09. <WARNING>
[a07] Audio overlaps for 7ms at playtime 1:10:47. <WARNING>
[a07] Audio overlaps for 23ms at playtime 1:13:47. <WARNING>
[a07] Audio overlaps for 24ms at playtime 1:15:20. <WARNING>
[a07] Audio overlaps for 18ms at playtime 1:25:56. <WARNING>
[a08] Audio overlaps for 27ms at playtime 0:05:51. <WARNING>
[a08] Audio overlaps for 23ms at playtime 0:27:55. <WARNING>
[a08] Audio overlaps for 16ms at playtime 0:32:31. <WARNING>
[a08] Audio overlaps for 9ms at playtime 0:34:13. <WARNING>
[a08] Audio overlaps for 20ms at playtime 0:48:54. <WARNING>
[a08] Audio overlaps for 18ms at playtime 0:49:56. <WARNING>
[a08] Audio overlaps for 17ms at playtime 0:52:28. <WARNING>
[a08] Audio overlaps for 29ms at playtime 0:53:10. <WARNING>
[a08] Audio overlaps for 31ms at playtime 0:57:34. <WARNING>
[a08] Audio overlaps for 8ms at playtime 1:04:46. <WARNING>
[a08] Audio overlaps for 25ms at playtime 1:07:09. <WARNING>
[a08] Audio overlaps for 7ms at playtime 1:10:47. <WARNING>
[a08] Audio overlaps for 23ms at playtime 1:13:47. <WARNING>
[a08] Audio overlaps for 24ms at playtime 1:15:20. <WARNING>
[a08] Audio overlaps for 18ms at playtime 1:25:56. <WARNING>
[s26] Creating file "00123 - 26 - Subtitle (PGS), Russian.sup"...
[s28] Creating file "00123 - 28 - Subtitle (PGS), English.sup"...
[a04] The original audio track has a constant bit depth of 16 bits.
[a06] The original audio track has a constant bit depth of 16 bits.
[a05] Starting 2nd pass...
[a05] Realizing (E-)AC3 gaps...
[a05] Creating file "00123 - 5 - AC3, English, 5.1 channels, 448kbps, 48kHz.ac3"...
[a07] Starting 2nd pass...
[a07] Realizing (E-)AC3 gaps...
[a07] Creating file "00123 - 7 - AC3, Russian, 5.1 channels, 448kbps, 48kHz.ac3"...
[a08] Starting 2nd pass...
[a08] Realizing (E-)AC3 gaps...
[a08] Creating file "00123 - 8 - AC3, Ukrainian, 5.1 channels, 448kbps, 48kHz.ac3"...
Video track 2 contains 147077 frames.
Video track 3 contains 147077 frames.
Subtitle track 10 contains 1125 normal and 7 forced captions.
Subtitle track 11 contains 1364 captions.
Subtitle track 12 contains 1090 normal and 9 forced captions.
Subtitle track 13 contains 880 captions.
Subtitle track 14 contains 887 captions.
Subtitle track 15 contains 1143 captions.
Subtitle track 16 contains 1004 captions.
Subtitle track 17 contains 1097 captions.
Subtitle track 18 contains 1145 captions.
Subtitle track 19 contains 1145 captions.
Subtitle track 20 contains 1095 captions.
Subtitle track 21 contains 869 captions.
Subtitle track 22 contains 1145 captions.
Subtitle track 23 contains 1732 captions.
Subtitle track 24 contains 1651 captions.
Subtitle track 25 contains 1625 captions.
Subtitle track 26 contains 16 forced captions.
Subtitle track 27 contains 312 forced captions.
Subtitle track 28 contains 20 forced captions.
eac3to processing took 26 minutes, 45 seconds.
Done.

tebasuna51

30th December 2010, 12:01

...
do you know why the following messages are displayed on -demux?

[a06] Decoding with libav/ffmpeg...
[a06] Encoding AC3 <640kbps> with libAften...

I would have expected no encoding/decoding here. Is there a problem?...

Check if your extracted .thd+ac3 have the embedded AC3 with 640 kbps and decide if is a problem for you.

Kisa_AG

30th December 2010, 14:24

It's not bug. eac3to never was programmed for such downmix (3.0->2.0). Only 5.1->2.0 and only DPLII (even 5.0->2.0 doesn't work).

Ok, thanks, now it's clear. But IMHO in such case it should be mentioned in eac3to help.

a1adin

31st December 2010, 15:59

Hi guys,

first of all, thanks for this very nice program.
sorry if this is a noob question:
I like to convert 5.1 aac to AC3.
Please let me know the correct command line to do it.

since I do not have nero, first I convert the aac to AC3 using vlc media player,
then I get the AC3 but with the wrong channel mapping.
Then I use eac3to to correct the mapping:
0 1 2 3 4 5
5.1 AAC FC FL FR SL SR LFE
5.1 AC3 FL FC FR SL SR LFE

so the command line is:
eac3to input.ac3 output.ac3 -1,0,2,3,4,5

However, when I listen to it, I still get the wrong channel mapping.
Please let me know what is wrong and how to fix it.
Thanks.

tebasuna51

31st December 2010, 18:41

...
I like to convert 5.1 aac to AC3.
Please let me know the correct command line to do it.
since I do not have nero,...
Without Nero plugin for eac3to you can use Faad and Aften:
Faad -w -b 4 input.aac | Aften -readtoeof 1 -b 640 - output.ac3

... first I convert the aac to AC3 using vlc media player, then I get the AC3 but with the wrong channel mapping.
With my test, always I convert AAC with VLC 1.1.4 I get a wrong channel map like this:
FL, FR, SL, SR, FC, LFE

no mater if the output is WAV, FLAC or AC3.
With AC3 the SR channel is empty.

Then I recommend you convert the AAC to FLAC (or WAV), to avoid the SR problem and two lossy encodes, with VLC and after use:
eac3to intermediate.flac output.ac3 -0,1,4,5,2,3

a1adin

1st January 2011, 09:46

Hi tebasuna51,

Thanks for a very informative reply! :D

With my test, always I convert AAC with VLC 1.1.4 I get a wrong channel map like this:
FL, FR, SL, SR, FC, LFE

so that's the culprit I could not make it right.
thanks for letting me know, I almost die with frustration figuring that out. :D

Without Nero plugin for eac3to you can use Faad and Aften:
Faad -w -b 4 input.aac | Aften -readtoeof 1 -b 640 - output.ac3

could you give the link for FAAD?

Also, I am going to install nero7.
What part of nero should I install to be able to use the aac decoder for eac3to?
Please advise.

robpdotcom

1st January 2011, 09:56

Also, I am going to install nero7.
What part of nero should I install to be able to use the aac decoder for eac3to?
Please advise.

Nero Burning ROM is all I needed.

tebasuna51

1st January 2011, 12:11

...
could you give the link for FAAD?

Also, I am going to install nero7.
What part of nero should I install to be able to use the aac decoder for eac3to?
For Faad decoder (FAAD2v20100614 CVS), and other free decoders/encoders, you can go to http://www.rarewares.org/

For Nero 7 you need install also the Blu-Ray/HD DVD plugin.

a1adin

1st January 2011, 18:06

@ tebasuna51 & robpdotcom
thanks for the reply! :D

what is the best nero for eac3to: 7.8.5.0 or 7.10.1.0 ?

Faad -w -b 4 input.aac | Aften -readtoeof 1 -b 640 - output.ac3

sorry if this is a noob question:
what is the "straight line" before Aften means?

tebasuna51

2nd January 2011, 07:14

...
what is the "straight line" before Aften means?
Is the "pipe" command, the standard output (STDOUT, -w parameter) of Faad is redirected to standard input (STDIN, input replaced with '-') of Aften.

The wav output of Faad is send to Aften through internal SO buffers (fast), without write the file to disk (slow).

Thunderbolt8

3rd January 2011, 20:37

got a problem with a 2.0 DTS-HD MA track, always get this error message when trying to convert it to flac:

eac3to v3.24
command line: eac3to track.dtsma track.flac
------------------------------------------------------------------------------
DTS Master Audio, 2.0 channels, 24 bits, 48kHz
(core: DTS, 2.0 channels, 24 bits, 960kbps, 48kHz)
Decoding with ArcSoft DTS Decoder...
Encoding FLAC with libFlac...
Creating file "track.flac"...
The ArcSoft DTS Decoder reported an error while decoding. <ERROR>
Aborted at file position 1871970304. <ERROR>

jj666

3rd January 2011, 21:08

960kb/sec core? I've never seen an official disk using that. Is it some homebrewn source?

