eac3to - audio conversion tool [Archive] - Page 13

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tebasuna51

3rd January 2013, 15:59

No matter if is aac, mp3 or wav. Your receiver can restore the surround and center channels if have the DPL II decoder inside.

LigH

3rd January 2013, 16:37

Dolby ProLogic is an analog technology. It analyzes phase directions and delays in signals as electrical voltage fluctuations, not as bits and bytes.

There is no technical relation between Dolby ProLogic (analog surround sound mixing) and Dolby Digital (digital audio compression), except for the same company which developed them.

A third different, unrelated technology developed by the Dolby Labs are noise reduction techniques for audio on magnetical tapes.

And a fourth, even more unrelated remark: Thomas Dolby made a song about Science! (http://www.youtube.com/watch?v=m42KM_2br4c) But his stage name was only inspired by the noise reduction techniques, he was not a member of Dolby Labs...

Boulder

3rd January 2013, 17:18

• Proper ac3 decoding. Whilst a freeware prog like azid can properly decode ac3, it's a real shame such powerful instrument as eac3to uses nero/libav for that, because both decoders has defects (http://forum.doom9.org/showthread.php?t=161654).Are those libav problems reported to the devs?

paradoxical

3rd January 2013, 17:22

The decoder author responded in the thread. The problem is that the results in that thread are based on the libav decoder as it was more than a year and a half ago. Which is why madshi asked if those results were even still relevant.

The test you're linking to is 1.5 years old and the guy reponsible for the libav AC3 decoder has seen and commented on the test results those 1.5 years ago, too, so I would say a retest would be needed to make sure those results are still valid today. And if they are, a bug report should be created for libav/ffmpeg, so that the problem is fixed.

bilditup1

3rd January 2013, 18:57

We can't support downloaded (and fake) files.

Right, didn't mean to offend anyone or feel entitled

phate89

3rd January 2013, 19:05

No matter if is aac, mp3 or wav. Your receiver can restore the surround and center channels if have the DPL II decoder inside.
Dolby ProLogic is an analog technology. It analyzes phase directions and delays in signals as electrical voltage fluctuations, not as bits and bytes.

There is no technical relation between Dolby ProLogic (analog surround sound mixing) and Dolby Digital (digital audio compression), except for the same company which developed them.

A third different, unrelated technology developed by the Dolby Labs are noise reduction techniques for audio on magnetical tapes.

And a fourth, even more unrelated remark: Thomas Dolby made a song about Science! But his stage name was only inspired by the noise reduction techniques, he was not a member of Dolby Labs...

Ok i finally understood. Thanks for the help :thanks:

Thunderbolt8

3rd January 2013, 23:02

thanks for the update! :thanks:

Groucho2004

4th January 2013, 09:55

No matter if is aac, mp3 or wav. Your receiver can restore the surround and center channels if have the DPL II decoder inside.

Picking up on that once more - I don't have a surround setup (and probably never will), just a plain stereo system. In this case I should always use "-downStereo", right?
Also, if I downmix DTS tracks, should I use the "phaseshift" option or does the downmix algorithm take care of that? How can I even determine if the back channels have a phase shift?

tebasuna51

4th January 2013, 12:17

Picking up on that once more - I don't have a surround setup (and probably never will), just a plain stereo system. In this case I should always use "-downStereo", right?
Yes.
Also, if I downmix DTS tracks, should I use the "phaseshift" option or does the downmix algorithm take care of that? How can I even determine if the back channels have a phase shift?
This is a old question.

A correct 5.1, no matter if is AC3 or DTS, must have surround channels already "phaseshifted" over same info in front channels, by soft or when are recorded with microphones at different places, if you do a new "phaseshift" you can finish with 180º than convert the mathematical "+", in mix functions, in a "-" and cancel many info.

Like you don't know how the 5.1 was generated my recommendation is don't use "phaseshift", most the times is the correct way. And only try "phaseshift" when the previous mix sounds strange.

Groucho2004

4th January 2013, 12:24

Yes.

This is a old question.

A correct 5.1, no matter if is AC3 or DTS, must have surround channels already "phaseshifted" over same info in front channels, by soft or when are recorded with microphones at different places, if you do a new "phaseshift" you can finish with 180º than convert the mathematical "+", in mix functions, in a "-" and cancel many info.

Like you don't know how the 5.1 was generated my recommendation is don't use "phaseshift", most the times is the correct way. And only try "phaseshift" when the previous mix sounds strange.

OK, thanks!

madshi

4th January 2013, 13:32

... the azid.dll should still be available from many BeSweet related sources.
Yes, but there's no matching header anywhere to be found, and the dll exports a bunch of APIs without any names or descriptions. I don't feel like reverse engineering the dll.

Is that still accurate or is libavcodec on par now?
Argh, I changed the first page of this thread, but didn't update the same description in the help.

Curious, why the change from Nero to libav as the preferred DD decoder?
Because there were some problems with the Nero decoder. Mostly, disabling DRC didn't work for all tracks.

Errors that libavcodec spits out break the progress bar
Doesn't happen for me!! Did you maybe replace the libav/ffmpeg dlls with a different version? Can anybody reproduce this problem?

This still doesn't appear to work with 5.0 DTS files.
is't true eac3to 3.25 don't downmix 5.0 files (tested DTS, AC3 and WAV).
Yes, a stupid mistake. I added all the code to make the downmixing work for all possible channel configurations, but then forgot to remove some code which disables downmixing for less than 6 channels. I'll release a new build, but I'll wait a couple of days, in case some other bugs show up. So guys, do testing *now*, because the next eac3to build will probably have to do for a few months again.

• avi, mp4, mov, mpg, flv demuxing(maybe even muxing, that would be fantastic). Basically, avi/mp4 support is most important feature I think eac3to is lacking, along with
Yeah, that would be nice, but it would be a lot of work, so it won't come anytime soon.

• Proper ac3 decoding. Whilst a freeware prog like azid can properly decode ac3, it's a real shame such powerful instrument as eac3to uses nero/libav for that, because both decoders has defects (http://forum.doom9.org/showthread.php?t=161654).
There's no header available for the azid.dll. I think the best solution would be to first double check whether the libav decoder still suffers from the same problem today (that test you mention is 1.5 years old after all), and if it does, post a bug report to the libav devs.

• and also, downmix to mono and ac3/dts dialnorm applying - minor, but still useful features.
What would you need dialnorm for? I believe it's a fundamentally broken concept.

Snowknight26

4th January 2013, 14:32

Doesn't happen for me!! Did you maybe replace the libav/ffmpeg dlls with a different version? Can anybody reproduce this problem?

If I extracted the zip to the same folder as previous instances of eac3to, would the latest version of libavcodec be used (avcodec-54.dll) or would old ones be used if they were present (such as avcodec.dll)?

tebasuna51 had a similar result (http://forum.doom9.org/showpost.php?p=1608764&postcount=96) but he used -progressNumbers or whatever the command is. Strange how mine shows the same thing as ffmpeg "[truehd @ 00000000]" whereas his shows "[libav]".

I'll see what happens when I clear out the eac3to folder..

madshi

4th January 2013, 14:38

I used the ones included in the zip. tebasuna51 had a similar result (http://forum.doom9.org/showpost.php?p=1608764&postcount=96) but he used -progressNumbers or whatever the command is. Strange how mine shows the same thing as ffmpeg "[truehd @ 00000000]" whereas his shows "[libav]".
tebasuna51's results are fundamentally different. "[libav]" means eac3to got the log message from libav. "[truehd @ ...]" means ffmpeg/libav bypassed eac3to and wrote the log directly to the command line window. Which ffmpeg/libav should not do because eac3to told it to forward all log messages to eac3to instead of writing them to the command line window. So the logging system is broken for you, but not for tebasuna51, nor for me.

Snowknight26

4th January 2013, 14:43

Well, I'm not sure what to say but having gone through the thread I see a few people have different results.

For example:
'[libav]': http://forum.doom9.org/showpost.php?p=1599868&postcount=11866
'[truehd @ 00000000]': http://forum.doom9.org/showpost.php?p=1600129&postcount=11883

Edit:
Just tried a corrupt file and eac3to only showed '[libav]' so I guess it was using older dlls that were present in the folder. Case closed. :p

nickintheforest

4th January 2013, 16:14

Yes, but there's no matching header anywhere to be found, and the dll exports a bunch of APIs without any names or descriptions. I don't feel like reverse engineering the dll.
Ok, azid hasn't source code. But liba52 has. liba52 in group "A" too (http://forum.doom9.org/showthread.php?p=1508819#post1508819).

paradoxical

4th January 2013, 16:19

Ok, azid hasn't source code. But liba52 has.

The source code isn't needed or the issue. The issue is the required header to actually call into the DLL.

liba52 in group "A" too (http://forum.doom9.org/showthread.php?p=1508819#post1508819).

Yes, based on a test that is 1.5 years out of date. You and get8p seem to gloss over the part where madshi has repeatedly stated that you should be checking that the issue is even still relevant today since the decoder has had changes made to it since that test. Then, if the issue still exists, report them to libav so they can be fixed. This would be the best solution.

madshi

4th January 2013, 18:20

liba52 is GPL, not LGPL, so I can't use it.

mrr19121970

4th January 2013, 18:25

Perhaps there is an issue with the new version?

http://forum.slysoft.com/showpost.php?p=361367&postcount=2976

Used eac3to 3.25 to successfully process Brave and Finding Nemo with ClownBD. However, the playback of Brave contains audio drops (3 in the first 5 minutes) like in Total Recall (2012). I have not checked Finding Nemo yet.

madshi

4th January 2013, 18:33

Not sure what is meant with "contains audio drops". I've personally tested Brave and had no problems with playback whatsoever.

sl1pkn07

4th January 2013, 19:29

according to http://forum.doom9.org/showthread.php?t=23921 and http://web.archive.org/web/20020205230410/http://www.doom9.org/sources.htm and http://web.archive.org/web/20020414191825/http://www.doom9.org/sources.htm

http://web.archive.org/web/20020205230410/http://www.doom9.org/Soft21/Sources/azid-dll-1.7.1.zip
http://web.archive.org/web/20020414191825/http://www.doom9.org/Soft21/Sources/azid-dll-1.8.zip

have sources to api call dll

but.... don't download

EDIT: http://web-beta.archive.org/web/20090228063605/http://dspguru.notrace.dk/BeSweetv1.5b23DLL.zip

is possible use azid trought beesweet dll? (the zip contains headers) (i don't know if beesweet.dll uses azid.dll)

Chumbo

5th January 2013, 01:38

@madshi/tebasuna51,
Thanks so much for your efforts. Great to see a new version. :)

madshi

5th January 2013, 12:04

madshi

5th January 2013, 12:26

eac3to v3.26 released

http://madshi.net/eac3to.zip

* fixed: downmixing of less than 6 channels to stereo failed
* patched libav AC3 decoder to properly decode high frequencies
* added support for floating point volume changes (e.g. -0.5db)
* dialnorm is no longer removed from DTS-HD tracks (didn't work, anyway)

hubblec4

5th January 2013, 13:15

thanks a lot madshi, you are the best.

filler56789

5th January 2013, 13:18

eac3to v3.26 released

http://madshi.net/eac3to.zip

* fixed: downmixing of less than 6 channels to stereo failed

* patched libav AC3 decoder to properly decode high frequencies

* added support for floating point volume changes (e.g. -0.5db)

* dialnorm is no longer removed from DTS-HD tracks (didn't work, anyway)

:thanks: ( again ) :)

get8p

5th January 2013, 15:53

Thanks! Ac3 fix is a huge plus. Finally ac3 demux and decode is possible in one line.
Never got the answer, is there any thoughts on avi, mp4, mov, mpg, flv demuxing(maybe even muxing) and downmix to mono?

Overdrive80

5th January 2013, 16:14

Never got the answer, is there any thoughts on avi, mp4, mov, mpg, flv demuxing(maybe even muxing) and downmix to mono?

It would be major improvements, and madshi only will do minor changes, for now.

get8p

5th January 2013, 16:37

Well, then I'd like madshi to tell if he's putting it in future to-do list, so I'd just stop asking.
And downmixing to mono, it can't be that big of a deal?

nickintheforest

5th January 2013, 17:13

> * patched libav AC3 decoder to properly decode high frequencies
thank you, madshi :)

Snowknight26

6th January 2013, 07:15

* encoder bitdepth for lossy DTS tracks is no longer displayed

It still is for DTS core tracks:
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)

madshi

6th January 2013, 09:27

It still is for DTS core tracks:
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
Oh well... Maybe next time...

madshi

6th January 2013, 09:28

eac3to v3.27 released

http://madshi.net/eac3to.zip

* fixed: raw processing cut away 16 samples sometimes
I'll probably be gone for a while now.

VincAlastor

6th January 2013, 10:11

thank you madshi for eac3to to keeping alive!!! the new release makes me very happy ^^

if you want to expand features in future, maybe you take a look at qaac (neroaac development stopped for years) and flaccl (better compression and sometimes faster than flac.exe; i don't know if flaccl can handle 24bit stream now)

thanks again :)

DarkSpace

6th January 2013, 10:30

Thank you for the new releases, and for taking the time to implement float Volume changes as well!

tebasuna51

6th January 2013, 10:38

Thanks madshi!

if you want to expand features in future, maybe you take a look at qaac (neroaac development stopped for years) and flaccl (better compression and sometimes faster than flac.exe; i don't know if flaccl can handle 24bit stream now)

You can always use qaac (I don't know flaccl) like external encoder with:

eac3to input stdout.wav | qaac -V 99 --ignorelength -o output.m4a -

06_taro

7th January 2013, 03:39

flaccl can handle up to 32bit pcm IIRC, you can always use pipe method: eac3to.exe input stdout.wav | CUETools.FLACCL.cmd.exe -o output.flac -

VincAlastor

7th January 2013, 03:55

tebasuna51

7th January 2013, 11:54

thank you for your help. i use eac3to often with megui and staxrip automatically... in cmd i think it's not practically for daily using. and why should eac3to come out with outdated encoders (neroaac, flac) and not with newer, often better encoders?

- You can't prove than qaac is better than NeroAacEnc at default eac3to -quality=0.5

- If you don't like command line maybe you can test the GUI UsEac3to.

- MeGUI support qaac now.

LigH

7th January 2013, 14:55

From the archive:

dg providem me with v1.9 :)
THX for that DSPguru

but i have read the documentation from the azid files and the author (midas) wants me to add splash screens, and some other things which i personally dont really like.
...

The azid.dll could possibly be used via the besweet.dll, but I believe that will only shift the issue to another level.

Midas (Azid), DSPguru (BeSweet), DarkAvenger (HeadAC3he) — Hopping from one dead horse to another... :o

And there is also a package with azid.exe CLI decoders (P3/P4 opt.), but you may try to avoid intermediate files?

Chumbo

7th January 2013, 15:02

eac3to v3.27 released

http://madshi.net/eac3to.zip

* fixed: raw processing cut away 16 samples sometimes
I'll probably be gone for a while now.
Thanks again. Hey, before you go, would you update the first post to reflect version 3.27 please?

madshi

7th January 2013, 15:14

From the archive:

The azid.dll could possibly be used via the besweet.dll, but I believe that will only shift the issue to another level.

Midas (Azid), DSPguru (BeSweet), DarkAvenger (HeadAC3he) — Hopping from one dead horse to another... :o

And there is also a package with azid.exe CLI decoders (P3/P4 opt.), but you may try to avoid intermediate files?
Doesn't matter anymore, anyway. See v3.26 release notes.

Atak_Snajpera

8th January 2013, 19:05

does new version still need MSVCP71.dll in system folder?

http://forum.doom9.org/showthread.php?p=1607304#post1607304

rapscallion

8th January 2013, 20:09

Thanks madshi for new version....however, Nemo 7.1 THD, decode to wavs, still fails lossless check :

[libav] Lossless check failed - expected 00, calculated f4. <WARNING>
[libav] Lossless check failed - expected 00, calculated 65. <WARNING>
[libav] Lossless check failed - expected 00, calculated f6. <WARNING>
[libav] Lossless check failed - expected 00, calculated 88. <WARNING>
The original audio track has a constant bit depth of 20 bits.

I've never seen the 20 bits depth before, besides this nemo track.

Edit: Correction, the previous libav (ver 3.25) would crash completely, while now it creates the wavs w/the warnings above. This is the same thing when creating flac via ffmpeg, so maybe it's a somehow corrupted stream.

Overdrive80

9th January 2013, 02:56

Hi, one question... Does eac3to decode to do "slowdown"???

tebasuna51

9th January 2013, 03:20

Hi, one question... Does eac3to decode to do "slowdown"???
Yes, slowdown-resample work over uncompressed audio data then eac3to must decode compressed audio before.

Furiousflea

11th January 2013, 09:54

Is there still a need to remap channels when decoding DTS 6.1 tracks with Arcsoft decoder with latest eac3to.

The changelog is ambiguous if that has been fixed or do we still need to specify a channel layout?

tebasuna51

11th January 2013, 10:38

Is there still a need to remap channels when decoding DTS 6.1 tracks with Arcsoft decoder

Decode was always fine, only when you use -down6 was wrong.
And fixed now.

dansrfe

12th January 2013, 08:49

Furiousflea

12th January 2013, 10:11

Decode was always fine, only when you use -down6 was wrong.
And fixed now.

Thanks for explaining that, much appreciated.

tebasuna51

12th January 2013, 11:55

So I recently bought a bluray which has 16-bit, 96Khz TrueHD and 24-bit, 48Khz DTS-HD/MA audio tracks. My question is: if space is not an issue and playback will always involve decoding to PCM via arcsoft's decoder or the like, which track should I keep in the mkv encode?

Seems than standard human ear can distinguise until 20 bit precission audio but not much over 20 KHz frequency (samplerate 40 KHz).

Also, as a side question, if I will always play back from my computer then is it better to encode lossless audio from DTS-HD/MA or TrueHD to FLAC and maintain the original file's bit-depth and sample rate or should I just keep the original DTS-HD-MA/TrueHD file?

If your PC can send to receiver, through HDMI, PCM multichannel audio (decoded from FLAC or DTS-HD-MA/TrueHD) then the quality is always the same. Flac files have less size.

There are external players (PopCorn, Xtreamer, ...) than can send also Flac decoded to PCM multichannel.

messi magician

12th January 2013, 16:56

I seem to be experiencing strange problems with some of my blu ray rips when converting Dts-HD Master to AC3 640 using eac3to

When playing back the m2ts files using Arcsoft player the audio says Dolby AC3 along with some odd kbps figures - 31kbps or 129kbps ect

Usually films just say Dolby AC3 in Arcsoft, i've even updated to the new version of Arcsoft and it still displays these odd numbers

When i ripped the Hunger Games which come out as 31kbps the audio sounded quieter compared to some of my other films so i have to turn up the volume a few notches

Checking the files though with mediainfo it shows the track as 640kbps English 48.0 KHz, 16bits, 6 channels, AC3 which is correct

Anyone else experieced this ? is this normal or is this a known problem with eac3to not converting the audio properly ?

I'm using the latest version of eac3to and arcsoft decoder 1.1.0.0

Snowknight26

14th January 2013, 01:06

tebasuna51's results are fundamentally different. "[libav]" means eac3to got the log message from libav. "[truehd @ ...]" means ffmpeg/libav bypassed eac3to and wrote the log directly to the command line window. Which ffmpeg/libav should not do because eac3to told it to forward all log messages to eac3to instead of writing them to the command line window. So the logging system is broken for you, but not for tebasuna51, nor for me.

Had the same behavior happen on a different system.

With the sample I've PMed you, simply running
eac3to part1.m2ts+part2.m2ts 2: test.flac
is enough to show the issue.

Vick

14th January 2013, 16:26

rapscallion

14th January 2013, 23:23

jj666

14th January 2013, 23:41

Vick...why would he want him to? If you're talking about *.dtshd files created by DTS-MA suite, that header is only used by authoring houses. For our purposes, we do NOT want the header to be there.

It can be easily removed by demuxing the track/stream via Tsmuxer. The resulting demux will have a *.dts extension, just like on a BD disc.

As a reminder, you still need to have a -21ms delay when removing .dtshd MA suite headers with EAC3TO, this bug didn't get fixed yet.

Cheers,

-jj-

rapscallion

14th January 2013, 23:46

@jj, that's why I use Tsmuxer.

filler56789

14th January 2013, 23:55

If you're talking about *.dtshd files created by DTS-MA suite, that header is only used by authoring houses. For our purposes, we do NOT want the header to be there.

I think there are people who would like to decompress the DTS-HD files to .WAVs through the "official" StreamPlayer (simply because ArcSoft's DTSdecoderdll.DLL not always works as it should), and in this case too, the DTS-HD headers are mandatory.

robertcollier4

15th January 2013, 02:09

Getting a line of output per second from the Nero encoder? Nero reporting every single second seems to be making eac3to think Nero is stuck. How can we disable Nero AAC encoder from reporting second by second processing?

command line: eac3to S01E01.mkv S01E01-downDpl.aac -downDpl
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 1:26:26, 24p /1.001
1: h264/AVC, English, 720p (16:9)
2: AC3, English, 5.1 channels, 384kbps, 48kHz, dialnorm: -27dB
[v01] The video track doesn't contain framerate information. <WARNING>
Track 2 is used for destination file "S01E01-downDpl.aac".
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Decoding with libav/ffmpeg...
[a02] Downmixing multi channel audio to stereo...
[a02] Reducing depth from 64 to 32 bits...
[a02] Encoding AAC <0.50> with NeroAacEnc...
[a02] Clipping detected, a 2nd pass will be necessary. <WARNING>
[a02] The Nero AAC encoder seems to be stuck... <ERROR>
[a02] [NeroAacEnc] Processed 0 seconds...
[a02] [NeroAacEnc] Processed 1 seconds...
.... (LINES CUT OUT FOR FORUM... THERE IS A LINE PER SECOND) ...
[a02] [NeroAacEnc] Processed 5053 seconds...
[a02] [NeroAacEnc] Processed 5054 seconds...
[a02] [NeroAacEnc] Process
Aborted at file position 2801484127. <ERROR>

Vick

15th January 2013, 08:12

.... For our purposes, we do NOT want the header to be there. .....
I think there are people who would like to decompress the DTS-HD files to .WAVs through the "official" StreamPlayer (simply because ArcSoft's DTSdecoderdll.DLL not always works as it should), and in this case too, the DTS-HD headers are mandatory.
You're right.
For StreamPlayer the header is necessary. And for re-authoring.

tebasuna51

15th January 2013, 10:46

@robertcollier4
Try the workaround suggested in first post here (http://forum.doom9.org/showthread.php?t=166487):

eac3to S01E01.mkv 2: stdout.wav -downDpl | NeroAacEnc -q 0.5 -ignorelength -if - -of S01E01-downDpl.m4a

tebasuna51

15th January 2013, 11:17

@madshi
There is description of *.dtshd file in Annex E (http://www.atsc.org/cms/standards/cs_documents/S13-1-329r18-WD-changes-to-NRT-CS-cln.pdf). Can you add header (and other chunks) to DTS-MA and DTS-HR file?

I think this is not a problem related to eac3to, you can make a independent tool to do the conversion.

At least for DTS-MA you can always recode the wavs supplied by eac3to with DTS Master Audio Suite.

Vick

15th January 2013, 13:00

....
Interesting. It seems there's a whole file structure. It's very easy to understand, though, so it should be easy adding support for that. Well, "easy" means "not difficult". It does cost some time, though, so it may have to wait a while.
....

I think this is not a problem related to eac3to, you can make a independent tool to do the conversion.
....

I would like to simplify process of developing...

tebasuna51

15th January 2013, 14:00

I am not opposed to Madshi do the job, only encourage others interested guys to do it.
I think that eac3to is not necesary to do it.

EuropeanMan

17th January 2013, 21:27

I have a bluray of an indian movie, but for some reason the audio is just messed up in the DTS/MA channel. voices tend to be in the background and music in the foreground.

HOW can i fix this? please...I'd like to RE-ENCODE the audio to AC3. I'll assume that I must separate the DTS into 6wav'files and then remux in a different order into AC3...correct? If you could please show me in short...I'd appreciate it. Thank you. I did this process long long long long ago...so have forgotten.

tebasuna51

18th January 2013, 01:42

1) Decode to wav's with eac3to:
eac3to input output.wavs

2) Encode to AC3 with EncWAVtoAC3, or Aften.exe, with the correct order of channels.

EuropeanMan

18th January 2013, 02:04

^ Thanks for that...but how do I actually change or perhaps ENHANCE the channel that has the main dialogue because it seems to me, it is set to background and not foreground...that's my real question. :) Appreciate the heads-up above!

tebasuna51

18th January 2013, 02:49

You can edit the wavs between 1) and 2) but we can't know the problem without listen a sample.

Maybe you can make a new thread with a sample, but not here.

madshi

20th January 2013, 17:43

eac3to now has a bug tracker:

http://eac3to.bugs.madshi.net

You can browse the bug reports without being logged in. In order to create bug reports you need to login. Please in the future if you have a clear bug to report, use the bug tracker. Currently please use the tracker only for bug reports. Feature wishes will be silently deleted.

kylix999

21st January 2013, 12:16

Is it possible to encode wav to eac3 format but stored in mp4?
I am able to do that with aac with ffmpeg but ffmpeg can't save eac3 in mp4 :( so maybe this program can.

LigH

21st January 2013, 12:58

Maybe a current MP4Box can. But eac3to is not responsible for unusual content in strictly specified containers, rather for the raw content instead.

kylix999

21st January 2013, 13:05

MP4box can't do that, I asked on their forum and it is not supported and do not plan to make it possible. Also it is listed in this table: http://mp4ra.org/codecs.html
so it is possible to do, but don't know how

LigH

21st January 2013, 13:14

In the german doom9/Gleitz board, we just discussed how to multiplex DTS-HD in MP4; if MP4Box can't, maybe l-smash can. So check if l-smash knows E-AC3 too.

kylix999

22nd January 2013, 09:17

In the german doom9/Gleitz board, we just discussed how to multiplex DTS-HD in MP4; if MP4Box can't, maybe l-smash can. So check if l-smash knows E-AC3 too.

you saved my ass, :thanks:

robertcollier4

27th January 2013, 10:33

Can eac3to handle AAC 5.1 / AAC 6 Ch audio input? If not - does anyone know of another tool which can?

robertcollier4

27th January 2013, 10:57

I am unable to get eac3to to read an Audio AAC 5.1 File that I have demuxed from mp4box. I am trying to downmux the AAC 5.1 audio to AAC 2.0 audio.

Media Info Reports:
Audio
ID : 2
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : LC
Codec ID : 40
Duration : 2h 16mn
Bit rate mode : Variable
Bit rate : 420 Kbps
Maximum bit rate : 430 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Compression mode : Lossy
Stream size : 409 MiB (100%)
Language : English

eac3to reports the Nero AAC decoder is installed
>eac3to -test
eac3to (v3.27) is up to date
Nero Audio Decoder (Nero 7) works fine

If I use Yamb-mp4box to demux the audio into an mp4 file, then run eac3to:
eac3to v3.27
command line: eac3to.exe "The.Amazing.Spider-Man.2012.1080p.BDRip.x264.AAC-26K_track2.mp4" test.wav -downDpl
------------------------------------------------------------------------------
The format of the source file could not be detected. <ERROR>

Or, if I use Yamb-mp4box to demux the audio into RAW Format, then run eac3to:
eac3to v3.27
command line: eac3to.exe "The.Amazing.Spider-Man.2012.1080p.BDRip.x264.AAC-26K_track2.aac" test2.wav -downDpl
------------------------------------------------------------------------------
AAC, 5.1 channels, 48kHz
Decoding with DirectShow (Nero Audio Decoder 2)...
The DirectShow audio decoder didn't accept the input stream. <ERROR>
Aborted at file position 262144. <ERROR>

Overdrive80

28th January 2013, 03:12

@robertcollier4 The containers of input only can be: MKV, EVO, M2TS, TS y VOB.

Audio format: RAW, (L)PCM, WAV (PCM, DTS, AC3), W64, RF64, AC3, E-AC3, DTS, DTS-ES, DTS-96/24, DTS-HD Hi-Res, DTS-HD Master Audio, MLP, TrueHD, TrueHD/AC3, FLAC.

tebasuna51

28th January 2013, 14:09

AAC must be also supported:

For best AAC decoding you need:
(1) Nero 7 (Nero 8 won't work!)

But don't work propperly in my test. My recommendation is don't use Nero 7 at all.

Use instead:

- BeHappy-AviSynth-BassAudio to decode, DownDPL and recode.

- Decode first to wav with Foobar2000 or UsEac3to (NeroAacDec for .m4a/.mp4 or Faad for .aac) and after eac3to for DownDPL and recode.

robertcollier4

29th January 2013, 13:04

Thanks I used the following steps to downmix from 6ch AAC to 2ch AAC:
(Note: qaac not necessary, but I prefer it since it can make AAC files with no encoder delay).

1. Demux audio into m4a file using mp4box (Yamb GUI).
2. neroaacdec -if "audio6chaac.m4a" -of "audio6chwav.wav"
3. eac3to.exe "audio6chwav.wav" stdout.wav -downDpl | qaac.exe --tvbr 127 --quality 2 --rate keep --ignorelength --no-delay - -o "audio2chaac.m4a"
4. Remux "audio2chaac.m4a" audio with original video track using mp4box (Yamb GUI)

06_taro

29th January 2013, 15:52

aac decoding works well for me with nero 7...

tebasuna51

30th January 2013, 00:21

1. Demux audio into m4a file using mp4box (Yamb GUI).
2. neroaacdec -if "audio6chaac.m4a" -of "audio6chwav.wav"
3. eac3to.exe "audio6chwav.wav" stdout.wav -downDpl | qaac.exe --tvbr 127 --quality 2 --rate keep --ignorelength --no-delay - -o "audio2chaac.m4a"
4. Remux "audio2chaac.m4a" audio with original video track using mp4box (Yamb GUI)

Ok.

For next time you don't need demux the audio from mp4.
You can use:

1. neroaacdec -if "video+audio.mp4" -of "audio6chwav.wav"

Atak_Snajpera

30th January 2013, 14:51

I want to confirm something. Does eac3to still need to decode TrueHD/DTSMA (-> wav,flac) in order to fix audio gaps?

tebasuna51

30th January 2013, 22:13

Always if there are gaps, overlaps sometimes can be solved deleting duplicated frames.

frencher

3rd February 2013, 01:23

Hi all and madshi,

I wanted to know if others have had the same problem as me for demuxing 3D Bluray PGS subtitles tracks (.sup) and if an update will fix this problem ?

Thanks

XadoX

4th February 2013, 15:19

I have a strange problem. I am trying to demux a playlist file.

With BDinfo there is show the correct lenght of the movie. But eac3to shows a wrong value and does a wrong demux.
eac3to adds the same stream multiple times to the demuxed file. mpc-hc is playing the playlist fine.

Snowknight26

4th February 2013, 18:13

You're probably processing the wrong playlist with eac3to.

XadoX

5th February 2013, 07:22

You're probably processing the wrong playlist with eac3to.
No. It is the same playlist.

Snowknight26

5th February 2013, 16:02

And which playlist would that be since it's not shown in the eac3to log?

Snowknight26

5th February 2013, 20:06

That won't be of any help.

Provide the output of
eac3to folder
followed by
eac3to folder x)
where x is the playlist index and we'll go from there.

XadoX

6th February 2013, 07:25

eac3to.exe g:\BDMV\PLAYLIST\00000.mpls
1) 00000.mpls, 0:21:26
[0+0+0+0+8].m2ts
- VC-1, 1080i60 /1.001 (16:9)
- DTS, German, stereo, 48kHz
- DTS, English, stereo, 48kHz

eac3to.exe g:\BDMV\PLAYLIST\00000.mpls 1)
M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 1:30:00, 60i /1.001
1: VC-1, 1080i60 /1.001 (16:9)
2: DTS, German, 2.0 channels, 384kbps, 48kHz
3: DTS, English, 2.0 channels, 384kbps, 48kHz
4: Subtitle (PGS), German

tebasuna51

6th February 2013, 11:19

1) 00000.mpls, 0:21:26
[0+0+0+0+8].m2ts
Seems a bug (or file corrupt) reading the mpls.

A workaround is use:
eac3to.exe g:\BDMV\STREAM\00000.m2ts + g:\BDMV\STREAM\00008.m2ts

XadoX

6th February 2013, 11:31

Somehow playing the playlist results in shorter playtime than only playing the m2ts files. So the workaround is not that easy I think.

tebasuna51

6th February 2013, 13:18

Somehow playing the playlist results in shorter playtime than only playing the m2ts files.

Please provide the playtime of the mpls and the extracted mkv using the workaround.

There are also a strange FILES info (same file used 4 times) in your bdinfo.txt:
FILES:

Name Time In Length
---- ------- ------
00000.M2TS 0:00:00.000 0:01:31.991
00000.M2TS 0:01:31.991 0:08:43.856
00000.M2TS 0:10:15.848 0:10:26.092
00000.M2TS 0:20:41.940 0:00:43.243
00008.M2TS 0:21:25.183 0:00:00.333

madshi

6th February 2013, 13:26

XadoX

6th February 2013, 13:26

I have asked neuron2 also. Maybe this (http://neuron2.net/board/viewtopic.php?f=8&t=264&sid=da4b63faa6ca216efc052509a2e5dbf0&p=2339#p2339) helps?

LigH

6th February 2013, 13:27

Looks like having an M2TS file in a playlist does not necessarily mean to play the whole M2TS file, but possibly only several ranges in them (e.g. skipping some "Director's Cut" scenes)?

XadoX

6th February 2013, 13:28

Such playlists (using defined sub-sections of m2ts files) are perfectly legal, according to the Blu-Ray spec, but they're only very very rarely used in real life. As a result eac3to always uses the full m2ts files, ignoring the "Time In" and "Length" information in the playlist.
Somehow this is great so it was not my fault doing something wrong :)
Any chance of a workaround?

madshi

6th February 2013, 13:59

Support for playlists with sub-sections of m2ts files is not planned for eac3to anytime soon. Implementing that would not be as easy as it might sound. An m2ts file consists of many different streams, all with their own timestamps. So figuring out which streams to cut when and where is not straightforward, especially because after a cut you want to start with a key frame and there's no guarantee that the "Time In" position in the playlist points to a key frame.

tebasuna51

6th February 2013, 14:00

Such playlists (using defined sub-sections of m2ts files) are perfectly legal, according to the Blu-Ray spec, but they're only very very rarely used in real life. As a result eac3to always uses the full m2ts files, ignoring the "Time In" and "Length" information in the playlist.

Yes, but XadoX say:
eac3to adds the same stream multiple times to the demuxed file.
...
M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 1:30:00, 60i /1.001

And don't work:
eac3to.exe g:\BDMV\STREAM\00000.m2ts + g:\BDMV\STREAM\00008.m2ts

madshi

6th February 2013, 14:06

The playlist references the 00000.m2ts file 4 times, so it's correct that eac3to shows [0+0+0+0+8]. The key problem is that the playlist doesn't want the whole 00000.m2ts file 4 times, instead it wants 4 different subsections of 00000.m2ts, and that's not supported by eac3to at the moment. I don't know how long 00000.m2ts and 00008.m2ts are exactly, but I suppose that the playlist doesn't cover the whole 00000.m2ts and 00008.m2ts files, but just a part of them. That's why using "eac3to 00000.m2ts+00008.m2ts" produces a wrong length, too.

XadoX

6th February 2013, 14:09

...I don't know how long 00000.m2ts and 00008.m2ts are exactly, but I suppose that the playlist doesn't cover the whole 00000.m2ts and 00008.m2ts files, but just a part of them. That's why using "eac3to 00000.m2ts+00008.m2ts" produces a wrong length, too.

Absolutly right!

Shevek

6th February 2013, 15:14

So, for playlists like this, until fully supported (if at all possible in future), should eac3to then report that this is a timed playlist and either refuse to work on it, or display an additional caveat to the user?

Guest

6th February 2013, 15:54

I suppose that the playlist doesn't cover the whole 00000.m2ts No, the BDInfo below shows that the referenced segments are contiguous:

00000.M2TS 0:00:00.000 0:01:31.991 0 0
00000.M2TS 0:01:31.991 0:08:43.856 0 0
00000.M2TS 0:10:15.848 0:10:26.092 0 0
00000.M2TS 0:20:41.940 0:00:43.243 0 0
00008.M2TS 0:21:25.183 0:00:00.333 0 0

The first time is the start time and the second is the duration. So, adding the duration to the start time gives the start time of the following segment.