Cheers,

-jj-

palpyhat

3rd January 2011, 21:31

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

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

And how do I get an unaltered full DTS-HD Master Audio track? What extension do I want to specify for output?

The resulting audio track will then be fed to mkvmerge gui, to be multiplexed in an existing matroska container.

Thunderbolt8

3rd January 2011, 22:23

960kb/sec core? I've never seen an official disk using that. Is it some homebrewn source?

Cheers,

-jj-its the commentary track of the Master's of Cinema version of Metropolis (and the error happens somewhere halfway during the process, not directly at the beginning; demuxing this track was fine, as well as remuxing the video and all other tracks)

dansrfe

4th January 2011, 10:53

And how do I get an unaltered full DTS-HD Master Audio track? What extension do I want to specify for output?

The resulting audio track will then be fed to mkvmerge gui, to be multiplexed in an existing matroska container.

eac3to "path\to\input.extension" 3: "save\path\to\output.dtshd"

Where input.extension can be some m2ts, evo, mpls or whatever source has the dts-hd audio. The "3:" in this instance is the specific track that the dts-hd audio is on. You can find the track number that holds the dts-hd file in your particular file by running:

eac3to "path\to\input.extension"

TinTime

4th January 2011, 14:15

The extension .dtsma also works for DTS Master Audio, if you need to distinguish them from DTS Hi-Res (.dtshr).

dansrfe

4th January 2011, 14:39

.dtsma is probably a better choice, correct.

palpyhat

4th January 2011, 15:50

Thank you.

Also a couple of technical questions.

1. In a "DTS-HD Master Audio" audio track, which data is stored first in each packet/frame? The DTS core, or the required deltas to reconstruct the DTS-HD-MA lossless stream? I assume it is the former, but I am not sure.

2. If we assume that C=core, D=Deltas and L=Lossless then we have L=C+D, and in each "DTS-HD Master Audio" audio track we have the following information -> C, D. Is it possible, from any existing piece of software to rebuild the L stream? If so, can it be muxed into a Matroska container, and will any decoder understand it?

kypec

4th January 2011, 16:05

2. If we assume that C=core, D=Deltas and L=Lossless then we have L=C+D, and in each "DTS-HD Master Audio" audio track we have the following information -> C, D. Is it possible, from any existing piece of software to rebuild the L stream? If so, can it be muxed into a Matroska container, and will any decoder understand it?
:confused: What you have described here looks like regular decode process -> DTS-HD MA aware software decoder (ArcSoft for instance) decodes compressed data into lossless stream. This stream you can save as LPCM with eac3to and mux into MKV later of course.

therealjoeblow

4th January 2011, 18:24

TDiTP_

8th January 2011, 14:04

After this (http://forum.doom9.org/showthread.php?p=1462166#post1462166) post of tebasuna51 i decided to check all possible DTS-HD 7.1-layouts. I suppose that for all configurations Arcsoft makes correction for standard location of speakers in a 7.1 audio system and i suppose that we should use Wave_Format_Ex with MaskChannel = 0x063F for all layouts. I calculated the coefficients for every mix (some test samples were used and i have a DTS-HD M.A.S.).

I do not know the BD on which layouts from 1 till 5 were used but it can appear in the future.

=============================================
Results:

0).
http://thumbnails13.imagebam.com/11421/5f0592114200579.jpg (http://www.imagebam.com/image/5f0592114200579)
Not compliant with Wave_Format_Ex; eac3to/arcsoft decodes it to Standart Mask = 0x063F (FL FR FC LFE BL BR SL SR). Mix:

FL = 0.68039*L FR = 0.68039*R
FC = 0.68039*C LFE = 0.68039*LFE
SL = 0.60065*Ls SR = 0.60065*Rs
BL = 0.31953*Ls + 0.68039*Lsr BR = 0.31953*Rs + 0.68039*Rsr

In other words (in dBs terms):

FL = L(-3.3 dB) FR = R(-3.3 dB)
FC = C(-3.3 dB) LFE = LFE(-3.3 dB)
SL = Ls(-4.4 dB) SR = Rs(-4.4 dB)
BL = Ls(-9.9 dB) + Lsr(-3.3 dB) BR = Rs(-9.9 dB) + Rsr(-3.3 dB)

The two conditions were satisfied:

Decoder avoids overflow: sum of the coefficients for every speaker less or equal 1.
Decoder preserves acoustic power contribution (~0.4629) for every speaker:

L: 0.68039^2 = 0.4629 LFE: 0.68039^2 = 0.4629
R: 0.68039^2 = 0.4629 C: 0.68039^2 = 0.4629
Ls: 0.31953^2 + 0.60065^2 = 0.4629 Rs: 0.31953^2 + 0.60065^2 = 0.4629
Lsr: 0.68039^2 = 0.4629 Rsr: 0.68039^2 = 0.4629

So.. this is correct mix.

1).
http://thumbnails35.imagebam.com/11421/b0499e114200582.jpg (http://www.imagebam.com/image/b0499e114200582)
Not compliant with Wave_Format_Ex; eac3to/arcsoft decodes it to Standart Mask = 0x063F (FL FR FC LFE BL BR SL SR). Mix:

FL = 0.58579*L + 0.41421*Lw FR = 0.58579*R + 0.41421*Rw
FC = 0.58579*C LFE = 0.58579*LFE
SL = 0.51713*Ls + 0.41421*Lw SR = 0.51713*Rs + 0.41421*Rw
BL = 0.27514*Ls BR = 0.27514*Rs

In other words (in dBs terms):

FL = L(-4.6 dB) + Lw(-7.6 dB) FR = R(-4.6 dB) + Rw(-7.6 dB)
FC = C(-4.6 dB) LFE = LFE(-4.6 dB)
SL = Ls(-5.7 dB) + Lw(-7.6 dB) SR = Rs(-5.7 dB) + Rw(-7.6 dB)
BL = Ls(-11.2 dB) BR = Rs(-11.2 dB)

The two conditions were satisfied:

Decoder avoids overflow: sum of the coefficients for every speaker less or equal 1.
Decoder preserves acoustic power contribution (~0.3431) for every speaker:

L: 0.58579^2 = 0.3431 LFE: 0.58579^2 = 0.3431
R: 0.58579^2 = 0.3431 C: 0.58579^2 = 0.3431
Lw: 0.41421^2 + 0.41421^2 = 0.3431 Ls: 0.51713^2 + 0.27514^2 = 0.3431
Rw: 0.41421^2 + 0.41421^2 = 0.3431 Rs: 0.51713^2 + 0.27514^2 = 0.3431

I do not give a similar calculations for the other layouts below, you can double-check the figures yourselves

This is correct mix too.

2).
http://thumbnails37.imagebam.com/11421/1e0aaf114200585.jpg (http://www.imagebam.com/image/1e0aaf114200585)
This layout is compatible with Wave_Format_Ex with Mask = 0x560F (FL FR FC LF SL SR TFL TFR), but eac3to/arcsoft decodes it to Mask = 0x063F. Mix:

FL = 0.58579*L + 0.41421*Lh FR = 0.58579*R + 0.41421*Rh
FC = 0.58579*C LFE = 0.58579*LFE
SL = 0.51713*Ls SR = 0.51713*Rs
BL = 0.27514*Ls BR = 0.27514*Rs

In other words (in dBs terms):

FL = L(-4.6 dB) + Lh(-7.6 dB) FR = R(-4.6 dB) + Rh(-7.6 dB)
FC = C(-4.6 dB) LFE = LFE(-4.6 dB)
SL = Ls(-5.7 dB) SR = Rs(-5.7 dB)
BL = Ls(-11.2 dB) BR = Rs(-11.2 dB)

Decoder avoids overflow and preserves acoustic power contribution (~0.3431) for all speakers except Lh, Rh (~0.1716).