It's possible that there is extra unreferenced material at the end of 00000.M2TS but it seems unlikely.

madshi

6th February 2013, 18:53

@neuron2, I've been told "eac3to 00000.m2ts+00008.m2ts" does not produce the expected runtime, so there *must* be extra unreferenced material at the end of either 00000.m2ts or 00008.m2ts. That's the only explanation that makes sense to me.

Guest

6th February 2013, 19:11

The other explanation is that the OP's estimate of the run time is wrong.

The only way to be sure is for the OP to give us the two M2TS files.

XadoX

6th February 2013, 19:13

They have around 3GB.

madshi

6th February 2013, 19:25

If the files as listed in the playlist had no extra material, we could calculate the overall runtime of [0+0+0+0+8] like this:

00000.m2ts: 0:21:25.183
00008.m2ts: 0:00:00.333

Overall runtime of [0+0+0+0+8] = 4 * 00:21:25.183 + 00:00:00.333 = 5141.065 seconds = 01:25:41.065

However, eac3to reported an overall runtime for [0+0+0+0+8] of exactly 01:30:00.000, so there must be some extra material somewhere, not referenced by the playlist.

madshi

6th February 2013, 19:27

@XadoX, what is the output of "eac3to 00000.m2ts" and "eac3to 00008.m2ts"?

XadoX

6th February 2013, 19:28

eac3to.exe f:\BDMV\STREAM\00000.m2ts
M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 0:22:30, 60i /1.001
1: VC-1, 1080i60 /1.001 (16:9)
2: DTS, German, 2.0 channels, 384kbps, 48kHz
3: DTS, English, 2.0 channels, 384kbps, 48kHz
4: Subtitle (PGS), German

eac3to.exe f:\BDMV\STREAM\00008.m2ts
M2TS, 1 video track, 0:00:00
1: VC-1, 1080i60 /1.001 (16:9)

madshi

6th February 2013, 19:32

Yeah, so 00000.m2ts has about 1 extra minute of playback not referenced by the playlist.

XadoX

6th February 2013, 19:36

General
ID : 0 (0x0)
Complete name : F:\BDMV\STREAM\00000.m2ts
Format : BDAV
Format/Info : Blu-ray Video
File size : 2.93 GiB
Duration : 23mn 0s
Overall bit rate mode : Variable
Overall bit rate : 18.2 Mbps
Maximum Overall bit rate : 48.0 Mbps

Video
ID : 4113 (0x1011)
Menu ID : 1 (0x1)
Format : VC-1
Format profile : Advanced@L3
Codec ID : 234
Duration : 22mn 30s
Bit rate : 16.7 Mbps
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16:9
Frame rate : 29.970 fps
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Interlaced
Scan order : Top Field First
Compression mode : Lossy
Bits/(Pixel*Frame) : 0.269
Stream size : 2.63 GiB (90%)

General
ID : 0 (0x0)
Complete name : F:\BDMV\STREAM\00008.m2ts
Format : BDAV
Format/Info : Blu-ray Video
File size : 54.0 KiB
Duration : 270ms
Overall bit rate : 1 491 Kbps
Maximum Overall bit rate : 48.0 Mbps

Video
ID : 4113 (0x1011)
Menu ID : 1 (0x1)
Format : VC-1
Format profile : Advanced@L3
Codec ID : 234
Duration : 334ms
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16:9
Frame rate : 29.970 fps
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Interlaced
Scan order : Top Field First
Compression mode : Lossy

robertcollier4

7th February 2013, 05:34

I have a question about the 24-bit-depth outputted by eac3to for use in lossy formats such as AAC which use floating point bit-depth. I am usually downmixing 6ch DTS files to 2ch AAC through piping to the qaac encoder. I am wondering if there would be better quality through having eac3to pass qaac its original 64-bit-depth file instead of a dithered 24-bit-depth file. I am trying to maintain best quality possible by doing as few bit-depth dithering operations as possible.

eac3to.exe "movie_6chdts.mkv" 2: stdout.wav -downDpl | qaac.exe --tvbr 127 --quality 2 --rate keep --ignorelength --no-delay - -o "2chaac.m4a"
eac3to tells me:
[a02] Reducing depth from 64 to 24 bits
[a02] Writing WAV

Even using the internally used neroaacenc does a down dithering operation. Why not pass the aac encoder the original 64-bit-depth data?
[a02] Reducing depth from 64 to 32 bits
[a02] Encoding AAC <1.00> with NeroAacEnc

I have read that lossy formats such as AAC use floating point and so while they do not technically have a "bit depth" they are able to deliver near equivalent of 64-bit-depth due to use of floating point. So in this case - wouldn't it better to have eac3to give qaac an undithered 64-bit-depth output so that qaac can encode the original undithered 64-bit-depth output from eac3to into floating point?

Reference: http://wiki.jriver.com/index.php/Audio_Bitdepth
Bitdepth of Lossy Formats
Lossy formats like MP3 [and AAC] use floating point math to build their output values. There is no true or correct bitdepth in this case.
Media Center preserves the full 64bit precision when converting from the lossy format to PCM and uses as much precision as possible during output.
In other words, taking a 16bit input file, encoding as MP3, and then playing it in Media Center will cause 64bit data to be delivered to the playback engine. This does not mean that the file has improved, only that Media Center does the best job possible dealing with the MP3 (or other lossy) data that is not inherently precise to some number of bits.

Reference: http://en.wikipedia.org/wiki/Talk%3AAudio_bit_depth
In a course on mastering that I took at Berklee College of Music, it was recommended that dither not be applied when lossy formats, like mp3 and AAC, were the destination. The reason being that dither to combat quantization is really only applicable to PCM because formats like mp3 do not really "quantize" in the same sense, especially if they are floating point.

Since eac3to operates internally in 64-bit-depth anyways, is there any way I can retain this and have eac3to give me its original 64-bit-depth wav as its output? (no dithering).

nevcairiel

7th February 2013, 07:48

Most encoders only accept up to 32-bit floating point ("float"), and not 64-bit ("double"), which is most likely why you're seeing the conversion for Nero.
If you think qaac supports 64-bit floating point input, you can use the "-full" switch, which lets eac3to output 64-bit float (or so i remember)

Note that most decoders also only produce 32-bit floating point (however eac3to might produce 64-bit after the downmixing)

LigH

7th February 2013, 08:14

An intermediate 64-bit result based on the decoding of an already lossy source, with even more loss due to the downmixing...

Will someone rather be able to hear the difference between 24 and 32 bit samples, or between original and lossy encoding instead? — Does he have a separate home cinema room with audiophile wall covers, or just a living room with furnitures?

And most important: Does he use gold-capped digital cables?

Get that steam out of here ... ;)

robertcollier4

7th February 2013, 11:46

If you think qaac supports 64-bit floating point input, you can use the "-full" switch, which lets eac3to output 64-bit float (or so i remember)

Thank you nevcairiel - the -full switch is exactly what I was looking for. eac3to DPLII algorithm works internally with and outputs 64-bit-depth and so this is the best thing to pass to qaac compressor.

eac3to.exe "video_6chdts.mkv" 2: stdout.wav -downDpl -full | qaac.exe --tvbr 127 --quality 2 --rate keep --ignorelength --no-delay - -o "2chaac.m4a"

Works great. eac3to now no longer reduces bit-depth. And qaac accepts the 64-bit-depth piped input just fine and a down dithering operation was avoided. I prefer using qaac because it has a special "--no-delay" switch (https://sites.google.com/site/qaacpage/news/qaacrelease212refalac112) (otherwise neroaacenc adds an encoder delay of 44ms).

I recommend adding the -full switch to the documentation (I could not find it anywhere). Also I advise anyone piping to qaac to use the -full switch for better quality - down dithering operations make a difference and if it can be avoided with a simple switch it should be. The m4a filesize produced by qaac with the "eac3to -full" switch is also approx. same - it just feeds qaac a more accurate source for it to apply its compression to.

filler56789

7th February 2013, 17:05

@ madshi:

Info

- Document all the working parameters.

Did you miss that, or you just think it would be an "exceedingly-major" fix/improvement? :sly:

Thank you nevcairiel - the -full switch is exactly what I was looking for. eac3to DPLII algorithm works internally with and outputs 64-bit-depth and so this is the best thing to pass to qaac compressor.

XadoX

7th February 2013, 17:33

Yeah, so 00000.m2ts has about 1 extra minute of playback not referenced by the playlist.
Any new ideas of a workaround?

Guest

7th February 2013, 18:21

Chop off the extra stuff using 'Output Trimmed TS' in DGIndexNV. Then load the trimmed 00000.m2ts and 00008.m2ts (if that 1/3 second of video is crucial!) in DGIndexNV and off you go.

Alternatively, trim out the extra material in your Avisynth script.

XadoX

7th February 2013, 19:23

Chop off the extra stuff using 'Output Trimmed TS' in DGIndexNV...
Thx I will use your workaround.

frencher

9th February 2013, 01:27

Hi all and madshi,

I wanted to know if others have had the same problem as me for demuxing 3D Bluray PGS subtitles tracks (.sup) and if an update will fix this problem ?

Thanks

Sorry I do not understand English, are you fix ? :confused:
If I missed the topic that interests me here talking :rolleyes:

tebasuna51

9th February 2013, 09:20

frencher

10th February 2013, 00:35

@frencher
About your question: "I wanted to know if others have had the same problem as me for demuxing 3D Bluray PGS subtitles tracks (.sup)"
Seems the answer is "Nope", at least not for me.

Please put the log file, if show any <ERROR> or <WARNING>, or more info about your problem.
One example from original PANASONIC - AVATAR 3D
My cmd line with eac3to 3.27 is:
"F:\Temp\eac3to.exe" "P:\BDMV\PLAYLIST\00854.mpls" -demux
Works without errors with tsMuxeR GUI 1.10.6

My log:
eac3to v3.27
command line: "F:\Temp\eac3to.exe" "P:\BDMV\PLAYLIST\00854.mpls" -demux
------------------------------------------------------------------------------
M2TS, 2 video tracks, 4 audio tracks, 7 subtitle tracks, 2:41:42, 24p /1.001
1: Chapters, 53 chapters
2: h264/AVC (left eye), 1080p24 /1.001 (16:9)
3: h264/AVC (right eye), 1080p24 /1.001 (16:9)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz, -9ms
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
5: AC3, French, 5.1 channels, 384kbps, 48kHz, dialnorm: -27dB, -9ms
6: AC3, German, 5.1 channels, 384kbps, 48kHz, -9ms
7: AC3, Italian, 5.1 channels, 384kbps, 48kHz, -9ms
8: Subtitle (PGS), English
9: Subtitle (PGS), French
10: Subtitle (PGS), German
11: Subtitle (PGS), Italian
12: Subtitle (PGS), Dutch
13: Subtitle (PGS), Turkish
14: Subtitle (PGS), English
Creating file "00001 - Chapters.txt"...
[s14] Extracting subtitle track number 14...
[a07] Extracting audio track number 7...
[a04] Extracting audio track number 4...
[s13] Extracting subtitle track number 13...
[s10] Extracting subtitle track number 10...
[s08] Extracting subtitle track number 8...
[a05] Extracting audio track number 5...
[a06] Extracting audio track number 6...
[s11] Extracting subtitle track number 11...
[v02] Extracting video track number 2...
[s12] Extracting subtitle track number 12...
[s09] Extracting subtitle track number 9...
[v03] Extracting video track number 3...
[v03] Creating file "00001 - 3 - h264 (right eye), 1080p24.h264"...
[a06] A remaining delay of -9ms could not be fixed.
[a04] Applying DTS delay...
[a05] Removing AC3 dialog normalization...
[a04] A remaining delay of +1ms could not be fixed.
[a07] A remaining delay of -9ms could not be fixed.
[a05] A remaining delay of -9ms could not be fixed.
[v02] Creating file "00001 - 2 - h264 (left eye), 1080p24.h264"...
[a04] Creating file "00001 - 4 - DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz.dtsma"...
[a06] Creating file "00001 - 6 - AC3, German, 5.1 channels, 384kbps, 48kHz.ac3"...
[a07] Creating file "00001 - 7 - AC3, Italian, 5.1 channels, 384kbps, 48kHz.ac3"...
[a05] Creating file "00001 - 5 - AC3, French, 5.1 channels, 384kbps, 48kHz.ac3"...
[s08] Creating file "00001 - 8 - Subtitle (PGS), English.sup"...
[s09] Creating file "00001 - 9 - Subtitle (PGS), French.sup"...
[s14] Creating file "00001 - 14 - Subtitle (PGS), English.sup"...
[s08] [0:00:02] The source file seems to be damaged (discontinuity). <WARNING>
[s09] [0:00:02] The source file seems to be damaged (discontinuity). <WARNING>
[s13] Creating file "00001 - 13 - Subtitle (PGS), Turkish.sup"...
[s10] Creating file "00001 - 10 - Subtitle (PGS), German.sup"...
[s11] Creating file "00001 - 11 - Subtitle (PGS), Italian.sup"...
[s12] Creating file "00001 - 12 - Subtitle (PGS), Dutch.sup"...
[s08] [2:33:19] The source file seems to be damaged (discontinuity). <WARNING>
[s09] [2:41:01] The source file seems to be damaged (discontinuity). <WARNING>
[s09] [2:41:03] The source file seems to be damaged (discontinuity). <WARNING>
[s09] [2:41:11] The source file seems to be damaged (discontinuity). <WARNING>
[s08] [2:41:11] The source file seems to be damaged (discontinuity). <WARNING>
Video track 2 contains 232607 frames.
Video track 3 contains 232607 frames.
Subtitle track 8 contains 1524 normal and 83 forced captions.
Subtitle track 9 contains 1337 normal and 82 forced captions.
Subtitle track 10 contains 1379 normal and 86 forced captions.
Subtitle track 11 contains 1385 normal and 86 forced captions.
Subtitle track 12 contains 1465 captions.
Subtitle track 13 contains 1469 captions.
Subtitle track 14 contains 136 captions.
eac3to processing took 14 minutes, 54 seconds.
Done.

tebasuna51

10th February 2013, 10:29

@frencher
Well, maybe there are corrupt data or other problem with this BD.
I can help you with this, use tsMuxeR if work fine.

frencher

10th February 2013, 15:22

Overdrive80

12th February 2013, 04:32

Hi, one question. If I use "-edit=0:01:30.054,1000ms" for insert delay on ac3 file, eac3to doesnt transcoding, do it?

In short, my question is whether decodes and encodes to insert the delay, with the consequent loss of quality

tebasuna51

12th February 2013, 12:01

With a 48 KHz AC3 only insert 31 ac3 frames (31 x 32 ms = 992 ms), the default for 1000ms is add silence frames, if you want copy the precedent frames instead (loop) use the parameter -loop.

v3.0 (changelog)
* for gaps, edits & repairs > 1000ms eac3to now inserts silence by default
* for gaps, edits & repairs < 1000ms eac3to now loops audio by default
* option "-silence" forces eac3to to insert silence instead of looping audio
* option "-loop" forces eac3to to loop audio instead of inserting silence

tebasuna51

12th February 2013, 15:02

Here is a screenshot of the subtitle in question is the subtitle 3D is perhaps that eac3to to create the error.

The link to download the subtitles (PGS) .sup => HERE (http://ul.to/g9f6xeur)

Without problems with Subtitle Edit 3.2.7 and "00001 - 8 - Subtitle (PGS), English, 1524 normal captions, 83 forced captions.sup" and your image (more dificult when special font in native dialogs).

Seems a OCR issue, not related with eac3to.

robertcollier4

12th February 2013, 17:23

I recommend adding the -log="filename.log" switch to the eac3to help output as well in an effort to document all working switches.

Overdrive80

12th February 2013, 18:04

frencher

12th February 2013, 20:47

Without problems with Subtitle Edit 3.2.7 and "00001 - 8 - Subtitle (PGS), English, 1524 normal captions, 83 forced captions.sup" and your image (more dificult when special font in native dialogs).

Seems a OCR issue, not related with eac3to.

eac3to v3.27
command line: "F:\Temp\eac3to.exe" "P:\BDMV\PLAYLIST\00854.mpls" -demux

I did not say there was a problem with the subtitles but when démuxing and if too much error in "eac3to v3.27" stops.
It is true that I use a translator because I am French and I have trouble with English so that my application does not mean the same thing if google translator mistranslated.
The best will be to test the BD ISO or Original 3DBD PANASONIC AVATAR.

m00c0w

14th February 2013, 12:02

I'm having a problem with downmixing DTS 5.1 to Stereo AC3. Every time I downmix I get a negative audio gain which makes the resulting audio too low when I remux it with the original video file.

Here are the different syntaxes that I've tried:

"G:\Apps\AV Tools\eac3to\eac3to.exe" track2_eng.dts output.ac3 -down2 -192
"G:\Apps\AV Tools\eac3to\eac3to.exe" track2_eng.dts output.ac3 -down2 -192 -normalize
"G:\Apps\AV Tools\eac3to\eac3to.exe" track2_eng.dts output.ac3 -down2 -192 +0dB

The same file is produced for each of the above command lines and the result is always the same:

DTS, 5.1 channels, 1:58:53, 24 bits, 1509kbps, 48kHz
Decoding with ArcSoft DTS Decoder...
Downmixing multi channel audio to stereo...
Encoding AC3 <192kbps> with libAften...
Creating file "output.ac3"...
Clipping detected, a 2nd pass will be necessary. <WARNING>
Starting 2nd pass...
Decoding with ArcSoft DTS Decoder...
Downmixing multi channel audio to stereo...
Encoding AC3 <192kbps> with libAften...
Applying -3.22dB gain...
Creating file "output.ac3"...
eac3to processing took 54 minutes, 32 seconds.
Done.

What am I doing wrong?

Many thanks.

nevcairiel

14th February 2013, 12:20

eac3to tells you that there was clipping in the audio, and to avoid that clipping, it has to apply a negative gain (reduce volume so it doesn't clip anymore). Downmixing always has this risk.
Nothing is wrong, this is how its supposed to behave ;)

m00c0w

14th February 2013, 12:40

Ah I see :( Is there no other way to avoid clipping other than reducing the volume of the whole audio? When I play the DTS 5.1 audio through my stereo speaker system on my PC the audio volume is fine, but after downmixing it's a lot lower. What does my PC do differently to process the 5.1 audio through stereo output?

tebasuna51

14th February 2013, 16:31

Ah I see :( Is there no other way to avoid clipping other than reducing the volume of the whole audio?
Nope, is the correct way.

When I play the DTS 5.1 audio through my stereo speaker system on my PC the audio volume is fine, but after downmixing it's a lot lower. What does my PC do differently to process the 5.1 audio through stereo output?

A lot?
We don't know what your PC does, but something wrong.

LigH

14th February 2013, 17:05

Depends on the downmixing strategy.

The pessimistic strategy is: Downmix 6 channels with 1/6 of each volume to ensure there is no clipping. Disadvantage: Probably too quiet.

The optimistic strategy is: Downmix 6 channels with 1/sqrt(6) of each volume and trust the probability. Disadvantage: Clipping is possible.

The elaborate strategy is: Downmix 6 channels in the second pass with the factor gathered in a statistical first pass (2-pass normalization). Disadvantage: Takes more time, not real-time.

The dynamic strategy is: Downmix 6 channels and keep changing the factor with a brief look-ahead (Automatic Gain Control). Disadvantage: Pumping volume.

Asmodian

15th February 2013, 02:39

eac3to uses the "optimistic" strategy but switches to the "elaborate" strategy when clipping is detected, as in this case.

tebasuna51

15th February 2013, 12:48

Sparktank

16th February 2013, 00:44

I've been getting a lot of quieter sources too when downmixing.
I use the simple stereo downmix because I have just my laptop speakers plus my TV. (My receiver/amplifier just had it's volume dial fall off :()
Quite a few movies have such a huge difference between the dialog (which seems to be "creative" and "artistic" if they whisper) and loud action sfx.
I also notice that the music seems to be in the same volume levels as the sfx. Dialog is just a lost art in movies.

Often I'll get -7db for normalization. And quite often, clipping is always detected so a second pass is necessary.
I really don't want clipping so I allow the second pass normalization.
However, the overall drop in volume makes it quieter for playback. I really have to crank the volume on the TV set to hear everything.

I've found some helpful techniques using SoX's compand function in this thread...
http://forum.doom9.org/showthread.php?p=1589644#post1589644

It works great, for the most part.
I don't use the wavbooster app as it converts the sample rate to 44.1kHz. I don't want that. Ever.

I'm still experimenting, as each movie has different mastering applied so no single setting will fit all movies.

----
The Quesion:

Should I downmix with eac3to using -downstereo (in WAV) then use SoX to do the compand?
Or can SoX compand a 6ch WAV and then use eac3to to downmix?

What's the logical approach to downmixing to simple stereo and companding?
It almost seems weird to downmix to simple stereo with negative normalization applied (to avoid clipping) and then raise it again.
Wouldn't some of the data be lost or altered when normalized to quieter decibels, in effect creating odd audio gains during the compand?

Or am I overthinking this?
I want to downmix and compand with the least "logical" steps to alter the data as much as possible.
I have the space to do everything in WAV before converting to the final AC3 product. (Or if I get better at remembering the stdout/stdin)
I don't mind the processing time either.

robertcollier4

16th February 2013, 01:06

eac3to uses the "optimistic" strategy but switches to the "elaborate" strategy when clipping is detected, as in this case.
Yes, and the "optimistic" strategy is here: http://forum.doom9.org/showthread.php?p=1600695#post1600695

This is only for -down2 right? What is the strategy for -downDpl used by eac3to? Is there any eac3to command line switch to have eac3to always use the 2-pass "elaborate" downmixing strategy by default?

Also, does using Nero 7 with -nero switch if available improve the quality of a downmix by using a different downmixing DPLII algorithm? I was under the impression that if Nero 7 AC3 decoder is used for downmixing then it might be using a higher quality commercial proprietary Dolby DPLII algorithm - is this true?

Asmodian

16th February 2013, 02:38

The link provided by tebasuna51 actually shows the downmix from all multichannel audio formats with both -down2 and -downDpl.

I believe the result from the "elaborate" method can be forced by using -normalize.

madshi

16th February 2013, 08:25

The latest eac3to version always uses the "elaborate" method, for all downmixing variants.

tebasuna51

16th February 2013, 12:11

The Quesion:

Should I downmix with eac3to using -downstereo (in WAV) then use SoX to do the compand?
Or can SoX compand a 6ch WAV and then use eac3to to downmix?

If you use sox compand, with a profile like "Film standard" in AC3 Dynamic Range Compression, you can finish with max peak at -11 dB, and you need Normalize after.

I think is better use sox compand over the 6 channels and after use eac3to with -downStereo -normalize.

Also, if compand is used over each channel, you can improve the dialogs volume in Center channel. If compand is used over the downmix the dialogs can be masked with global volume.

nautilus7

16th February 2013, 14:59

Hi, I got this output when demuxing Cars (2006) Blu-ray Disc (us version). It's a seamless branched disc.
eac3to 1) -demux -keepdialnorm
M2TS, 1 video track, 10 audio tracks, 7 subtitle tracks, 1:56:35, 119.881p
1: Chapters, 32 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48kHz
(embedded: AC3 EX, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB)
4: AC3 EX, English, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
5: AC3 EX, French, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
6: AC3 EX, Spanish, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
7: AC3, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
8: AC3, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
9: AC3, French, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
10: AC3, French, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
11: AC3, Spanish, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
12: AC3, Spanish, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
13: Subtitle (PGS), English
14: Subtitle (PGS), French
15: Subtitle (PGS), Spanish
16: Subtitle (PGS), French
17: Subtitle (PGS), Spanish
18: Subtitle (PGS), English
19: Subtitle (PGS), English
Creating file "00145 - Chapters.txt"...
s15 Extracting subtitle track number 15...
a07 Extracting audio track number 7...
a05 Extracting audio track number 5...
a04 Extracting audio track number 4...
s14 Extracting subtitle track number 14...
a11 Extracting audio track number 11...
a03 Extracting audio track number 3...
v02 Extracting video track number 2...
a06 Extracting audio track number 6...
s16 Extracting subtitle track number 16...
s18 Extracting subtitle track number 18...
a09 Extracting audio track number 9...
a12 Extracting audio track number 12...
a08 Extracting audio track number 8...
s19 Extracting subtitle track number 19...
s13 Extracting subtitle track number 13...
a10 Extracting audio track number 10...
a03 Extracting audio track number 3...
s17 Extracting subtitle track number 17...
a03 Extracting TrueHD stream...
a03 Extracting TrueHD stream...
a03 Decoding with libav/ffmpeg...
a03 Remapping channels...
a03 Encoding AC3 <640kbps> with libAften...
v02 Creating file "00145 - 2 - h264, 1080p24.h264"...
a03 Creating file "00145 - 3 - TrueHD+AC3, English, 5.1 channels, 48kHz.thd+ac3"...
a06 Creating file "00145 - 6 - AC3 EX, Spanish, 5.1 channels, 640kbps, 48kHz.ac3"...
a05 Creating file "00145 - 5 - AC3 EX, French, 5.1 channels, 640kbps, 48kHz.ac3"...
a04 Creating file "00145 - 4 - AC3 EX, English, 5.1 channels, 640kbps, 48kHz.ac3"...
s18 Creating file "00145 - 18 - Subtitle (PGS), English.sup"...
a07 Creating file "00145 - 7 - AC3, English, 2.0 channels, 192kbps, 48kHz.ac3"...
a08 Creating file "00145 - 8 - AC3, English, 2.0 channels, 192kbps, 48kHz.ac3"...
a10 Creating file "00145 - 10 - AC3, French, 2.0 channels, 192kbps, 48kHz.ac3"...
a11 Creating file "00145 - 11 - AC3, Spanish, 2.0 channels, 192kbps, 48kHz.ac3"...
a12 Creating file "00145 - 12 - AC3, Spanish, 2.0 channels, 192kbps, 48kHz.ac3"...
a09 Creating file "00145 - 9 - AC3, French, 2.0 channels, 192kbps, 48kHz.ac3"...
s19 Creating file "00145 - 19 - Subtitle (PGS), English.sup"...
a03 Skipping identical AC3 frames (seamless branching)...
a03 Skipping identical AC3 frames (seamless branching)...
a06 Skipping identical AC3 frames (seamless branching)...
a04 Skipping identical AC3 frames (seamless branching)...
a05 Skipping identical AC3 frames (seamless branching)...
a10 Skipping identical AC3 frames (seamless branching)...
a09 Skipping identical AC3 frames (seamless branching)...
a08 Skipping identical AC3 frames (seamless branching)...
a12 Skipping identical AC3 frames (seamless branching)...
a07 Skipping identical AC3 frames (seamless branching)...
a11 Skipping identical AC3 frames (seamless branching)...
s13 Creating file "00145 - 13 - Subtitle (PGS), English.sup"...
s14 Creating file "00145 - 14 - Subtitle (PGS), French.sup"...
s15 Creating file "00145 - 15 - Subtitle (PGS), Spanish.sup"...
s17 Creating file "00145 - 17 - Subtitle (PGS), Spanish.sup"...
a03 Audio overlaps for 5ms at playtime 0:19:08.
a03 Audio overlaps for 5ms at playtime 1:20:15.
a03 The audio gaps/overlaps can't be removed from the TrueHD bitstream.
a03 In order to remove them you'll have to transcode to another format.
a04 Audio has a gap of 1ms at playtime 0:01:15.
a04 Audio has a gap of 2ms at playtime 0:04:24.
a04 Audio has a gap of 11ms at playtime 1:33:56.
a04 Audio has a gap of 7ms at playtime 1:42:04.
a04 Audio has a gap of 10ms at playtime 1:42:36.
a05 Audio has a gap of 1ms at playtime 0:01:15.
a05 Audio has a gap of 2ms at playtime 0:04:24.
a05 Audio has a gap of 11ms at playtime 1:33:56.
a05 Audio has a gap of 7ms at playtime 1:42:04.
a05 Audio has a gap of 10ms at playtime 1:42:36.
a06 Audio has a gap of 1ms at playtime 0:01:15.
a06 Audio has a gap of 2ms at playtime 0:04:24.
a06 Audio has a gap of 11ms at playtime 1:33:56.
a06 Audio has a gap of 7ms at playtime 1:42:04.
a06 Audio has a gap of 10ms at playtime 1:42:36.
a07 Audio has a gap of 1ms at playtime 0:01:15.
a07 Audio has a gap of 2ms at playtime 0:04:24.
a07 Audio has a gap of 11ms at playtime 1:33:56.
a07 Audio has a gap of 7ms at playtime 1:42:04.
a07 Audio has a gap of 10ms at playtime 1:42:36.
a08 Audio has a gap of 1ms at playtime 0:01:15.
a08 Audio has a gap of 2ms at playtime 0:04:24.
a08 Audio has a gap of 11ms at playtime 1:33:56.
a08 Audio has a gap of 7ms at playtime 1:42:04.
a08 Audio has a gap of 10ms at playtime 1:42:36.
a09 Audio has a gap of 1ms at playtime 0:01:15.
a09 Audio has a gap of 2ms at playtime 0:04:24.
a09 Audio has a gap of 11ms at playtime 1:33:56.
a09 Audio has a gap of 7ms at playtime 1:42:04.
a09 Audio has a gap of 10ms at playtime 1:42:36.
a10 Audio has a gap of 1ms at playtime 0:01:15.
a10 Audio has a gap of 2ms at playtime 0:04:24.
a10 Audio has a gap of 11ms at playtime 1:33:56.
a10 Audio has a gap of 7ms at playtime 1:42:04.
a10 Audio has a gap of 10ms at playtime 1:42:36.
a11 Audio has a gap of 1ms at playtime 0:01:15.
a11 Audio has a gap of 2ms at playtime 0:04:24.
a11 Audio has a gap of 11ms at playtime 1:33:56.
a11 Audio has a gap of 7ms at playtime 1:42:04.
a11 Audio has a gap of 10ms at playtime 1:42:36.
a12 Audio has a gap of 1ms at playtime 0:01:15.
a12 Audio has a gap of 2ms at playtime 0:04:24.
a12 Audio has a gap of 11ms at playtime 1:33:56.
a12 Audio has a gap of 7ms at playtime 1:42:04.
a12 Audio has a gap of 10ms at playtime 1:42:36.
s16 Creating file "00145 - 16 - Subtitle (PGS), French.sup"...
a03 The original audio track has a constant bit depth of 24 bits.
a04 Starting 2nd pass...
a04 Realizing (E-)AC3 gaps...
a04 Creating file "00145 - 4 - AC3 EX, English, 5.1 channels, 640kbps, 48kHz.ac3"...
a05 Starting 2nd pass...
a05 Realizing (E-)AC3 gaps...
a05 Creating file "00145 - 5 - AC3 EX, French, 5.1 channels, 640kbps, 48kHz.ac3"...
a06 Starting 2nd pass...
a06 Realizing (E-)AC3 gaps...
a06 Creating file "00145 - 6 - AC3 EX, Spanish, 5.1 channels, 640kbps, 48kHz.ac3"...
a07 Starting 2nd pass...
a07 Realizing (E-)AC3 gaps...
a07 Creating file "00145 - 7 - AC3, English, 2.0 channels, 192kbps, 48kHz.ac3"...
a08 Starting 2nd pass...
a08 Realizing (E-)AC3 gaps...
a08 Creating file "00145 - 8 - AC3, English, 2.0 channels, 192kbps, 48kHz.ac3"...
a09 Starting 2nd pass...
a09 Realizing (E-)AC3 gaps...
a09 Creating file "00145 - 9 - AC3, French, 2.0 channels, 192kbps, 48kHz.ac3"...
a10 Starting 2nd pass...
a10 Realizing (E-)AC3 gaps...
a10 Creating file "00145 - 10 - AC3, French, 2.0 channels, 192kbps, 48kHz.ac3"...
a11 Starting 2nd pass...
a11 Realizing (E-)AC3 gaps...
a11 Creating file "00145 - 11 - AC3, Spanish, 2.0 channels, 192kbps, 48kHz.ac3"...
a12 Starting 2nd pass...
a12 Realizing (E-)AC3 gaps...
a12 Creating file "00145 - 12 - AC3, Spanish, 2.0 channels, 192kbps, 48kHz.ac3"...
Video track 2 contains 167736 frames.
Subtitle track 13 contains 1732 captions.
Subtitle track 14 contains 1525 captions.
Subtitle track 15 contains 1494 captions.
Subtitle track 16 contains 83 captions.
Subtitle track 17 contains 89 captions.
Subtitle track 18 contains 1815 captions.
Subtitle track 19 contains 1582 captions.
eac3to processing took 25 minutes, 57 seconds.
Done.

It seems to me like the 1 (identical) frame that is removed, causes these gaps later on. I say so, because if you sum the duration of the gaps for each audio tracks you have 31ms (almost equal to 1 ac3 frame).

Any ideas? I have never come across a disc that has overlaps and gaps at the same time...

nautilus7

16th February 2013, 16:18

I noticed another thing as well.

If I demux a matroska file (.mkv) which contains a FLAC audio track, then eac3to decodes the FLAC and encodes it again as FLAC. Why does this happen?

Sparktank

17th February 2013, 05:44

I think is better use sox compand over the 6 channels and after use eac3to with -downStereo -normalize.

Also, if compand is used over each channel, you can improve the dialogs volume in Center channel. If compand is used over the downmix the dialogs can be masked with global volume.

Thank you! I'll give this a whirl next time. I had thoughts that this might be more efficient.

sl1pkn07

17th February 2013, 15:33

about:

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

why no use NeroAacDec from NeroAACCodec?

06_taro

17th February 2013, 15:51

Didn't do any further tests, but AAC is encoded in floating point precision yet NeroAacDec limits its decoded data to only 16-bit, while Nero's directshow decoder outputs 24-bit

sl1pkn07

17th February 2013, 16:19

ok

pity nero +7 never work on linux (through wine)

frencher

20th February 2013, 09:09

robertcollier4

21st February 2013, 17:13

I noticed that it is said that eac3to 5.1ch -DownDpl algorithm does not use any of the information from the LFE track??

Is it recommended to use -mixlfe with -downDpl?

From: http://forum.doom9.org/showthread.php?p=1600695#post1600695
3/2 or 3/2.1
Stereo
FL' = FL + 0.7071 x FC + BL
FR' = FR + 0.7071 x FC + BR
Dpl
FL' = FL + 0.7071 x FC + 0.8660 x BL + 0.5000 x BR
FR' = FR + 0.7071 x FC - 0.5000 x BL - 0.8660 x BR

http://forum.doom9.org/showthread.php?p=1600695#post1600695
If exist the parameter -mixlfe add:
FL' = ... + 0.7071 x LFE
FR' = ... + 0.7071 x LFE

tebasuna51

21st February 2013, 23:36

Is it recommended to use -mixlfe with -downDpl?

Dolby Digital recommend NOT use LFE in DPL dowmix.

But, of course, it's your choice.

LigH

22nd February 2013, 08:43

I preferred to mix the LFE with -3 dB when I used Azid.

robertcollier4

22nd February 2013, 09:49

After many downmixes to play AC3 5.1 movies on my stereo hardware player, here are the settings I have found optimal. A batch file included if it might be useful to others.

My findings:
1) I prefer the Nero 7 AC3 decoder (if you have it installed), thus I force it over ffmpeg using the -nero switch. I have not had any problems with the "bug in disabling Nero DRC" in a few hundred encodes.
2) I prefer -downStereo over -downDpl because it gives cleaner dialogues. Although -downDpl effects might sound a little better due to the 'matrix' echo effect in left channel, I found dialogues to be cleaner in -downStereo.
3) I prefer -mixlfe as it includes bass that gets picked up by the crossover on my 2.1 speaker set.
4) I use -full to let eac3to pass qaac 64 bit-depth data since that is what is generated by the eac3to downmixer anyways. No need to do a down dithering operation since qaac will take care of that when it converts the audio data to floating point.
5) I use qaac because it has --no-delay which removes the 44ms encoder delay introduced by all AAC encoders. Thus the audio is insured to be in sync with the original video.