3).
http://thumbnails27.imagebam.com/11421/9e87ac114200587.jpg (http://www.imagebam.com/image/9e87ac114200587)
Not compliant with Wave_Format_Ex; eac3to/arcsoft decodes it to Standart Mask = 0x063F (FL FR FC LFE BL BR SL SR). Mix:

FL = 0.62897*L FR = 0.62897*R
FC = 0.62897*C LFE = 0.62897*LFE
SL = 0.55525*Ls + 0.44474*Lhs SR = 0.55525*Rs + 0.44474*Rhs
BL = 0.29543*Ls BR = 0.29543*Rs

In other words (in dBs terms):

FL = L(-4.0 dB) FR = R(-4.0 dB)
FC = C(-4.0 dB) LFE = LFE(-4.0 dB)
SL = Ls(-5.1 dB) + Lhs(-7.0 dB) SR = Rs(-5.1 dB) + Rhs(-7.0 dB)
BL = Ls(-10.6 dB) BR = Rs(-10.6 dB)

Decoder avoids overflow and preserves acoustic power contribution (~0.3956) for all speakers except Rhs, Lhs (~0.1978).

4).
http://thumbnails19.imagebam.com/11421/729b9f114200588.jpg (http://www.imagebam.com/image/729b9f114200588)
Compliant with Wave_Format_Ex, is decoded by eac3to/arcsoft to Mask = 0x0F0F (FL FR FC LFE BC SL SR TC).

FL = 0.58579*L FR = 0.58579*R
FC = 0.62897*C + 0.41421*Ch LFE = 0.58579*LFE
SL = 0.27514*Ls + 0.41421*Cs SR = 0.27514*Rs + 0.41421*Cs
BC = 0.51713*Ls TC = 0.51713*Rs

In other words (in dBs terms):

FL = L(-4.6 dB) FR = R(-4.6 dB)
FC = C(-4.0 dB) + Ch(-7.6 dB) LFE = LFE(-4.6 dB)
SL = Ls(-11.2 dB) + Cs(-7.6 dB) SR = Rs(-11.2 dB) + Cs(-7.6 dB)
BC = Ls(-5.7 dB) TC = Rs(-5.7 dB)

Decoder avoids overflow and preserves acoustic power contribution (~0.3433) for all speakers except C, Ch (~0.3956 and ~0.1716).

Looks like ArcSoft decodes as for standard MaskChannel = 0x063F and eac3to uses incorrect Mask.
After such decoding we need rename channels: SL->BL, SR->BR, BC->SL, TC->SR and it will be for "standard location of speakers in a 7.1 audio system".

5).
http://thumbnails16.imagebam.com/11421/52856f114200589.jpg (http://www.imagebam.com/image/52856f114200589)
Compliant with Wave_Format_Ex, is decoded by eac3to/arcsoft to Mask = 0x270F (FL FR FC LF BC SL SR TFC).

FL = 0.72317*L FR = 0.72317*R
FC = 0.72317*C LFE = 0.72317*LFE
SL = 0.33967*Ls + 0.51135*Cs SR = 0.33967*Rs + 0.51135*Cs
BC = 0.63841*Ls + 0.36158*Oh TFC = 0.63841*Rs + 0.36158*Oh

In other words (in dBs terms):

FL = L(-2.8 dB) FR = R(-2.8 dB)
FC = C(-2.8 dB) LFE = LFE(-2.8 dB)
SL = Ls(-9.4 dB) + Cs(-5.8 dB) SR = Rs(-9.4 dB) + Cs(-5.8 dB)
BC = Ls(-3.9 dB) + Oh(-8.8 dB) TFC = Rs(-3.9 dB) + Oh(-8.8 dB)

Decoder avoids overflow and preserves acoustic power contribution (~0.5230) for all speakers except Oh (~0.2615).

Again, looks like ArcSoft decodes as for standard MaskChannel = 0x063F and eac3to uses incorrect Mask.
After such decoding we need rename channels: SL->BL, SR->BR, BC->SL, TFC->SR and it will be for "standard location of speakers in a 7.1 audio system".

=============================================
Conclusion:

For all layouts Arcsoft makes correction for standard location of speakers in a 7.1 audio system and i suppose that we should use Wave_Format_Ex with MaskChannel = 0x063F for all layouts and remap (rename) channels for some layouts.

BTW, ArcSoft 1.1.0.0 can decode "normal setup" (-logdts: $84b), but it can't decode 16-bit non-standart 7.1-layouts correctly (noise in FL and FR channels). ArcSoft 1.1.0.7 and 1.1.0.8 decodes all 7.1-layouts correctly.

Taddeusz

8th January 2011, 14:35

I have a movie, Edward Scissorhands, which has a DTS-HD MA 4.0ch track. Trying to convert the track to Dolby Digital but I receive the error:

a03 The AC3 encoder received a non-supported data format (pcm, 4, 24, -).

Can libaften not encode a 4ch track with no LFE?

Thunderbolt8

8th January 2011, 16:12

BTW, ArcSoft 1.1.0.0 can decode only "normal setup" (-logdts: $84b). ArcSoft 1.1.0.7 and 1.1.0.8 decode all configurationis this really true? I guess that would be the first thing the other 2 versions can do better than 1.1.0.0. guess, in that case we'd need to switch between different versions then.

TDiTP_

9th January 2011, 07:01

is this really true? I guess that would be the first thing the other 2 versions can do better than 1.1.0.0. guess, in that case we'd need to switch between different versions then.

Yes, we need switch in this situation.
In case of non-standart scheme decoder 1.1.0.0 give garbage in some channels. So, i tried other two versions and they are decoded properly. If you have DTS-HD M.A.S. you can try yourself. If not, try decode test sample (http://www.mediafire.com/?kagbxe5zg7tdzg0) of DTS-HD 7.1 (scheme 2) and compare results.
AFAIK, there are only two differences between 1.1.0.0 and 1.1.0.8:
- 1.1.0.0 can decode DTS(-HD) 6.1/6.0 but can't decode non-standart 7.1
- 1.1.0.8 can't decode DTS(-HD) 6.1/6.0 but can decode non-standart 7.1
Both decode DTS(-HD) 1.0 correctly, unlike 1.1.0.7.

Midzuki

10th January 2011, 01:29

<SNIP>

AFAIK, there are only two differences between 1.1.0.0 and 1.1.0.8:
- 1.1.0.0 can decode DTS(-HD) 6.1/6.0 but can't decode non-standart 7.1
- 1.1.0.8 can't decode DTS(-HD) 6.1/6.0 but can decode non-standart 7.1
Both decode DTS(-HD) 1.0 correctly, unlike 1.1.0.7.

argh :(

deathlord

10th January 2011, 19:59

Yes, we need switch in this situation.
In case of non-standart scheme decoder 1.1.0.0 give garbage in some channels. So, i tried other two versions and they are decoded properly. If you have DTS-HD M.A.S. you can try yourself. If not, try decode test sample (http://www.mediafire.com/?kagbxe5zg7tdzg0) of DTS-HD 7.1 (scheme 2) and compare results.
AFAIK, there are only two differences between 1.1.0.0 and 1.1.0.8:
- 1.1.0.0 can decode DTS(-HD) 6.1/6.0 but can't decode non-standart 7.1
- 1.1.0.8 can't decode DTS(-HD) 6.1/6.0 but can decode non-standart 7.1
Both decode DTS(-HD) 1.0 correctly, unlike 1.1.0.7.

Maybe madshi could make it so eac3to automatically takes the correct decoder for each format, provided the right version is present?

GollyJer

10th January 2011, 21:01

Is there any way to get eac3to to use the MPC-HC (Gabest) splitter instead of Haali?

Snowknight26

11th January 2011, 01:40

It doesn't use any splitters. It only uses the MKV muxer that comes packaged with Haali Media Splitter.

harry.rb

11th January 2011, 09:34

I have Nero 7 (premium) installed, along with the BluRay/HDDVD Plugin, they show as being properly licensed in the Nero ProductSetup tool, on Win7x86; however eac3to -test still reports that "Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed".
This same version was working fine with no tweaks on WinXP, but not after a clean install of Win7 and Nero.
Can someone please enlighten me as to how to get eac3to to recognize the decoder on Win7x86?

I had the same problem with win7. After a lot of reading on internet en trying things i finaly got it working. Try this:
Install your Nero 7 on the c: drive. Register theBluRay/HD plugin. Copy or install the latest Eac3to to c:\eac3to.
Find the files: AdvrCntr2.dll, NeAudio2.ax and neEacDec.dll in c:\program files\common files\ahead\lib and c:\program files\common files\ahead\dsfilter, and copy those files to the eac3to directory. The file NeAudio2.ax wasn't installed on my win7 pc, i copied the file from my XP pc. If you don't have the file e-mail me and i can send it. Then run cmd.exe with administrator rights (rightclick mouse). In the command prompt go to the c:\eac3to directory and register the 3 files: "regsvr32 advrCntr2.dll" and so on for the other 2. Then do eac3to -test and see if its working....Let us know if it is working....

xkodi

12th January 2011, 17:16

AFAIK, there are only two differences between 1.1.0.0 and 1.1.0.8:
- 1.1.0.0 can decode DTS(-HD) 6.1/6.0 but can't decode non-standart 7.1
- 1.1.0.8 can't decode DTS(-HD) 6.1/6.0 but can decode non-standart 7.1
Both decode DTS(-HD) 1.0 correctly, unlike 1.1.0.7.

i'm busy at the moment and i can't check it myself, but i'm wondering how exactly people who are reporting those results check to confirm them and if they checked every single byte that the decoded information is bit-by-bit perfect.

also, TDiTP asked me in private message to check again my results reported here:

http://forum.doom9.org/showthread.php?p=1426223#post1426223

which i'm very sure about, because i checked them many times before, but since several people claimed using the "fake" header approach results in bit-by-bit perfect decode, i'm going to check it again.

in my opinion, maybe, it's good idea several people who are willing to do the work, to collaborate, prepare all possible DTS-HD (MA) samples and build a table with what is working and what not and with which version of which decoder, etc. but those people have to be credible and to know what they are doing, because otherwise it's pointless and there are enough speculations already.

[EDIT] please, stop speculating - i wasted 15-20 minutes to check:

1.1.0.8 ... but can decode non-standart 7.1

and it's totally not true and if someone claims otherwise, i advice him/her to first learn how to use hex-editor before try to mislead all of us with false information.

my test was run on the following:

"DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 96kHz"

and in fact 1.1.0.8 output is the same as 1.1.0.0

TDiTP_

15th January 2011, 16:06

Please, stop your panic.

1.

[EDIT] please, stop speculating - i wasted 15-20 minutes to check:

Quote:
1.1.0.8 ... but can decode non-standart 7.1
and it's totally not true and if someone claims otherwise, i advice him/her to first learn how to use hex-editor before try to mislead all of us with false information.

my test was run on the following:

"DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 96kHz"

and in fact 1.1.0.8 output is the same as 1.1.0.0

Where did i say anything about "DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 96kHz" ? In this case indeed results are identical. i can confirm it.
But i was talking about "DTS Master Audio, 7.1 (strange setup) channels, 16 bits, 48kHz". In this case results are not identical and 1.1.0.0 produces bad PCM (FL and FR channels).
For those who has any doubt i can encode DTS in DTS-HD M.A.S. from any test WAVs and upload it to any fileshare. Everyone can double-check it.

i think the difference is due to 16-bit and 24-bit. 24-bit decoded by 1.1.0.0 as by 1.1.0.8 - ok; 16-bit DTS-HD 7.1 preperly decoded only by 1.1.0.0.
2.

also, TDiTP asked me in private message to check again my results reported here:

http://forum.doom9.org/showthread.ph...23#post1426223

which i'm very sure about, because i checked them many times before, but since several people claimed using the "fake" header approach results in bit-by-bit perfect decode, i'm going to check it again.

Very nice that you are talking about it here. I'm sure that you're mistaken at least in the case 5.1. I checked the 3 BD-discs with DTS-HD MA 5.1 (but I think that result with 7.1 would be the same, i will be able to confirm it later) and WAVs (StreamPlayer) were identical WAVs (ArcSoft, Sonic). i don't know how to prove it, do you want me to screen HexEditor for you? i'm not the only one who disagrees with the results of your test.

3.
I made only one mistake. Accept my apologies.
ArcSoft 1.1.0.8 can't normal decode DTS(-HD) 1.0. Only ArcSoft 1.1.0.0 can do it.

4.
New information. For DTS (5.1 16/48 1510) [sample (http://www.mediafire.com/?tgfa7uc0kx8iqtl)] i think
- libavcodec, NicDtsSource and Tranzcode doesn't correct decode DTS (5.1, 16-bit, 48 kHz) - too low level of LFE. In case of NicDtsSource and Tranzcode also other channels have lower (~0.5 dB) level.
- ArcSoft 1.1.0.0 produce result differs from 1.1.0.8.
Here comparison table with some other dts-decoders and Decoder of DTS-HD StreamPlayer (reference):

Total RMS Power: SNR for L.wav:
LFE.wav L.wav

DTS.inc reference decoder -25.35 dB -19.37 dB 57.2 dB (Gain for File B = 0.99994)
ArcSoft(1.1.0.0)-eac3to -25.35 dB -19.37 dB 57.2 dB (Gain for File B = 0.99994)
ArcSoft(1.1.0.8)-eac3to -25.35 dB -19.37 dB 57.2 dB (Gain for File B = 0.99994)
Sonic(4.3.0.169)-eac3to -25.35 dB -19.37 dB 57.2 dB (Gain for File B = 0.99994)
Nero7-eac3to -25.35 dB -19.37 dB 57.2 dB (Gain for File B = 0.99994)
libav-eac3to -28.11 dB -19.37 dB 56.346 dB (Gain for File B = 1.0002)
libav-eac3to.-full -28.11 dB -19.37 dB 56.346 dB (Gain for File B = 1.0002)
libav-ffmpeg -28.11 dB -19.37 dB 56.32 dB (Gain for File B = 1.0002)
NicDtsSource-BeHappy -28.11 dB -19.88 dB 56.346 dB (Gain for File B = 1.0608)
Tranzcode 0.40b -28.11 dB -19.88 dB 56.346 dB (Gain for File B = 1.0608)

Average frequency graphs:
http://thumbnails9.imagebam.com/11524/e367a9115231836.jpg (http://www.imagebam.com/image/e367a9115231836) http://thumbnails32.imagebam.com/11524/a7a3b0115231842.jpg (http://www.imagebam.com/image/a7a3b0115231842) http://thumbnails34.imagebam.com/11524/ad94cf115231850.jpg (http://www.imagebam.com/image/ad94cf115231850)
It's all right with frequencies, but in case of libav-ffmpeg one can see a bit higher level after audio bandwidth (~22.7 kHz).

WAV of Arcsoft 1.1.0.8, Sonic and Nero are identical byte-in-byte (i'm using HxD (http://mh-nexus.de/en/hxd/)) to each other and not idential to Arcsoft 1.1.0.0.

Again, i can encode to everyone any DTS from his sample-WAVs for recheck.