@echo off
set PATH_EAC3TO=D:\PortableApps\VideoProcessing\eac3to327
set PATH_MKVMERGE=D:\PortableApps\VideoProcessing\mkvtoolnix-unicode-6.0.0
set PATH_QAAC=D:\PortableApps\VideoProcessing\qaac
set PATH=%PATH%;%PATH_EAC3TO%;%PATH_MKVMERGE%;%PATH_QAAC%

for /f %%a IN ('dir /b *.mkv') do (
echo ---------------------- BATCH STARTING %%~na%%~xa
eac3to.exe "%%~na%%~xa" -log="%%~na.eac3to.log" 2: stdout.wav -full -nero -downStereo -mixlfe -normalize | qaac.exe --tvbr 127 --quality 2 --rate keep --ignorelength --no-delay - -o "%%~na.m4a" 1>"%%~na.log" 2>&1
mkvmerge.exe -o "%%~na-2chremux.mkv" --no-audio "%%~na%%~xa" "%%~na.m4a" 1>>"%%~na.log" 2>&1
)
pause

tebasuna51

22nd February 2013, 11:39

I preferred to mix the LFE with -3 dB when I used Azid.
Is the same than:
http://forum.doom9.org/showthread.ph...95#post1600695
If exist the parameter -mixlfe add:
FL' = ... + 0.7071 x LFE
FR' = ... + 0.7071 x LFE

tebasuna51

22nd February 2013, 12:20

1) I prefer the Nero 7 AC3 decoder (if you have it installed), thus I force it over ffmpeg using the -nero switch. I have not had any problems with the "bug in disabling Nero DRC" in a few hundred encodes.
Apply DRC is not a problem when you want downmix, normalize and want more dialogs volume.
Is only a problem when you want preserve the original quality of your source.

2) I prefer -downStereo over -downDpl because it gives cleaner dialogues. Although -downDpl effects might sound a little better due to the 'matrix' echo effect in left channel, I found dialogues to be cleaner in -downStereo.
Of course -downStereo is preferred when you never play the audio with a 5.1 system with DPL decoder.

BTW:
- the echo effect must be in both left and right channels
- the contribution of Center channel (dialogues) is the same in -downStereo and -downDpl mix.

3) I prefer -mixlfe as it includes bass that gets picked up by the crossover on my 2.1 speaker set.
Maybe when you play with a 2.1 audio system, never recommended when play with TV speakers.
Add the LFE channel reduce the global volume of the rest of channels to avoid clip when normalize.
Also Dolby Digital say than mathematical add LFE to front channels can produce, sometimes, cancels or not desired effects.

4) I use -full to let eac3to pass qaac 64 bit-depth data since that is what is generated by the eac3to downmixer anyways. No need to do a down dithering operation since qaac will take care of that when it converts the audio data to floating point.
OK.
BTW preserve 64 bit precission is not very important after apply DRC, downmix and normalize, 24 bit integer is enough to convert to a lossy format.

5) I use qaac because it has --no-delay which removes the 44ms encoder delay introduced by all AAC encoders. Thus the audio is insured to be in sync with the original video.
NeroAacEnc put the delay in .m4a metadata and MkvMerge apply the needed delay in the AAC stream.

Overdrive80

26th February 2013, 01:37

Hi, I have .tsv file than if I rename to .ts, mediainfo show this details (http://pastebin.com/zJAGBM0d). But eac3to launch error. The video file is playback with mpc properly, premiere import without troubles. Mkvmerge import and all fine.

The source file is TV capture.

eac3to v3.27
command line: "C:\Program Files (x86)\MeGUI\tools\eac3to\eac3to.exe" "C:\Users\Isra\Desktop\INAZUMA ELEVEN_ '¡LA PRUEBA DEL CAPITAN!'(26022013_191942).TS" -progressnumbers -log="C:\Program Files (x86)\UsEac3to\UsEac3To.log"
------------------------------------------------------------------------------
The format of the source file could not be detected. <ERROR>

EDIT: DGindex is able to load too without troubles.

EDIT2: I attach example file (http://db.tt/H6OiU8p9)

EDIT3: I did contact with support department of device, and they said that I can handle tsv files with avidemux, and its true.

PowerGamer

28th February 2013, 18:51

When I want to extract DTS-HD MA 7.1 audio track from BluRay disc in its "original form" (for muxing into mkv later) I specify .dtsma extension for the output file (for ex.: "eac3to.exe 00800.mpls 3:audio.dtsma") and I don't need to have ArcSoft DTS Decoder installed, right? In other words, ArcSoft DTS Decoder needs to be installed only if I want eac3to to convert DTS-HD MA 7.1 audio track into some other format?

tebasuna51

28th February 2013, 22:49

@PowerGamer

Yes, is correct.

You can use .dts or .dtshd extension also, instead .dtsma, the stardard dts is extracted only if you put the parameter -core.

Anakunda

8th March 2013, 13:31

HI! I have finally managed to make working ArcSoft DTS decoder with eac3to but to my surprise the result is unusable.
Firstly I don't understand why the intermediate WAV by ArcSoft is 2-3x smaller than WAV made by ffmpeg, and secondly the result otherwise retains all 7.1 channels but the actors voice is missing at all and moreover after muxing it with video track the sound is (in KMP) simply ugly. What's wrong with ArcSoft? (I used DtsDec.dll and dtsdecoderdll.dll from latest TMT for this).

eac3to v3.24
command line: D:\media\eac3to\eac3to.exe Source.dtshd stdout.wav -normalize
------------------------------------------------------------------------------
DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48kHz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48kHz)
CAUTION: Decoding this track with ArcSoft results in low volume. <WARNING>
Decoding with ArcSoft DTS Decoder...
Applying RAW/PCM delay...
Writing WAV...
Creating file "stdout.pass1.wav"...
The original audio track has a constant bit depth of 24 bits.
Starting 2nd pass...
Reading WAV...
Reducing depth from 64 to 24 bits...
Writing WAV...
Applying 0.92dB gain...
Creating file "stdout.wav"...
The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 50 minutes, 16 seconds.
Done.

Boulder

8th March 2013, 13:36

Have you tried updating eac3to for starters? You are using an old version, the latest one is v3.27.

Anakunda

8th March 2013, 13:38

No yet, I noticed there's new version just now. I'll made a rip with new e3t and will see..

Anakunda

8th March 2013, 16:34

Wov it works with 3.27 :rolleyes: now why the temporary wav file for 2hour 8 channel audio track only has about 8GiB when the same 6 channel temporary wav from same track had almost 20gigs with ffmpeg? Does Arscoft decode lossy? This is what mediainfo writes about it:
Audio
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Signed
Codec ID : 00001000-0000-0100-8000-00AA00389B71
Bit rate mode : Constant
Bit rate : 9 216 Kbps
Channel(s) : 8 channels
Channel positions : Front: L C R, Side: L R, Back: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 24 bits

And why the encoded track is so quiet? This in log:

eac3to v3.27
command line: D:\media\eac3to\eac3to.exe Source.dtshd stdout.wav -normalize
------------------------------------------------------------------------------
DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48kHz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48kHz)
CAUTION: Decoding this track with ArcSoft results in low volume. <WARNING>
Decoding with ArcSoft DTS Decoder...
Applying RAW/PCM delay...
Writing WAV...
Creating file "stdout.pass1.wav"...
The original audio track has a constant bit depth of 24 bits.

Can this be fixed please?

rapscallion

8th March 2013, 17:38

HI! I have finally managed to make working ArcSoft DTS decoder with eac3to but to my surprise the result is unusable.

You don't say which version of AS decoder you're using.

Those files, from Ver v1.1.0.0, work perfectly. Anything newer, not so much.

Plus, 20gb for just an audio track. That's not right, considering the complete file, with video, is 27gb !

Edit: is source.dtshd a result of encoding via DTS Master Audio Suite? If so, you have to remove the header hex by demuxing, via Tsmuxer , before running through eac3to.

Anakunda

8th March 2013, 17:45

You don't say which version of AS decoder you're using.

Those files from Ver v1.1.0.0, work perfectly. Anything newer, not so much.

Plus, 20gb for just an audio track. That's not right !

Yes I got such a sizes with stadard dts decoder, don't know where's the difference. Maybe it was floating point while AS does 24 bit fixedpoint? I don't remember the std decoder output params already.

And where can I get the old AS decoder files which "work perfectly"? I extracted the mines from Totalmediatheatre 6.0.1.119 installation. Are these ok?

Edit: is source.dtshd a result of encoding via DTS Master Audio Suite? If so, you have to remove the header hex by demuxing, via Tsmuxer , before running through eac3to.
No this was extracted right from bluray disc by eac3to. Do I need to remove the header anyway?

rapscallion

8th March 2013, 17:57

No this was extracted right from bluray disc by eac3to. Do I need to remove the header anyway?

No, however, I would rename it to "dts"

PM sent

rapscallion

8th March 2013, 20:29

I just realized you were decoding to wav/pcm, so 8gb would not be unusual.

What are you trying to accomplish ?

Sparktank

9th March 2013, 06:26

I extracted the mines from Totalmediatheatre 6.0.1.119 installation. Are these ok?

For ArcSoft 1.1.0.0, the "dtsdecoderdll.dll" has two different dates you should watch out for.

(From TMT 2.1.6.120)
Good: v1.1.0.0 25/04/2008
Bad: v1.1.0.0 21/04/2008

More info here:
http://forum.doom9.org/showthread.php?p=1266679#post1266679

DarkSpace

9th March 2013, 12:56

now why the temporary wav file for 2hour 8 channel audio track only has about 8GiB when the same 6 channel temporary wav from same track had almost 20gigs with ffmpeg?

I just did some rough calculations, but (8 GB / 24 bit) * 64 bit equals roughly 21 GB, so I'd guess that ffmpeg decodes the lossy DTS core track at 64 bit floating point precision whereas ArcSoft decodes the lossless DTS-HD track at its specified precision of 24 bit. Comparing file sizes is not always a reliable way to find out which file is better, it only serves to show which file holds more (theoretical) information.

Anakunda

9th March 2013, 13:11

I guessed that libavcodec uses floating point too, this may explain the huge filesize difference. But I still get awfully sounding conversion on output with
eac3to Source.dtshd stdout.wav -normalize | qaac --ignorelength -o Destination.m4a - whichever ArcSoft files I use (now testing ArcSoft package 1.1.0.0)
What I don't understand if I open the dtshd file in foobar2000 using eac3to as decoder wrapper, the file plays very good, no annoying hiss and not quiet even in 7.1 layout. Simply no artifacts. Why the conversion is so poor, I use the same command line???

rapscallion

9th March 2013, 15:47

As I asked above, just what is it that you're trying to accomplish????

@Sparktank, he has v1.1.0.0 25/04/2008

Anakunda

9th March 2013, 15:50

I want to convert dts hd 8 channels track to aac 8 channels track

rapscallion

9th March 2013, 16:07

So, I assume, that you can play the BD on an Apple device.

I use Arcsoft Media Coverter for that. One step process and works every time.

Any further with eac3to, maybe someone else can chime in.

Anakunda

9th March 2013, 16:10

Oh yes I dont have apple device :cool:.
Maybe someone else can explain why the 8channels aac sound so distorted.

Boulder

9th March 2013, 16:14

I have sometimes had problems with outputting stuff if ffdshow is set to output 32bit floating point audio, there's static and distortion. Have you checked that the problem is not in the decoder of the result instead of something else. Maybe you could put a sample somewhere so we can test it too.

DarkSpace

9th March 2013, 16:52

Maybe someone else can explain why the 8channels aac sound so distorted.
As mentioned in the comment by Boulder, you need to make sure that isn't just your decoder making things sound ugly. Aside from that, I propose that you separate your command line into two steps and use an intermediate WAV file and play that back to verify that it's not eac3to that's making your audio sound ugly (remove the -full switch if you don't have enough free space).
eac3to "Source.dtshd" "Intermediate.wav" -normalize -full
qaac --ignorelength -o "Destination.m4a" "Intermediate.wav"
Also, I don't know how qaac works, so make sure that it's getting the input it expects (e.g. don't give it floating point data when it expects integer).

rapscallion

9th March 2013, 18:12

Oh yes I dont have apple device :cool:.
Maybe someone else can explain why the 8channels aac sound so distorted.

OK, you've really got me curious, what's the point/purpose of doing this conversion?

Chumbo

9th March 2013, 19:16

I have a few dts files I'm converting to ac3 but the ArcSoft decoder is reporting an error. What are the best ways to get the file converted using an alternate method? For now, I'm basically checking if a failure occurs on the default setting, then use -sonic and if that fails, use -nero. Is this a good approach? Seems to be working fine. Am I using the recommended next-in-line to the ArcSoft decoder by going to sonic first and then nero? Thanks.

Furiousflea

11th March 2013, 01:12

Chumbo

11th March 2013, 03:27

Yea, Arcsoft -> Sonic -> Nero

(Best avoid nero altogether ;) for any DTS-HD)
Thanks, that's what I thought. Thankfully, not a dts-hd track. ;)

Anakunda

11th March 2013, 12:54

I propose that you separate your command line into two steps and use an intermediate WAV file and play that back to verify that it's not eac3to that's making your audio sound ugly (remove the -full switch if you don't have enough free space).
I did so now and got strange results.
The dtshd track reported length by MI is none. In foobar2000 it's reported only 1h02m which is even not a half of true length.

ediainfo for dtshd: Audio
Format : DTS
Format/Info : Digital Theater Systems
Format profile : MA / Core
Mode : 16
Format settings, Endianness : Big
Bit rate mode : Variable
Bit rate : Unknown / 1 509 Kbps
Channel(s) : 8 channels / 6 channels
Channel positions : Front: L C R, Side: L R, Back: L R, LFE / Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 24 bits
Compression mode : Lossless / Lossy

Now I decoded it to wav using eac3to source.dtshd intermediate.wav -normalize -full

and got this 22Gigs big WAV file. If I open it in foobar2000 it seems to play all channels properly but is reported length only 10m29s

Audio
Format : PCM
Format profile : Float
Codec ID : 00001000-0000-0300-8000-00AA00389B71
Codec ID/Hint : IEEE
Duration : 2h 6mn
Bit rate mode : Constant
Bit rate : 24.6 Mbps
Channel(s) : 8 channels
Channel positions : Front: L C R, Side: L R, Back: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 64 bits
Stream size : 21.8 GiB (100%)

Also trying to convert this to aac results in only 10m29s track :(

DarkSpace

11th March 2013, 15:09

The dtshd track reported length by MI is none. In foobar2000 it's reported only 1h02m which is even not a half of true length.
I have no idea exactly what foobar2000 is doing, but I have read several times already that guessing the duration of an Elementary Stream with variable bitrate can be tricky (there is no container to tell you the true duration, after all).

Now I decoded it to wav using eac3to source.dtshd intermediate.wav -normalize -full and got this 22Gigs big WAV file. If I open it in foobar2000 it seems to play all channels properly but is reported length only 10m29s
Some players have problems with WAV files larger than 2 GB. If your player doesn't play the whole file, try using the .w64 format for conversion instead of .wav (no worries, both formats are lossless). If your player plays the whole file but displays a wrong duration, you don't have to convert to .w64, although of course it shouldn't harm things.

Also trying to convert this to aac results in only 10m29s track :(
Same as above, WAV files greater than 2 GB may be problematic. However, before I mention that you may need to look for a switch to convert the whole file to AAC, make sure the --ignorelength parameter is present. Alternatively, you can also use the .w64 approach mentiond above, if qaac can handle .w64 files.

dvbt

12th March 2013, 11:27

hello friends when I convert a file into ac3 EAC3 here is the error in EAC3 tools

eac3to v3.27
command line: "C:\Users\le pc de jax\Desktop\e\eac3to.exe" "D:\PVR SMART DVB\smartdvb DVB-C-DVB-T\numero23.eac3" "D:\PVR SMART DVB\smartdvb DVB-C-DVB-T\15.ac3" -192
------------------------------------------------------------------------------
E-AC3 Surround, 2.0 channels, 0:42:52, 128kbps, 48kHz, dialnorm: -23dB
Removing E-AC3 dialog normalization...
Decoding with libav/ffmpeg...
Encoding AC3 <192kbps> with libAften...
[libav] exponent out-of-range <WARNING>
[libav] error decoding the audio block <WARNING>
[libav] exponent out-of-range <WARNING>
[libav] error decoding the audio block <WARNING>
Creating file "D:\PVR SMART DVB\smartdvb DVB-C-DVB-T\15.ac3"...
[libav] exponent out-of-range <WARNING>
[libav] error decoding the audio block <WARNING>
[libav] exponent out-of-range <WARNING>
[libav] error decoding the audio block <WARNING>
[libav] exponent out-of-range <WARNING>
[libav] error decoding the audio block <WARNING>
eac3to processing took 20 seconds.
Done.

tebasuna51

12th March 2013, 16:53

Where is the <ERROR>?
I only see <WARNING>'s

The proccess can finish OK, maybe with some gaps because imperfect TV capture.
The output AC3 have similar length and play OK?

dvbt

12th March 2013, 17:40

hello the ts and perfect without any error

dvbt

12th March 2013, 17:44

here is the log of ts
# # File = D: \ SMART PVR DVB \ POUCHIN TV DVB-T PVR \ NUMBER 23 2013-01-14 12-06-13.ts
PIDs included (* = PCR) = 210, 220 *, 230, 231, 232, 240, 241
PIDs found root = 0
First and last PCR = 24:16:01.8, 25:53:44.3 (in pid 220)
Duration = 01:37:42.5
-------------------------------------------------- -------------------------------------------------
Current audit
Made.
Summary for this file:
Number of packets = 23201252
Continuity errors = 0
Error flags = 0
Error packets = 0
Total time = 01:37:42.5
Note that this term is only based on the time marker
found in the TS stream. If parts are missing in the file
output, the displayed value will likely be inaccurate.
verification is complete

I have a glitch when the bed by my EAC3 vlc here is the the problem is the first to second http://dl.free.fr/getfile.pl?file=/CeIa83pB

tebasuna51

13th March 2013, 00:41

Yes, there are a audible glitch at the begining. That hapens with TV capture.
Decode to wav, edit the problem and recode after to ac3.

Boulder

14th March 2013, 20:27

Does Sonic downmix 7.1 material to 5.1 when decoding or what does the "decodes only 5.1" mean in the first post? I have a 7.1 DTS-HD MA track which refuses to be decoded without errors with Arcsoft and I'd like to use the whole lossless track for encoding to AAC.

tebasuna51

15th March 2013, 13:10

I don't use Sonic decoder but you can try decoding this Channel_Test_DTS-MA_7.1 (http://www.sendspace.com/file/fva4xn) (only 1.3 MB) and see what hapens.
All channels have a max peak at 100% (LFE only 50%), analyzing the output wav you can know how work the decoder.

Infineon

20th March 2013, 20:33

I've been trying to convert DTS-HD to lossless WAV/FLAC for my media library. I've done some reading up on the requirements and have been using eac3to with the blu-ray stream extractor GUI but cannot get a suitable DTS-HD decoder working and just end up with 48Khz output from the DTS core. I gather you need to source files from an old version of TMT (2?) for the setup to work. I've been unable to find this, even a trial download.

Can anyone point me / msg me where I might get the required DLLs or is there another way to do this now?

LigH

21st March 2013, 08:27

Looking for the ArcSoft Codec Unlocker / ASAudioHD+ (http://forum.doom9.org/showthread.php?p=1392750)?

Or simply install RipBot264 (http://forum.doom9.org/showthread.php?t=151692)... :rolleyes:

Furiousflea

21st March 2013, 10:58

I use Arcsoft decoder to decode DTS-MA and encode to FLAC using eac3to.

However, I just read this thread and the person talks about changing default settings for the Arcsoft decoder...
http://forum.doom9.org/showthread.php?p=1392750

This makes me worry that the FLAC that is output from decoding the DTS-MA track has dynamic range compression applied.

Could someone confirm that this is or isn't the case and there is no need to mess around with any settings to use the Arcsoft decoder correctly (apart from having it recognized by eac3to of course)....

:eek:

Thanks in advance.

nevcairiel

21st March 2013, 11:10

eac3to should set the options for the decoder, you don't need to worry about it.

Furiousflea

21st March 2013, 11:25

eac3to should set the options for the decoder, you don't need to worry about it.

Thanks mate...was worrying about hundreds of hours of work ahead :o

Infineon

21st March 2013, 14:55

I got a trial copy of TMT 3 Platinum Retail 3.0.1.160 and installed it. Got RipBot264 setup and I copied the required ax and DLL files into the eac3to folder from the RipBot264 Tools\ax\ArcSoft DTS Decoder\ folder. I then registered the ASAudioHD.ax file within the eac3to folder (confirmed registered successfully) but still I get:

"[a02] The ArcSoft and Sonic decoders don't seem to work, will use libav instead."

The version of the dtsdecoderdll.dll currently in my eac3to folder (coped from RipBot) is 1.1.0.0. The CheckActivate I copied over is 1.0.0.2.

Any ideas where I've screwed up?

filler56789

21st March 2013, 16:16

^ @Infineon: make sure your eac3to folder is included in the PATH (http://en.wikipedia.org/wiki/PATH_(variable)) (environment-variable).

Infineon

21st March 2013, 19:01

Thanks I've just added it but it hasn't made a difference. This is frustrating :(

heerschop

22nd March 2013, 17:38

Thanks I've just added it but it hasn't made a difference. This is frustrating :(

In order, for me, to get arcsoft working with EAC3to I also have additional files in my arcsoft directory. I checked and when I remove one of these files, the decoder doesn't work anymore.
Here is the list of all the arcsoft decoder files:
- ASAudioHD.ax
- checkactivate.dll
- DtsDec.dll
- dtsdecoderdll.dll
- MagCore.dll
- MagPCMac.dll
- MagUIEngine.dll
- MagUIInter.dll

regsvr32.exe C:\xxx\Arcsoft\ASAudioHD.ax
add to path => C:\xxx\Arcsoft and Reboot

Hope this will help you,

Greetz

Infineon

23rd March 2013, 11:37

In my setup I have copied the files from a trial version of ArcSoft into my eac3to.exe folder root (rather than into an 'ArcSoft' sub-folder for example). The only file I didn't copy from your list is 'DtsDec.dll'. That wasn't mentioned in any of the setup guides I've seen.

I think my biggest worry is that I don't have the correct versions of these files as I wasn't able to source the earlier trial downloads of TMT (versions 2 or 3). Would someone be kind enough to zip a copy of these and provide? (So I can be sure this isn't the issue). The version of eac3to I'm using is: 3.27.0.0 but I'm running it using version 3 of the 'HdBrStreamExtractor'.

Overdrive80

23rd March 2013, 16:00

@Infineon @heerschop Please, see rule 6.

You have that put .dlls files in system32 or syswow64. If you OS is 32 bit then system32, and if is 64 bit in both, syswow64 and system32

Nico8583

24th March 2013, 00:16

Hi :)
Could you implement MKV AVC/MVC on next release ?
Thanks !

LigH

24th March 2013, 08:12

How exactly is AVC/MVC related to audio?

frencher

24th March 2013, 12:01

How exactly is AVC/MVC related to audio?

Same output of MakeMKV with Bluray 3D

Tuturial with MakeMKV HERE (http://www.hdfever.fr/2012/09/02/tutoriel-riper-un-blu-ray-3d-en-mkv-avcmvc-full-hd-3d-avec-makemkv/)

LigH

24th March 2013, 12:07

AVC is a video format.
MVC is a video subformat.

eac3to is an audio converter. What exactly is it supposed to support? I'd expect eac3to to ignore video completely, instead.

frencher

24th March 2013, 13:15

AVC is a video format.
MVC is a video subformat.

eac3to is an audio converter. What exactly is it supposed to support? I'd expect eac3to to ignore video completely, instead.

What is nice is that eac3to can combine in .mkv or streams .h264 same as MakeMKV and readable with stereo player.
eac3to can already pick and remux into a video stream .mkv and extract by .mpls or .SSIF fux the AVC/MVC.
eac3to also covers video and is not only dedicated audio has been developped even if the basis for that.
Being able to combine directly. Developer mkv can help other applications mvc encoders, readers free 3D bluray, and many are seeking a solution as there only to count the number of post http://forum.doom9.org/showthread.php?t=155246 this topic
I work elsewhere on player encoder and stream BD 3D AVC/MVC which is in my signature

Infineon

24th March 2013, 13:21

Overdrive, I've just checked and these files were put into the SysWow64 already but were not in System32 so I copied them in. Unfortunately this didn't change anything. I think I might have a PATH issue though, I've added the path to the System variables rather than user, hopefully this is correct? However, I notice that if I try to run any of the exe files in my eac3to folder by just typing their name in a command prompt such as 'C:\eac3to.exe' they do not launch, instead it says - 'eac3to' is not recognized as an internal or external command, operable program or batch file'.

If I type 'PATH' into a command prompt here is the output, I can see that the variable I've added is in fact the correct path for my eac3to executables folder (it's the last entry at the end):

C:\>path
PATH=C:\Program Files (x86)\NVIDIA Corporation\PhysX\Common;C:\Program Files\Com
mon Files\Microsoft Shared\Microsoft Online Services;C:\Program Files (x86)\Comm
on Files\Microsoft Shared\Microsoft Online Services;C:\Program Files\Common File
s\Microsoft Shared\Windows Live;C:\Program Files (x86)\Common Files\Microsoft Sh
ared\Windows Live;C:\Windows\system32;C:\Windows;C:\Windows\System32\Wbem;C:\Win
dows\System32\WindowsPowerShell\v1.0\;C:\Program Files (x86)\Windows Live\Shared
;C:\Program Files (x86)\QuickTime\QTSystem\; D:\Installs\Multimedia\Ripping\eac3
to\eac3to327

I'm wondering why this path doesn't work... any thoughts?

Infineon

24th March 2013, 13:35

Sorry I've just seen there was a space before the start of the path entry for eac3to, removing it fixed the problem and it's now working. Sadly HdBrStreamextractor still doesn't see the ArcSoft decoder and resorts to libav. I wonder if I should try just using the standard eac3to.exe command line executable to run the extraction? It's a bit complex for me though, I also have this GUI version: 'yr_eac3to_more_gui.exe' even this looks complex for me but I'll give it a try :)

filler56789

24th March 2013, 13:45

Sorry I've just seen there was a space before the start of the path entry for eac3to, removing it fixed the problem and it's now working.

Me thinks you deserve to be punished. http://forum.videohelp.com/attachments/2671-1279232225/uglylol.gif

Enjoy :p

Infineon

24th March 2013, 13:50

Probably ;) but it's still not working. I've just tried running eac3to using a GUI, this was the output:

eac3to v3.27
command line: "D:\Installs\Multimedia\Ripping\eac3to\eac3to327\eac3to.exe" "C:\Users\MyUser\Desktop\00003.m2ts" 3: "C:\Users\MyUser\Desktop\test
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 1:30:08, 60i /1.001
1: h264/AVC, 1080i60 /1.001 (16:9)
2: RAW/PCM, 2.0 channels, 24 bits, 96kHz
3: DTS Master Audio, 5.1 channels, 24 bits, 96kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
This audio conversion is not supported. <ERROR>

Infineon

24th March 2013, 14:09

I had left the 'wav' type off the end of that command to make it 'test.wav', when I fixed that and ran again I got: 'The program can't start because MSVCP71.dll is missing from your computer. Try reinstalling the program to fix this problem.' - But it worked !!! OMG...

Now this noob just needs to work out how to cut up a 9GB WAV file :) I think I can get the chapter data out into txt file using the hdbrextractor ui easily enough. Can anyone recommend a good way to chop up a wav based on chapter info txt file? Then I just need to FLAC encode - simple enough, even for me!

DarkSpace

24th March 2013, 22:33

Can anyone recommend a good way to chop up a wav based on chapter info txt file?
Take a look at mkvmerge (http://www.bunkus.org/videotools/mkvtoolnix/downloads.html), it allows you to load the Chapters.txt and split the output at chapter points.

Infineon

24th March 2013, 22:37

Thanks I'll do that. Been finding Foobar pretty useful too using cue files.

frencher

25th March 2013, 01:22

What is nice is that eac3to can combine in .mkv or streams .h264 same as MakeMKV and readable with stereo player.
eac3to can already pick and remux into a video stream .mkv and extract by .mpls or .SSIF fux the AVC/MVC.
eac3to also covers video and is not only dedicated audio has been developped even if the basis for that.
Being able to combine directly. Developer mkv can help other applications mvc encoders, readers free 3D bluray, and many are seeking a solution as there only to count the number of post http://forum.doom9.org/showthread.php?t=155246 this topic
I work elsewhere on player encoder and stream BD 3D AVC/MVC which is in my signature

Nobody is interested in developing a BluRay Player free...
Many of us seek a solution to some BluRay Disc capricious whose structure is complicated.
We would welcome author of eac3to has been about our problem reading and re-encoding that eac3to could solve for a small additional function.
cordially

LigH

25th March 2013, 09:27

eac3to is an audio converter.

I doubt it will ever be a complete Blu-ray authoring tool.

But I may have misunderstood. You may like to explain in french again, your english translation partially makes little sense...

LeXXuz

25th March 2013, 11:59

robertcollier4

25th March 2013, 17:38

I frequently downmix DTS 5.1 audio to 2.0 stereo for playing on my hardware player using "eac3to -downStereo" or "eac3to -downDpl". However with both I frequently have the problem (with DTS sources specifically) that explosions and sound effects are super loud and the dialogue are too quiet. What I want to do is basically "boost" Center channel.

The current co-efficients being used (http://forum.doom9.org/showthread.php?p=1600695#post1600695) by -downStereo are:
FL' = FL + 0.7071 x FC + BL
FR' = FR + 0.7071 x FC + BR

What I want is something like:
FL' = FL + 0.85 x FC + BL
FR' = FR + 0.85 x FC + BR

Can I somehow use custom co-efficients with eac3to? or what tool can I use to convert DTS to WAV with custom co-efficients? Thanks. I am using Windows and would prefer a command line utility.

---------------------------------------------------------------
UPDATE: I have found custom channel mixing from DTS can be done with ffmpeg.exe. For tracks in which eac3to downmixing coefficients result in too quiet dialogue, I have done extensive testing and found that mixing 0.25 coefficient Left and Right channels with 1 coefficient Center channels gives the best results for me for watching a DTS movie on a stereo system. Analytically this may seem like a very low amount of FL and FR - but it emphasizes crisp clear Center channel dialogues along with music which "accompanies" the movie instead of overpowering it.

This command will convert the first audio track in an mkv file into a stereo downmixed AAC M4A file. It will do the conversion in 32-bit float mode which is what libdts uses internally and so will avoid dithering before passing to AAC compressor. You need to have ffmpeg.exe and qaac.exe on your path.
ffmpeg.exe -report -loglevel verbose -i "moviename.mkv" -map 0:a:0 -f wav -acodec pcm_f32le -ac 2 -af pan=stereo:FL=0.25*FL+FC:FR=0.25*FR+FC - | qaac.exe --tvbr 127 --quality 2 --rate keep --ignorelength --no-delay - -o "moviename-audio-dialoguedownmix.m4a"

---------------------------------------------------------------

frencher

27th March 2013, 02:31

Warning !! Bad translation by Google:
That would be nice is that eac3to can combine in .mkv or stream .h264 as is already MakeMKV as readable and compatible with Stereoscopic player.
eac3to remux already allows a video stream in a container .mkv rip tracks but also by .mpls or .ssif flow AVC/MVC so eac3to handles video well even if it does not reach here was created and the database audio remains.
What good will is that eac3to can combine directly .mkv and or .h264 stream AVC/MVC a Bluray 3D.
We are looking for a more developer solution such as recoder MVC 3D bluray players, 3D video camcorder, and many are seeking a solution, but in the field of free there is no big thing, just count the number of positions about Guide to convert BD 3D to 3D Left+Right Stereoscopic and Anaglyph (http://forum.doom9.org/showthread.php?t=155246) that remains active but the solutions are very rare.
I also work on a drive, bluray recoder 2D & 3D + 3D BD flow AVC/MVC which is in my signature and would welcome madshi discusses about us... :thanks:

Ce qui serait bien, c'est que eac3to puisse combiner en .mkv ou flux .h264 comme le fait MakeMKV car déjà lisible et compatible avec Stereoscopic player.
eac3to permet déjà de remuxer un flux vidéo dans un conteneur .mkv mais aussi extraire les pistes par .mpls ou .SSIF les flux AVC/MVC donc eac3to gère bien la vidéo même s'il n'à pas été créé pour çà et que la base reste bien l'audio.
Ce qui serai bien c'est que eac3to puisse combiner directement en .mkv et, ou .h264 les 2 flux AVC/MVC d'un Bluray 3D.
Nous sommes plusieurs développeur à chercher une tel solution tel que réencodeur MVC, lecteurs bluray 3D, vidéos de camèscope 3D, et beaucoup cherchent une solution mais dans le domaine du gratuit on ne trouve pas grand chose, il suffit de compter le nombre de postes au sujet Guide to convert BD 3D to 3D Left+Right Stereoscopic and Anaglyph (http://forum.doom9.org/showthread.php?t=155246) qui reste actif mais les solutions sont trés rares.
Je travaille d'ailleurs sur un lecteur, réencodeur bluray 2D & 3D + flux 3D BD AVC/MVC qui est dans ma signature et serait heureux que madshi se penche sur notre sujet... :thanks:
http://i46.tinypic.com/2qanwcj.png

Kurtnoise

27th March 2013, 06:44

Ce qui serai bien c'est que eac3to puisse combiner directement en .mkv et, ou .h264 les 2 flux AVC/MVC d'un Bluray 3D.
eac3to uses the haali muxer to create mkv files, which is not compatible with MVC contents. Besides, this filter seems to be abandoned for a while...So, try something else.

frencher

27th March 2013, 14:59

OK thanks,

Sample with MakeMKV (http://forum.doom9.org/showthread.php?p=1620266#post1620266), do you have an idea to do the same work for free ?
We have "CombineMVC.exe" but this is the format for "transport stream" ( .ts .m2ts .mts) and not CLI

Kurtnoise

28th March 2013, 13:56

I'm not aware of such tool but why not using x264 within frame packing switch & mkv as output ?

Then, you should be able to mux your A/V streams with mkvtoolnix after that...just a guess.

frencher

28th March 2013, 19:29

I'm not aware of such tool but why not using x264 within frame packing switch & mkv as output ?

Then, you should be able to mux your A/V streams with mkvtoolnix after that...just a guess.

Try CombineMVC.exe (http://ul.to/edpwnxkm) and open the ouptut "combinedMVC.h264" with media info ;)

You are Aix en provence (13), i was born in Toulon (83) Tchii, Tchii, Tchii... :D

Anakunda

4th April 2013, 10:46

Thanks to all for the feedback about my conversion. Now I'm back to the problem. To resume in short the goal is to make a proper conversion from Dts-HD MA with correct channel mappings to Apple AAC with preserving all channels and using Arcsoft decoder.

Now I got this track

General
Complete name : english.dtshd
Format : DTS
Format/Info : Digital Theater Systems
Format profile : MA / Core
File size : 3.21 GiB
Overall bit rate mode : Variable

Audio
Format : DTS
Format/Info : Digital Theater Systems
Format profile : MA / Core
Mode : 16
Format settings, Endianness : Big
Bit rate mode : Variable
Bit rate : Unknown / 1 509 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

[dts @ 0000000002127520] max_analyze_duration 5000000 reached at 5002667 microseconds
[dts @ 0000000002127520] Estimating duration from bitrate, this may be inaccurate
Input #0, dts, from 'english.dtshd':
Duration: 04:59:34.15, start: 0.000000, bitrate: 1535 kb/s
Stream #0:0: Audio: dts (DTS-HD MA), 48000 Hz, 5.1(side), fltp, 1536 kb/s

(The true duration is 2h45m)

Using eac3to english.dtshd temp.wav -normalize -60ms I get this

General
Complete name : temp.wav
Format : Wave
File size : 7.98 GiB
Duration : 2h 45mn
Overall bit rate mode : Constant
Overall bit rate : 6 912 Kbps

Audio
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Signed
Codec ID : 00001000-0000-0100-8000-00AA00389B71
Duration : 2h 45mn
Bit rate mode : Constant
Bit rate : 6 912 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 24 bits
Stream size : 7.98 GiB (100%)

[wav @ 0000000002127160] max_analyze_duration 5000000 reached at 5003458 microseconds
Input #0, wav, from 'temp.wav':
Duration: 01:22:30.49, bitrate: 13852 kb/s
Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1(side), s32, 6912 kb/s

If I open in Audition all channels seem present and unique. Now I would need a tool that pipes the temporary wav for qaac so it makes a proper conversion.