DivxBr

15th January 2011, 17:42

xkodi

16th January 2011, 12:48

Please, stop your panic.
.........

hehe, i'm far away from being panicked, because at least i know well what i'm doing and i'm going to spend time explaining to you why you're wrong again, only because i see that you're at least interested in the subject.

first of all here:

File name: 71_test_wavs.rar File size: 344.99 KB (http://www.fileserve.com/file/7mcEfae)

i uploaded 8 mono 16-bit, 48 kHz WAV files that are simple enough samples to make the testing both easy and conclusive, the files are as follows:

* "8_Channels_CH0_L.wav" : contains the sound of pronouncing the words "Front Left", the rest is silence or in hex editor that is zeroes

* "8_Channels_CH1_R.wav" : contains the sound of pronouncing the words "Front Right", the rest is silence or in hex editor that is zeroes

* "8_Channels_CH2_C.wav" : contains the sound of pronouncing the word "Center", the rest is silence or in hex editor that is zeroes

* "8_Channels_CH3_LFE.wav" : contains short bass sound at last few seconds, the rest is silence or in hex editor that is zeroes

* "8_Channels_CH4_BL.wav" : contains the sound of pronouncing the words "Back Left", the rest is silence or in hex editor that is zeroes

* "8_Channels_CH5_BR.wav" : contains the sound of pronouncing the words "Back Right", the rest is silence or in hex editor that is zeroes

* "8_Channels_CH6_SL.wav" : contains the sound of pronouncing the words "Side Left", the rest is silence or in hex editor that is zeroes

* "8_Channels_CH7_SR.wav" : contains the sound of pronouncing the words "Side Right", the rest is silence or in hex editor that is zeroes

as i mentioned those WAVs are perfect for testing for at least 2 reasons - first most of them is silence and the sound is short and easy to distinguish, which allows to understand what is decoded, because most of the file bytes are zeroes and that makes it really very easy to compare source WAV and decoded WAV in hex editor; second - the sound in each WAV is on different time, which also allows to understand better what is decoded

so, here is "DTS Master Audio, 7.1 (strange setup) channels, 16 bits, 48kHz" sample encoded by the above WAVs:

File name: dts_hd_ma_strange_setup_test.rar File size: 848.70 KB (http://www.fileserve.com/file/fuc9dJu)

now, if you decode the above DTS-MA sample with Arcsoft 1.1.0.0 and 1.1.0.8 you will get:

1) Left ("L") channel decoded by Arcsoft 1.1.0.0 =/= Left ("L") channel decoded by Arcsoft 1.1.0.8: the sound decoded by Arcsoft 1.1.0.0 is just some distorted noise on the same time frame as words "Front Left" of the source WAV, the sound decoded by Arcsoft 1.1.0.8 is indeed the words "Front Left", but they are not decoded bit-perfectly to the original source WAV

2) Right ("R") channel decoded by Arcsoft 1.1.0.0 =/= Right ("R") channel decoded by Arcsoft 1.1.0.8: the sound decoded by Arcsoft 1.1.0.0 is just some distorted noise on the same time frame as words "Front Right" of the source WAV, the sound decoded by Arcsoft 1.1.0.8 is indeed the words "Front Right", but they are not decoded bit-perfectly to the original source WAV

3) Center ("C") channel decoded by Arcsoft 1.1.0.0 == Center ("C") channel decoded by Arcsoft 1.1.0.8: indeed the sound is the word "Center" on the same time frame as the source WAV, but not decoded bit-perfectly to the original source WAV

4) LFE channel decoded by Arcsoft 1.1.0.0 == LFE channel decoded by Arcsoft 1.1.0.8: indeed the sound is the bass sound on the same time frame as the source WAV, but not decoded bit-perfectly to the original source WAV

5) Back Left ("BL") channel decoded by Arcsoft 1.1.0.0 == Back Left ("BL") channel decoded by Arcsoft 1.1.0.8: the sound is words "Back Left, Side Left" or Arcsoft mixed the BL and SL of the source WAVs during the decoding. also, the decoded information is not bit-perfect to the original source WAVs

6) Back Right ("BR") channel decoded by Arcsoft 1.1.0.0 == Back Right ("BR") channel decoded by Arcsoft 1.1.0.8: the sound is words "Back Right, Side Right" or Arcsoft mixed the BR and SR of the source WAVs during the decoding. also, the decoded information is not bit-perfect to the original source WAVs

7) Side Left ("SL") channel decoded by Arcsoft 1.1.0.0 == Side Left ("SL") channel decoded by Arcsoft 1.1.0.8: the sound is words "Back Left" or of course not bit-perfect to SL source WAV as well as not even bit-perfect to BL source WAV

8) Side Right ("SR") channel decoded by Arcsoft 1.1.0.0 == Side Right ("SR") channel decoded by Arcsoft 1.1.0.8: the sound is words "Back Right" or of course not bit-perfect to SR source WAV as well as not even bit-perfect to BR source WAV

so, i hope now you understand better what i'm talking about. also, here are the decoded WAVs for your reference respectively by Arcsoft 1.1.0.0 and 1.1.0.8:

File name: 1100.rar File size: 302.04 KB (http://www.fileserve.com/file/T9CkVXb)

File name: 1108.rar File size: 389.27 KB (http://www.fileserve.com/file/vz2ZtSe)

in very short there is no version of Arcsoft decoder that can handle DTS-MA 7.1 "strange setup" files in bit-perfect way, i agree that there is small improvement regarding "L" and "R" channel with version 1.1.0.8, but still the decoded information is not correct in terms of bit-perfect decode. the only way to get bit-perfect decode from DTS-MA 7.1 "strange setup" files is to use Sonic decoder and convert it to correct 5.1 WAV file as i tried to explain here:

http://forum.doom9.org/showthread.php?p=1443085#post1443085

(and in some of the subsequent posts).

if you want me to check something more or have objections, let me know, i believe it's in interest of the whole community to have clarity what and how works and what doesn't.

TinTime

16th January 2011, 14:04

Hi,
I tried hard to find an answer, but couldn't.
Is there any way to overcome the 15 minutes limitation (only are shown the 'mpls' with more than 15 min) when scanning playlists in BD folders?
Thanks.

Not directly. I wrote a batch script to call eac3to for each playlist and output the lengths of each to a text file. Possibly one of the GUIs for eac3to might offer similar functionality? I'm not sure.

DivxBr

16th January 2011, 18:04

Not directly. I wrote a batch script to call eac3to for each playlist and output the lengths of each to a text file. Possibly one of the GUIs for eac3to might offer similar functionality? I'm not sure.

Yes, I'd tried this, but when I make this, in some disks, I get some odds and/or fake responses, like this:

1) 00643.mpls, 2:12:38
[643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643+643+643+643+643
+643+643+643+643+643+643+643+643+643].m2ts
- Chapters, 451 Chapters
- h264/AVC, 1080p24 /1.001 (16:9)

who, despite the fact that it shows a duration of more than 15min isn't show (correctly) by eac3to when I use it with the whole folder.

quartex

17th January 2011, 11:51

Hi,
Since I have experienced BD-R degradation (http://forum.doom9.org/showthread.php?p=1471875), I' now demuxing again all my BD-R (to check if they are good or not). The BD-R content is the result of demux/remux (using eac3to and tsmuxer). So I'm running eac3to against these BD-R contents.

I've noticed that sometime I get this warning message:
The video framerate is correct, but rather unusual. <WARNING>

and sometime this:
Audio overlaps for xx ms at playtime xx

Is something to worry about?

In the degraded BD-R I get
the source file seems to be damaged (sync byte missing)
and when I get this, I'm sure the BD-R is bad

What about the other warning messages?
Thanks!

rapscallion

18th January 2011, 23:11

For the first time I'm demuxing an LPCM audio track.

When selected, eac3to identifies the audio track as "raw/pcm".

So, in order to retain the original audio, for remuxing later, which output to I select, LPCM, PCM, Raw, or WAV ?

Edit: Or, should I just select "wavs" and then re encode them to DTS-MA ?

Thunderbolt8

19th January 2011, 17:41

guess that depends if you want to have it as dtsma or not. for pure sake of being lossless, it shouldnt matter.

rapscallion

19th January 2011, 17:48

Thanks. I played around with it and did demux to wavs and re encode to MA.

Significant space saving vs pcm, with really no loss of quality.

Lyle_JP

19th January 2011, 21:15

Thanks. I played around with it and did demux to wavs and re encode to MA.

Significant space saving vs pcm, with really no loss of quality.

If you had re-encoded to TrueHD you would have saved even more space with identical quality.

rapscallion

19th January 2011, 21:22

If you had re-encoded to TrueHD you would have saved even more space with identical quality.
Except that I don't have a True HD professional encoder.
What would you use for results equivalent to DTD-MAS ?

The Dolby Media Producer is $3995.00 vs DTS-MAS @ $ 1495.00.
Plus , the Dolby Suite only runs on Mac.

Adub

19th January 2011, 21:25

FLAC possibly, but that is only useful if you are going to keep your output on your computer, or just want to use it as a intermediary format.

rapscallion

19th January 2011, 21:27

FLAC possibly, but that is only useful if you are going to keep your output on your computer, or just want to use it as a intermediary format.

Which I'm not.

I'm backing up to BD25/50 for playback on a SA BD player.

Adub

19th January 2011, 21:30

Okay, that's what I thought. There really aren't that many audio formats that you can choose from, as they are inherently limited by the Blu-ray spec.

martyseilh

20th January 2011, 01:42

"And eac3to will no longer do anything to the "fixed" files."

I agree because when I am using eac3to i thought i did something wrong but the problem is eac3to is not really fixing files.

----------------
Auto Traffic Monopoly System (http://www.squidoo.com/automate-auto-traffic-monopoly-review)

jmone

20th January 2011, 06:28

Hi - just a clarification on DD+ (E-AC3) to FLAC conversion as I've a bunch of HD-DVD that is time to rip to MKV. Anyway, madshi's opening post describes that the Nero 7 filters are "best" followed by libav, yet I've read other posts in this thread that say don't bother with Nero 7 anymore as the Arcsoft HD Audio filters are best.
Thanks
Nathan

FYI
- I own a lic for Arcsoft TMT 3 and 5 and eac3to sees this as working fine
- I own a lic for Nero 7 Premium but eac3to reports that there is somehting wrong (presumably it is that the "Nero HD DVD / Blu-Ray plugin" need to be isntalled and activated)

TDiTP_

20th January 2011, 07:42

xkodi. Sorry, i just now saw your post. almost missed.

i'm doing and i'm going to spend time explaining to you why you're wrong again, only because i see that you're at least interested in the subject.
So.. where i'm was wrong? nowhere. you got the same results that i. i didn't use word "bit-perfect", i know about "strange setup". Did you read my previous post? http://forum.doom9.org/showthread.php?p=1469401#post1469401
i agree with tebasuna51 that for all scheme Arcsoft make correction for standard location of speakers in a 7.1 audio system. And wrote that correct mix can't be obtrained with 1.1.0.0. it's all :) . Your experiment confirms this.

there is no version of Arcsoft decoder that can handle DTS-MA 7.1 "strange setup" files in bit-perfect way, i agree that there is small improvement regarding "L" and "R" channel with version 1.1.0.8, but still the decoded information is not correct in terms of bit-perfect decode. the only way to get bit-perfect decode from DTS-MA 7.1 "strange setup" files is to use Sonic decoder and convert it to correct 5.1 WAV file as i tried to explain here

main problem, i think, that you didn't see this (http://forum.doom9.org/showthread.php?p=1469401#post1469401) my post. There i gave a link to your message (http://forum.doom9.org/showthread.php?p=1443085#post1443085) too.

if you want me to check something more or have objections, let me know, i believe it's in interest of the whole community to have clarity what and how works and what doesn't.
let's talk about StreamPlayer Decoder. In other thread.

PS. May be small bug in eac3to. "-shutdown" doesn't work. not only on my PC.

jmone

20th January 2011, 08:31

Another Q - Any opinions on how E-AC3 at 1,536kbps compares with DTS at 1,536kbps as a target (instead of FLAC) for the conversion of these HD-DVD audio tracks?

nurbs

20th January 2011, 09:24

AC3 at 640 kbps should give the same perceptual quality as DTS at 1.5 Mbps. E-AC3 should have better quality than AC3 at the same bitrate. Make of that what you will.

WonderCsabo

20th January 2011, 17:15

Hi guys!

Noob question: i have an mkv file and a H.264 and DTS-HD MA stream within. Mediainfo says the length is 1m 1s 229ms.
I demux the DTS-HD MA stream with eac3to, then Mediainfo says the demuxed stream is 1m 1s 472ms.

Why does it happen?

TheLion

22nd January 2011, 16:12

Please help!

I am about to rip my Blu-Ray collection to mkv files using eac3to. (Anydvd, transcoding to flac and mkv for video, all the latest revisions - Arcsoft DTS decoder 1.1.0.8 (1.1.0.0 has the same results)

Lately a lot of discs failed to be processed by eac3to. Examples are The Expendables and Aliens (Alien 1 is processed without a hitch).

See the following log as an example:

C:\eac3to>eac3to d: 1) 1: c:\alien\b.txt 2: c:\alien\b.mkv 3: c:\alien\b.flac 8:
c:\alien\bger.flac 13: c:\alien\b.sup
M2TS, 1 video track, 10 audio tracks, 18 subtitle tracks, 2:34:26, 24p /1.001
1: Chapters, 44 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
4: AC3, English, 3/1.1 channels, 448kbps, 48kHz, dialnorm: -27dB
5: AC3 Surround, English, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
6: AC3, Portuguese, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
7: DTS, French, 5.1 channels, 24 bits, 768kbps, 48kHz
8: DTS, German, 5.1 channels, 24 bits, 768kbps, 48kHz
9: AC3, Spanish, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
10: AC3, English, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
11: AC3, English, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
12: AC3, English, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
13: Subtitle (PGS), English
14: Subtitle (PGS), Portuguese
15: Subtitle (PGS), Danish
16: Subtitle (PGS), Finnish
17: Subtitle (PGS), French
18: Subtitle (PGS), German
19: Subtitle (PGS), Spanish
20: Subtitle (PGS), Dutch
21: Subtitle (PGS), Norwegian
22: Subtitle (PGS), Swedish
23: Subtitle (PGS), English
24: Subtitle (PGS), Danish
25: Subtitle (PGS), Finnish
26: Subtitle (PGS), French
27: Subtitle (PGS), German
28: Subtitle (PGS), Dutch
29: Subtitle (PGS), Norwegian
30: Subtitle (PGS), Swedish
Creating file "c:\alien\b.txt"...
s13 Extracting subtitle track number 13...
v02 Extracting video track number 2...
v02 Muxing video to Matroska...
a08 Extracting audio track number 8...
a08 Decoding with ArcSoft DTS Decoder...
a03 Extracting audio track number 3...
a08 Encoding FLAC with libFlac...
a03 Decoding with ArcSoft DTS Decoder...
a03 Encoding FLAC with libFlac...
a08 Creating file "c:\alien\bger.flac"...
a03 Creating file "c:\alien\b.flac"...
s13 Creating file "c:\alien\b.sup"...
s13 0:51:25 The source file seems to be damaged (sync byte missing).
v02 0:54:54 The source file seems to be damaged (sync byte missing).
a08 0:54:54 The source file seems to be damaged (sync byte missing).
a03 0:54:54 The source file seems to be damaged (sync byte missing).
s13 0:51:25 The source file seems to be damaged (sync byte missing).
v02 0:54:54 The source file seems to be damaged (discontinuity).
a08 0:54:54 The source file seems to be damaged (sync byte missing).
a03 0:54:54 The source file seems to be damaged (sync byte missing).
s13 0:51:25 The source file seems to be damaged (sync byte missing).
v02 0:54:54 The source file seems to be damaged (sync byte missing).
a08 0:54:54 The source file seems to be damaged (sync byte missing).
a03 0:54:54 The source file seems to be damaged (sync byte missing).
s13 0:51:25 The source file seems to be damaged (sync byte missing).
v02 0:54:54 The source file seems to be damaged (sync byte missing).
a08 0:54:54 The source file seems to be damaged (sync byte missing).
a03 0:54:54 The source file seems to be damaged (sync byte missing).
s13 0:51:25 The source file seems to be damaged (sync byte missing).
v02 0:54:54 The source file seems to be damaged (sync byte missing).
a08 0:54:54 The source file seems to be damaged (sync byte missing).
a03 0:54:54 The source file seems to be damaged (sync byte missing).
s13 0:51:25 The source file seems to be damaged (sync byte missing).
v02 0:54:54 The source file seems to be damaged (sync byte missing).
a08 0:54:54 The source file seems to be damaged (sync byte missing).
a03 0:54:54 The source file seems to be damaged (sync byte missing).
v02 0:54:54 The source file seems to be damaged (sync byte missing).
a08 0:54:54 The source file seems to be damaged (discontinuity).
a03 The ArcSoft DTS Decoder reported an error while decoding.
a08 The ArcSoft DTS Decoder reported an error while decoding.
Aborted at file position 15497293824.

What to do? Thanks!

TinTime

22nd January 2011, 18:48

Might be a read error. Can you rip the whole disc using AnyDVD? If so does eac3to show the same errors when running against the files on your HDD?