This ffmpeg -i temp.wav -f f32le - | qaac --ignorelength -o English.m4a --tvbr 100 - generates an error. Advices please.

Here's the complete list of acceptable inputs for qaac:

LC 8000Hz Mono -- 8,12,16,20,24
LC 8000Hz Stereo (L R) -- 16,20,24,28,32,40,48
LC 8000Hz Quadraphonic -- 32,40,48,56,64,72,80,96
LC 8000Hz 4.0 (C L R Cs) -- 32,40,48,56,64,72,80,96
LC 8000Hz 5.0 (C L R Ls Rs) -- 40,48,56,64,72,80,96,112
LC 8000Hz 5.1 (C L R Ls Rs LFE) -- 40,48,56,64,72,80,96,112
LC 8000Hz 6.0 (C L R Ls Rs Cs) -- 48,56,64,72,80,96,112,128,144
LC 8000Hz 6.1 (C L R Ls Rs Cs LFE) -- 48,56,64,72,80,96,112,128,144
LC 8000Hz 7.0 (C L R Ls Rs Rls Rrs) -- 56,64,72,80,96,112,128,144,160
LC 8000Hz 7.1 (C Lc Rc L R Ls Rs LFE) -- 56,64,72,80,96,112,128,144,160
LC 8000Hz 8.0 (C L R Ls Rs Rls Rrs Cs) -- 64,72,80,96,112,128,144,160,192
LC 11025Hz Mono -- 8,12,16,20,24,28,32
LC 11025Hz Stereo (L R) -- 16,20,24,28,32,40,48,56,64
LC 11025Hz Quadraphonic -- 48,56,64,72,80,96,112,128
LC 11025Hz 4.0 (C L R Cs) -- 48,56,64,72,80,96,112,128
LC 11025Hz 5.0 (C L R Ls Rs) -- 64,72,80,96,112,128,144,160
LC 11025Hz 5.1 (C L R Ls Rs LFE) -- 64,72,80,96,112,128,144,160
LC 11025Hz 6.0 (C L R Ls Rs Cs) -- 72,80,96,112,128,144,160,192
LC 11025Hz 6.1 (C L R Ls Rs Cs LFE) -- 72,80,96,112,128,144,160,192
LC 11025Hz 7.0 (C L R Ls Rs Rls Rrs) -- 96,112,128,144,160,192,224
LC 11025Hz 7.1 (C Lc Rc L R Ls Rs LFE) -- 96,112,128,144,160,192,224
LC 11025Hz 8.0 (C L R Ls Rs Rls Rrs Cs) -- 96,112,128,144,160,192,224,256
LC 12000Hz Mono -- 12,16,20,24,28,32
LC 12000Hz Stereo (L R) -- 24,28,32,40,48,56,64
LC 12000Hz Quadraphonic -- 48,56,64,72,80,96,112,128
LC 12000Hz 4.0 (C L R Cs) -- 48,56,64,72,80,96,112,128
LC 12000Hz 5.0 (C L R Ls Rs) -- 64,72,80,96,112,128,144,160
LC 12000Hz 5.1 (C L R Ls Rs LFE) -- 64,72,80,96,112,128,144,160
LC 12000Hz 6.0 (C L R Ls Rs Cs) -- 72,80,96,112,128,144,160,192
LC 12000Hz 6.1 (C L R Ls Rs Cs LFE) -- 72,80,96,112,128,144,160,192
LC 12000Hz 7.0 (C L R Ls Rs Rls Rrs) -- 96,112,128,144,160,192,224
LC 12000Hz 7.1 (C Lc Rc L R Ls Rs LFE) -- 96,112,128,144,160,192,224
LC 12000Hz 8.0 (C L R Ls Rs Rls Rrs Cs) -- 96,112,128,144,160,192,224,256
LC 16000Hz Mono -- 12,16,20,24,28,32,40,48
LC 16000Hz Stereo (L R) -- 24,28,32,40,48,56,64,72,80,96
LC 16000Hz Quadraphonic -- 48,56,64,72,80,96,112,128,144,160,192
LC 16000Hz 4.0 (C L R Cs) -- 48,56,64,72,80,96,112,128,144,160,192
LC 16000Hz 5.0 (C L R Ls Rs) -- 64,72,80,96,112,128,144,160,192,224
LC 16000Hz 5.1 (C L R Ls Rs LFE) -- 64,72,80,96,112,128,144,160,192,224
LC 16000Hz 6.0 (C L R Ls Rs Cs) -- 72,80,96,112,128,144,160,192,224,256,288
LC 16000Hz 6.1 (C L R Ls Rs Cs LFE) -- 72,80,96,112,128,144,160,192,224,256,288
LC 16000Hz 7.0 (C L R Ls Rs Rls Rrs) -- 96,112,128,144,160,192,224,256,288,320
LC 16000Hz 7.1 (C Lc Rc L R Ls Rs LFE) -- 96,112,128,144,160,192,224,256,288,320
LC 16000Hz 8.0 (C L R Ls Rs Rls Rrs Cs) -- 96,112,128,144,160,192,224,256,288,320,384
LC 22050Hz Mono -- 16,20,24,28,32,40,48,56,64
LC 22050Hz Stereo (L R) -- 32,40,48,56,64,72,80,96,112,128
LC 22050Hz Quadraphonic -- 64,72,80,96,112,128,144,160,192,224,256
LC 22050Hz 4.0 (C L R Cs) -- 64,72,80,96,112,128,144,160,192,224,256
LC 22050Hz 5.0 (C L R Ls Rs) -- 80,96,112,128,144,160,192,224,256,288,320
LC 22050Hz 5.1 (C L R Ls Rs LFE) -- 80,96,112,128,144,160,192,224,256,288,320
LC 22050Hz 6.0 (C L R Ls Rs Cs) -- 96,112,128,144,160,192,224,256,288,320,384
LC 22050Hz 6.1 (C L R Ls Rs Cs LFE) -- 96,112,128,144,160,192,224,256,288,320,384
LC 22050Hz 7.0 (C L R Ls Rs Rls Rrs) -- 112,128,144,160,192,224,256,288,320,384,448
LC 22050Hz 7.1 (C Lc Rc L R Ls Rs LFE) -- 112,128,144,160,192,224,256,288,320,384,448
LC 22050Hz 8.0 (C L R Ls Rs Rls Rrs Cs) -- 128,144,160,192,224,256,288,320,384,448,512
LC 24000Hz Mono -- 16,20,24,28,32,40,48,56,64
LC 24000Hz Stereo (L R) -- 32,40,48,56,64,72,80,96,112,128
LC 24000Hz Quadraphonic -- 64,72,80,96,112,128,144,160,192,224,256
LC 24000Hz 4.0 (C L R Cs) -- 64,72,80,96,112,128,144,160,192,224,256
LC 24000Hz 5.0 (C L R Ls Rs) -- 80,96,112,128,144,160,192,224,256,288,320
LC 24000Hz 5.1 (C L R Ls Rs LFE) -- 80,96,112,128,144,160,192,224,256,288,320
LC 24000Hz 6.0 (C L R Ls Rs Cs) -- 96,112,128,144,160,192,224,256,288,320,384
LC 24000Hz 6.1 (C L R Ls Rs Cs LFE) -- 96,112,128,144,160,192,224,256,288,320,384
LC 24000Hz 7.0 (C L R Ls Rs Rls Rrs) -- 112,128,144,160,192,224,256,288,320,384,448
LC 24000Hz 7.1 (C Lc Rc L R Ls Rs LFE) -- 112,128,144,160,192,224,256,288,320,384,448
LC 24000Hz 8.0 (C L R Ls Rs Rls Rrs Cs) -- 128,144,160,192,224,256,288,320,384,448,512
LC 32000Hz Mono -- 24,28,32,40,48,56,64,72,80,96
LC 32000Hz Stereo (L R) -- 48,56,64,72,80,96,112,128,144,160,192
LC 32000Hz Quadraphonic -- 96,112,128,144,160,192,224,256,288,320,384
LC 32000Hz 4.0 (C L R Cs) -- 96,112,128,144,160,192,224,256,288,320,384
LC 32000Hz 5.0 (C L R Ls Rs) -- 128,144,160,192,224,256,288,320,384,448
LC 32000Hz 5.1 (C L R Ls Rs LFE) -- 128,144,160,192,224,256,288,320,384,448
LC 32000Hz 6.0 (C L R Ls Rs Cs) -- 144,160,192,224,256,288,320,384,448,512,576
LC 32000Hz 6.1 (C L R Ls Rs Cs LFE) -- 144,160,192,224,256,288,320,384,448,512,576
LC 32000Hz 7.0 (C L R Ls Rs Rls Rrs) -- 192,224,256,288,320,384,448,512,576,640
LC 32000Hz 7.1 (C Lc Rc L R Ls Rs LFE) -- 192,224,256,288,320,384,448,512,576,640
LC 32000Hz 8.0 (C L R Ls Rs Rls Rrs Cs) -- 192,224,256,288,320,384,448,512,576,640,768
LC 44100Hz Mono -- 32,40,48,56,64,72,80,96,112,128,144,160,192,224,256
LC 44100Hz Stereo (L R) -- 64,72,80,96,112,128,144,160,192,224,256,288,320
LC 44100Hz Quadraphonic -- 128,144,160,192,224,256,288,320,384,448,512,576,640
LC 44100Hz 4.0 (C L R Cs) -- 128,144,160,192,224,256,288,320,384,448,512,576,640
LC 44100Hz 5.0 (C L R Ls Rs) -- 160,192,224,256,288,320,384,448,512,576,640,768
LC 44100Hz 5.1 (C L R Ls Rs LFE) -- 160,192,224,256,288,320,384,448,512,576,640,768
LC 44100Hz 6.0 (C L R Ls Rs Cs) -- 192,224,256,288,320,384,448,512,576,640,768,960
LC 44100Hz 6.1 (C L R Ls Rs Cs LFE) -- 192,224,256,288,320,384,448,512,576,640,768,960
LC 44100Hz 7.0 (C L R Ls Rs Rls Rrs) -- 224,256,288,320,384,448,512,576,640,768,960
LC 44100Hz 7.1 (C Lc Rc L R Ls Rs LFE) -- 224,256,288,320,384,448,512,576,640,768,960
LC 44100Hz 8.0 (C L R Ls Rs Rls Rrs Cs) -- 256,288,320,384,448,512,576,640,768,960,1280
LC 48000Hz Mono -- 32,40,48,56,64,72,80,96,112,128,144,160,192,224,256
LC 48000Hz Stereo (L R) -- 64,72,80,96,112,128,144,160,192,224,256,288,320
LC 48000Hz Quadraphonic -- 128,144,160,192,224,256,288,320,384,448,512,576,640
LC 48000Hz 4.0 (C L R Cs) -- 128,144,160,192,224,256,288,320,384,448,512,576,640
LC 48000Hz 5.0 (C L R Ls Rs) -- 160,192,224,256,288,320,384,448,512,576,640,768
LC 48000Hz 5.1 (C L R Ls Rs LFE) -- 160,192,224,256,288,320,384,448,512,576,640,768
LC 48000Hz 6.0 (C L R Ls Rs Cs) -- 192,224,256,288,320,384,448,512,576,640,768,960
LC 48000Hz 6.1 (C L R Ls Rs Cs LFE) -- 192,224,256,288,320,384,448,512,576,640,768,960
LC 48000Hz 7.0 (C L R Ls Rs Rls Rrs) -- 224,256,288,320,384,448,512,576,640,768,960
LC 48000Hz 7.1 (C Lc Rc L R Ls Rs LFE) -- 224,256,288,320,384,448,512,576,640,768,960
LC 48000Hz 8.0 (C L R Ls Rs Rls Rrs Cs) -- 256,288,320,384,448,512,576,640,768,960,1280
HE 32000Hz Mono -- 12,16,24,32,40
HE 32000Hz Stereo (L R) -- 24,32,40,48,56,64,80
HE 32000Hz Quadraphonic -- 48,56,64,80,96,112,128,160
HE 32000Hz 5.1 (C L R Ls Rs LFE) -- 64,80,96,112,128,160,192
HE 32000Hz 7.1 (C Lc Rc L R Ls Rs LFE) -- 96,112,128,160,192,224,256
HE 44100Hz Mono -- 16,24,32,40
HE 44100Hz Stereo (L R) -- 32,40,48,56,64,80
HE 44100Hz Quadraphonic -- 64,80,96,112,128,160
HE 44100Hz 5.1 (C L R Ls Rs LFE) -- 80,96,112,128,160,192
HE 44100Hz 7.1 (C Lc Rc L R Ls Rs LFE) -- 112,128,160,192,224,256
HE 48000Hz Mono -- 16,24,32,40
HE 48000Hz Stereo (L R) -- 32,40,48,56,64,80
HE 48000Hz Quadraphonic -- 64,80,96,112,128,160
HE 48000Hz 5.1 (C L R Ls Rs LFE) -- 80,96,112,128,160,192
HE 48000Hz 7.1 (C Lc Rc L R Ls Rs LFE) -- 112,128,160,192,224,256
HE 88200Hz Mono -- 32,40,48,56,64,80,96,112,128,160
HE 88200Hz Stereo (L R) -- 64,80,96,112,128,160,192,224,256,320
HE 88200Hz Quadraphonic -- 128,160,192,224,256,320,448,640
HE 88200Hz 5.1 (C L R Ls Rs LFE) -- 160,192,224,256,320,448,640
HE 88200Hz 7.1 (C Lc Rc L R Ls Rs LFE) -- 224,256,320,448,640,1120
HE 96000Hz Mono -- 32,40,48,56,64,80,96,112,128,160
HE 96000Hz Stereo (L R) -- 64,80,96,112,128,160,192,224,256,320
HE 96000Hz Quadraphonic -- 128,160,192,224,256,320,448,640
HE 96000Hz 5.1 (C L R Ls Rs LFE) -- 160,192,224,256,320,448,640
HE 96000Hz 7.1 (C Lc Rc L R Ls Rs LFE) -- 224,256,320,448,640,1120

Btw. ffmpeg as well as qaac doesnot recognize the input format properly. Which input format should I force to ffmpeg for temp.wav?
s16le is wrong.

kypec

4th April 2013, 11:26

Now I would need a tool that pipes the temporary wav for qaac so it makes a proper conversion.

This ffmpeg -i temp.wav -f f32le - | qaac --ignorelength -o English.m4a --tvbr 100 - generates an error. Advices please.

What's wrong with piping directly from eac3to to qaac?eac3to inputfile.dtshd stdout.wav | qaac --ignorelength -o English.m4a --tvbr 100 -

Anakunda

4th April 2013, 11:35

What's wrong with piping directly from eac3to to qaac?eac3to inputfile.dtshd stdout.wav | qaac --ignorelength -o English.m4a --tvbr 100 -

That's wrong the backend encoder doesnot seem to make proper conversion of the temp wavfile ie. it doesnot seem to like 24bit little endiand signed wav. When I used builtin decoder which produced the big 20GB or so temp file the conversion went fine.

kypec

4th April 2013, 13:34

That's wrong the backend encoder doesnot seem to make proper conversion of the temp wavfile ie. it doesnot seem to like 24bit little endiand signed wav. When I used builtin decoder which produced the big 20GB or so temp file the conversion went fine.
Did you try to use -down16 option in eac3to?

Anakunda

4th April 2013, 13:41

No, I give it a try, thank U for the hint.

Sparktank

4th April 2013, 18:58

I've never had any problems getting QAAC to use 24 bit depth, piping from eac3to.
I even had success encoding 64bit IEE fp with QAAC.

Maybe move the output switch the end?
"eac3to.exe" "source.dtshd" "stdout.wav" | "qaac.exe" -V127 -q 2 -i --no-delay -r keep --threading - -o "output.m4a"

Anakunda

4th April 2013, 19:33

This 5.1ch track I have successfully converted after downgrading to 16bit, on previous track which had 7.1 config I was unable to make the conversion anyway.

"eac3to.exe" "source.dtshd" "stdout.wav" | "qaac.exe" -V127 -q 2 -i --no-delay -r keep --threading - -o "output.m4a"
Is this syntax (with Arcsoft decoder) working for you on DTSHD 7.1 too?
And what is --no-delay good for.

Anyway thanks for the commandline, I'll try this as soon as I will have 8 channel track.

Sparktank

4th April 2013, 19:55

Yes, it works for 8ch DTSHD as well.
QAAC has some weird issue on how it handles delays something or other.

https://sites.google.com/site/qaacpage/news/qaacrelease212refalac112
[qaac] release 2.12 (refalac 1.12)
posted Jan 16, 2013, 5:07 AM by nu 774
Add --no-delay option. (Read the discussion at HA thread from here).

--no-delay will compensate encoder delay (2112 samples) by prepending silence of 960 samples before sending input to encoder, then trimming 3 AAC frames at beginning (2112 + 960 = 3072 = 1024 * 3, where 1024 is the frame length of AAC. So total amount of delay will be exactly equals to length of 3 AAC frames). Note that these numbers are doubled in case of SBR.

This option is meant for video as a mean to resolve A/V sync issue. The resultant AAC will have exactly zero-delay, but might have pops/clicks at the beginning. Use with care.

I don't get any of the pops/clicks however.

the_weirdo

5th April 2013, 06:02

@Anakunda

You can try this: Download this libsndfile (https://dl.dropbox.com/u/18695757/Software%20Binaries/libsndfile-1.dll) and put it in qaac directory.

Anakunda

5th April 2013, 06:16

Great I'll try this in case of further problems. After all when I checked the former track params the encoding problems were not on qaac neither eac3to but nonstd layout of audio track.

--(AUDIO)----

Format..........: DTS-HD MA
Channels........: 7.1 (strange setup)
Bit depth.......: 24 bits
Sample rate.....: 48 kHz
Bit rate........: 4702 kbps
Language........: English

robertcollier4

5th April 2013, 11:47

This ffmpeg -i temp.wav -f f32le - | qaac --ignorelength -o English.m4a --tvbr 100 - generates an error. Advices please.

Try this:
ffmpeg.exe -report -loglevel verbose -i "moviename.mkv" -map 0:a:0 -f wav -acodec pcm_f32le - | qaac.exe --tvbr 127 --quality 2 --rate keep --ignorelength --no-delay - -o "moviename-audio.m4a"

Regarding the --no-delay switch for qaac. Most compressors such as AAC and MP3 prepend 2112 samples to the beginning. Read here (http://developer.apple.com/library/mac/#documentation/QuickTime/qtff/QTFFAppenG/QTFFAppenG.html) the section titled "Historical Solution—Implicit Encoder Delay". Normally all AAC encoders (including neroaacenc and Quicktime) will add 2112 samples (44 ms at 48Khz samples/sec) to the beginning of the audio file and store this information in a tag. If you remux with mkvmerge or play with a supported player - then it should detect that the audio file has an encoder delay in the beginning of it and skip it - however, to increase compatability with all muxers and players qaac can chop off the encoder delay at the beginning to make it so that the beginning of your audio file is the same as the input which then makes it irrelevant of whether your muxer/player will be able to detect and remove the encoder delay or not.

If trying to pipe directly from eac3to to qaac and qaac is not accepting the file, try the -full switch for eac3to which will instruct it to send qaac the full internally used conversion bit-depth rate without applying any down dithering at all.
"eac3to.exe" "source.dtshd" "stdout.wav" -full | "qaac.exe" --tvbr 127 --quality 2 --rate keep --ignorelength --no-delay - -o "output.m4a"

tebasuna51

5th April 2013, 13:33

Try this:
ffmpeg.exe -report -loglevel verbose -i "moviename.mkv" -map 0:a:0 -f wav -acodec pcm_f32le - | qaac.exe --tvbr 127 --quality 2 --rate keep --ignorelength --no-delay - -o "moviename-audio.m4a"

But ffmpeg can't decode DTS-HD, only the standard dts 'core'.

Anakunda

5th April 2013, 13:38

Yes sure the use of ffmpeg was only workaround for converting 16bit integer wav to float on the fly but since 24bit input shows no problem for qaac then ffmpeg is superfluous.

sneaker_ger

5th April 2013, 22:23

Try this:

Regarding the --no-delay switch for qaac. Most compressors such as AAC and MP3 prepend 2112 samples to the beginning. Read here (http://developer.apple.com/library/mac/#documentation/QuickTime/qtff/QTFFAppenG/QTFFAppenG.html) the section titled "Historical Solution—Implicit Encoder Delay". Normally all AAC encoders (including neroaacenc and Quicktime) will add 2112 samples (44 ms at 48Khz samples/sec)

Neroaacenc does not use 2112 samples. The delay varies from encoder to encoder.

LeXXuz

6th April 2013, 12:53

Is eac3to having problems with partitions >2TB?

I found anything but one post in this long thread where someone reports almost the same issue I seem to have.

I am unable to demux streams from a BD structure if those files are on a 3TB NTFS partition.

Usually I use eac3to in megui but I also tried to demux directly via command line. It's no use. Browsing BD structure, listing streams and playlists, all works fine.

However, when I start a demuxing job eac3to stops right at the beginning after writing just some bytes to the destination path and that's it. No more disc activity, no error message, nothing.

I tested this with several external 3TB drives by now. It does not work, even if the destination path is my 'small' internal hard drive. :(

Okay it it's not eac3to it seems to be the Matroska muxer causing this problem. eac3to uses Haali's media splitter AFAIR. Latest version I can find is dated 03/03/2011. And this one is not working with partitions that big, at least not for me. :(

Anyone know about a more recent version or can anyone at least confirm this issue?

filler56789

6th April 2013, 13:35

Anyone know about a more recent version

http://forum.doom9.org/showthread.php?p=1616085#post1616085

or can anyone at least confirm this issue?

Sorry, but I stopped using Haali filters many moons ago.

LeXXuz

8th April 2013, 11:16

http://forum.doom9.org/showthread.php?p=1616085#post1616085

Sorry, but I stopped using Haali filters many moons ago.

Does eac3to work with other filters to mux to mkv? If so, which one?
I always read it requires Haali's filter to work with mkv...

LigH

8th April 2013, 13:07

My recommendation would be to use eac3to as (elementary stream) audio converter only — and leave the MKV multiplexing to the specialist in this branch: mkvmerge.

LeXXuz

9th April 2013, 07:07

Like most people I use eac3to to demux BD/HD DVD structures. And like most people I need to have the video elementary streams in a container for further processing, as for AVC/VC1 preferably in mkv. Of course I can use any other multiplexer in a 2nd step after demuxing the source disc. But this is unnecessary stress for my disk drive and a waste of time if it 'could' be done right in the 1st step. :)

madshi

9th April 2013, 07:36

There's also the question about timestamps. mkvtoolnix used to be worse at writing the correct timestamps into the MKV for some video files, compared to eac3to. I suppose mkvtoolnix got better with that in the past months/years, but I'm not sure how good it really is now. Creating the proper timestamps is not always easy, and at least in the past eac3to was better at that. Just yesterday I received an MKV sample from a madVR user, muxed with mkvtoolnix. I noticed some playback problems. After remuxing it with eac3to it played better for me. And just for the record, the specialist for muxing MPEG2, VC-1 and h264 video tracks into MKV is eac3to, not mkvmerge. eac3to had proper VC-1 muxing support long before mkvmerge. I was the one pioneering VC-1 into MKV muxing (Haali tried it first, but didn't get it to work properly). And when I implemented h264 muxing support, mkvmerge was buggy as hell with h264 muxing. JFWIW...

gp2221

9th April 2013, 14:42

Doesn't 'anothereac3to gui' use mkvmerge to reassemble the mkv? Why doesn't it just use eac3to?

I've been using 'anothereac3to GUI' to rip my blu-rays because sometimes I want a separate copy of the audio track (in flac). I use makemkv for other discs. Are there better options available now?

madshi

9th April 2013, 14:46

eac3to can only mux the video track to MKV. In order to add the audio tracks, too, you have to use mkvmerge. However, if you feed mkvmerge with the video MKV created by eac3to, mkvmerge will reuse the timestamps etc muxed by eac3to, AFAIK.

cyberbeing

9th April 2013, 15:23

Anyone know about a more recent version?

The latest beta release of CCCP codec pack (http://www.cccp-project.net/forums/index.php?topic=6604.msg43371#msg43371) contain a pre-release Haali Splitter 1.13.97.23 from March 4th, 2013, which also includes a new build of gdsmux.exe

innounp -x (http://innounp.sourceforge.net/) can be used to extract out the Haali folder if you don't want to use the CCCP installer.

Edit: JEEB posted the Haali 1.13.97.23 installer http://forum.doom9.org/showpost.php?p=1623522&postcount=114

lazyn00b

10th April 2013, 07:12

Okay it it's not eac3to it seems to be the Matroska muxer causing this problem. eac3to uses Haali's media splitter AFAIR. Latest version I can find is dated 03/03/2011. And this one is not working with partitions that big, at least not for me. :(

Anyone know about a more recent version or can anyone at least confirm this issue?

I'm having the same issue when trying to use eac3to (via Staxrip) on source files located on a 4 TB Windows 8 Storage Space. I found another guy on the Videohelp forums who has the same issue, although he's using MKV2AC3 as the front end to eac3to. I tried replacing eac3to 3.24 with 3.27, but it didn't make a difference. Moving the source files from the Storage Space to a single (smaller) drive solved the issue.

BTW, the failure happens pretty much immediately, and eac3to does not seem to produce a log file.

superhil

10th April 2013, 09:55

i don't understand how "-edit=0:00:00,0ms" works
could you help me how is the command if i want to take first ten seconds as a sample of the audio?

tebasuna51

10th April 2013, 10:49

-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

Then:
-edit=H:MM:SS,Xms -silence
Add X ms of silence at H:MM:SS

-edit=H:MM:SS,Xms -loop
Repeat last X ms at H:MM:SS

-edit=H:MM:SS,-Xms
Delete X ms at H:MM:SS

You can't extract the first ten seconds.
There are a parameter:

-Xmb

Than stop extract audio after reach X MB of the source, but you can't know easily how many seconds you extract.
You can use DelayCut to do the job for AC3/DTS/MP3/WAV audio.

superhil

10th April 2013, 14:21

Dear tebasuna51
Thanks for the explanation.
I think i missunderstood about the function of "-edit" option

LeXXuz

16th April 2013, 19:07

The latest beta release of CCCP codec pack (http://www.cccp-project.net/forums/index.php?topic=6604.msg43371#msg43371) contain a pre-release Haali Splitter 1.13.97.23 from March 4th, 2013, which also includes a new build of gdsmux.exe

innounp -x (http://innounp.sourceforge.net/) can be used to extract out the Haali folder if you don't want to use the CCCP installer.

Edit: JEEB posted the Haali 1.13.97.23 installer http://forum.doom9.org/showpost.php?p=1623522&postcount=114

Thank you for the link. Unfortunately the newer version does not change anything. :(

I'm having the same issue when trying to use eac3to (via Staxrip) on source files located on a 4 TB Windows 8 Storage Space. I found another guy on the Videohelp forums who has the same issue, although he's using MKV2AC3 as the front end to eac3to. I tried replacing eac3to 3.24 with 3.27, but it didn't make a difference. Moving the source files from the Storage Space to a single (smaller) drive solved the issue.

BTW, the failure happens pretty much immediately, and eac3to does not seem to produce a log file.

It seems it is NOT the partition size in general. I removed lots of stuff from my 3TB drive, reduced partition size to 1.5TB and created a 2nd partition of approx. 650GB (so both partitions stay below 2.1TB in total).

However, eac3to still refuses to write to this partition when the video output container is set to mkv. I tried this manually and with MeGUI as front-end.

Just to get this straight: eac3to does not write/create mkv files on it's own. It has to rely on a DS filter, is that correct?

Is it possible to make eac3to work with a different filter to get rid of the outdated Haali filter once and for all?

Snowknight26

16th April 2013, 19:15

Not until madshi makes quite a few changes to eac3to. It's been suggested by me and a few others to ditch gdsmux in favor of ffmpeg but nothing is set in stone. If he does decide to do it (being as busy as he is) it'll be quite a while before we see the results.

madshi

16th April 2013, 19:23

eac3to writes MKV files through the Haali MKV Muxer DirectShow filter. However, all other files are created and written to directly by eac3to. Replacing the Haali MKV Muxer would be quite a lot of work, so don't expect that anytime soon.

Sparktank

17th April 2013, 00:23

mindbomb

23rd April 2013, 15:31

For demuxing subtitles with zlib compression in an mkv, eac3to demuxes the file sucessfully, but it doesnt remove the compression.

cafevn

24th April 2013, 02:13

hi, if i only need extract DTS-HDMA5.1 audio i don't need arc soft decoder?

mindbomb

24th April 2013, 03:07

hi, if i only need extract DTS-HDMA5.1 audio i don't need arc soft decoder?

correct.

cafevn

24th April 2013, 15:33

correct.

thank you, i only need remux blu-ray, not encode anythings

kypec

25th April 2013, 09:14

thank you, i only need remux blu-ray, not encode anythings
MakeMKV (http://www.makemkv.com/) is probably more suitable for your purpose than eac3to if all you need is demux/remux Bluray streams. ;)

cafevn

25th April 2013, 13:45

MakeMKV (http://www.makemkv.com/) is probably more suitable for your purpose than eac3to if all you need is demux/remux Bluray streams. ;)

i have 2-disc blu-ray want to remux to 1 mkv file (titanic 3d)

i thing i need demux 2 disc and use mkvmerge to append to 1 mkv file

that right :D

dansrfe

25th April 2013, 19:26

Is there support planned for MP4 files?

DigitalfreakNYC

28th April 2013, 17:44

Every time I try to convert an uncompressed wav to thd, I get "audio not supported." Any suggestions?

filler56789

28th April 2013, 20:41

Every time I try to convert an uncompressed wav to thd, I get "audio not supported." Any suggestions?

eac3to cannot encode to TrueHD. From the help screen:

Decoded audio data can be stored as / encoded to:
(1) RAW, (L)PCM
(2) WAV (PCM only), W64, RF64, AGM
(3) WAVs (multiple mono WAV files, PCM only)
(4) AC3
(5) DTS
(6) AAC
(7) FLAC

dansrfe

30th April 2013, 02:25

Will there be support for stdin any time soon? I need to transcode audio that resides within an MP4 container so I need to pipe MP4Box's audio extract output to eac3to's input then pipe that back to MP4Box to mux the audio and encoded video to MP4.

Snowknight26

30th April 2013, 02:27

Use ffmpeg instead?

thecross

30th April 2013, 23:54

I really like this tool. Only thing I wish it could handle was multithreading. No matter what I do it seems to saturate only one "core", so on a "quad core" CPU eac3to will max out at 25%. On an "octal core" CPU it will max out at 12.5%. Perhaps there is a command line argument to set threading options or the number of threads to use, or a special super secret "build from source" multithreaded version I am unaware of, but I can't seem to find a way to make eac3to use my entire CPU.

Q-the-STORM

1st May 2013, 05:10

any way to speed up audio to a specific framerate?
this works fine:

eac3to source.wav output.wav -25.000 -changeTo24.000

but I would need something like this
eac3to source.wav output.wav -25.000 -changeTo25.650

seems like eac3to only supports the common framerates, but wouldn't it be possible/easy to open it up to more specific values?

Shevek

1st May 2013, 12:22

tebasuna51

2nd May 2013, 01:05

Or sox:

sox source.wav output.wav speed 1.026

Where 1.026 = 25.650/25

dansrfe

2nd May 2013, 01:09

Is there any particular reason why eac3to can't read from and mux to an MP4 container? Is it deliberate by design?

Kurtnoise

2nd May 2013, 09:35

why not remux your MP4 into MKV instead ?

MKV support in eac3to is not perfect, yes, but it's much better than nothing.

tebasuna51

2nd May 2013, 09:56

eac3to was born to manage E-AC3 audio (for that the name) in EVO container.
And many other audio conversions.
After was added support for m2ts, ts, VOB, ...
Later a partial support for mkv files.

MP4 support was never added.
Maybe because there are other tools to do the job, maybe because patent conflicts, standard full specs, different formats, ...

MerolaC

2nd May 2013, 13:59

Hello.
madshi, Thank you a lot EAC3To. It's awesome!

I'm having a doubt. Which is the last version of Nero 7 that works for EAC3To. I have the version 7.10.0 (Activated and all to Premium Edition) and when I do the test in EAC3To, it still says that I don't have Nero 7 installed.
I'm on Windows 7 Ultimate x64
Is that the problem?

In any case, thank you a ton!

sephirotic

2nd May 2013, 16:37

Please forgive me if this is a common question, i've spent a lot of time searching for answers for my problem with no avail.

So i succesfully decoded a DTS-ES file (6.1 channel) into a flac. I wasn´t satisfied with the file size so i decided to go lossy to save space. At first i considered going with AC3 but 640kbps wouldn´t fit DVD9 discs and i don´t want to reencode my video track again. That's when i started getting worried that AC3 448kbps quality wouldn´t satisfy me enough so i decided going on AAC multichannel. This was the first time i was encoding a 7th channel audio AAC instead of a simple 5.1, eac3to gave me the warning when encoding the audio:
"neroaacenc doesn´t support 6.1 encoding. Will double the 7th channel".
While i do know that AAC is much more efficient than AC3 i started worrying that this extra useless (doubled) channel would kill all the purpose of extra efficiency of the AAC. 8 channels vs 6 channels would represent roughly 30% of extra audio data.
Do people actually transcode 6.1 DTS-ES to AAC or should i stick with DD 448?

Compatibility is not a issue to me since the video track itself is a 10bit encode so i will use a software decoding pc in my living room for playing the file.

Another question is: eac3to does not support Adaptive bit rate of nero? -q settings are getting me very inconsistent sizes/bitrates.
(also, is ABR considerable much less efficient than VBR encoding? should i stay with VBR even if it gives 20kb/s less than a target abr setting?)

Thanks a lot.

tebasuna51

2nd May 2013, 21:17

Which is the last version of Nero 7 that works for EAC3To.

I have Nero 7.11.10 and work.
BTW, now Nero decoder is needed only for AAC.

MerolaC

3rd May 2013, 02:43

I have Nero 7.11.10 and work.
BTW, now Nero decoder is needed only for AAC.

Thanks for the answer. I will try reinstalling. But now that you say that. Is the Nero decoder obsolete? Or why is not needed anymore? The AC3 decoder I mean. I already have the one AAC working.

Edit:
Well, I re-read the OP and DRC is applied sometimes, I guess I know now.... Again, thank you tebasuna.

tebasuna51

3rd May 2013, 12:30

...This was the first time i was encoding a 7th channel audio AAC instead of a simple 5.1, eac3to gave me the warning when encoding the audio:
"neroaacenc doesn´t support 6.1 encoding. Will double the 7th channel".

Preserve a 6.1 audio is only usefull when you have a 6.1 audio equipment (not very common).
If you have 5.1 speakers is better use -down6, the BC channel is mixed in surround channel and you still have the BC like phantom channel.

BTW, NeroAacEnc can encode 6.1. Try this:

eac3to input6.1.dts stdout.wav | NeroAacEnc -q 0.5 -ignorelength -if - -of output.m4a

The problem here is the order in back/rear channels. Maybe your AAC decoder can't restore the correct mapping.

Another question is: eac3to does not support Adaptive bit rate of nero? -q settings are getting me very inconsistent sizes/bitrates.

The NeroAacEnc options are:

Quality/bitrate control:
-q <number> : Enables "target quality" mode.
<number> is a floating-point number in 0...1 range.
-br <number> : Specifies "target bitrate" mode.
<number> is target bitrate in bits per second.
-cbr <number> : Specifies "target bitrate (streaming)" mode.
<number> is target bitrate in bits per second.
When neither of above quality/bitrate options is used,
the encoder defaults to equivalent of -q 0.5

Multipass encoding:
-2pass : Enables two-pass encoding mode.
Note that two-pass more requires a physical file as input,
rather than stdin.

To have a efficient -br encode you need -2pass and need a physical file. eac3to can't do this, decode your input to wav and use NeroAacEnc with the -2pass option.

buffyangel108

3rd May 2013, 14:56

Does anyone know how I can pipe eac3to output to flac.exe?

I know eac3to has in-built flac encoding (using libFlac), but compression is seemingly set at level 8 which takes a lot longer than level 0 (and I don't need the extra compression).

I've tried:

eac3to file.dts stdout.wav | flac -0 - -o file.flac

...which works on short files but not on movie-length ones, throwing up a foreign metadata error towards the end of encoding. I'm guessing this could be where the stdout.wav temp file reaches the 4GB WAV header limit?

I've also tried doing it in two steps: dts>wav (with eac3to) then wav>flac (with flac), but again wav files larger than 4GB fail to encode in flac. :-(

Does anyone know a workaround? If not, could eac3to.exe or libFlac.dll conceivably be patched to output at level 0 flac compression instead of level 8?

nhakobian

3rd May 2013, 19:46

...which works on short files but not on movie-length ones, throwing up a foreign metadata error towards the end of encoding. I'm guessing this could be where the stdout.wav temp file reaches the 4GB WAV header limit?

This is both true and false at the same time.

When you pipe something, there is no file created, so there is no filesystem bound filesize limit.

However, the WAV file format internally has a limit of 4GB filesizes due to how data is stored in its internal headers, which are passed between the two programs. Maybe the crash is occurring when this value overflows.

the_weirdo

4th May 2013, 06:14

@buffyangel108

You can try adding --ignore-chunk-sizes to FLAC command line and see if that can help. Also, you may try this version (http://lists.xiph.org/pipermail/flac-dev/2013-April/004057.html) of FLAC.

superhil

10th May 2013, 08:32

i found a bug, "-edit=0:00:0X,10000ms -silence" produce "invalid edit format" for X=0,1,..,10
but, start from X=11, e.g. "-edit=0:00:11,10000ms -silence" works well.

tebasuna51

10th May 2013, 11:36

i found a bug, "-edit=0:00:0X,10000ms -silence" produce "invalid edit format" for X=0,1,..,10
but, start from X=11, e.g. "-edit=0:00:11,10000ms -silence" works well.

It's a know bug. The amount to insert must be lower than the current position.
If you want insert 20 sec. the position must be at last 0:00:21

Maybe eac3to check first if can do a -loop option and show the <ERROR> because don't have enough audio to repeat.
But for -silence option should admit it.

madshi

10th May 2013, 12:58

If it's a known bug why is it not in the bug tracker? ;)

http://eac3to.bugs.madshi.net

superhil

10th May 2013, 15:24

If it's a known bug why is it not in the bug tracker? ;)

http://eac3to.bugs.madshi.net

then you should add it into bug tracker :D

because "-edit=0:00:00,10000ms -silence" is valid input

Chumbo

10th May 2013, 15:34

then you should add it into bug tracker :D

because "-edit=0:00:00,10000ms -silence" is valid input
You find it, you log it is how it works.

superhil

10th May 2013, 19:48

You find it, you log it is how it works.
I'm sorry I don't understand what you mean :confused:

That option should be a valid input, right? adding 10s of silence start from 00:00:00 until 00:00:10
But eac3to recognizes it as invalid input

I understand that "-edit=00:00:00,10000ms -loop" is invalid input, but "-silence" should work

tebasuna51

10th May 2013, 21:40

That option should be a valid input, right? adding 10s of silence start from 00:00:00 until 00:00:10

For that is enough. +10000ms

superhil

11th May 2013, 06:29

For that is enough. +10000ms
i understand it, for that purpose +10000ms will work well.

i just want to point out that "-edit=0:00:0X,10000ms -silence" for X=0,1,..,10 should be a valid input. ;)

madshi

11th May 2013, 07:20

If you want the bug fixed you need to add it to the bug tracker. It's as simple as that.

Groucho2004

11th May 2013, 09:06

superhil

11th May 2013, 16:31

If you want the bug fixed you need to add it to the bug tracker. It's as simple as that.

Maybe my explanation is not good, but as the developer i'm sure that you understand it very well :D

http://bugs.madshi.net/view.php?id=61

superhil

11th May 2013, 16:33

@madshi
Is there a chance you could expand the "normalize" functionality so one can specify the peak level (instead of always normalizing to 0dB)? I sometimes like to do some post processing and having some headroom would be nice.
Maybe something like "-normalize -3dB".

Is it similar to add gain after normalize?
Normalize to 0dB means that you add some positive or negative gain so that the peak become at 0dB.
If you want the peak at +2dB you can simply normalize and add +2dB gain

tebasuna51

11th May 2013, 17:45

If you want the peak at +2dB...

You can't have peaks at +2dB with the standar output 24 bit int.
The max value is 0 dB.

superhil

11th May 2013, 18:20

You can't have peaks at +2dB with the standar output 24 bit int.
The max value is 0 dB.
I'm sorry, I don't know about that :D
But, couple days ago i converted .ac3 audio to wav with eac3to. In logfile i saw that it applies -XdB gain (i don't remember exact value). Does it mean original audio has a positive peak value (+X dB)?

Groucho2004

11th May 2013, 18:22

Is it similar to add gain after normalize?More or less. I just want to skip the additional step of adding gain.

If you want the peak at +2dB
As already pointed out by tebasuna51 - :confused:

Groucho2004

11th May 2013, 18:24

DarkSpace

11th May 2013, 18:54

I'm sorry, I don't know about that :D
But, couple days ago i converted .ac3 audio to wav with eac3to. In logfile i saw that it applies -XdB gain (i don't remember exact value). Does it mean original audio has a positive peak value (+X dB)?

The maximum level for PCM data is 0 dB, no matter what bit depth.

I think that with floating point samples, it is possible to have peaks above 0 dB. Also, since eac3to internally decodes ac3 as 64-bit floating point, this is quite likely what happened, and eac3to lowered the volume to prevent clipping.

Groucho2004

11th May 2013, 20:55

I think that with floating point samples, it is possible to have peaks above 0 dB.
No, it's not.

Edit:
Some light reading about the subject. (http://wiki.multimedia.cx/index.php?title=PCM)

sephirotic

11th May 2013, 21:04

@tebasuna51

BTW, NeroAacEnc can encode 6.1. Try this:

eac3to input6.1.dts stdout.wav | NeroAacEnc -q 0.5 -ignorelength -if - -of output.m4a

The problem here is the order in back/rear channels. Maybe your AAC decoder can't restore the correct mapping.

The NeroAacEnc options are:

Quality/bitrate control:
-q <number> : Enables "target quality" mode.
<number> is a floating-point number in 0...1 range.
-br <number> : Specifies "target bitrate" mode.
<number> is target bitrate in bits per second.
-cbr <number> : Specifies "target bitrate (streaming)" mode.
<number> is target bitrate in bits per second.
When neither of above quality/bitrate options is used,
the encoder defaults to equivalent of -q 0.5

Multipass encoding:
-2pass : Enables two-pass encoding mode.
Note that two-pass more requires a physical file as input,
rather than stdin.

To have a efficient -br encode you need -2pass and need a physical file. eac3to can't do this, decode your input to wav and use NeroAacEnc with the -2pass option.

Thank you very much for your answers. The eac3 unnoficial-faq neither the first page of this thread had a full list of options for neroaac parameters so this post was very useful. May i suggest adding a -h into the eac3to with all available commands in a future update? (is there another argument for showing help built in with the eac3to.exe x86 build? sorry if this was a dumb question)

Anw, thanks again.

Snowknight26

11th May 2013, 21:08

Why should another program's parameters be listed by eac3to?

LigH

12th May 2013, 05:54

IEEE floating point samples (single or double precision, MS codec 2CC = 0003h) would be able to store denormalized sample values beyond ±1.0; still, it should be avoided, the output from sound cards will usually be based on integer sample values, the D/A converter won't produce a higher voltage. But it may be useful for storing raw footage to be finalized.

There are several discussions if a floating point PCM audio with samples beyond 0 dBFS is "valid"; some say it was a purpose to use this format for cases where exceeding results are probable. I certainly share this opinion (http://forum.cockos.com/showpost.php?p=252083&postcount=8):

There's nothing "mad" about having 32-bit floating point files that go over the -1.0 - 1.0 range in the intermediate stages of audio production...But I do agree anything to be listened as the final product should be as standard as possible.

madshi

12th May 2013, 08:21

You can use the "-full" eac3to parameter to force eac3to to create a 64bit floating point file which can then store peaks above -1.0 - 1.0. When you're done with (external) processing, another eac3to run without "-full" will finally produce a 24bit file with no clipping. However, not many tools understand 64bit floating point wav files. Maybe the "-float32" switch (instead of "-full") has better compatability with a wider range of external tools, but it results in a small loss of precision, naturally.

tebasuna51

12th May 2013, 08:59

I think that with floating point samples, it is possible to have peaks above 0 dB. Also, since eac3to internally decodes ac3 as 64-bit floating point, this is quite likely what happened, and eac3to lowered the volume to prevent clipping.
Yes, is correct.
But take in mind than any output above 0 dB is always a error in the lossy process of encode/decode, because the original source never can have peaks above 0 dB.

This problem never occurs decoding lossless formats.

The eac3 unnoficial-faq neither the first page of this thread had a full list of options for neroaac parameters so this post was very useful. May i suggest adding a -h into the eac3to with all available commands in a future update? (is there another argument for showing help built in with the eac3to.exe x86 build? sorry if this was a dumb question)

eac3to don't support all parameters of internal encoders (Aften, NeroAacEnc and Flac) then can't be showed like eac3to parameters.

But you can use the 'pipe' way to use the full parameters of external encoders Aften.exe, NeroAacEnc.exe, Flac.exe and others:

eac3to input stdout.wav <eac3to parameters> | Ext_Encoder <Ext_encoder parameters>

You can know all parameters of external encoders with:
Aften -longhelp
NeroAacEnc -help
Flac --help
...

wiggaz

13th May 2013, 10:40

Hi,
I'd like to know how to cut an audio at a specific time and keep the first part of the cut.
I searched a lot, but I didn't found anything.
Thanks in advance.

rhaz

13th May 2013, 11:45

How to use eac3to to mux 6 mono wavs to DTS? I have a surcode encoder for DTS, I tried More GUI, but I totally don't understand anything there, there is no selection for dts encoder neither. I tried BeSweet GUI, it its all bugs in Windows 8, it gives blank screen in the GUI, I can't make any settings or options selection.

tebasuna51

13th May 2013, 11:47

I'd like to know how to cut an audio at a specific time and keep the first part of the cut.
AFAIK eac3to can't be used for that.
There are specific tools to do the job for each audio codec.
Like here is off topic open other thread with a MediaInfo of your audio file.

LigH

13th May 2013, 13:23

@ tebasuna51:

Did you reply to wiggaz, to rhaz, or to both?

@ wiggaz:

It seems that eac3to does not yet offer support for processing only # seconds while transcoding. A combination of a negative delay and a maximum transcoding duration could allow something similar to Trim() in AviSynth, but it may have to be implemented yet...

@ rhaz:

As far as I know, open the six channel WAVs in your SurCode DVD-DTS encoder. It has an own GUI. See the screenshot on the Minnetonka website (http://www.minnetonkaaudio.com/index.php?option=com_content&view=article&id=54&Itemid=87). You don't feed them via eac3to when they are already six mono WAVs, only when a surround audio source is transcoded.

RamGuy

13th May 2013, 13:54

I'm trying to use Eac3To to convert DTS-HD MA into FLAC libav isn't able to do more than the regular DTS embedded in the DTS-HD MA track so I started fiddling with the latest trial version of ArcSoft TotalMedia Theatre as that seems to be the way to get this going.

I've moved all the dll's and ax files to my Eac3To folder, manually registered all the necessary files using elevated CMD prompt using regsvr32 but it's still not working for me.

I only get the message "The ArcSoft DTS decoder doesn't seem to work, will use libav insetad." and we are back to basic...

LigH

13th May 2013, 14:43

eac3to used to support the DLLs provided by TMT3, but the most current version (TMT5) is not supported anymore (it doesn't use a "checkactivation.dll" which was required in the older supported version). You may be able to find a package of all supported DLLs, but sharing them separately (like in RipBot264) may not comply with the ArcSoft EULA...

rack04

22nd May 2013, 01:10

I'm trying to back-up a copy of a 3D Blu-ray and I notice that eac3to reports left eye and right eye. If I want to back-up to 2D does it matter which one I select?

Shevek

22nd May 2013, 01:18

I'm trying to back-up a copy of a 3D Blu-ray and I notice that eac3to reports left eye and right eye. If I want to back-up to 2D does it matter which one I select?

very much so - I can never remember which is which, but one eye is the normal 2D movie and the other eye is only the "extra" info needed to create the 3D image. There won't be an audio track in the "extra" eye track.

rack04

22nd May 2013, 15:34

very much so - I can never remember which is which, but one eye is the normal 2D movie and the other eye is only the "extra" info needed to create the 3D image. There won't be an audio track in the "extra" eye track.

Thanks. I'm going to assume the left eye is the normal 2D movie because the right eye choked DGDecNV and the resultant file size was much less than the left eye stream.

r0lZ

24th May 2013, 13:11

I have an example of a commercial 3D BD with the left and right views inverted. There are at least two examples currently available: Ice Age 3 and Sammy II. In both cases, eac3to prints this:

eac3to.exe "Z:\BDMV\PLAYLIST\00852.mpls" 1)

M2TS, 2 video tracks, 11 audio tracks, 12 subtitle tracks, 1:34:02, 24p /1.001
1: Chapters, 20 chapters
2: h264/AVC (left eye), 1080p24 /1.001 (16:9)
3: h264/AVC (right eye), 1080p24 /1.001 (16:9)
[...]

It should be this:

2: h264/AVC (right eye), 1080p24 /1.001 (16:9)
3: h264/AVC (left eye), 1080p24 /1.001 (16:9)

Also, track 3 is not an AVC stream, but a MVC stream. So, ideally, the output should be:

2: h264/AVC (right eye), 1080p24 /1.001 (16:9)
3: h264/MVC (left eye), 1080p24 /1.001 (16:9)

However, that's not really important, as long as the MVC stream is always after the AVC stream. I suppose it's the case, but I'm not absolutely sure. If in this case it is before the AVC stream, the correct output should be:

2: h264/MVC (left eye), 1080p24 /1.001 (16:9)
3: h264/AVC (right eye), 1080p24 /1.001 (16:9)

Currently, it is therefore not possible to know for sure what's the correct order of the views to convert the 3D movie to side-by-side or top-bottom.
Is it possible to fix that bug? Thanks in advance.

[EDIT] I did a demux of track 2, and it is the AVC stream. So, apparently, that confirms that the AVC track is always "before" the MVC track. But there is still no way to know what track contains the left or right view.

frencher

25th May 2013, 12:59

I have same problem of r0lZ with commercial 3D BD with the left and right views inverted "Le voyage extraordinaire de Samy"

http://cdn-ibb.ladmedia.fr/var/ibb/storage/images/sorties/insolite/le-voyage-extraordinaire-de-samy-a-la-geode/8432676-1-fre-FR/Le-Voyage-extraordinaire-de-Samy-a-la-Geode_fiche_sortie_portrait_184.jpg

eac3to prints this:

eac3to.exe "H:\BDMV\PLAYLIST\00011.mpls"

M2TS, 2 video tracks, 1 audio track, 1:25:35, 24p /1.001
1: Chapters, 12 chapters
2: h264/AVC (left eye), 1080p24 /1.001 (16:9)
3: h264/AVC (right eye), 1080p24 /1.001 (16:9)
[...]

It should be this:
2: h264/AVC (right eye), 1080p24 /1.001 (16:9)
3: h264/MVC (left eye), 1080p24 /1.001 (16:9)

Same thanks ;)

yesgrey

25th May 2013, 18:44

Hi,

Is it possible to use madFlac to decode a flac file and then feed it into eac3to?

I am asking this because I am testing flaccl and would like to verify if the files created by it are bit perfect.

If I use eac3to to convert a flac file created by flaccl I get the following error:
"An error in the stream caused the FLAC decoder to lose synchronization.
Aborted at file position 3145728."

However, if I try playing the file using madFlac it plays fine, so, if we could use madFlac instead of eac3to's internal flac decoder I think I would be able to do the tests I want...

The only way I can think about is to use graphedit for converting the flac file with madFlac to 6 wav files (since it's a movie soundtrack) and then convert the 6 wav files to one single flac using eac3to. Would this be the only option?

madshi

26th May 2013, 14:33

If madFlac can play it, but eac3to not, this might be a bug in eac3to, or a bug in the audio file. I'd suggest that you create a bug entry here: http://eac3to.bugs.madshi.net and attach a short test file with which eac3to complains. Then if I find some time for eac3to again, I'll give it a try. As for feeding madFlac (or any other DirectShow decoder output) into eac3to: I'm not aware of a direct/simple way to do that.

yesgrey

26th May 2013, 17:25

Bug entry created. However, due to problems with it not accepting flac files it ended up creating two bug entries. Could you please delete the one that has no file attached?

I wanted to start using flaccl instead of flac inside eac3to, because it will save me a lot of time, but I want to be sure that it gives accurate results, hence why I wanted to use eac3to to convert a flaccl generated file to a flac eac3to flac, to compare with the original one created with eac3to, which I consider as the reference.

The tests I did using graphstudio and madflac to output pcm files were not very good to flaccl, because not only the files are different than the eac3to generated one, but they are also different between them if I compare two different compression levels. It seems flaccl is messing the data in someway... However, I think a comparison using eac3to would be more appropriate.

Now, it's just waiting for you to find the time to look at it... :)

r0lZ

26th May 2013, 18:14

Madshi, if you want the info on how to determine if the AVC stream of a 3D BD is the left or right view, Jdobbs has just poster the answer here (http://forum.doom9.org/showthread.php?p=1630291#post1630291).

AngelGraves13

29th May 2013, 20:39

Can eac3to do TrueHD to WAVS? I tried it on Ghostbusters and it threw me an error using the Stream Extractor tool. Should I just extract the THD file and try it via command line?

tebasuna51

29th May 2013, 23:51

Yes, it can.
Try with:

eac3to input.thd output.wavs

and put the log if there are problems.

Sparktank

1st June 2013, 10:00

madshi,

Will you be changing the FLAC library to the newer revisions made by Xiph.org?

FLAC has just updated to 1.3.0 (http://xiph.org/flac/changelog.html) very recently.

Also the source tarballs containing the newer Xiph updates can be downloaded from...
http://downloads.xiph.org/releases/flac/

They also moved their development to their own git repository (http://git.xiph.org/?p=flac.git;a=summary).

EDIT: I actually just replaced the libFLAC.dll from RareWares.com (v1.3.0) in the eac3to directory and it seems to have converted audio to FLAC without problems.

EDIT2: I think I understand the FLAC update.
It's mostly for the front-end CLI encoders (flac.EXE) and not the library itself (libflac.DLL) and since eac3to deals with the library directly, the update isn't needed.

I just read a post over at Illustrate's forum (dBpoweramp) about someone requesting a FLAC update and the Illustrate devs explained it pretty well.

Thunderbolt8

1st June 2013, 19:25

dtsdecoderdll.dll v1.1.0.9 has been released with ArcSoft TotalMedia Theatre 6.0.1.123.

anyone able to do an extensive test with all that dts (ma/hd) decoding with all those bitdepth and channel options to see whether there are any advantages compared to the former versions?

Chumbo

2nd June 2013, 20:34

Help clarify something for me please. When slowing down audio, is the displayed message supposed to show the actual parameters as requested on the command or is it just a generic message? I ask because you can see, below, that I'm slowing down 24 to 23.976 but the message shows Changing FPS from 25.000 to 23.976.
Running: eac3to "test.mkv" 2: "test.ac3" -24.000 -changeTo23.976 -384
MKV, 1 video track, 1 audio track, 0:42:57, 25p
1: MPEG2, English, 576p25 (16:9) with pulldown flags
2: AC3, English, 5.1 channels, 384kbps, 48kHz, dialnorm: -27dB "3/2+1"
v01 The video bitstream framerate field doesn't match the container framerate.
a02 Extracting audio track number 2...
a02 Removing AC3 dialog normalization...
a02 Decoding with libav/ffmpeg...
a02 Changing FPS from 25.000 to 23.976...
a02 Remapping channels...
a02 Encoding AC3 <384kbps> with libAften...
a02 Creating file "test.ac3"...

tebasuna51

2nd June 2013, 22:48

When the input file have a fps defined:

MKV, 1 video track, 1 audio track, 0:42:57, 25p

the -24.000 is ignored.
Extract the ac3 if you need a duration change -24.000 -changeTo23.976

Chumbo

2nd June 2013, 23:40

When the input file have a fps defined:

MKV, 1 video track, 1 audio track, 0:42:57, 25p

the -24.000 is ignored.
Extract the ac3 if you need a duration change -24.000 -changeTo23.976
Okay, that's a good idea. Yeah, the original video stream actually has its FPS as 24 but eac3to is picking up the container FPS. Thanks T.

AnryV

3rd June 2013, 09:34

Is there any news about E-AC3 7.1 decoding?

eac3to.exe M.eac3 M.wavs
E-AC3, 7.1 channels, 1:47:46, 768kbps, 48kHz
(core: AC3, 5.1 channels, 1:47:46, 448kbps, 48kHz)
Extracting E-AC3 core...
Decoding with libav/ffmpeg...
Reducing depth from 64 to 24 bits...
Writing WAVs...
Creating file "M.L.wav"...
Creating file "M.R.wav"...
Creating file "M.LFE.wav"...
Creating file "M.SR.wav"...
Creating file "M.SL.wav"...
Creating file "M.C.wav"...

r0lZ

10th June 2013, 12:41

In a 3D BD, the left and right eye streams are stored independently in two different M2TS files, and together in the corresponding SSIF file. These files are referenced by the MPLS file. eac3to can demux all of them, and that's nice. However, there is a big bug when demuxing the subtitles from the SSIF or MPLS file of many 3D-BDs.

The M2TS file containing the AVC video stream (usually the left eye view) contains also all audio and subtitle streams. Usually, the M2TS file containing the MVC video stream (right eye) contains ONLY the video stream. The audio streams are never included in the dependent M2TS. But in some BDs, the subtitle streams are ALSO included in that dependent M2TS file. It's with that BDs that the bug occurs.

When there are two identical subtitle streams in the two M2TS files, and the user demuxes the SSIF or MPLS file, the bug occurs. I think it's because the SAME stream is included TWICE in the SSIF file. So, when eac3to demuxes the subtitles, it processes two times the same subtitle stream, and that produces a lot of "The source file seems to be damaged (discontinuity)" error messages, and finally the demux operation fails with "The SUP reader received unknown data. Aborted at file position #."

It is possible to successfully demux the subtitle stream from the left or right view M2TS file, but it cannot be demuxed from the SSIF file (or indirectly from the MPLS). It's a pity, as of course, it is possible to demux only one of the two video streams at a time from the M2TS (and you cannot create the chapter file).

Is it possible to fix this bug?
Let me know if you need additional information.
Thanks!

frencher

10th June 2013, 17:20

madshi

10th June 2013, 17:31

Guys, if there's a bug in (or problem with) eac3to, why don't you use the eac3to bug tracker? That's your one and only chance to get bugs fixed in the next eac3to build (whenever that may come). If you post a bug in the bug tracker, a sample would always be nice, so that I can actually reproduce the problem on my PC. Thanks.

rapscallion

10th June 2013, 21:11

I have a BD with a Dolby True HD track (with a -32ms delay) and I want to add a DTS-Master Audio track.

When extracting the DTHD to wavs (Nero decoder selected) eac3to states "Applying RAW/PCM delay..."

Using DTS MA suite, will the raw/pcm delay still be intact when the DTS-HD track is created?

I did a 10 min test mux, via Tsmuxer, with no dealy selected, and the audio/dialog seems to be in sync,
but would like to confirm.

DarkSpace

10th June 2013, 22:49

When extracting the DTHD to wavs (Nero decoder selected) eac3to states "Applying RAW/PCM delay..."

Using DTS MA suite, will the raw/pcm delay still be intact when the DTS-HD track is created?
No. eac3to decodes the track and fixes the delay value (that's the line above), which means that after decoding, the track has a delay of 0 ms (no delay).
In case you want to keep the delay for whatever strange reasons, try adding +32ms to your commandline. It should then look something like this:
eac3to "00001.mpls" 2: "Path\00001.wavs" +32ms
This way, the -32ms delay of the track that eac3to wants to remove are compensated for by the +32ms delay added to the track by eac3to.

hubblec4

12th June 2013, 11:01

madshi would you like to update FLAC for eac3to

Brazil2

12th June 2013, 13:16

madshi would you like to update FLAC for eac3to
http://forum.doom9.org/showthread.php?p=1631141#post1631141
I actually just replaced the libFLAC.dll from RareWares.org (v1.3.0) in the eac3to directory and it seems to have converted audio to FLAC without problems.

hubblec4

12th June 2013, 14:13

thanks, i will try it.

mindbomb

13th June 2013, 03:21

didn't they change something with 6.1 and 7.1 flac audio with the new encoder?
Or is it consistent with how eac3to has been doing it all along?

madshi

13th June 2013, 07:32

http://bugs.madshi.net/view.php?id=88

Chumbo

16th June 2013, 03:21

I just bought the new 2nd edition of the Spears & Munsil audio/video calibration BD. I wanted to rip it so I can put some of the calibrations on my media server rather than having to always pop the BD into the player. However, as with other calibration discs, many of the clips are short, i.e., 2 to 6 minutes in length. So eac3to shows "HD DVD / Blu-Ray disc structure not found." which I believe is due to the large number of play lists and IIRC there's a minimum length?

Is there any way to override this minimum length because this title, even though has many short clips and play lists, is valid and would be nice to be able to process it with eac3to. If there isn't an override, do you want me to post this as a bug in bug tracker or as a feature request in the request thread? Thank you.

tebasuna51

16th June 2013, 09:28

@Chumbo
You can use each mpls like input for eac3to.

Boulder

16th June 2013, 10:50

I've been trying to encode the audio track of Edward Scissorhands to flac (using the internal libflac) and AAC (piping to qaac) but the channel order is incorrect. Apparently the track has 3 front channels and one rear channel (centre?) but I don't know how to configure the encoders. Can anyone help?

Chumbo

16th June 2013, 13:13

@Chumbo
You can use each mpls like input for eac3to.
I didn't even know you can do that. Thank you tebasuna as always. :)

SchnittenGott

16th June 2013, 13:18

Hello everyone!

I'm new but I already have a question/problem.
I'm trying to perform a rather simple task (at least I thought) with eac3to:
Convert a DTS-HD MA 6.1 ES track into mono wavs.
The BD at hand is "Revenge of the Sith":

eac3to v3.27
command line: eac3to P:\Revenge\ 1) 3: C:\BDE\Revenge.wavs
------------------------------------------------------------------------------
M2TS, 1 video track, 8 audio tracks, 29 subtitle tracks, 2:20:06, 24p /1.001
1: Chapters, 50 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz)
4: AC3, English, 5.1 channels, 448kbps, 48kHz
5: AC3, Spanish, 5.1 channels, 448kbps, 48kHz
6: AC3, French, 5.1 channels, 448kbps, 48kHz
7: DTS, French, 5.1 channels, 768kbps, 48kHz
8: AC3, Portuguese, 5.1 channels, 448kbps, 48kHz
9: AC3, English, 2.0 channels, 224kbps, 48kHz
10: AC3, English, 2.0 channels, 224kbps, 48kHz
11: Subtitle (PGS), English
12: Subtitle (PGS), Spanish
13: Subtitle (PGS), French
14: Subtitle (PGS), Dutch
15: Subtitle (PGS), Portuguese
16: Subtitle (PGS), Finnish
17: Subtitle (PGS), Swedish
18: Subtitle (PGS), Norwegian
19: Subtitle (PGS), Danish
20: Subtitle (PGS), English
21: Subtitle (PGS), English
22: Subtitle (PGS), Spanish
23: Subtitle (PGS), Spanish
24: Subtitle (PGS), French
25: Subtitle (PGS), French
26: Subtitle (PGS), Dutch
27: Subtitle (PGS), Dutch
28: Subtitle (PGS), Portuguese
29: Subtitle (PGS), Portuguese
30: Subtitle (PGS), Finnish
31: Subtitle (PGS), Finnish
32: Subtitle (PGS), Swedish
33: Subtitle (PGS), Swedish
34: Subtitle (PGS), Norwegian
35: Subtitle (PGS), Norwegian
36: Subtitle (PGS), Danish
37: Subtitle (PGS), Danish
38: Subtitle (PGS), English
39: Subtitle (PGS), English
[a03] Extracting audio track number 3...
[a03] Remapping channels...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Writing WAVs...
[a03] Creating file "C:\BDE\Revenge.L.wav"...
[a03] Creating file "C:\BDE\Revenge.R.wav"...
[a03] Creating file "C:\BDE\Revenge.SL.wav"...
[a03] Creating file "C:\BDE\Revenge.LFE.wav"...
[a03] Creating file "C:\BDE\Revenge.SR.wav"...
[a03] Creating file "C:\BDE\Revenge.BC.wav"...
[a03] Creating file "C:\BDE\Revenge.C.wav"...
[a03] The original audio track has a constant bit depth of 24 bits.
Video track 2 contains 201538 frames.
eac3to processing took 37 minutes, 6 seconds.
Done.

As the result I have the mono wavs, but they have incorrect length! (2h 40min.)
MediaInfo:

Complete name : C:\BDE\Revenge.R.wav
Format : Wave
File size : 1,29 GiB
Duration : 2h 40min
Overall bit rate mode : konstant
Overall bit rate : 1 152 Kbps

Audio
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Signed
Codec ID : 1
Duration : 2h 40min
Bit rate mode : konstant
Bit rate : 1 152 Kbps
Channel(s) : 1 Kanal
Sampling rate : 48,0 KHz
Bit depth : 24 bits
Stream size : 1,29 GiB (100%)

In comparison, the same action performed with the german bd of the same movie is working flawlessly:

eac3to v3.24
command line: eac3to P:\Rache\ 1) 6: C:\BD\Rache.wavs
------------------------------------------------------------------------------
M2TS, 1 video track, 8 audio tracks, 19 subtitle tracks, 2:20:06, 24p /1.001
1: Chapters, 50 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS-ES, English, 6.1 channels, 24 bits, 768kbps, 48kHz
4: DTS, Spanish, 5.1 channels, 24 bits, 768kbps, 48kHz
5: AC3, Spanish, 5.1 channels, 448kbps, 48kHz
6: DTS Master Audio, German, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
7: AC3, Hungarian, 5.1 channels, 448kbps, 48kHz
8: AC3, Polish, 5.1 channels, 448kbps, 48kHz
9: AC3, English, 2.0 channels, 224kbps, 48kHz
10: AC3, English, 2.0 channels, 224kbps, 48kHz
11: Subtitle (PGS), English
12: Subtitle (PGS), Spanish
13: Subtitle (PGS), Portuguese
14: Subtitle (PGS), German
15: Subtitle (PGS), Chinese
16: Subtitle (PGS), Arabic
17: Subtitle (PGS), Modern Greek
18: Subtitle (PGS), Hungarian
19: Subtitle (PGS), Icelandic
20: Subtitle (PGS), Hebrew
21: Subtitle (PGS), Polish
22: Subtitle (PGS), Spanish
23: Subtitle (PGS), Spanish
24: Subtitle (PGS), German
25: Subtitle (PGS), German
26: Subtitle (PGS), Chinese
27: Subtitle (PGS), Chinese
28: Subtitle (PGS), English
29: Subtitle (PGS), English
[a06] Extracting audio track number 6...
[a06] Decoding with ArcSoft DTS Decoder...
[a06] Writing WAVs...
[a06] Creating file "C:\BD\Rache.L.wav"...
[a06] Creating file "C:\BD\Rache.R.wav"...
[a06] Creating file "C:\BD\Rache.C.wav"...
[a06] Creating file "C:\BD\Rache.SR.wav"...
[a06] Creating file "C:\BD\Rache.LFE.wav"...
[a06] Creating file "C:\BD\Rache.SL.wav"...
[a06] The original audio track has a constant bit depth of 24 bits.
Video track 2 contains 201538 frames.
eac3to processing took 23 minutes, 37 seconds.
Done.

The MediaInfo provides the correct length (2h 20min.):

General
Complete name : C:\BD\Rache.R.wav
Format : Wave
File size : 1.13 GiB
Duration : 2h 20mn
Overall bit rate mode : Constant
Overall bit rate : 1 152 Kbps

Audio
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Signed
Codec ID : 1
Duration : 2h 20mn
Bit rate mode : Constant
Bit rate : 1 152 Kbps
Channel(s) : 1 channel
Sampling rate : 48.0 KHz
Bit depth : 24 bits
Stream size : 1.13 GiB (100%)

The only difference I noticed between the 2 logs is the "[a03] Remapping channels..." step at the US BD.
So my question is:
Is there any way you can turn off/disable the "Remapping Channels" step?
Or is there another way to get rid of the problem?
Thanks in advance for helping me out!

tebasuna51

16th June 2013, 16:22

SchnittenGott

17th June 2013, 08:41

@SchnittenGott
There are a problem, but isn't related with "Remapping Channels" that is needed for a 6.1 (english) and not needed for 5.1 (german).

Please run:
eac3to -test
to know software versions.

I installed only the ones I needed for dts decoding, so not all are present:

eac3to v3.27
command line: eac3to -test
------------------------------------------------------------------------------
eac3to (v3.27) is up to date
Nero Audio Decoder (Nero 7) is not working correctly
http://www.nero.com/eng/store-blu-ray.html
CAUTION: You need Nero 7. Nero 8 won't work with eac3to.
ArcSoft DTS Decoder (1.1.0.8) works fine
Sonic Audio Decoder (3.27.0.0) doesn't seem to be installed
Haali Matroska Muxer (2009-01-11) is installed
There's a new version (2011-03-03) 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 doesn't seem to be installed
http://www.surcode.com

Is there a new version of ArcSoft DTS Decoder?

tebasuna51

17th June 2013, 18:54

...
Is there a new version of ArcSoft DTS Decoder?

Not always a new version is better, I still use 1.1.0.0 without problems.

filler56789

18th June 2013, 15:20

eac3to v3.27
command line: eac3to -test
------------------------------------------------------------------------------
eac3to (v3.27) is up to date
ArcSoft DTS Decoder (1.1.0.8) works fine

Is there a new version of ArcSoft DTS Decoder?

Yes, 1.1.0.9

http://forum.doom9.org/showthread.php?p=1631214#post1631214

maxmercy

20th June 2013, 02:17

Hey everyone,

Is there a way to adjust individual channel levels in eac3to when changing formats?

I would like to take an HDAudio source and modify the individual channel levels prior to converting to/writing a .w64 file.

Thanks for your help,

JSS

Mark Regalo

20th June 2013, 02:21

Is there a list of all the .dlls and .ax files I need and what I need to register for all the decoders/encoders used in eac3to?

tebasuna51

20th June 2013, 06:14

Is there a way to adjust individual channel levels in eac3to when changing formats?
Nope, with eac3to you can only modify all channels with the same value.

maxmercy

20th June 2013, 13:00

Thanks for the prompt reply!

JSS

frencher

22nd June 2013, 13:48

Hello all,

What are all the extensions supported by eac3to ?
I know that one
.h264 .vc1 .ac3 .eac3 .thd .thd+ac3 .dts .dtshd .wav .wavs .sup .srt .txt

Y there a list of all those used by eac3to for demux ?

tebasuna51

22nd June 2013, 19:25

eac3to can output:

Video: .h264 .vc1 .m2v (all can be muxed to .mkv with Haali)
Uncompressed audio (extract or decode): .wav (.wavs) .w64 .rf64 .pcm .raw
Compressed audio (extract or encode): .ac3 .flac .dts (encode with SurCode) .m4a (aac audio in mp4 container encoded with NeroAacEnc)
Other compressed audio (only extract): .eac3 .thd (.thd+ac3) .dtshd .aac .mp2 .mp3
Subtitles: .sup .srt .ass .ssa
Chapters: .txt

frencher

23rd June 2013, 22:45

Already thank you very much for the reply;)
By cons I think it is missing, I just found one. Mpeg2 and it seems there are other ...
Madshi, could you provide a list of standard format demux function in eac3to -help

tebasuna51

24th June 2013, 00:49

By cons I think it is missing, I just found one. Mpeg2 and it seems there are other ...

What?

MPEG2 video -> .m2v
MPEG2 audio -> .mp2 .mp3 (maybe .mp1 but I don't found any sample to test)

frencher

24th June 2013, 05:20

Thanks tebasuna51,

It's strange, works same :confused:

MPEG2 video (http://ul.to/78qidjok) -> .m2v and .mpeg2

Why ?