Thunderbolt8

22nd January 2011, 19:16

rerip the disc. Ive converted aliens just fine.

DVDBob

22nd January 2011, 20:16

Is there an alternative decoder for AAC, so I can avoid to install Nero???

Thunderbolt8

22nd January 2011, 21:24

just get it here and only use the corresponding .dll(? or .exe) http://www.nero.com/eng/downloads-nerodigital-nero-aac-codec.php

no need to install anything

DVDBob

22nd January 2011, 22:46

Ok thank you.

TheLion

23rd January 2011, 00:13

Might be a read error. Can you rip the whole disc using AnyDVD? If so does eac3to show the same errors when running against the files on your HDD?

Thank you for the response. I certainly tried to rip it on HD first using AnyDVD or tsMuxer (demuxing, transcoding into combined stream...) - all with the same result.

So this might be a problem with my BD-drive with certain seamless branching discs...I will try a firmwire update or something...

Mark_A_W

23rd January 2011, 00:48

I have to rip seamless branching discs to the hard drive before processing with eac3to. If I try and process direct from the disc I get these errors - for seamless branching titles, normal titles are ok from the disc.

I had the same issue - about 20 pages ago by now...

TheLion

23rd January 2011, 00:57

Mark_A_W

23rd January 2011, 02:34

Do you rip them with AnyDVD? Which drive do you have - mine is the LG BH10LS30?

Yes, Anydvd rip "VideoDVD to Harddisk"

Then use eac3to.

I have the LG Combo BD/HD-DVD drive GGC-H20L drive.

d9rook

24th January 2011, 21:26

I'm hoping someone can help me a little. Trying to take an E-AC3 file with random bit rates to a AC3 448kbps bit rate file for playback over a 360?

nurbs

24th January 2011, 22:43

What's the problem?
Basically the command you need to use is something like
eac3to input.eac3 ouput.ac3 -448

Cheshire Cat

25th January 2011, 05:27

Hi! I'm new using eac3to. I used to use different tools to do the job of converting DTS tracks to AC3 but they all seemed to have some problem/drawback.

I've been reading about eac and from what I've seen it does always introduce some kind of modification when processing the audio. More generally the dialog normalization, correct?

I would like to ask if isn't there any way the audio conversion can be made respecting the -db values of the original track? I mean, without changing the track from the original. Just transcoding. It is possible or with eac I'm stuck with normalization?

tebasuna51

25th January 2011, 11:47

...
I would like to ask if isn't there any way the audio conversion can be made respecting the -db values of the original track? I mean, without changing the track from the original. Just transcoding. It is possible or with eac I'm stuck with normalization?
eac3to respect the -db values of the original track. Just transcode.
The parameter -normalize is optional.

d9rook

26th January 2011, 04:09

What's the problem?
Basically the command you need to use is something like
eac3to input.eac3 ouput.ac3 -448

My original file is an mkv file. What is the command to rip the audio track from the mkv so that I have a seperate eac3 file? Or should I be ripping to a different format?I'm trying to end up with a wtv file. Thanks.

tebasuna51

26th January 2011, 10:20

If the eac3 audio is the T track in mkv:

eac3to T: input.mkv ouput.ac3 -448

nurbs

26th January 2011, 11:03

My original file is an mkv file. What is the command to rip the audio track from the mkv so that I have a seperate eac3 file? Or should I be ripping to a different format?I'm trying to end up with a wtv file. Thanks.
You type
eac3to input.mkv
This will list all the tracks in the mkv file with numbers next to them. Let's say your eac3 track is the second track in the file. Then you type
eac3to input.mkv 2:output.ac3 -448

d9rook

26th January 2011, 17:08

Thank you for the steps.

Tried, got the following error.

D:\eac3to>eac3to d:/movies/300/300.mkv 2:output.ac3 -448
MKV, 1 video track, 1 audio track, 1:56:33, 30p /1.001
1: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
2: EAC3, English, 5.1 channels, 48kHz, -83ms
"3/2+1"
Bitstream parsing for track 2 failed.
Demuxing this track may still produce correct results - or not.
This audio conversion is not supported.

D:\eac3to>

nurbs

26th January 2011, 17:41

I think your slashes are the wrong way. It should be "\", not "/". Also try putting the paths in brackets.
D:\eac3to>eac3to "d:\movies\300\300.mkv" 2:"d:\movies\300\output.ac3" -448

Snowknight26

26th January 2011, 20:59

It's most likely a mangled MKV or E-AC3 stream as eac3to has no problems parsing E-AC3.

d9rook

26th January 2011, 22:39

Thanks for the replies. I tried 3 other mkvs ripped using makemkv, all gave the identical error. Then I d/led ClownBD and used the gui and it gave the same error on all 3 as well. Should I rip with some other tool? has anyone specifically used Makemkv to take to mkv then eac3to to convert the audio on a HD-dvd?

nurbs

26th January 2011, 22:53

d9rook

28th January 2011, 20:53

You can try to extract the track first, for instance:
eac3to input.mkv 2:output.eac3
or
eac3to input.mkv -demux
and do the reencoding afterwards. There is also mkvextract if you want to demux the streams, but I haven't used that myself.
I personally have no experience with MakeMKV, so I can't help you specifically with that. I use AnyDVD HD, but there is no reason you shouldn't be able to get MakeMKV to work.

Also make sure that you have the latest version of eac3to. Maybe MakeMKV uses Header Removal Compresson, which eac3to only supports since version 3.23.

Ok I extracted the track as you said using -demux. Then I used the 300.eac3 300.ac3 -448 command. Then I muxed them back together to a .mkv. Is this the only way I will be able to do this? It seems like I doing something too many times.
Anyway to speed up the process? Thank you so much for the help, cant wait to see if this works on the 360 with no stutter.

d9rook

28th January 2011, 21:05

I used eac3to 300.vc1+300.ac3 300.mkv
Was this incorrect? Not hearing audio.

DVDBob

29th January 2011, 19:32

I hope someone will post a step by step guide to how i can get AAC decoding to work in eac3to???

d9rook

3rd February 2011, 17:35

You can try to extract the track first, for instance:
eac3to input.mkv 2:output.eac3
or
eac3to input.mkv -demux
and do the reencoding afterwards. There is also mkvextract if you want to demux the streams, but I haven't used that myself.
I personally have no experience with MakeMKV, so I can't help you specifically with that. I use AnyDVD HD, but there is no reason you shouldn't be able to get MakeMKV to work.

Also make sure that you have the latest version of eac3to. Maybe MakeMKV uses Header Removal Compresson, which eac3to only supports since version 3.23.

What do you recommend for reencoding? I got the streams seperate and tried to put em back together as a mkv but there is no sound..

nurbs

4th February 2011, 12:43

d9rook

4th February 2011, 15:32

I kind of lost track of the discussion, sorry. Could you sum up what you want to do and how far you got again?

Do you mean you got no sound after you reencoded the track ac3 and then muxed to a mkv again, or did you mux the eac3 track and that didn't play?

Overall goal: Take a HD-DVD disc to a format that plays stutterfree over the 360.

I rip the hd-dvd using Makemkv, the result is 1: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
2: EAC3, English, 5.1 channels, 48kHz, -83ms
"3/2+1"

This file stutters like crazy when I try to playback via 360.
I had to seperate then do eac3to input.eac3 ouput.ac3 -448
So now I have the vc1 file and ac3 file. What is the best way to put these back togther properly? My end desired format is .wtv so that I can ff/rw on the 360, but I dont think most people here are using 360 as extender so they dont have this issue. Thanks.

nurbs

4th February 2011, 16:01

I'm not familiar with WTV, but maybe this thread (http://thegreenbutton.com/forums/t/81225.aspx?PageIndex=1) can help. The way I see it you'd have to mux the files (video and ac3) via MKVMergeGUI (turn off header removal compression in the options to be safe) and then do step 6 in the thread I linked. That's assuming you can mux VC-1 the same way as H.264.

rapscallion

4th February 2011, 21:03

Sorry guys but couln't see an appropriate place to ask this question so I'm trying here.

I have 2 versions of the Blade Blu-ray. The Alliance (Canadian) release which is NOT the OAR (1.88:1),
but with an excellent DTS-MA track and the Dutch release which is the OAR, but with a low bitrate DTS-MA,
just barely above the core br.