What's missing here and what's wrong here for extract only ? ( -demux)

Keyword eac3to demux .extension

** VC-1 = .vc1
** h264/AVC = .h264
** MPEG2 = .m2v works with .mpeg2
** AC3 = .ac3
** E-AC3 = .eac3
** AC3 Surround = .ac3
** TrueHD = .thd
** TrueHD/AC3 = .thd+ac3
** DTS = .dts
** DTS Master Audio = .dtsma
** DTS Hi-Res = .dtshr
** DTS-HD (no core) = .dtsnc
** DTS Express = .dtslbr
** FLAC = .flac
** RAW/PCM = .pcm
** MLP = .mlp
** WAV = .wav
** W64 = .w64
** RF64 = .rf64
** DTSWAV = .dtswav
** AC3WAV = .ac3wav
** AAC = .aac
** MP1 = .mp1
** MP2 = .mp2
** MP3 = .mp3
** (PGS) = .sup
** (DVD) = .sup
** (SRT) = .srt
** (ASS) = .ass
** (SSA) = .ssa
** (VobSub) = ?

LigH

24th June 2013, 08:29

Depending on your purpose, you may want to support '.wavs' too (demultiplexes/decodes and de-interleaves a multi-channel audio stream to separate mono .wav files per audio channel).

VobSub consists of .sub (RLE-encoded DVD subpicture stream) + .idx (timecode and stream information text file).

tebasuna51

24th June 2013, 11:36

It's strange, works same :confused:

MPEG2 video (http://ul.to/78qidjok) -> .m2v and .mpeg2

Why ?
Because the extension is not mandatory for some formats.
The format of the file is defined in the file header.

For instance you can use .dts for all of this:
** DTS = .dts
** DTS Master Audio = .dts
** DTS Hi-Res = .dts
** DTS Express = .dts
The use of other extension is optional and only to remind you the internal format, but there are soft than only accept .dts

To extract the core of a dtshd you need add the -core parameter, if you put only the extension .dts, without -core, you obtain a dtshd.

BTW, DTS-HD (no core) don't exist, a dtshd must have always a core.

- .pcm and .raw have the same audio data but with different byte and channel order. A .raw is exact to a .wav/.w64/.rf64 without header.

- Graphic subtitles.
Only (PGS) = .sup is usefull

With (DVD), subs from VOB container, the .sup extracted is not supported by mkvmerge to mux or subrip/suprip/subtitleedit to convert to text subtitles.
If you want convert to text subs better use subrip/subtitleedit directly from VOB files.
BTW the .sup extracted by eac3to (or PgcDemux) can be used, with the .IFO file from DVD structure, by BDsup2sub to obtain the graphic subtitles sub/idx accepted by MkvMerge. But there are other tools to extract sub/idx directly from VOB's

The (VobSub), subs sub/idx in mkv container, aren't supported by eac3to to extract. Use MkvExtract instead.

EDIT:
** DTSWAV = .dtswav
** AC3WAV = .ac3wav
Aren't supported like output by eac3to.
Some times (not always) are supported like input to extract the .dts/.ac3 included in the fake wav container.

filler56789

24th June 2013, 19:41

...

BTW, DTS-HD (no core) don't exist, a dtshd must have always a core.

...

Only if one wants/needs compliance with the Blu-Ray specifications. In the so-called "Digital Delivery" mode, the Master Audio Encoder Suite can output a pure lossless stream, but only for certain channel layouts (2.0, 5.1 and 7.1).

nevcairiel

24th June 2013, 20:25

Even on Blu-ray you have DTS-HD streams without core, the so called DTS Express uses a low-bitrate HD stream without core to be able to produce a VBR stream with low bitrate for mostly voice (commentary tracks)

frencher

25th June 2013, 05:43

OK Thank's all :)
If update or error(s) send me update ;)

## Video
** VC-1 = .vc1
** h264/AVC = .h264
** MPEG1 = .mpeg1
** MPEG2 = .m2v
** MS/VFW/FOURCC/XVID = .xvid
## Audio
** AC3 = .ac3
** E-AC3 = .eac3
** AC3 Surround = .ac3
** TrueHD = .thd
** TrueHD/AC3 = .thd+ac3
** DTS = .dts
** DTS Master Audio = .dts
** DTS Hi-Res = .dts
** DTS-HD (no core) = .dts
** DTS Express = .dts
** FLAC = .flac
** RAW/PCM = .pcm
** MLP = .mlp
** WAV = .wav
** W64 = .w64
** RF64 = .rf64
** DTSWAV = .dtswav
** AC3WAV = .ac3wav
** AAC = .aac
** MP1 = .mp1
** MP2 = .mp2
** MP3 = .mp3
## Subtitles
** (PGS) = .sup
** (DVD) = .sup
** (SRT) = .srt
** (ASS) = .ass
** (SSA) = .ssa
** (VobSub) = .sup

tebasuna51

25th June 2013, 09:06

Only if one wants/needs compliance with the Blu-Ray specifications. In the so-called "Digital Delivery" mode, the Master Audio Encoder Suite can output a pure lossless stream, but only for certain channel layouts (2.0, 5.1 and 7.1).
Like I never see this kind of files please confirm:
- If this stream can be muxed in any container and extracted by eac3to.
- If eac3to can decode this stream.

Even on Blu-ray you have DTS-HD streams without core, the so called DTS Express uses a low-bitrate HD stream without core to be able to produce a VBR stream with low bitrate for mostly voice (commentary tracks)
OK, if we can call DTS Express like DTS High Definition.
BTW, DTS Express was already listed.

OK Thank's all :)
If update or error(s) send me update ;)

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

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

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

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

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

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

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

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

** (VobSub) =
eac3to can't extract this kind of sub.

filler56789

25th June 2013, 14:13

Like I never see this kind of files please confirm:
- If this stream can be muxed in any container and extracted by eac3to.
No.
- If eac3to can decode this stream.
No, and I serious doubt any DirectShow decoder available a.t.m. can do that as well.

BTW, neither mkvmerge nor LAV Splitter support A_DTS/LOSSLESS thus far.

Besides,

http://matroska.org/technical/specs/codecid/index.html says:

Please, when reading through this list, always keep in mind that the intention behind it is NOT to list all existing audio and video codecs, but more to list those codecs that are currently supported in matroska (or will be supported soon),

so the problem of course, is what they mean by "soon" :)

http://lists.matroska.org/pipermail/matroska-devel/2011-June/004051.html

IanD

26th June 2013, 10:33

DarkSpace

26th June 2013, 12:05

I recently tried converting an HD-DVD with TrueHD audio to Bluray format using first ClownBD 0.81 and then eac3to 3.27, but each software insists on decoding and re-encoding the AC3 core of the TrueHD track.

Does anyone know why this might be occurring and how to stop it happening?

From what I remember, TrueHD tracks on HD-DVD don't have an AC3 core, so eac3to will have to generate one from the decoded TrueHD track for the BluRay version. There is no "re-encoding" of the AC3 core, eac3to is just generating data that wasn't there previously.

IanD

26th June 2013, 15:34

From what I remember, TrueHD tracks on HD-DVD don't have an AC3 core, so eac3to will have to generate one from the decoded TrueHD track for the BluRay version. There is no "re-encoding" of the AC3 core, eac3to is just generating data that wasn't there previously.

Thanks for the clarification: I didn't realise that was the case.

A number of HD-DVD have better video and/or audio than the equivalent Bluray, still, and since I no longer have an HD-DVD player it's good to be able to get them into a playable format again.

yesgrey

27th June 2013, 15:23

I wanted to start using flaccl instead of flac inside eac3to, because it will save me a lot of time, but I want to be sure that it gives accurate results, hence why I wanted to use eac3to to convert a flaccl generated file to a flac eac3to flac, to compare with the original one created with eac3to, which I consider as the reference.

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

Now I can remux my movies much faster... :)

rapscallion

1st July 2013, 21:43

Video frames have a duration of 40 ms (25 fps) then delays of +/- 20 ms are absolutely undetectable.
From an old post, however I couldn't find the answer to this:
What is the video frame duration @23.976 fps?

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

The log shows (note lines 2 and 7):

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

I then converted the wavs to DTS-MA.

So, does the "applying RAW/PCM delay..." mean that I don't have to apply any delay when muxing the DTS track back with the video?

tebasuna51

1st July 2013, 21:56

What is the video frame duration @23.976 fps?
1000 ms / 23.976 = 41.7 ms

So, does the "applying RAW/PCM delay..." mean that I don't have to apply any delay when muxing the DTS track back with the video?
Correct.

rapscallion

1st July 2013, 22:18

@tebasuna51

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

I'm always a bit confused by audio/video delays.
Also, this is the first time I've seen "This track begins with a non-major frame. <WARNING>"

tebasuna51

2nd July 2013, 08:43

Also, this is the first time I've seen "This track begins with a non-major frame. <WARNING>"
I can't help you with this.
I don't know TrueHD specs.

rapscallion

2nd July 2013, 20:41

A TrueHD stream is composed of Major sync frames and non-major frames. Apparently tracks are supposed to start with a major frame.

This may have accounted for the -32ms delay and the warning.

DoctorM

3rd July 2013, 21:10

Love how useful this tool is. I have 2 feature requests:

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

2) Dolby Pro Logic II/IIx/IIz Upmix. I often want to expand the number of tracks on a stereo source to 5.1 or 6.1. Manually doing this is labor intensive and requires multiple programs and in between conversions. Be nice to just: Eac3to StereoTrack.wav SurroundTrack.ac3 -upDpl6

filler56789

3rd July 2013, 21:39

Nico8583

3rd July 2013, 22:01

Hi :)
Any news about fix for left view / right view with MVC flag ?
This case is open in Mantis : http://bugs.madshi.net/view.php?id=86
Thanks !

DoctorM

4th July 2013, 06:08

Thanks, filler56789, float me a loan? ;-)

frencher

5th July 2013, 06:08

Like I never see this kind of files please confirm:
- If this stream can be muxed in any container and extracted by eac3to.
- If eac3to can decode this stream.

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

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

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

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

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

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

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

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

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

** (VobSub) =
eac3to can't extract this kind of sub.

OK thanks tebasuna51 ;)

frencher

5th July 2013, 06:17

Hi :)
Any news about fix for left view / right view with MVC flag ?
This case is open in Mantis : http://bugs.madshi.net/view.php?id=86
Thanks !

Temporary While waiting for a update eac3to
Small tool for detect AVC/MVC <=> MVC/AVC order (http://forum.doom9.org/showthread.php?p=1635457#post1635457)

maxmercy

10th July 2013, 21:19

EDIT - Careful searching found that for any 6.1 format the remap must be done to accomplish -double7 processing correctly.

Hey everyone,

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

I was wanting to just use:

eac3to input.w64 output.w64 -double7

Will this work?

Thanks in advance,

JSS

tebasuna51

11th July 2013, 08:16

After my test seems you need:

eac3to input.w64 output.w64 -0,1,2,3,5,6,4 -double7

maxmercy

12th July 2013, 12:33

Thanks! It worked great!

JSS

Overdrive80

12th July 2013, 15:52

Would be possible implement option for output wav on 32 bit float??

Bigmango

14th July 2013, 17:00

Could we please get a replacement of the FLAC encoder with the one in ffmpeg 2.0 with "-compression_level 12" ?

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

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

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

Thanks.

Boulder

14th July 2013, 17:11

Can't you utilise piping to ffmpeg?

Bigmango

14th July 2013, 17:53

Can't you utilise piping to ffmpeg?

I am using the HD-DVD/Blu-Ray Stream Extractor (the MeGUI version as it allows to queue tasks - this way I can set up 20 or more MKV to FLAC taks in the queue and have it run in one go), so I don't think it's possible to setup piping with the gui.

Thanks for the feedback.

Boulder

14th July 2013, 19:31

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

gnol009

15th July 2013, 09:46

Anyone know how to creat qpfile.txt?

filler56789

15th July 2013, 16:27

Anyone know how to creat qpfile.txt?

Looks like you have posted in the wrong forum AND in the wrong thread :)

http://forum.doom9.org/forumdisplay.php?f=77

jpsdr

6th August 2013, 19:40

I have a little question...
If i do something like : "eac3to 00001.m2ts 3: c:\File.wavs"
Track 3 is DTS-HD MA, either 2.0, 5.1 or 7.1. I have the decoderdtsdll.dll installed.
Will the wave files be created only from the DTS core (so from lossy audio), or will they be created from the full DTS-HD MA information (the lossless audio) ?
Thanks.

rapscallion

6th August 2013, 20:44

I Use a GUI with eac3to and do the same kind of extractions and always get the full lossless wavs.

Sparktank

7th August 2013, 00:24

It will extract the core if you tell it to with "-core" switch.
Otherwise, if Arcsoft decoder registered, then it will extract the lossless DTS.

DoctorM

7th August 2013, 05:40

nevcairiel

7th August 2013, 07:10

I oddly find that when I'm decoding AC3 I have to force the switch -Nero. Yes, Nero 7 is properly installed, eac3to -test KNOWS it is installed (and says it works fine) and it seems to work properly when I add -Nero to my line. For some reason it doesn't seem to be the default decoder though.

Any thoughts on this?

The default decoder for AC3 is ffmpeg/libav, so that behaviour is expected (as documented in the first post in this thread)
I don't think the Nero decoder does a better job at it, so the default choice here makes sense, IMHO.

DoctorM

7th August 2013, 16:57

Thank you. I misread that. Nero is best for AAC not AC3. In fact nothing is listed as 'best for ac3". I guess I just assumed.

Sparktank

8th August 2013, 01:53

Thank you. I misread that. Nero is best for AAC not AC3. In fact nothing is listed as 'best for ac3". I guess I just assumed.

The Wikibooks (http://en.wikibooks.org/wiki/Eac3to/How_to_Use#Audio_Decoders) says that Nero should offer better sound quality over libav/ffmpeg.
As a result Nero7 is the default decoder for AC3 format.
Outdated since version 3.25...
* libav is now the default decoder for AC3 and E-AC3

But shortly before release of version 3.26, someone asked about this and Madshi said there was a problem where Nero didn't always remove DRC for some files.

Curious, why the change from Nero to libav as the preferred DD decoder?Because there were some problems with the Nero decoder. Mostly, disabling DRC didn't work for all tracks.

Overdrive80

8th August 2013, 12:00

The Wikibooks (http://en.wikibooks.org/wiki/Eac3to/How_to_Use#Audio_Decoders) says that Nero should offer better sound quality over libav/ffmpeg.

Outdated since version 3.25...

But shortly before release of version 3.26, someone asked about this and Madshi said there was a problem where Nero didn't always remove DRC for some files.

Somebody users reports issues decoding ac3 files with nero decoder, mainly with multitracks files (5.1). For this reason, libav is decoder for default.

This Wikibooks is outdated.

Mark Regalo

10th August 2013, 22:42

What files are used by eac3to from Nero to decode aac files? I have an old Nero 7 cd which I bought a long time ago yet I really don't want to install Nero on my computer.

Snowknight26

10th August 2013, 22:43

Then why do you want to use Nero to decode AAC?

Mark Regalo

10th August 2013, 23:16

Then why do you want to use Nero to decode AAC?

Isn't that what eac3to uses as the "best" method to decode aac? I just read what madshi wrote about eac3to in the OP.

Mark Regalo

12th August 2013, 01:43

So, any idea on how to use Nero with eac3to without installing it?

Sparktank

12th August 2013, 02:58

So, any idea on how to use Nero with eac3to without installing it?

There was a batch script created a long time ago that would do it for you.

Try a google search and limit to this forum only.
site:forum.doom9.org something something something eac3to

utenteanonimo64

15th August 2013, 14:40

Is there any software that can take the information created by eac3to in "chapters.txt" and apply it to a WAV/W64 file to split it into separate files with the appropriate WAV/W64 headers for each chapter?

Shevek

15th August 2013, 15:15

utenteanonimo64

15th August 2013, 18:38

Perhaps Medieval CUE Splitter?

Thanks for the quick reply. I will have to look if there is an easy way to turn "chapters.txt" into a CUE format file...

kevmitch

17th August 2013, 10:58

I noticed that decoding the same surround source file with the -downmix option produced different output files each time (the samples weren't the same). I'm pretty sure that this is because there is dithering which is randomly chosen. Sure enough, when I specify "-full" as suggested by madshi (http://forum.doom9.org/showthread.php?p=1401084#post1401084), the output is identical every time.

My question is how exactly is the random noise seeded/generated and why not just use a method (i.e., same seed) that produces identical results each time? I know it would put my mind at ease a little.

kevmitch

17th August 2013, 11:52

laserfan

18th August 2013, 16:58

I recently needed to re-work a BD and keep the TrueHD audio track, and the demuxed result wouldn't re-mux properly. I instead used tsMuxeR to demux and that worked perfectly.

I wonder: does eac3to demux TrueHD tracks (AS TrueHD tracks, not just the AC3 core) properly, and I just had a one-off glitch of some sort?

If demuxing TrueHD *is* a problem for eac3to, are there any other such demuxing gotchas that are known?

tebasuna51

18th August 2013, 17:17

I don't know problems demuxing TrueHD tracks with eac3to.

laserfan

18th August 2013, 18:17

I don't know problems demuxing TrueHD tracks with eac3to.
Thanks.

Bigmango

22nd August 2013, 06:41

kukushka

22nd August 2013, 13:31

Same problem here.

probably it's worth to give it a try with sonic too. but if every possible decoder decode same distorted original, it probably was there initially..

Bigmango

22nd August 2013, 16:03

While waiting to be able to post here for the first time, I posted over at avsvorums (http://www.avsforum.com/t/1482702/oblivion-blu-ray-official-avsforum-review/150#post_23626101) about a problem I noticed trying to decode the DTS HD MA sound from the Obilivion blu ray. In short, the center channel becomes distorted in loud bits. Since that posting, I have realised that specifying '-core' option to eac3to fixes the problem but of course its not "lossless", which my decode of the MA track clearly isn't either in this case.

I have uploaded a sample (https://dl.dropboxusercontent.com/u/60598588/oblivion_distort.zip) .dtshd file from 8:10-8:20 of the film which exhibits the problem. Also included with the sample are my eac3to decodes of the centre channel with and without "-core" as well as the libav decode, which also exhibits the distiortion. And of course all the logs and eac3to -test results.

I get the same results (bit-for-bit) using eac3to with the Arcsoft decoder versions 1.1.0.0, 1.1.0.7 and 1.1.0.8. I also hear the distortion when decoding the .dtshd directly in mplayer, vlc and audacity (which I think use the same code as libav anyway).

EDIT: On second thought, the distortion is there in core track as well. I don't know how I didn't hear that before.

After some more testing with the full movie, this seems to be a problem with the ffmpeg decoder.

BTW, as you are saying the tiny distortion you are getting in your sample is in the original track as well. But, starting at the same time up to 20 seconds later (not in your sample) there often is A LOT of sound clipping (and in other parts of the movie too).

I seem to be getting this with ffmpeg based decoders, and then not every time. PowerDVD plays it fine (also the FLAC).

So, as far as I can see, this is a bug in ffmpeg.

nevcairiel

22nd August 2013, 16:31

How is it a bug in ffmpeg if the original problem was the ArcSoft decoder causing the issue? :)

Bigmango

22nd August 2013, 17:00

How is it a bug in ffmpeg if the original problem was the ArcSoft decoder causing the issue? :)

I first thought Arcsoft was the issue (I played my Oblivion with FLAC backup on XBMC), but after some more testing the ffmpeg based players (tested XBMC, MPC-HC have the problem) seem to be the only ones having the playback problem (the tested XBMC which is playing back the FLAC as PCM on a hardware receiver is on a totaly different system than the tested MPC-HC, totaly different hardware and drivers - both have the problem.).

Furthermore, PowerDVD plays the eac3to+Arcsoft transcoded FLAC fine whitout the clipping that's sometimes produced by the ffmpeg players, and Oblivion is the only movie having this issue (in a media library of ~300 movies).

So, if PowerDVD plays the (Arcsoft decoded) FLAC fine but 2 different ffmpeg players on 2 different systems don't, ffmpeg is the issue.

nevcairiel

22nd August 2013, 17:16

So you're saying ffmpegs FLAC decoder is the issue, right? Not actually related to DTS directly at all?
I tried listening to the FLAC center channel samples posted above, and i couldn't really hear any obvious distortions, but maybe i just dont know what to listen for (or my player is fixed)

Sparktank

22nd August 2013, 23:11

Regarding distortion: I've been meaning to start a new thread with heavy investigation into this.
Intense testing with libav/arcsoft/sonic/etc.

I didn't observe any (for Oblivion) using Foobar2000 with the center-FLAC.
I can't remember if it was apparent in MPC-HC (FFDshow), but I don't remember there being any.
The distortion wasn't immensely apparent for me as it was for you.

Even opening in Audacity it didn't show any clipping (marked in red). It does show that it's cut off at .8 ("The vertical scale displays amplitude when showing the waveform" from Audacity Help)
http://i.imgur.com/KNnn2lO.png

Using PaulStretch (to slow down the audio by repeating samples numerous times) shows clipping when slowed down to 10x the normal rate.
http://i.imgur.com/ICQa6Od.png

Which is where you hear the perceived distortion ("perceived" as in it's not apparent to everyone).

----

This could be a result of various setups of software/hardware. (why some can hear it and some cannot)
I've disabled all "effects" that my soundcard has to offer (ie, "loudness equalization", etc).
I've even disabled any normalization or regain effects in FFDShow when playing back the FLAC in MPC-HC.
Also MPC-HC has all its internal effects disabled to play back the audio with the lest altering/manipulation.

I haven't done any testing with the elementary stream (dts-hd ma) though.
I have Arcsoft 1.0.0.9 and can do a test to compare but I don't have any detailed knowledge or software to really examine the differences between the other versions.
But Foobar2000 has a "compare two files" to detect any offset in volume, etc.

----

I observed the same thing with Tron Legacy. Although, with TL, the distortion is much more apparent than that of Oblivion.
After looking at the Oblivion samples (which played fine with barely any distortion in Foobar), I went to rip my BD of TL to test for distortion.

It seems that depending on the player it (Tron Legacy) did have severe bass distortion.
I observed distortion on different instances. Notably when playing back on MPC-HC with FFShow.

Playing back a converted WAV (5.1 (down6) or 7.1) on Foobar doesn't have any distortion but playing the WAVs on MPC-HC did however (FFDShow again).
I have not tried using LAV filters though or other.
Downmixing (either 7.1 or converted 5.1) to 2.0 (-downStereo; -7.xxdb automatically applied with 2nd pass) did playback on all media players without any distortion.
---
I'm still playing around with different setups to test with the most variables.
I'm even running through SoX for "stat" and "stats" (though that info is quite useless to me until I learn it).

But the distortion seemed very conditional.
I just need to remember to document everything I did to see where distortion was obvious (and not, in case of Foobar??)

Though I'll be creating a new thread for this with mega details.
I'm really curious about some movies having this weird distortion anomaly (fact: some hear it; some don't).

X-Men: First Class has it at times as well.
I can hear it when watching the movie on my system (but since I upgraded my BD player, it might be a different experience now).

Sorry for the off-topic details, will create a new thread.
This is highly intriguing.

madshi

22nd August 2013, 23:18

If the problem is in the ffmpeg decoder then using madFlac should work without issues?

nevcairiel

23rd August 2013, 01:36

The ffmpeg FLAC decoder is perfectly lossless, so its more likely some other combination of the playback chain.

Bigmango

23rd August 2013, 04:51

So you're saying ffmpegs FLAC decoder is the issue, right? Not actually related to DTS directly at all?

Yes, or something else in ffmpeg.

I tried listening to the FLAC center channel samples posted above, and i couldn't really hear any obvious distortions, but maybe i just dont know what to listen for (or my player is fixed)

Yes, these samples don't have the distortion. There is some kind of "stuttering" at one point , but it is also present in the original DTS track.

What I am however experiencing, is sound "clipping" (some very high and loud fast "pops" or "ticks") which happens several times starting at around 08:05 to 08:30 in the movie (and at some other places in the movie too when there's some high pitched sound).

edit: I think this is what Sparktank explains above.

Bigmango

23rd August 2013, 04:59

The ffmpeg FLAC decoder is perfectly lossless, so its more likely some other combination of the playback chain.

If this is really the case (?) then it should be something else in the ffmpeg playback chain as Openelec (XBMC) playing back as PCM on a hardware receiver and MPC-HC are both having the issue, on totally different hardware (+ Openelec is Linux and MPC-HC is on Windows).

Bigmango

23rd August 2013, 05:06

Which is where you hear the perceived distortion ("perceived" as in it's not apparent to everyone).

It comes out very lound and clear on my 7.1 home theater system (pay back with Openelec XBMC mediacenter).

On windows MPC-HC with headphones it is less noticeable, but still there. However, the strange thing is that sometimes the clipping is there, and sometimes it isn't. This needs more testing...

tebasuna51

23rd August 2013, 11:03

I make some test over the dtshd sample included. And I want post my opinion about this.

1) Decoded to wavs directly (ArcSoft 1.1.0.0 and libav) result the same output than decode the Flacs included in the sample.
For me don't exist problems related to Flac encoder/decoders.

2) Decoding the "core" with ArcSoft produce the same (more or less) output than decoding with ffmpeg/libav (or libdts with NicAudio).
Then I can't suspect problems with ffmpeg/libav decoder.

3) There are clip (see attached image) in the wave form.
This can be listened like distort sound, more or less clear depending on audio system chain.
But the clip isn't at 0dB, that suggest the problem was in the original source and isn't a decoder problem.

4) But there are a problem that I never see. The lossless output from DTS-MA decoded with ArcSoft is 3dB loud than the decoded core (ArcSoft or libav).
Not only on clip parts but over the full sample.

I checked all my DTS-MA samples and always the volume is the same decoding lossless or core.

The attached image is from my decoded wavs but is the same than flacs included in the sample.

Anakunda

23rd August 2013, 15:29

Hello, may I get some help how to properly setup eac3to for conversion 7.1 channels audio, whenever I try to convert 8-channel track (DtsHD master audio) to AAC it always produces useless 8channel digital noise. All 6-channel tracks convert nicely so the problem must have to do something with the channels layout. Maybe wrong channels mapping? Or is that encoder bug? Is Arcsoft decoder decoding the audio improperly?
Using pipe, the only way I know:

C:\Users\Anakunda\AppData\Local\Temp>benice "I:\video\(workset)\MARVELS_IRON_MAN_3_BD_BLUEBIRD\english71.dtshd" -d -55 --down16
Executing eac3to "I:\video\(workset)\MARVELS_IRON_MAN_3_BD_BLUEBIRD\english71.dtshd" stdout.wav -55ms -down16 -normalize | D:\media\AAC\qaac.exe --ignorelength --threading --verbose --no-delay --rate keep --quality 2 --tvbr 100 -o "I:\video\(workset)\MARVELS_IRON_MAN_3_BD_BLUEBIRD\english71.m4a" -

eac3to v3.27
command line: D:\media\eac3to\eac3to.exe "I:\video\(workset)\MARVELS_IRON_MAN_3_BD_BLUEBIRD\english71.dtshd" stdout.wav -55ms -normalize
------------------------------------------------------------------------------
DTS Master Audio, 7.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
Decoding with ArcSoft DTS Decoder...
Applying RAW/PCM delay...
Writing WAV...
Creating file "stdout.pass1.wav"...
The original audio track has a constant bit depth of 24 bits.
Starting 2nd pass...
Reading WAV...
Reducing depth from 64 to 24 bits...
Writing WAV...
Applying -0.09dB gain...
Creating file "stdout.wav"...
The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 49 minutes, 59 seconds.
Done.

qaac 2.19, CoreAudioToolbox 7.9.8.3
english71.m4a
Prepend 960 samples (0.02 seconds)
Using default channel layout.
Output layout: 7.1 (C Lc Rc L R Ls Rs LFE)
Enable threading
AAC-LC Encoder, TVBR q100, Quality 96
2:10:32.839 (4.6x)
375976304/-1 samples processed in 28:09.990
Overall bitrate: 665.123kbps
7812/7812 chunks written (optimizing)
Encoding done.

tebasuna51

23rd August 2013, 21:22

Hello, may I get some help how to properly setup eac3to for conversion 7.1 channels audio, whenever I try to convert 8-channel track (DtsHD master audio) to AAC it always produces useless 8channel digital noise.
Without problems here with a simple channels test DTS-MA 7.1.

The ArcSoft (1.1.0.0) output is correct (FL FR FC LF BL BR SL SR) the qaac (2.19) Output layout: 7.1 (C Lc Rc L R Ls Rs LFE) is the default like you can see here (https://github.com/nu774/qaac/wiki/Multichannel--handling), but is played correctly in my 7.1 system with mpc-hc, and decoded correctly with NeroAacDec.

Try without --threading to see if there are problems in your system with pipe and multithread with big files.
Try also decode the DTS-MA to wav and encode the wav file.

Anakunda

23rd August 2013, 23:21

Try without --threading to see if there are problems in your system with pipe and multithread with big files.

Thanks dude, don't know what's --threading exactly for but it seems without it the output now is perfect :cool:
For some reason with 6channel audio qaac with threading support worked correctly.

carlmart

28th August 2013, 14:33

It's been a long time since I could use eac3to. I wonder what might be happening.

I have already installed both eac3to and put the GUI on the same folder. But when I load a file as I remember I did, something seems to go wrong because it doesn't accept any command any longer.

Sorry if this question is so primary.

Snowknight26

28th August 2013, 20:50

eac3to doesn't come with a GUI.

What's the question?

Sparktank

28th August 2013, 21:57

the GUI on the same folder

Which GUI? There's a couple.
Try downloading a fresh copy of the GUI and installing it in the existing folder.

Maybe you lost some dependencies (MSV? etc).

Or when was the last time you used it?
eac3to is now 3.27 and one of the GUI's had to update (for reasons I can't remember) so try downloading a newer/updated version?

carlmart

29th August 2013, 15:35

Sorry for not including more details.

GUI was < yr_eac3to_more_gui>, supposedly latest version.

eac3to is January 6, 2013 version, so it is 3.27

How do I check my "dependencies"? What's MSV?

LigH

29th August 2013, 15:41

Possibly "Microsoft Visual ...(C++ Runtime Libraries)"...

Sparktank

29th August 2013, 19:03

yes, some GUI's might need Microsoft Visual.
I'm not sure, I don't use GUI's.

But that GUI seems to have stopped development. It may not be compatible with v3.27.

Switch to something more recent and still active like BDClown or UsEac3to v1.0.0.

Clown BD: (from first post in this thread)
http://forum.slysoft.com/showthread.php?t=25818

UsEac3to v1.0.0: a different GUI for eac3to
http://forum.doom9.org/showthread.php?t=145574

All other GUI's seem to have died off in 2010.

laserfan

30th August 2013, 23:17

While waiting to be able to post here for the first time, I posted over at avsvorums (http://www.avsforum.com/t/1482702/oblivion-blu-ray-official-avsforum-review/150#post_23626101) about a problem I noticed trying to decode the DTS HD MA sound from the Obilivion blu ray. In short, the center channel becomes distorted in loud bits. Since that posting, I have realised that specifying '-core' option to eac3to fixes the problem but of course its not "lossless", which my decode of the MA track clearly isn't either in this case.

I have uploaded a sample (https://dl.dropboxusercontent.com/u/60598588/oblivion_distort.zip) .dtshd file from 8:10-8:20 of the film which exhibits the problem. Also included with the sample are my eac3to decodes of the centre channel with and without "-core" as well as the libav decode, which also exhibits the distiortion. And of course all the logs and eac3to -test results.

I get the same results (bit-for-bit) using eac3to with the Arcsoft decoder versions 1.1.0.0, 1.1.0.7 and 1.1.0.8. I also hear the distortion when decoding the .dtshd directly in mplayer, vlc and audacity (which I think use the same code as libav anyway).

EDIT: On second thought, the distortion is there in core track as well. I don't know how I didn't hear that before.
I just racked this one up and my Arcsoft TMT 3 which uses (I believe) dtsdecoderdll.dll 1.1.0.7 has serious problems with the DTS-MA audio. But on my setup it's not as you describe, but rather a problem with the background music and afaict not the center channel. In any case it is a problem throughout.

Using instead my Media Player Classic HC, version 1.6.8.7417, playback of the soundtrack is perfect. I dunno how to determine what decoding method is being used in MPC-HC.

An interesting problem; I have never seen anything like it before, so I wonder what is different about this disc.

EDIT: Bah, I played with this some more today, and Arcsoft TMT 3 (mine's 3.0.1.185) does indeed play this disc fine--I don't know what the magic is though. I had looked at it a half dozen times, and fiddled with settings, but today I fiddled with settings again, both the Audio Sound Card/Output Mode settings and the Audio Effect mode, and it just started sounding correct. I can't seem to "go back" i.e. change settings back and get the bad audio again. Sorry for the bandwidth--I don't know what the problem might have been before.

Sparktank

30th August 2013, 23:43

dtsdecoderdll.dll 1.1.0.7

Have you tried 0 or 8? (or even 9)

While playing back in MPC-HC you can go the Menu Bar > Play > Filters and see what is being used to decode (most likely FFDShow; core audio for DTS-HD MA).

The previous comments from others largely suggests the process chain has a huge effect on the perceivable distortion yet it does seem to be on the disc itself.
Looking at other threads about similar problems, it seems that hardware also has different ways of experiencing the clipping/distortion.

If you read the thread about Tron: Legacy, you'll see endless posts about software VS hardware and even the probability of getting just a bad disc from a "bad batch".

Although, I've yet to see any hard evidence supporting any such claims of "bad batches" where the audio is malformed.
In most cases of "bad batches", the disc itself deteriorates and makes the entire disc unreadable.

Or just the transfer itself is poorly mastered (in the case of the UK Boxset for David Lynch where many customers complained and received a newly transferred boxset free of charge upon producing a receipt within a constrained time limit).

Tron Legacy and X-Men First Class exhibit similar clipping/distortion, from what I've seen.

A most recent extraction of a sample from Tron: Legacy (using Arcsoft versions 0, 8 and 9) indicate the clipping/distortion is on the movie itself, and while others experience it, I don't believe it to be secluded to a single disc or a small batch.

laserfan

31st August 2013, 13:12

Have you tried 0 or 8? (or even 9)
No. I do have TMT5 and 6 also but they are not installed atm. Can I just swap-in to TMT3 the newer dlls?

To be clear, while I've posted in the eac3to thread, I have not attempted (yet anyway) to convert in any way the DTS-MA soundtrack. I posted merely to say that 1.) Oblivion ripped w/o errors, so whatever problems the disc might exhibit are not due to disc defects, and 2.) whatever problems one might have with the sound are due to the player/playback software, cuz MPC-HC plays it perfectly.

I've yet to play the disc on any of my set-top players, but fully expect the sound to be perfect.

When I buy a disc, I tend to try ripping it onto my PC in order to confirm it is not defective (and before committing 2+ hours to it in my HT).

IanD

1st September 2013, 04:03

Is there an option in eac3to to combine a TrueHD soundtrack and AC3 soundtrack into a .thd+ac3 combination?

I have a situation where a .thd+ac3 soundtrack was accidentally demuxed into its separate components and TSmuxer does not seem to want to mux a standalone TrueHD track.

At the moment, eac3to decodes the TrueHD track to AC3 and uses that to create a .thd+ac3, but I would prefer to use the original AC3.

tebasuna51

1st September 2013, 13:31

@IanD
I think is not possible now.

Stereodude

7th September 2013, 18:49

Are there any known issues with ArcSoft DTS Decoder - version 1.1.0.0 when handling DTS Master Audio, English, 6.1 channels, 24 bits, 48kHz (core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz) with v3.27 of eac3to?

I tried a -downDpl -mixlfe and -downDpl and got a stereo wave file in both cases with a ~6dB channel imbalance between L and R and the file doesn't sound anything like what I hear when playing the disc in ArcSoft TotalMedia Theatre 5 having it do a 2 channel mixdown.

DPLII mixdown w/ LFE:
http://imageshack.us/a/img822/985/oe80.png

DPLII mixdown no LFE:
http://imageshack.us/a/img835/360/a5nm.png

I extracted each track to a separate .wav using .wavs and eac3to is correct that the Back Center channel is empty (as shown in the log files below). However, there isn't any left/right volume imbalance in the remaining 6 individual wave files, so I'm unsure why the 2 channel mixdown is off. :confused:

Individual channels - click for full size:
http://imageshack.us/a/img28/3706/og7p.th.png (http://imageshack.us/a/img28/3706/og7p.png)

"LOGS":
C:\HDTV Tools\eac3to>eac3to.exe i: 1) 3: e:\TempBD\CSH\DTSMA_w_LFE.wav -downDpl -mixLFE
M2TS, 1 video track, 2 audio tracks, 1:35:56, 24p /1.001
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz)
4: RAW/PCM, English, 2.0 channels, 24 bits, 48kHz
a03 Extracting audio track number 3...
a03 Remapping channels...
a03 Decoding with ArcSoft DTS Decoder...
a03 Downmixing multi channel audio to stereo...
a03 Reducing depth from 64 to 24 bits...
a03 Writing WAV...
a03 Clipping detected, a 2nd pass will be necessary.
a04 Creating file "e:\TempBD\CSH\stereo.wav"...
a03 Creating file "e:\TempBD\CSH\DTSMA_w_LFE.wav"...
a03 Original audio track, L+R+C+LFE+BL+BR: constant bit depth of 24 bits.
a03 Original audio track, BC: no audio data.
a03 The processed audio track has a constant bit depth of 24 bits.
a03 Starting 2nd pass...
a03 Extracting audio track number 3...
a03 Remapping channels...
a03 Decoding with ArcSoft DTS Decoder...
a03 Downmixing multi channel audio to stereo...
a03 Reducing depth from 64 to 24 bits...
a03 Writing WAV...
a03 Applying -14.42dB gain...
a03 Creating file "e:\TempBD\CSH\DTSMA_w_LFE.wav"...
a03 The processed audio track has a constant bit depth of 24 bits.
Video track 2 contains 138004 frames.
eac3to processing took 25 minutes, 5 seconds.
Done.

C:\HDTV Tools\eac3to>eac3to.exe i: 1) 3: e:\TempBD\CSH\DTSMA.wav -downDpl
M2TS, 1 video track, 2 audio tracks, 1:35:56, 24p /1.001
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz)
4: RAW/PCM, English, 2.0 channels, 24 bits, 48kHz
a03 Extracting audio track number 3...
a03 Remapping channels...
a03 Decoding with ArcSoft DTS Decoder...
a03 Downmixing multi channel audio to stereo...
a03 Reducing depth from 64 to 24 bits...
a03 Writing WAV...
a03 Clipping detected, a 2nd pass will be necessary.
a03 Creating file "e:\TempBD\CSH\DTSMA.wav"...
a03 Original audio track, L+R+C+LFE+BL+BR: constant bit depth of 24 bits.
a03 Original audio track, BC: no audio data.
a03 The processed audio track has a constant bit depth of 24 bits.
a03 Starting 2nd pass...
a03 Extracting audio track number 3...
a03 Remapping channels...
a03 Decoding with ArcSoft DTS Decoder...
a03 Downmixing multi channel audio to stereo...
a03 Reducing depth from 64 to 24 bits...
a03 Writing WAV...
a03 Applying -9.71dB gain...
a03 Creating file "e:\TempBD\CSH\DTSMA.wav"...
a03 The processed audio track has a constant bit depth of 24 bits.
Video track 2 contains 138004 frames.
eac3to processing took 25 minutes, 7 seconds.
Done.

DoctorM

7th September 2013, 19:59

Each version of ArcSoft has it's own issues:
I tested 5 different versions of dtsdecoderdll.dll along with LAV Audio, here are results.

Versions tested: 1.1.0.0, .1, .5, .7, .8.

Samples tested:
1) Lossy DTS 2.0, 2.1, 5.1, 6.0, 6.1 configs, 16/48, 24/48 and 24/96 bitdepth/sample rate;
2) DTS-MA 1.0, 2.0, 2.1, 5.1, 6.1, 7.1 configs including 7.1 "strange setup" config (eac3to term), 16/48, 24/48, 24/96 variants.
My results differ from results that some people posted in eac3to thread, I guess there may be some eac3to specific problems.

lossy DTS
1) 1.1.0.0 and 1.1.0.1 always decode lossy DTS as 24 bit, and their decoding results differ from each other.
2) 1.1.0.5 and up decode in proper bitdepth.
3) Versions 1.1.0.5 and up decode lossy DTS identically.
4) 1.1.0.1 and 1.1.0.5 decode 24 bit lossy DTS identically.
5) Decoding of 16 bit lossy DTS was changed from 1.1.0.0 to 1.1.0.1 and from 1.1.0.1 to 1.1.0.5.
6) 1.1.0.7 and 1.1.0.8 decode 6.0 without back center channel.

The conclusion for lossy: I assume the best version for lossy DTS is 1.1.0.5. It decodes all configurations I've tested, in proper bitdepth, and decoding algorithm didn't change since this version (except 6.0 bug in .7 and .8).

DTS-MA
All versions output identical streams except mono case. Mono decoding was broken in 1.1.0.5 and partially fixed in 1.1.0.8 (outputs as stereo). Note that MA 7.1 16/48 "strange setup" is decoded as 24/48 by all versions.

The conclusion for lossless: the best versions are 1.1.0.0 and 1.1.0.1.

Of course, to be sure that decoding is done properly some "reference proper" decoder is needed. I don't have it.

EDIT: Forgot to mention about 2.1 bug. None of the versions can decode LFE channel in 2.1 files.

Stereodude

7th September 2013, 19:59

Frankly, I'm not sure what's going on... I took a closer look at the channels that came out when I decoded with .wavs. The supposed LFE channel is full range and sounds like a back center channel.

As a sanity check I played the Blu-Ray back from my HTPC through my Pioneer Elite receiver which supports HDMI bitstreaming. It sees a 6.1 DTS-HD MA audio track and there is content in all 6.1 channels, no empty channels.

eac3to tells me I have the ArcSoft DTS decoder (1.1.0.0)

C:\HDTV Tools\eac3to>eac3to.exe -test
eac3to (v3.27) is up to date
Nero Audio Decoder (Nero 7) works fine
ArcSoft DTS Decoder (1.1.0.0) works fine

Stereodude

7th September 2013, 20:19

Each version of ArcSoft has it's own issues:How are you determining the revision? When I right click on ASAudioHD.ax bring up the properties and click on the version tab I see numbers like

1.3.2.87
1.3.3.127
1.3.3.283

eac3to tells me "ArcSoft DTS Decoder (1.1.0.0) works fine" for all 3 of those versions. :confused:

DoctorM

7th September 2013, 20:21

Check the version of dtsdecoderdll.dll. "Eac3to -test" should report the correct version as well.

Stereodude

7th September 2013, 20:31

Check the version of dtsdecoderdll.dll. "Eac3to -test" should report the correct version as well.Oh, okay. I was checking the wrong file.

I between my various computers I have 1.1.0.0 and 1.1.0.8. I apparently have two different 1.1.0.8 versions though since the file sizes are different and they are not binary identical.

Stereodude

7th September 2013, 21:05

Do I just need to change out the .dll to change decoder versions? I'm getting 7 identical channels of corrupted / garbled audio when trying the 1.1.0.8 dll's. :confused:

Since I can't get 1.1.0.8 to work, I'll just report the results I got from 1.1.0.0

With just the DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz core you don't get the back center channel. If you decode the full DTS-HD MA, the channel that was LFE channel (with the core) turns into a full range channel. It seems to be the back center, but I'm can't confirm if it's been correct decoded or not. However, the LFE channel vanishes.

tebasuna51

7th September 2013, 21:15

...
I tried a -downDpl -mixlfe and -downDpl and got a stereo wave file in both cases with a ~6dB channel imbalance between L and R and the file doesn't sound anything like what I hear when playing the disc in ArcSoft TotalMedia Theatre 5 having it do a 2 channel mixdown.

Is not a problem with ArcSoft version, try adding -phaseShift parameter to -downDpl ( -mixlfe not recommended).

Stereodude

7th September 2013, 21:29

Is not a problem with ArcSoft version, try adding -phaseShift parameter to -downDpl ( -mixlfe not recommended).Well, ArcSoft / eac3to can't even correctly decode the 6.1 DTS-HD MA audio so I certainly wouldn't exactly expect it to downmix it correctly.

dansrfe

7th September 2013, 23:16

Is vorbis encoding/decoding support within eac3to in the foreseeable future?

Sparktank

8th September 2013, 04:57

Well, ArcSoft / eac3to can't even correctly decode the 6.1 DTS-HD MA audio so I certainly wouldn't exactly expect it to downmix it correctly.

Have you even tried his suggestion to use -phaseshift?
(omitting -mixlfe)

If you want to keep the LFE channel, try making a 2.1 (LFE separate channel not mixed into FL+FR channels) audio file instead.

tebasuna51

8th September 2013, 11:29

Well, ArcSoft / eac3to can't even correctly decode the 6.1 DTS-HD MA audio so I certainly wouldn't exactly expect it to downmix it correctly.

I don't have problems decoding 6.1 with ArcSoft 1.1.0.0 like you have.

Try with this test channels: http://www.sendspace.com/file/s04pgo

tebasuna51

8th September 2013, 11:43

Is vorbis encoding/decoding support within eac3to in the foreseeable future?

To encode you can use the pipe option:

eac3to input stdout.wav | OggEnc2 -q 3 --ignorelength -o %_.ogg -

To decode we need than madshi add the option, libav can decode vorbis.

Stereodude

8th September 2013, 13:03

Have you even tried his suggestion to use -phaseshift?
(omitting -mixlfe)

If you want to keep the LFE channel, try making a 2.1 (LFE separate channel not mixed into FL+FR channels) audio file instead.No, I haven't tried it yet. I'm still working to get it to decode all the channels instead of leaving some of them empty. Once it can decode all the channels I'll worry about the downmix.

I don't have problems decoding 6.1 with ArcSoft 1.1.0.0 like you have.

Try with this test channels: http://www.sendspace.com/file/s04pgo
That test file decodes fine.

Try this one:
http://www.sendspace.com/file/vw8tb3

tebasuna51

8th September 2013, 17:48

Try this one:
http://www.sendspace.com/file/vw8tb3

You are right, decoding this sample the BC channel is empty and the LFE channel are full of high frequencies.

Now the question is if there are a decoder than can decode correctly this sample. Try other ArcSoft versions or Sonic.

If don't found any maybe we can conclude than this dtshd was wrong encoded.

Decoding the core with Foobar2000 or NicDtsSource the LFE channel is low volume but with high frequencies also. I think something is wrong in this sample.

Stereodude

8th September 2013, 18:50

You are right, decoding this sample the BC channel is empty and the LFE channel are full of high frequencies.

Now the question is if there are a decoder than can decode correctly this sample. Try other ArcSoft versions or Sonic.

If don't found any maybe we can conclude than this dtshd was wrong encoded.

Decoding the core with Foobar2000 or NicDtsSource the LFE channel is low volume but with high frequencies also. I think something is wrong in this sample.Did you try the core with eac3to? When i did, the BC channel is empty but the LFE seemed okay.

The sample may be unusual, but it does decode fine on my receiver via HDMI. It shows DTS Master Audio with 6 channels + LFE. When I bypass all bass redirection and processing I get a full range signal in all 6 channel (none are empty), plus a solid LFE channel.

As I recall there were several variants of DTS-ES. One has the 6th channel matrixed into the two rear channels. The other had a discrete 6th channel. I haven't played the core via SPDIF to a receiver yet to see which this is.

I couldn't get 1.1.0.8 working on my system and don't have other versions. I'm not sure if I have Sonic.

tebasuna51

8th September 2013, 19:16

Did you try the core with eac3to? When i did, the BC channel is empty but the LFE seemed okay.
Yes, I decode the core with ArcSoft and libav and both output a BC empty and a LFE like Foobar2000 or NicDtsSource, low volume but with high frequencies, not normal.

The sample may be unusual, but it does decode fine on my receiver via HDMI. It shows DTS Master Audio with 6 channels + LFE. When I bypass all bass redirection and processing I get a full range signal in all 6 channel (none are empty), plus a solid LFE channel.
Do you have a 6.1 speaker system?

If you have 5.1 or 7.1 maybe the receiver make some mix to side/back channels.

You can't know if LFE have high frequencies, only low frequencies was send to subwoofer speaker.

As I recall there were several variants of DTS-ES. One has the 6th channel matrixed into the two rear channels. The other had a discrete 6th channel. I haven't played the core via SPDIF to a receiver yet to see which this is.
This sample is signaled like 6.1 discrete (Extended Coding Flag set to 1)

Nico8583

8th September 2013, 19:22

Stereodude

8th September 2013, 22:56

Yes, I decode the core with ArcSoft and libav and both output a BC empty and a LFE like Foobar2000 or NicDtsSource, low volume but with high frequencies, not normal.I went back and double checked... Libav and ArcSoft give me a real LFE channel with the core, not a full range channel at a low volume.

Here's an mp3 of each (I increased the volume by 10dB).

Libav (core):
http://www.sendspace.com/file/xnr773

ArcSoft v1.1.0.0 (core):
http://www.sendspace.com/file/6kxkst

Do you have a 6.1 speaker system?

If you have 5.1 or 7.1 maybe the receiver make some mix to side/back channels.No, I have a 7.1 speaker system. The receiver mirrors the BC into both of the rear speakers. It does not mix the sides into the rears. I used Pure Stream Direct when testing which disables any channel redirection, bass management, or processing of the audio when I tested.

This sample is signaled like 6.1 discrete (Extended Coding Flag set to 1)ArcSoft TMT 5 says it's ES Discrete when output via SPDIF. Apparently my Pioneer Elite receivers don't display whether it's Discrete or Matrix. My old Sony ES unit did.

Edit: Sonic 4.3.0.169 does the same thing as ArcSoft 1.1.0.0. The core leaves the BC channel empty, but decodes the LFE. The full DTS-MA HD stream seems to put the BC into the LFE channel.

I wonder if the file is incorrectly flagged as discrete but is actually matrix.

tebasuna51

9th September 2013, 08:49

tebasuna51

9th September 2013, 09:39

I went back and double checked... Libav and ArcSoft give me a real LFE channel with the core, not a full range channel at a low volume.
You are right, my first spectral analysis was wrong.

I wonder if the file is incorrectly flagged as discrete but is actually matrix.
Changing the Extended Coding Flag, in standard DTS header, to 0 eac3to show the file as:

DTS Master Audio, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48kHz)

DTS-ES, 5.1 is like eac3to show a 6.1 matrixed.

Decoding the DTS-MA the output is the same than before, but decoding the core the output is only 5.1 without the empty BC channel.

In order to obtain a -downDpl mix, like you want in your first post, I can only suggest extract the core and use BeHappy.

Or edit the core with a hex editor replacing all hex values:
7FFE8001FC3C7DB277001D
with
7FFE8001FC3C7DB277000D
and use eac3to.

jj666

9th September 2013, 11:14

Hi,
I have a problem when I try to extract video streams from "MONSTERS INC 3D".
I have this message one time for each video stream : [v02] The video bitstream framerate field doesn't seem to match the timestamps. <WARNING>
And several warnings like that : [v02] Video has a gap of 14 frames at playtime 0:00:00. <WARNING>
On playlist info, it is indicated "48p /1.001" but streams are 24p/1001.
I have also tried with the new TsMuxer and no problems.
Anyone can help me ?
Thanks !

If you wait until the end of the spam, it ends up with one extra frame in the MVC stream, which would mean it's not remuxable anyway (at least not in Scenarist, without manually editing the .VES file).

Video track 2 contains 132552 frames.
Video track 3 contains 132553 frames.

I noticed there are a lot of branched files on this disk, so perhaps this is the reason for the problem.

1) 00000.mpls, 1:32:09
[800+879+880+1002+1006+881+1008+1010+1014+1016+1020+1022+1026+1028+1032+883+1
034+1040+1042+1046+1048+1052+885+886+887+888+1054+1058+889+1060+1062+1066+891+89
2+908].m2ts

If Madshi is interested in checking this one out, have no issue to donate the original disk.

Cheers,

-jj-

laserfan

9th September 2013, 13:47

These posts from June 2011 (a different thread) inspires me to ask: does anyone know which dlls came from which Arcsoft TMT versions?

I have a couple dozen Setup files for TMTs including v2 thru v6, but I only have one version actually installed to a PC. Would be nice to know what dlls came with which vs. trying to install each and hunt them down...!

:eek:

EDIT: Here's one data point:

I added in my past post (http://forum.doom9.org/showthread.php?p=1508578#post1508578) information about 2.1 bug.

Also I ran eac3to with all 1.1.0.x versions of arcsoft decoder on lossy DTS files. As I understand eac3to patches streams to mark it 24 bit and to make all arcsoft versions output 24 bit. In this case outputs of all versions except 1.1.0.0 are identical.

So the final conclusion is following: the best version of dtsdecoderdll.dll is 1.1.0.1. It decodes all lossy streams to 24 bit, it doesn't have mono and 6.0 bugs, its decoding algorithm for lossy and lossless content wasn't changed in latter versions (except mono and 6.0 of course). The only problem of 1.1.0.1 (and all arcsoft versions) - silent LFE channel in 2.1 streams.

PS Version 1.1.0.1 came with the original release of TMT3, 3.0.1.120.

The elaboration posted earlier in this thread:

I tested 5 different versions of dtsdecoderdll.dll along with LAV Audio, here are results.

Versions tested: 1.1.0.0, .1, .5, .7, .8.

Samples tested:
1) Lossy DTS 2.0, 2.1, 5.1, 6.0, 6.1 configs, 16/48, 24/48 and 24/96 bitdepth/sample rate;
2) DTS-MA 1.0, 2.0, 2.1, 5.1, 6.1, 7.1 configs including 7.1 "strange setup" config (eac3to term), 16/48, 24/48, 24/96 variants.
My results differ from results that some people posted in eac3to thread, I guess there may be some eac3to specific problems.

lossy DTS
1) 1.1.0.0 and 1.1.0.1 always decode lossy DTS as 24 bit, and their decoding results differ from each other.
2) 1.1.0.5 and up decode in proper bitdepth.
3) Versions 1.1.0.5 and up decode lossy DTS identically.
4) 1.1.0.1 and 1.1.0.5 decode 24 bit lossy DTS identically.
5) Decoding of 16 bit lossy DTS was changed from 1.1.0.0 to 1.1.0.1 and from 1.1.0.1 to 1.1.0.5.
6) 1.1.0.7 and 1.1.0.8 decode 6.0 without back center channel.

The conclusion for lossy: I assume the best version for lossy DTS is 1.1.0.5. It decodes all configurations I've tested, in proper bitdepth, and decoding algorithm didn't change since this version (except 6.0 bug in .7 and .8).

DTS-MA
All versions output identical streams except mono case. Mono decoding was broken in 1.1.0.5 and partially fixed in 1.1.0.8 (outputs as stereo). Note that MA 7.1 16/48 "strange setup" is decoded as 24/48 by all versions.

The conclusion for lossless: the best versions are 1.1.0.0 and 1.1.0.1.

Of course, to be sure that decoding is done properly some "reference proper" decoder is needed. I don't have it.

EDIT: Forgot to mention about 2.1 bug. None of the versions can decode LFE channel in 2.1 files.

Nico8583

10th September 2013, 18:47

If you wait until the end of the spam, it ends up with one extra frame in the MVC stream, which would mean it's not remuxable anyway (at least not in Scenarist, without manually editing the .VES file).

Video track 2 contains 132552 frames.
Video track 3 contains 132553 frames.

I noticed there are a lot of branched files on this disk, so perhaps this is the reason for the problem.

1) 00000.mpls, 1:32:09
[800+879+880+1002+1006+881+1008+1010+1014+1016+1020+1022+1026+1028+1032+883+1
034+1040+1042+1046+1048+1052+885+886+887+888+1054+1058+889+1060+1062+1066+891+89
2+908].m2ts

If Madshi is interested in checking this one out, have no issue to donate the original disk.

Cheers,

-jj-
For me, streams contain the same number of frame...

mikeyakame

13th September 2013, 01:18

I've done a few tests with the reference DTS-HD decoder and it appears at least the version of dtsdecoder.dll from TMT 5.3 did somewhat decode a 5.1 ( L R C LFE Ls Rs ) HD-MA bit exact.

I say somewhat because Arcsoft's decoder padded the stream on output and the resulting length was around ~30-40ms longer than the original PCM audio which the Master Audio stream was encoded from.

To get the Arcsoft output to line up with the reference decoder output, I had to remove a bunch of empty samples before and after the bit exact region (With a trusty hex editor no less). Just for reference the decoded channels from the DTS-HD reference decoder (it only decodes to mono wav files) were bit for bit exact with the mono wavs I originally fed the encoder.

Sparktank

13th September 2013, 03:07

TMT 5.3

What version of the dtsdecoder.dll is that?

filler56789

13th September 2013, 06:07

mikeyakame

13th September 2013, 18:58

Looked into it just now, and the reason the Arcsoft Decoder is outputting "junk" is because this junk is supposed to be discarded when decoding back to pcm. Going off what I've learnt, I believe eac3to needs to be responsible for this step.

From the start of the DTS-HD file.

CORESSMD HEADER + 0x20

0x30 + 3 bytes = Core Sample Rate Hz
0x33 + 2 bytes = Core Bitrate Kbps
0x35 + 4 bytes = Channel Mask
0x39 + 2 bytes = Frame Payload in Bytes

EXTSS_MD HEADER + 0x3C

0x4C + 3 bytes = Extended Avg Bitrate Kbps
0x4F + 3 bytes = Extended Peak Bitrate Kbps
0x52 + 2 bytes = Peak Bitrate Smoothing Buffer Size Kb

AUPR-HDR + 0x54.

0x67 + 3 bytes = Source Sample Rate
0x6A + 4 bytes = DTS Frame Count
0x6E + 2 bytes = DTS Frame Size in Samples
0x70 + 1 byte = Appears to mean Codec Delay Samples [ delay_samples / frame_size frames ] are included in the DTS Frame Count when its value is 0.
0x71 + 4 bytes = Source Sample Count (Encoded from PCM)
0x75 + 2 bytes = Bitmask? [ 0x02 = Ext is Lossless, 0x01 is probably used for DTS Express where the Ext Stream is lossy and CBR ]
0x77 + 2 bytes = Codec Delay in Samples

That's pretty much all the data you need to calculate how many samples of the decoded PCM you need to discard from the beginning and the end.

I'll use my working file as an example.

Source Sample Count = 68154486
Arcsoft Decoded Sample Count = 68155904

Difference between the source PCM and the Arcsoft decoded PCM is 1418 samples. In this case it amounted to 2836 bytes for a single channel.

1418 samples * 2 byte depth = 2836 bytes

I binary compared the source and the decoded output and determined that there was 2048 bytes extra at the beginning, and 788 bytes extra at the end.

So now to our DTS-HD header values.

DTS Encoded Frames = 133117
DTS Frame Size = 512 samples
Codec Delay = 1024 samples

Source Sample Count = 68154486

Codec delay should be time allowed to index the audio and build the frame navigation table and what not, so it comes first.

codec_delay / dts_frame_size = 2 frames

source_samples / dts_frame_size = 133114 full frames + 118/512 frame

2 ( 512 codec_delay_samples ) + 133114 ( 512 source_samples ) + 1 ( 118/512 source_samples ) = 113117

The header says the encoded DTS-HD file has 113117 frames, so we got our maths right.

Back to our decoded differences.
We have +2048 bytes at the beginning.

Codec Delay = 1024 samples * 2 bytes depth = 2048 bytes.

We discard the first 2048 bytes of each channel because they are padding and not from the source pcm.

Now we have +788 bytes left at the end.

788 bytes / 2 bytes depth = 394 samples

These samples were not from our source PCM, so we must discard them too.

So, why 394 exactly?

1 full DTS Frame = 512 samples
Last DTS Frame = 118 samples

512 - 118 = 394

And that is where the mysterious 394 samples at the end appeared from.

Arcsoft's DTS decoder appears to decode all padding into the PCM output instead of discarding it like the Reference DTS-HD decoder does.

eac3to just needs to calculate if there are padded frames, and discard these when writing out the pcm data or after the fact.

Hope this helps, was kinda fun to figure out.

hdtv00

14th September 2013, 05:35

You guys way deep into this and I don't know what happened or if I'd just been lucky.

I rip my concert blu rays using eac3to,hdbrstreamextractor and playback using waspi. My older rips all seem to be the right 24 bit 96khz but last two I've tried have been down sampled for 48khz. New Bryan Adams and Bee Gee's. Did something break in my setup. Has DTS Master audio never ripped correctly. I see it trying the different decoders, it settles on libav, say arcsoft isn't doing it yet i have it installed.

mikeyakame

14th September 2013, 07:31

Well checked one of my Blu-rays and demuxed the DTS-HD stream, it appears the header and the navigation table are both stripped when muxed into m2ts.

In that case I'll do some more tests just to confirm whether the codec_delay is fixed at 2 frames and whether the frame_size is fixed at 512 samples (which they most likely are, but want to be 100% sure.) If that is the case, the first 1024 samples could probably be safely discarded. The only time I can see the partial frame and codec_delay becoming a problem is when seemless branching is used. Not too sure how this is handled.

tebasuna51

14th September 2013, 11:08

Well checked one of my Blu-rays and demuxed the DTS-HD stream, it appears the header and the navigation table are both stripped when muxed into m2ts.
Yes, that is know. The main header than encoder (like DTS MA Master Suite) put at the begining is a help to BD Authoring soft, and the muxer can transfer the data to container header but the stream itself can't contain a main header, each DTS frame have a header.

I think the muxer must also strip the delay frames because I never see DTS tracks with delay in BD's.

I can confirm than eac3to ignore this main header and the frame delay is not applied, then the output have a initial delay.

The problem don't exist with extracted DTS from any container and is not related with seemless branching.

The sample with I work have also 2 delay frames and framesize 512 samples.

I don't know if encoders always put 2 frames delay. Seems DTS MA Master Suite yes.

The frame size can have different values but 512 is the standard, I never see other.
BTW, the frame size is stored in each DTS frame header and is know without the main header.

Q-the-STORM

15th September 2013, 10:56

just got this after demuxing a blu-ray

[v02] Waiting for DirectShow decoder thread to finish. Please wait...

only posts I found containing this got the demux process "Aborted at file position XXX", not here though, no error messages, everything looks the same, with the exception of that line...

this is first time I saw it, seems to be straight forward to what it means, so I'm not really worried, just wanted to know if anyone else got this before, without any errors in the files....

tebasuna51

15th September 2013, 18:58

@Q-the-STORM
I never see that.
Please put the full log, to know, at least, the video track than need a "DirectShow decoder".

Q-the-STORM

15th September 2013, 19:52

I just deleted the log a minute ago...

but there was nothing special there... it was an episode of a TV show, all other episodes didn't have it, only one of them had the line, everything else was the same, so there's really no info to get from the log...

maxmercy

16th September 2013, 00:22

I have a question:

I have run into this issue on multiple occasions, where the LFE channel on a TrueHD track is severely clipped, but its AC3 counterpart is not.

Please see the attached pic, the top track is the AC3 LFE channel, and the bottom track is the TrueHD version.

I decoded to .wav for analysis with the libav decoder.

Are there any known issues with the libav decoder that would cause this?

JSS

kevmitch

18th September 2013, 11:16

...the DTS HD MA sound from the Obilivion blu ray. In short, the center channel becomes distorted in loud bits.

Sorry I seem to be late in checking back on this. Hopefully the initial responders are still watching this thread.

There really two possibilities here:

1) The soundtrack is badly mastered with hard limiting/compression or just plain clipping.

2) The arcsoft decoder is producing incorrect output from the .dtshd input.

The main thing I want to know is wether or not someone can reproduce the distortion with a licensed decoder (either hardware or software). This should be compared with my FLAC samples to make sure that you know what you're looking for. If you can confirm that the FLACS have the distortion, but the licenced decode of either the .dtshd file (or preferrably the original blu-ray) does not, then we have a decoding rather than mastering problem. Proving a true mastering problem may be difficult if multiple licenced decoders have a similar bug to arcsoft (maybe firmware upgrades are necessary?).

Have you tried the Arcsoft decoder 1.1.0.9 ?

Does anyone finally know what improvements or regressions version 1.1.0.9 brings?

It produces identical output sample for sample in this case.

BTW, as you are saying the tiny distortion you are getting in your sample is in the original track as well. But, starting at the same time up to 20 seconds later (not in your sample) there often is A LOT of sound clipping (and in other parts of the movie too).

I'm not hearing any problems here.

So, if PowerDVD plays the (Arcsoft decoded) FLAC fine but 2 different ffmpeg players on 2 different systems don't, ffmpeg is the issue.
So, as far as I can see, this is a bug in ffmpeg.

Once you've got the FLAC, it should either work on all players or none at all since it is lossess. Every flac decoder should produce identical output, which is not a hard problem since the spec is completely open. It's possible a given decoder has a bug, but I have not found any evidence for this. I also have the bare minimum of cruft between my decoder and speakers. No effects or anything. I hear the distortion in windows (vlc,winamp,MPC-HD). But I also get the same results in Linux (audacity, audacious, mplayer, vlc) with direct ALSA output.

I tried listening to the FLAC center channel samples posted above, and i couldn't really hear any obvious distortions, but maybe i just dont know what to listen for (or my player is fixed)

Between 2.5s and 3.0s in the above sample (8:15 in the film) is where it distorts. It's easy to miss because there's a lot of sound going on there. It presents as a high frequency scratching/crackling/rapid popping. I can hear it on numerous different audio/decoder setups, but it is most apparent with my etymotic in-ear earphones which have a very flat frequency response. Try using an equaliser to boost the high frequencies and it's unmistakable.

Even opening in Audacity it didn't show any clipping (marked in red). It does show that it's cut off at .8 ("The vertical scale displays amplitude when showing the waveform" from Audacity Help)

I also looked at the waveforms in audacity and visually saw the same "hard limiting" effect, which indeed seems to suggest bad mastering. However, I can't yet rule out the possibility that it's bad decoding - maybe arcsoft is mistakenly clipping at 0.8? The fact that the MA and core differ in amplitude (see next quote) gives me a bad feeling.

4) But there are a problem that I never see. The lossless output from DTS-MA decoded with ArcSoft is 3dB loud than the decoded core (ArcSoft or libav).
Not only on clip parts but over the full sample.

Interesting, I can confirm this!

1.) Oblivion ripped w/o errors, so whatever problems the disc might exhibit are not due to disc defects,

In both the bad mastering and bad decode scenarios, there would be no errors ripping from the disc. With bad mastering, a bad signal is correctly encoded to the disc. This would produce no errors except playing back the original bad signal. With bad decoding, the error hasn't happened until after ripping.

2.) whatever problems one might have with the sound are due to the player/playback software, cuz MPC-HC plays it perfectly.

I can still hear the distortion when I try the .dtshd file with MPC-HC. Or rather I had to mux the .dtshd to an .mka because MPC doesn't handle .dtshd directly.

I have a question:

I have run into this issue on multiple occasions, where the LFE channel on a TrueHD track is severely clipped, but its AC3 counterpart is not.

Please see the attached pic, the top track is the AC3 LFE channel, and the bottom track is the TrueHD version.

I decoded to .wav for analysis with the libav decoder.

Are there any known issues with the libav decoder that would cause this?

JSS
The following may be of interest

4) But there are a problem that I never see. The lossless output from DTS-MA decoded with ArcSoft is 3dB loud than the decoded core (ArcSoft or libav).
Not only on clip parts but over the full sample.

maxmercy

18th September 2013, 21:13

The following may be of interest
Quote:
Originally Posted by tebasuna51 View Post
4) But there are a problem that I never see. The lossless output from DTS-MA decoded with ArcSoft is 3dB loud than the decoded core (ArcSoft or libav).
Not only on clip parts but over the full sample.

I see the problem with both DTS-HDMA and DolbyTrueHD, and mainly in the LFE channel. Curiously, Here are the three tracks, Top is AC3, middle is THD with 120Hz steep lowpass and then normalized, and last is THD untouched. Curious...

JSS

mp3dom

19th September 2013, 09:27

Is it possible to change the filename convention for the surround tracks as Ls/Rs instead of SL/SR (at least for 5.1)
A lot of professional tools uses this naming convention (L/R/C/Lf (or Lfe)/Ls/Rs). For example, encoders like dts-HD MAS searches for that naming convention so when you feed the first channel (usually the left) to the encoder, it takes all the other channels automatically (this also avoid possible errors while passing manually the files).
Tnx.

kevmitch

20th September 2013, 11:08

I see the problem with both DTS-HDMA and DolbyTrueHD, and mainly in the LFE channel. Curiously, Here are the three tracks, Top is AC3, middle is THD with 120Hz steep lowpass and then normalized, and last is THD untouched. Curious...

JSS

Using my DTSHD-MA 7.1 channels, 24 bits, 48kHz DTSHD-MA sample (https://dl.dropboxusercontent.com/u/60598588/oblivion_distort.zip) from above (http://forum.doom9.orgshowthread.php?p=1640665#post1640665), I find that versions 1.1.0.0, 1.1.0.7, 1.1.0.8,1.1.0.9 all produce the same "lossless" decode as I mentioned previously

However, when I pull out the -core (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz), using all four versions listed above, they are all quieter than the lossless decode by approximately the same ammount. The most obvious sign is that the signal is clipped at 0.6 rather than 0.8. But it is also quiter in non-clipped parts as pointed out by tebasuna51 (http://forum.doom9.org/showthread.php?p=1641423#post1641423).

On top of this difference in amplitude between the lossy and lossless, the 1.1.0.0 decoder produces different core (lossy) output from 1.1.0.7, 1.1.0.8 and 1.1.0.9. The latter three produce identical lossy results.

It appears that all versions of the arcsoft decoder for DTS as well as the libav decoder for truehd (as you've shown) exhibit amplitude discrepancies between lossy and lossless output. That's two distinct decoders for two distinct codecs which is kind of spooky. Maybe it's "supposed" to be like that for some reason.

kevmitch

20th September 2013, 11:17

I should add that I've also tried -keepDialnorm, on both the lossless and lossy decodes, which produces identical files.

maxmercy

21st September 2013, 13:12

Using my DTSHD-MA 7.1 channels, 24 bits, 48kHz DTSHD-MA sample (https://dl.dropboxusercontent.com/u/60598588/oblivion_distort.zip) from above (http://forum.doom9.orgshowthread.php?p=1640665#post1640665), I find that versions 1.1.0.0, 1.1.0.7, 1.1.0.8,1.1.0.9 all produce the same "lossless" decode as I mentioned previously

However, when I pull out the -core (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz), using all four versions listed above, they are all quieter than the lossless decode by approximately the same ammount. The most obvious sign is that the signal is clipped at 0.6 rather than 0.8. But it is also quiter in non-clipped parts as pointed out by tebasuna51 (http://forum.doom9.org/showthread.php?p=1641423#post1641423).

On top of this difference in amplitude between the lossy and lossless, the 1.1.0.0 decoder produces different core (lossy) output from 1.1.0.7, 1.1.0.8 and 1.1.0.9. The latter three produce identical lossy results.

It appears that all versions of the arcsoft decoder for DTS as well as the libav decoder for truehd (as you've shown) exhibit amplitude discrepancies between lossy and lossless output. That's two distinct decoders for two distinct codecs which is kind of spooky. Maybe it's "supposed" to be like that for some reason.

Or maybe a way to phase out DD/DTS Lossy, as the Lossless tracks will be perceived to be 'better' due to higher levels....who knows. There are many DTSHD and DTHD decodes I have analyzed that do not clip the LFE channel like above, but some tracks do...maybe a mistake in post-production.

JSS

kevmitch

22nd September 2013, 04:03

Or maybe a way to phase out DD/DTS Lossy, as the Lossless tracks will be perceived to be 'better' due to higher levels....who knows. There are many DTSHD and DTHD decodes I have analyzed that do not clip the LFE channel like above, but some tracks do...maybe a mistake in post-production.

JSS
LOL. The sad thing is you're probably right.

DoctorM

28th September 2013, 23:21

A question. Applying slowdown to a PAL Dolby audio track as such:
eac3to "AudioFile_80.ac3" "NTSC.wav" -down16 -slowdown -25.000
AC3, 2.0 channels, 1:24:57, 192kbps, 48kHz, dialnorm: -27dB
Removing AC3 dialog normalization...
Decoding with libav/ffmpeg...
Changing FPS from 25.000 to 23.976...
Reducing depth from 64 to 16 bits...
Writing WAV...
Creating file "NTSC.wav"...
eac3to processing took 15 minutes, 4 seconds.
Done.

I then tried to load the resulting .wav into a better quality Dolby encoder, except it wouldn't load.
I loaded the .wav into Sony Sound Forge and resaving (making no changes). That file loads fine.