So I want to mux the Alliance DTS with the Dutch video, so I can have the best of both.
However the movie lenghts are slightly different, by 3 sec !
The framerate on both is 23.976

The Dutch= 2.00.16.0.........Alliance=2.00.13.0

Obviously, when I mux the 2 together, the audio is out of sync (certainly not by 3 sec)

Is there a way to make this in sync ? Thanks

Adub

4th February 2011, 22:14

Uh, that looks like 3 milliseconds. You shouldn't be able to hear that.

Unless you wrote the time wrong?

rapscallion

5th February 2011, 00:02

Yes, I wrote it wrong...corrected. It is 3 sec difference

jj666

5th February 2011, 00:09

Given both tracks are lossless, can you hear a difference?

The way I'd do it would be to demux both tracks with EAC3TO, demux to .WAVs and waveform compare in Audition to verify either an initial +/- delay and/or cuts necessary to sync.

Cheers,

-jj-

rapscallion

5th February 2011, 01:21

Thanks, good suggestion jj.

Difference? I'm not sure, maybe it's just psycological but one track is ~1576 kps, 16 bit, while the other is ~ 3700 kbps, 24 bit. Hard to believe that the first one is truly lossless, but I guess when I demux to wavs I'll see.

I've been playing around with the track sync in VLC and +900 ms puts it just about spot on, even when I slow down playback. But if I can get it exact in Audition, I'll give it a shot.

Edit : pm sent

kypec

7th February 2011, 12:50

I usually open both video sources via Avisynth -> StackHorizontal(a,b) and time/scene shifts can be immediately seen. I then cut/append blank frames as necessary while leaving audio track untouched.

rapscallion

7th February 2011, 23:24

I usually open both video sources via Avisynth -> StackHorizontal(a,b) and time/scene shifts can be immediately seen.

Thanks, but that's something I don't know how to do.

kypec

8th February 2011, 07:47

Well, then it's about time to start learning :p

Install Avisynth (http://sourceforge.net/projects/avisynth2/files/AviSynth%202.5/AviSynth%202.5.8/)
Download some universal source filter, I'd suggest FFMpegSource (http://forum.doom9.org/showthread.php?t=127037)
Download AvsPMod (http://forum.doom9.org/showthread.php?t=153248) or VirtualDub
Open AVS scriptLoadPlugin("path\to\plugins\ffms2.dll")
a=FFVideoSource("first video.mkv")
b=FFVideoSource("second video.mkv")
StackHorizontal(a,b)

rapscallion

9th February 2011, 23:07

I apologize again for the OT of this subject.

Thanks for the info kypec.
Already use #'s 1 & 2.

I've heard of VirtualDub but never used it because it doesn't support *.m2ts video sources, which is the subject of my op.

Midzuki

10th February 2011, 01:51

I've heard of VirtualDub but never used it because it doesn't support *.m2ts video sources,

I am sure phaeron won't mind if you write a .(M2)TS source plugin :)
Watches pelcome :D

kypec

10th February 2011, 06:52

I've heard of VirtualDub but never used it because it doesn't support *.m2ts video sources, which is the subject of my op.
Well, VirtualDub supports AVS scripts, right?
AVS supports FFMS2 plugin, right?
Finally, FFMS2 supports M2TS & TS streams (maybe not frame-accurate when seeking though), or you can try DGDecNV if you have compatible nVidia GPU which is frame-accurate with all supported file formats.

Weirdo

10th February 2011, 16:27

Eac3to reports: "Haali Matroska Muxer (2010-11-07) is installed, There's a new version (2010-11-21) available". However, I have installed the latest 21/11/2010 version from http://haali.su/mkv/.

jeremy33

12th February 2011, 14:51

Madshi,

Can you add my little GUI eac3to µGUI (http://www.homecinema-hd.com/eac3tougui_en.html) to your first post please ?

afonic

14th February 2011, 01:11

Hi,

sorry to "hack" this thread, but it is kind of eac3to related.

It reports I have ArcSoft:

[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding AC3 <448kbps> with libAften...
[a03] Creating file "D:\Movies\xxxx\1_3_audio.ac3"...
[a03] The original audio track has a constant bit depth of 24 bits.
Video track 2 contains 141706 frames.
eac3to processing took 11 minutes, 7 seconds.
Done.

And it appears I have. However I didn't install anything from them. Any idea what could have installed it? I have Nero, AnyDVD and DVD Fab from commercial applications.

Delta2

16th February 2011, 16:28

nurbs

16th February 2011, 16:51

Delta2

16th February 2011, 16:59

I guess eac3to always uses the same order (chapter,video,audio,subtitles) regardless of what the track order in the container is. If eac3to lists the audio as track 2 then that's what you'll have to use in your command line.

but if you make a gui that uses eac3to how do you know the track order, i.e., if there is more than 1 audio type ?

tebasuna51

16th February 2011, 17:19

First run:

eac3to v3.17
command line: "C:\eac3to\eac3to.exe" C:\SAMPLE~2\CTU-X2~1.MKV
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 0:01:05, 24p /1.001
1: h264/AVC, English, 720p24 /1.001
2: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, 2ms

and analize the log.

sl1pkn07

18th February 2011, 20:09

Eac3to reports: "Haali Matroska Muxer (2010-11-07) is installed, There's a new version (2010-11-21) available". However, I have installed the latest 21/11/2010 version from http://haali.su/mkv/.

same here

GRKNGLR

18th February 2011, 21:14

Midzuki

18th February 2011, 21:57

@ madshi (feature request) :

why not make eac3to look for mkvtoolnix in the folders covered by the "PATH" envinronment-variable? In the same way it looks for neroaacenc.exe. I am sure I'm not the only one who uses mmg.exe without the help from an "installer". :)

ketchup318

19th February 2011, 06:56

Hi

I'm trying to extract DTS-HD audio from a m2ts file so that I can mux it with my compressed video stream (MKV).

Using MediaInfo, the original m2ts file displays this for audio
Audio #1
ID : 4352 (0x1100)
Menu ID : 1 (0x1)
Format : DTS
Format/Info : Digital Theater Systems
Format profile : MA / Core
Muxing mode : Stream extension
Codec ID : 134
Duration : 50mn 23s
Bit rate mode : Variable
Bit rate : 3 688 Kbps / 1 510 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 24 bits
Compression mode : Lossless / Lossy
Language : English

But using eac3to, I only get this in my converted file
Audio
Format : DTS
Format/Info : Digital Theater Systems
Format profile : MA / Core
Duration : 1h 47mn
Bit rate mode : Variable
Bit rate : 1 747 Kbps / 1 510 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 24 bits
Compression mode : Lossless / Lossy

This is what I input for eac3to. The funny thing is that it doesn't seem to be using the Arcsoft Decoder that I bought with TMT.

eac3to.exe C:\makin.m2ts 2: C:\test.dts
M2TS, 1 video track, 4 audio tracks, 6 subtitle tracks, 0:50:23, 24p /1.001
1: VC-1, 1080p24 /1.001 (16:9)
2: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
3: AC3, French, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
4: AC3, Spanish, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
5: AC3, Portuguese, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
6: Subtitle (PGS), English
7: Subtitle (PGS), French
8: Subtitle (PGS), Spanish
a02 Extracting audio track number 2...
a02 Creating file "C:\test.dts"

When I convert it to flac, it says this. It seems like now it's using the decoder.

eac3to.exe C:\makin.m2ts 2: C:\test.dts
M2TS, 1 video track, 4 audio tracks, 6 subtitle tracks, 0:50:23, 24p /1.001
1: VC-1, 1080p24 /1.001 (16:9)
2: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
3: AC3, French, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
4: AC3, Spanish, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
5: AC3, Portuguese, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
6: Subtitle (PGS), English
7: Subtitle (PGS), French
8: Subtitle (PGS), Spanish
a02 Extracting audio track number 2...
a02 Decoding with ArcSoft DTS Decoder...
a02 Encoding FLAC with libFlac...
a02 Creating file "test.flac"...

Is it not possible to extract the full DTS-HD track in it's pure form? Is flac the only way to get the lossless track out or is reduced DTS track all I can get? Thanks in advance