Original File:
Audio
ID : 0
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Signed
Codec ID : 00001000-0000-0100-8000-00AA00389B71
Duration : 1h 28mn
Bit rate mode : Constant
Bit rate : 1 536 Kbps
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Stream size : 973 MiB (100%)

Resaved file:
Audio
ID : 0
Format : PCM
Format settings, Endianness : Little
Codec ID : 1
Duration : 1h 28mn
Bit rate mode : Constant
Bit rate : 1 536 Kbps
Channel(s) : 2 channels
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Stream size : 973 MiB (100%)

So where is the problem coming from? Is there something I should be changing on the eac3to command line to make it more compatible?

Groucho2004

28th September 2013, 23:40

So where is the problem coming from?
Possibly the extensible WAVE header that eac3to writes. Try adding the switch "-simple".

DoctorM

29th September 2013, 06:43

Bingo! Thanks.

Btw, has there been talk about doing something to allow eac3to to remove the dialog normalization during transcoding and then if the output is ac3 embed a normalization again?
Basically allow the source and encode to have the same normalization when finished?

tebasuna51

29th September 2013, 10:28

Basically allow the source and encode to have the same normalization when finished?

Apply the same Dialog Normalization than source when recode is your choice.

If you only play Dolby Digital files can be a good idea, but if you play different files (DTS, MP3, AAC, FLAC, ...) maybe is better put Dialog Normalization to -31 to avoid low volume.

DoctorM

29th September 2013, 18:55

Is there a switch to add a normalization to AC3 encodings? I haven't seen that. I don't mean -keepDialnorm (which alters the original audio), I mean something like -DialNorm 27.

tebasuna51

29th September 2013, 20:38

You need use the external Aften.exe encoder:

eac3to input stdout.wav | Aften -b 640 -readtoeof 1 -dnorm 27 - output.ac3

DoctorM

30th September 2013, 03:18

At that point it's just as easy to decode to the AC3 to WAV and then use a separate encoder.
Didn't know aften supported dnorm though.

Groucho2004

30th September 2013, 14:57

Is there a switch to add a normalization to AC3 encodings? I haven't seen that. I don't mean -keepDialnorm (which alters the original audio), I mean something like -DialNorm 27.
This (http://forum.doom9.org/showthread.php?t=56020) is a good (and necessary) read in case you're planning on adding dialog normalization and/or DRC.

laserfan

30th September 2013, 16:44

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

I must admit that the more I read about this, and the more I surf on it, the less I understand about it.

:o

I have always left eac3to alone, which of course means to remove dialnorm, but occasionally I notice that when I make backups of BD discs and keep the "director commentary" track, these always blow you out of your seat it seems, because they sound very much louder than the base program. Is dialnorm related to this in any way? Should I be keeping dialnorm for these tracks?

It seems Dialog Normalization must have a purpose or it wouldn't be in the standard--when is it needed??? Who here can explain this so mere mortals can understand it?

:confused:

DoctorM

30th September 2013, 17:51

The simple answer: An audio track is at whatever volume. The dialog normalization number tells your audio decoder that it is loud or soft by so much and that it must adjust the volume to keep it at a standard level.

In theory it should make all audio tracks sound roughly the same volume. Without it most audio tracks will be louder than they should be.

While I wouldn't want to use -keepDialnorm since that physically adjusts the track's volume when decoding, reapplying a dialog normalization number to an ac3 track after re-encoding make sense.

I always use a true Dolby encoder after Eac3to does its thing for this reason.
You could also try a program like VOBDNorm to add on a dial. norm. after the fact.

laserfan

30th September 2013, 19:23

I wouldn't want to use -keepDialnorm since that physically adjusts the track's volume when decoding, reapplying a dialog normalization number to an ac3 track after re-encoding make sense.

If Dialog Normalization is authored into AC3 tracks on BD discs, presumably by a smart sound guy in the studio's Authoring facility, to tell a player's/amp's DD decoder to adjust [the Center channel?] up-or-down, what is the reason for stripping it out when doing these demux/remux processes? Why would you want to strip it out and then reapply; what do you know that the original sound guy doesn't?

Sorry if these are dumb questions--I'm REALLY not getting this. :(

DoctorM

30th September 2013, 21:12

Dial. norm. adjust the global volume, not just the center. It is intended to do so in a manner that makes the dialog level a specific volume though.

In theory using -keepDialNorm would be more accurate if you're going back to AC3 (whereas encodings tend to boost the volume to the maximum it can reach without clipping, so the D.N. would be undesirable.)
It's more(?) lossy to reduce the volume of an audio track while decoding and then re-encoding with no dialog normilzation level (technically -31db).

Subtle detail may be lost, especially since AC3 tracks tend to already be mixed at a lower level (peaking 80% or less of maximum).

Anyone who has listened to an AC3 track and then a CD or mp3 will have noticed this. Honestly, there shouldn't be a difference, but mixing CD's at 100% peak is a throw back to analog recordings.

laserfan

30th September 2013, 22:50

Thank you DoctorM, I think I am getting closer. At least, it seems clear(er) to me now that dialnorm is an attempt to include in the recording some information that is useful if you are trying to maintain a reference sound level for the dialogue in a program, in order to have some consistency when listening to multiple programs and maintaining the same setting on the amplifier. Which is not the case (for me at least) in racking-up a BD and watching a movie or concert film (I adjust volume up or down as needed to the program).

The only circumstance that I can see dialnorm as beneficial then might be if one were to make a compilation disc of favorite movie scenes or song/concert performances, so as to (try to) maintain that consistency--though I'd still imagine there'd be wide differences at least with concert film clips.

Anyway thanks for replying to me, I appreciate it. :)

DarkSpace

30th September 2013, 23:37

tebasuna51

1st October 2013, 10:57

Like idea use DialNorm is good, the problem is when Dolby Digital tracks must compete with other sources like DTS (can have a DialNorm but is not used most the times), commercials in TV, CD audio (Loudness war (http://en.wikipedia.org/wiki/Loudness_war))...

There are many users than ask for more volume in DD tracks.
A standard value for DialNorm in DD tracks is -27, that mean than a DD decoder must attenuate all sound by 4 dB.
Many movies have DTS and DD tracks with different volume because that.

Not problem with DD tracks in different languages when the DialNorm is changed from -27 to -31 in all, maybe a commentary track can have a different DialNorm, check this before.

BTW Aften (included in eac3to) don't make DD compliant tracks, is only a A/52 (AC-3) audio encoder.
Who want a DD track please read the post recommended by Groucho2004 and use a commercial DD compliant encoder.

@DoctorM
To make a DD track after the slowdown you can use a DD encoder with the same DialNorm (and DRC) than source without problems.

laserfan

1st October 2013, 13:48

I think the main purpose of DialNorm is to achieve a consistent volume for multiple audio tracks on the same disc. Say, you've got a disc with audio in the movie's original language and your local language, then DialNorm is supposed to make switching between local and original audio possible without having to readjust volume levels on your playback equipment.

This makes a lot of sense, but it also argues for keeping dialnorm when using eac3to to demux-then-remux a disc with e.g. Director's Commentary track(s) as I do.

Now that I think of it, BD-RB uses tsMuxeR to demux and I imagine that tsMuxeR doesn't touch the dialnorm metadata.

Guess I have to re-think my processing methods. Recently I noticed the Dir Commentary audiotrack on a disc I had to re-do (for subtitle issues) was very much louder than the main audio so I used eac3to (and Aften I think) to lower it by some 6dB. Maybe if I'd kept dialnorm for these this would not have been necessary? :o

Q-the-STORM

19th October 2013, 19:56

I'm wondering... since FLAC was updated to 1.3.0 and 6.1/7.1 channel masks from input WAV files were fixed, shouldn't there be a an update of eac3to with a new libFLAC?

mastrboy

19th October 2013, 21:24

You can update it yourself by updating the libFLAC.dll file with a newer one.

sneaker_ger

19th October 2013, 22:27

Quoting madshi:
FLAC 1.3.0 has just officially sanctioned what eac3to has been doing all along. So there's nothing for me to change.

sl1pkn07

20th October 2013, 20:08

eac3to mkv muxer in linux (with wine) still don't work (yes, I have installed haali and -test says correct)

any help for why happen this?

greetings

tebasuna51

22nd October 2013, 13:39

53 posts moved to new thread How to run eac3to (http://forum.doom9.org/showthread.php?t=169530)

mindbomb

22nd October 2013, 22:52

i just noticed you can do audio delays in eac3to.
if im making an mkv, would it make more sense to delay the audio directly with eac3to or is it better to do it with mkvmerge?

the_weirdo

23rd October 2013, 06:58

Maybe you should delay the audio directly with eac3to if possible. Standalone players and some splitters/decoders may not hornor the delay information in Matroska. But it's just my opinion and I'm not an expert on this.

LigH

23rd October 2013, 08:09

If you do a (possibly lossy) conversion with encoding, like dts to AC3, it is certainly recommendable to handle the delay already during that conversion, to get a perfectly synced result.

But if you just extract core audio, without actually encoding (e.g. EAC3 to AC3, DTS-HD to Core dts), I would suggest to just keep the delay.

tebasuna51

23rd October 2013, 12:18

...
But if you just extract core audio, without actually encoding (e.g. EAC3 to AC3, DTS-HD to Core dts), I would suggest to just keep the delay.

I think you want say "TrueHD to AC3", because EAC3 to AC3 is a recode.

BTW, I always apply the delay even in extractions. Is your choice.

nevcairiel

23rd October 2013, 12:22

Technically, from Blu-ray EAC3 tracks, you can also extract a "core" AC3. It uses a concept of dependent frames, which means the 5.1 core is coded in basic AC3, and there are 4 channels coded as E-AC3 which replace the downmixed back channels in the 5.1 and add 2 more channels for full 7.1

LigH

23rd October 2013, 12:47

You probably know this part better than me, I have so little material to be converted...

Also it may depend on the technique used to fix the delay, and the sign. Omitting preceding audio frames is simple; prepending "known silence blocks" only works as long as they are really well known.

tebasuna51

23rd October 2013, 16:54

@nevcairiel
You are right, I forget this option added in v3.25:

v3.25
...
* added support for demuxing Blu-Ray primary E-AC3 tracks (AC3 core)

But BD's tracks TrueHD are more common than EAC3.

Q-the-STORM

24th October 2013, 00:34

Quoting madshi:FLAC 1.3.0 has just officially sanctioned what eac3to has been doing all along. So there's nothing for me to change.

thanks, good to know ^^

But if you just extract core audio, without actually encoding (e.g. EAC3 to AC3, DTS-HD to Core dts), I would suggest to just keep the delay.

why? eac3to just adds silent audioframes, it doesn't reencode the actual audiotrack, so it doesn't really matter if you're reencoding or not...

LigH

24th October 2013, 12:28

why? eac3to just adds silent audioframes, it doesn't reencode the actual audiotrack, so it doesn't really matter if you're reencoding or not...

If this feature is reliable, then I will revert my previous opinion.

People shall know, though, that audio frames usually have a different duration than video frame durations for all usual frame rates. Losslessly adding or removing whole audio frames at the beginning will probably not "fix" any arbitrary delay to the millisecond. Just to the next closest frame. Certainly below recognition threshold, though...

madshi

25th October 2013, 21:16

FWIW, I might get around to making a new build some time in the next couple of weeks. So if any bugs are known but not listed in the bug tracker yet, please someone add them to the tracker:

http://eac3to.bugs.madshi.net

Thanks. Just to make sure: The bug tracker is currently only open for bugs, not for feature requests.

sl1pkn07

25th October 2013, 21:31

madashi.

make work muxing mkv with eac3to in linux/wine is a bug (http://sl1pkn07.no-ip.com/paste/view/cec84682) or is a feature?

madshi

25th October 2013, 23:14

A feature.

therube

26th October 2013, 20:24

Hi. I've been using eac3to for a short bit of time now. Nice little program.
(I have not /yet/ read through this, eh, short, thread ;-).)

Most of what I've done so far are transcodes, with the final output to aac (m4a), low bitrates, 32-64 range, perhaps with an added delay (-###ms).

As far as I know Nero (neroAacEnc.exe v1.5.4) is the only AAC encoder you're supporting?
(I do not have Nero 7, or any "Nero" other then the command-line encoder.)

Reason I ask, my issue, is that I get "clipping", perhaps better described, when I jump through a clip, I'll typically hear a chirp (a pop) when I jump.
And I seem to recall a post, somewhere, or a wiki, that mentioned, something to that effect?

I do not get these artifacts if I use a higher bitrate (-quality=).

So, is it possible a wrong parameter, or conflicting parameter is being passed to Nero causing this?
Or can another aac encoder be used, fdkaac or whatever?

hubblec4

26th October 2013, 23:36

hi madshi

a short question.

Is it necessary that eac3to so deep/long scan a file(m2ts) or a BD-Structure when i want to demux only the chapters?

eac3to.exe "H:" 1) 1:chapter.txt

tebasuna51

27th October 2013, 09:52

...
Most of what I've done so far are transcodes, with the final output to aac (m4a), low bitrates, 32-64 range, perhaps with an added delay (-###ms)...

Maybe NeroAacEnc isn't the recommended encoder for these low bitrates. You can use other encoder than accept STDIN like input, for instance qaac, using the STDOUT of eac3to with this sintax:

eac3to input stdout.wav [eac3to-parameters] | qaac [qaac-parameters] -o output.m4a -

tebasuna51

29th October 2013, 09:20

10 posts about Clown BD moved to new thread Question about ClownBD v0.81 (http://forum.doom9.org/showthread.php?t=169646)

Richard1485

4th November 2013, 00:53

I didn't see this new option in 3.25 until now.

-downStereo

Does this mix the lfe channel or not?

a=wavsource()
flr = GetChannel(a, 1, 2)
fcc = GetChannel(a, 3, 3)
lrc = MixAudio(flr, fcc, 0.3694, 0.2612)
blr = GetChannel(a, 5, 6)
MixAudio(lrc, blr, 1.0, 0.3694)

If not and it's the same downmix as above, I would to thank madshi very much for adding it.

tebasuna51

4th November 2013, 02:19

@Jeff B
To mix lfe use also -mixlfe

See the downmix matrix here: http://forum.doom9.org/showthread.php?p=1600695#post1600695

With automatic normalization if mix go over 0 dB.

Richard1485

4th November 2013, 02:36

Thanks, tebasuna. Yes. I know how to mix lfe. I was just making sure that it wasn't included because I don't like it.

It seems that the stereo downmix is different from the custom one that you made for me for Behappy. I'll see how it sounds. :-)

EDIT: Automatic normalization is good.

LigH

4th November 2013, 09:13

Depends on the technique. Only 2-pass normalization can preserve the volume distribution over the whole play time. If it is done in one pass (like the "Automatic Gain Control" of "ancient" tape recorders), the loudness will change often.

tebasuna51

4th November 2013, 13:52

Of course is the 2-pass method.

phate89

7th November 2013, 03:52

Hi.. I'm using a lot lately eac3to to check recorded ts with transmission errors to detect where and how big are the errors. Even if the gap processing doesn't return a video and audio in sync (i actually don't care because i only need audio) i'm wondering how eac3to fill the gaps. It duplicates the frame near or simply insert a silence? And is it possible with some switch to avoid to create the gaps (that i don't need)?

Wile_E_Coyote

8th November 2013, 19:26

I've seen notes about adding -mixlfe which I usually do when I'm downmixing (-downDpl) but is there a way to keep the LFE channel when going from DTS to AC3 and just mixing the rears into the fronts?
The layout of the room I'm watching TV in isn't ideal (they rarely are) so I have to roll out the cables to connect the rear speakers of my 5.1 system and thought I'd see what just the front speakers plus my sub would sound like.
I invariably get the "clipping detected" message and second pass so wondered if this would help that too?

Richard1485

8th November 2013, 22:02

I've seen notes about adding -mixlfe which I usually do when I'm downmixing (-downDpl) but is there a way to keep the LFE channel when going from DTS to AC3 and just mixing the rears into the fronts?

You could run your 5.1 through Eac3to once to downmix.

Eac3to input.dts output.wavs -downDpl

Then run it through again to split it into individual wavs.

Eac3to input.dts output2.wavs

Be careful to give the output files from the second run a different name. After that, you could take the two wavs from the first run and the LFE from the second run and use WAV to AC3 encoder (http://www.videohelp.com/tools/EncWAVtoAC3) to make your 2.1. I always use the MUX Wizard to map my channels.

I don't see why this wouldn't work, but no doubt Tebasuna will correct me if I have overlooked something.

tebasuna51

9th November 2013, 03:26

@Wile_E_Coyote
I can't understand for what you want a downmix 5.1 -> 2.1, but you can test the Jeff B method.

When downmix the "clipping detected" message is normal (try add 5-6 channels in only 2) and means than automatic normalization works fine.

wiggaz

11th November 2013, 03:05

Hi everyone,
I have a problem with eac3to.
I'm trying to convert some dtshd files into AC3. I've done this a lot of times even between those files I'm talking about.
Here what I've done:
I've extracted those dtshd files and they are like this:
DTS Master Audio, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
Now, to convert them into AC3 I'm using this:
%_.ac3 -normalize
Well, after the 1st pass, eac3to get stuck at the .wav file and nothing happened then.
Here's the log:
eac3to v3.27
command line: "C:\Program Files (x86)\eac3to 3.27\eac3to.exe" "D:\audio.dtshd" "D:\audio.dtshd_.ac3"
-normalize -progressnumbers -log="C:\Program Files (x86)\eac3to 3.27\UsEac3To.log"
------------------------------------------------------------------------------
DTS Master Audio, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
Decoding with ArcSoft DTS Decoder...
Writing WAV...
Creating file "D:\audio.dtshd_.ac3.pass1.wav"...
What can I do? What do you suggest?
I've already tried without the "-normalize" and nothing changed.
Thanks in advance.

Stereodude

11th November 2013, 03:45

Maybe NeroAacEnc isn't the recommended encoder for these low bitrates. You can use other encoder than accept STDIN like input, for instance qaac, using the STDOUT of eac3to with this sintax:

eac3to input stdout.wav [eac3to-parameters] | qaac [qaac-parameters] -o output.m4a -For whatever it's worth, I found piping from eac3to to qaac to be extremely slow. CPU usage of each process (qaac and eac3to) were in the single digits (~6-9%) on my Q6600, but eac3to writing to a wave file has no problem saturating a core (25%) with the same decoding settings. For me it's a lot faster to go to an intermediate wave file and feed that to qaac.

shinchiro

12th November 2013, 12:24

Does eac3to support stdin?

Richard1485

12th November 2013, 21:42

"C:\Program Files (x86)\eac3to 3.27\eac3to.exe" "D:\audio.dtshd" "D:\audio.dtshd_.ac3

I've found that Eac3to sometimes does not like multiple stops in output file-names. Try this.

eac3to "D:\audio.dtshd" "D:\audio.ac3"

tebasuna51

13th November 2013, 10:30

I'm trying to convert some dtshd files into AC3. I've done this a lot of times even between those files I'm talking about.
...
Well, after the 1st pass, eac3to get stuck at the .wav file and nothing happened then.

If work with some dtshd but not with other maybe the last one is corrupt. (I supose you have enough disk space for the wav file)

Does eac3to support stdin?

Nope.

manolito

23rd November 2013, 12:06

Sorry if this has been asked before - but this thread is really long...

Last time I looked eac3to could not handle AAC audio files, probably because I do not have Nero 7 installed. Since I have no intention to install Nero 7 I would like to ask if current versions of eac3to can handle AAC sources by using other decoders like ffdshow or faad.

Cheers
manolito

LigH

23rd November 2013, 12:14

From the help screen:

eac3to v3.27, freeware by madshi.net
...

Supported source formats:
...
(6) AAC audio

So with the help of the included libavcodec, probably yes.

the_weirdo

23rd November 2013, 12:22

Actually, eac3to still need "Nero Audio Decoder 2" (from Nero 7) to decode AAC. Currently, it can't use any other DirectShow decoders or libavcodec to do that.

Lenmaer

23rd November 2013, 13:08

I was wondering if it was possible in the future to have eac3to read avisynth scripts?
Tha would be great to make audio trims using eac3to.

therube

23rd November 2013, 14:21

eac3to could not handle AAC audio files, probably because I do not have Nero 7 installed
You do not need Nero 7.
All you need is Nero AAC Codec, Free.
http://www.nero.com/eng/company/about-nero/nero-aac-codec.php

Download (~2 MB), unzip & drop the .exe's into your eac3to directory.

manolito

23rd November 2013, 14:53

You do not need Nero 7.
All you need is Nero AAC Codec, Free.
http://www.nero.com/eng/company/about-nero/nero-aac-codec.php

Download (~2 MB), unzip & drop the .exe's into your eac3to directory.

Sorry, you are posting incorrect information...:mad:

Of course I have tried your recommendation before I posted my question here, and it sure DOES NOT WORK.
eac3to v3.27
command line: eac3to i:\test.mp4 i:\testac3
------------------------------------------------------------------------------
The format of the source file could not be detected. <ERROR>

So please, next time you post, please verify that your information is correct...:p

Cheers
manolito

tebasuna51

23rd November 2013, 17:32

eac3to never support mp4 container like input.
Also .m4a (only aac audio in mp4 container) is not supported.

.aac input (with ADTS headers for instance) is only supported with Nero 7 installed.

LigH

23rd November 2013, 19:02

:o Sorry for a wrong hope then.

therube

24th November 2013, 14:56

Something like this certainly does work, with eac3to & nero encoder:

eac3to.exe INPUT OUTPUT OPTIONS DELAY
eac3to.exe riot_INPUT.wav riot_OUTPUT.w.18-1350ms.speedup.m4a -quality=0.18 -speedup -1350ms

eac3to.exe SN.ac3 sn+2300.m4a +2300ms -quality=0.18

eac3to.exe ie.wav ie-speedup+2100.m4a -speedup -quality=0.75 +2100ms

eac3to does not take mp4 as a source.

Supported source formats:
(1) RAW, (L)PCM
(2) WAV (PCM, DTS and AC3), W64, RF64
(3) AC3, E-AC3
(4) DTS, DTS-ES, DTS-96/24, DTS-HD Hi-Res, DTS-HD Master Audio
(5) MP1, MP2, MP3 audio
(6) AAC audio
(7) MLP, TrueHD, TrueHD/AC3
(8) FLAC
(9) EVO/VOB/(M2)TS and MKV

Decoded audio data can be stored as / encoded to:
(1) RAW, (L)PCM
(2) WAV (PCM only), W64, RF64, AGM
(3) WAVs (multiple mono WAV files, PCM only)
(4) AC3
(5) DTS
(6) AAC
(7) FLAC

manolito

24th November 2013, 20:29

.aac input (with ADTS headers for instance) is only supported with Nero 7 installed.

@therube
Seems like you did not read my question... ;)

I asked if eac3to handles aac audio without Nero 7 installed, and tebasuna answered my question in a very clear way. So the eac3to feature list is a little bit misleading.

Cheers
manolito

LigH

25th November 2013, 11:19

"Misleading", indeed. If it is written in such a general form like

(6) AAC audio

the "average user" would probably assume that it is generally supported without specific prerequisites. But after reading the longer explanation by tebasuna51, that is obviously not true. Just not every user will have read this post (http://forum.doom9.org/showthread.php?p=1654965#post1654965).

Therefore I would recommend to include such unexpected prerequisites in the help text:

(6) AAC audio (ADTS only, requires Nero 7)

At least as long as eac3to does not use LAV or L-SMASH "completely" as alternative splitter/decoder.

shh

26th November 2013, 11:50

Hi!
How do I extract the core AC3 stream of a combined E-AC3/AC3-stream?
My file looks exactly like a combined DTS-MA/DTS-core stream, but eac3to always want's to convert the file to AC3:
C:\eac3to\eac3to.exe german.ac3 out.ac3 -core
E-AC3, 7.1 channels, 1:44:16, 896kbps, 48kHz
(core: AC3, 5.1 channels, 1:44:16, 512kbps, 48kHz)
Extracting E-AC3 core...
Decoding with libav/ffmpeg...
Remapping channels...
Encoding AC3 <640kbps> with libAften...
Creating file "coreac3.ac3"...
Clipping detected, a 2nd pass will be necessary.
-------^C

Snowknight26

26th November 2013, 15:54

http://forum.doom9.org/showthread.php?t=169776

shh

26th November 2013, 19:50

http://forum.doom9.org/showthread.php?t=169776
Oops, thank you very much! :eek:

I got it to work:
You have to create an output.ec3, though the result is an AC3 actually. So you also have to rename your .ec3 to .ac3 afterwords.
Hmm. This should be considered as a bug: I'm creating a real AC3, no E-AC3. Why should I name my output ec3? And, additionally, where's the doc to this secret .ec3-switch? I never read about an ec3 file before...
For .dtsma -> .dts -core the engine works correctly.

Perhaps I should've used full paths in my first post :rolleyes:
(no audio issue for me)
D:\monster>C:\eac3to\eac3to.exe german.ec3 coreac3.ec3 -core
E-AC3, 7.1 channels, 1:44:16, 896kbps, 48kHz
(core: AC3, 5.1 channels, 1:44:16, 512kbps, 48kHz)
Extracting E-AC3 core...
Creating file "coreac3.ec3"...
eac3to processing took 5 seconds.
Done.

D:\monster>C:\eac3to\eac3to.exe coreac3.ec3
AC3, 5.1 channels, 1:44:16, 512kbps, 48kHz

Snowknight26

26th November 2013, 20:42

I wouldn't call it a bug per se but more of an unintended consequence of the relationship between E-AC3 and AC3, the fact that eac3to can natively encode AC3 (which it tries to do by default), and that eac3to only recently added support for extracting the E-AC3 core (from Blu-ray tracks only).

You don't have to name it .ec3.. you could choose .eac3 because, again, with .ac3 eac3to will try to transcode it by default, regardless of the -core switch. DTS, however, cannot be encoded by eac3to without additional software, hence the reason the -core switch works as you want it to.

Try the following and you'll see what I mean:

eac3to file.dtshd core.dtshd -core

As expected, the output is still the 'core' DTS track but with the .dtshd extension.

LigH

27th November 2013, 08:11

The fact that the work mode of eac3to is determined mainly by the filename extension of the output, appears to be a weakness in this case.

Snowknight26

27th November 2013, 16:31

The fact that the work mode of eac3to is determined mainly by the filename extension of the output, appears to be a weakness in this case.

True. Maybe when (or should I say 'if') eac3to starts using ffmpeg for processing/conversion will it be a good idea to switch to its command line arguments style of specifying formats (-acodec, -vcodec, -f, etc.).

Q-the-STORM

1st December 2013, 20:08

is there a way to force eac3to to encode AC3 with libav instead of aften? Justin Ruggles himself said that libav is better than aften and aften shouldn't be used anymore... he should know, since he developed aften and is now involved in libav... is there a way to force eac3to to use it? is there any reason not to use libav by default?

also on a similar matter, any way to get eac3to to use the DTS-HD Master Audio Suite (if you own it) for encoding DTS? would definitely be an improvement to surcode...

filler56789

1st December 2013, 23:47

is there a way to force eac3to to encode AC3 with libav instead of aften?

I think you can tell eac3to to write to stdout and pipe into ffmpeg or avconv...

also on a similar matter, any way to get eac3to to use the DTS-HD Master Audio Suite (if you own it) for encoding DTS? would definitely be an improvement to surcode...

If you donate a considerable amount of $$$$$ to madshi, he may think about that feature request ^_^ Seriously, the Master Audio Suite is a Java application, and unless I'm very misinformed, madshi is not a Java enthusiast :)

maxmercy

5th December 2013, 02:13

I have a 2 channel track that when played through an AVR will upmix to LCR and a single surround channel. Can eac3to 'upmix' this to individual L,C,R,S wavs?

JSS

DoctorM

5th December 2013, 08:05

maxmercy - I've previously requested that they take a look at a DPLII upmix option. There didn't seem to be much interest.

tebasuna51

5th December 2013, 10:15

Can eac3to 'upmix' this to individual L,C,R,S wavs?

Nope, eac3to can't do this upmix.

You can try Foobar2000 and the plugin foo_dsp_fsurround (http://www.hydrogenaudio.org/forums/index.php?showtopic=52235)

LigH

5th December 2013, 10:51

I am not certain, but I believe to remember that Selur was interested in such matrices too, it may be avaliable in his "Hybrid" converter.

Q-the-STORM

6th December 2013, 01:26

I think you can tell eac3to to write to stdout and pipe into ffmpeg or avconv...

thanks, I'll try that...

other question still remains, why is libav not used by default in eac3to?

maxmercy

6th December 2013, 12:59

Nope, eac3to can't do this upmix.

You can try Foobar2000 and the plugin foo_dsp_fsurround (http://www.hydrogenaudio.org/forums/index.php?showtopic=52235)

Thanks for the tip. Is there a way that foobar can write a .w64 or WAVEX after upmixing the stereo/prologic source?

JSS

torturesauce

9th December 2013, 12:40

The "eac3to and more" GUI does not work well at my Windows 8.1. It looks blown out of proportion and many menus/buttons are hidden. I tried disabling DPI scaling and it still doesn't look right. Is there any way to fix it? Or are there any other GUIs for eac3to that have the same amount of options?

the_weirdo

9th December 2013, 14:30

torturesauce

9th December 2013, 16:56

Thank you, it's perfect!

torturesauce

9th December 2013, 18:35

DarkSpace

9th December 2013, 22:01

One more thing: I'm trying to get eac3to to extract a 6.1 Discrete DTS-ES stream from a DVD (.VOB) into a 7-channel .dts file with no loss in quality. I have installed Arcsoft DTS decoder, but it will only work when I export it to .WAV or .FLAC (it's still lossless, but the exported files are, of course, unnecessarily large). Is there a bit-perfect method to demux the stream to a 7-channel .dts?
This sounds like you want no conversion at all going on. In this case, the command line is
eac3to "VTS_01_1.VOB"+"VTS_01_2.VOB" "output.dts"
(or whatever your input files may be). Assuming that, you probably need only select dts as output codec in the GUI and there should be no conversion (I don't use a GUI myself, so I'm guessing here, but it should work).

torturesauce

10th December 2013, 10:38

This sounds like you want no conversion at all going on. In this case, the command line is
eac3to "VTS_01_1.VOB"+"VTS_01_2.VOB" "output.dts"
(or whatever your input files may be). Assuming that, you probably need only select dts as output codec in the GUI and there should be no conversion (I don't use a GUI myself, so I'm guessing here, but it should work).

No such luck. I used this command, both with eac3to and the USeac3to GUI, and the resulting .dts file is 6-channel.

LigH

10th December 2013, 10:46

How certain are you that the dts audio stream in the VOBs actually is 6.1 DTS-ES, and that the demultiplexed file is not anymore? Which tools did you use to check their properties?

tebasuna51

10th December 2013, 10:52

No such luck. I used this command, both with eac3to and the USeac3to GUI, and the resulting .dts file is 6-channel.

Please put the extraction log file and the info than show UsEac3to when load the extracted dts.

torturesauce

10th December 2013, 17:18

Oh, my God! I was wrong. The resulting .dts DOES have the seventh channel as well, according to MediaInfo. Whoops. Thank you for your help, guys!

Lenmaer

30th December 2013, 14:56

Anyone can help me how to properly extract this into FLAC?

omplete name : F:\BDMV & DVDISO\Hal\BDMV\STREAM\00007.m2ts
Format : BDAV
Format/Info : Blu-ray Video
File size : 176 MiB
Duration : 34s 560ms
Overall bit rate mode : Variable
Overall bit rate : 42.7 Mbps
Maximum Overall bit rate : 48.0 Mbps

Video
ID : 4113 (0x1011)
Menu ID : 1 (0x1)
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L4.1
Format settings, CABAC : Yes
Format settings, ReFrames : 4 frames
Codec ID : 27
Duration : 34s 34ms
Bit rate mode : Variable
Bit rate : 39.1 Mbps
Maximum bit rate : 38.0 Mbps
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16:9
Frame rate : 23.976 fps
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.786
Stream size : 158 MiB (90%)

Audio #1
ID : 4352 (0x1100)
Menu ID : 1 (0x1)
Format : TrueHD / AC-3
Mode extension : CM (complete main)
Format settings, Endianness : Big
Muxing mode : Stream extension
Codec ID : 131
Duration : 34s 34ms
Bit rate mode : Variable / Constant
Bit rate : 448 Kbps
Maximum bit rate : 672 Kbps
Channel(s) : 2 channels
Channel positions : Front: C / Front: L R
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Stream size : 1.82 MiB (1%)

Because it seems there's a Front channel too but eac3to always extracts it as 2.0 Stereo FLAC.

Stereodude

30th December 2013, 17:24

Because it seems there's a Front channel too but eac3to always extracts it as 2.0 Stereo FLAC.It says it is 2 channel though.

Lenmaer

30th December 2013, 18:28

OK, thanks

Stereodude

30th December 2013, 18:45

My guess is that the L & R are mono (the same) and the center is different. Those are the two channels. You want a wave file with 3 channels?

Lenmaer

30th December 2013, 19:38

well, I wanted a flac file with 3 channels if the original really had 3 channels.

the_weirdo

31st December 2013, 03:44

From your posted MediaInfo, I think it says TrueHD track is mono and embeded AC-3 track is stereo, not that the audio track has 3 channels. I may be wrong though. Maybe you should upload a sample.

pottzman

1st January 2014, 05:30

Could somebody tell me how you get eac3to to detect sonic audio decoder without installing it? I can get Nero, arcsoft v6, and surcode all detected without installing the entire parent program but I am unable to do the same with sonic audio decoder.

Cheers

Stereodude

3rd January 2014, 23:46

On the change / improvement request front... It would be nice if eac3to reported durations out to the ms instead of the rounding to the second.

So instead of this:
AC3, 5.1 channels, 0:19:18, 384kbps, 48kHz
Decoding with libav/ffmpeg...
Reducing depth from 64 to 24 bits...
Writing W64...

You'd get:
AC3, 5.1 channels, 0:19:17.824, 384kbps, 48kHz
Decoding with libav/ffmpeg...
Reducing depth from 64 to 24 bits...
Writing W64...

Groucho2004

4th January 2014, 00:17

On the change / improvement request front... It would be nice if eac3to reported durations out to the ms instead of the rounding to the second.
I second that request.

Coren

6th January 2014, 13:07

So. You have an audio-only Blu-ray, or a concert Blu-ray, and you want to rip the audio onto your computer, perhaps into nicely track-separated FLAC files.
Ripping is the easy part. Problem is, eac3to gives you a single audio file and a chapters file, but you still have to chop that big file into bits following the timestamps in the chapters file. How I do it, is I turn the chapters file into a CUE file, which I then feed to foobar2000, which generates individual audio files following that CUE sheet. But making the CUE file is a pain in the backside, so I decided to write a little Python script that does it for me.

https://github.com/frederik-vaassen/audioTools/blob/master/makeCUE.py

All this does, basically, is read in the chapters file, extract the titles (if available) and the timestamps, and pour them back into a properly (I hope) formatted CUE file.
I made this purely for myself, and as such it's basic and probably won't work on everything, but if you run into errors, feel free to DM me and I'll try and have a look.

You will need a recent version of Python 2 (I use 2.7). Place the script in the folder your files are in (you can also specify the full path of each input and output file as arguments) and do this at the command line:

python makeCUE.py <chapter_file> <audio_file> <output_cue_file>

There are some extra options, try makeCUE.py -h to see them.

Hope this is of use to someone!

Lenmaer

6th January 2014, 13:11

I merge the m2ts (or the audio file extracted with eac3to) into an MKV or MKA (no need for the video track) with chapters (extracted with Chapter Grabber) for that.
Then open that MKV/MKA into foobar, it'll rip and split everything into separate audio files.

Lynx_TWO

10th January 2014, 15:42

Hey guys real quick question: If I decode 5.1/7.1 audio to 2-channel Dolby Pro Logic II using the -downDpl switch, plus the -mixlfe to mix in the LFE channel, am I doubling the .1 bass track, or does EAC3to ignore the -mixlfe switch when using the -downDpl switch? I would not want to mix in LFE if the -downDpl switch already does this...

Thanks!

LigH

10th January 2014, 16:11

The result will have 2 channels. There is no more separate LFE channel, low frequencies are mixed into Left and Right.

DoctorM

10th January 2014, 19:18

I think he might be asking if the -mixlfe is redundant and already done when using -downDpl.

LigH

10th January 2014, 20:30

I think I remember a "no" on this question which was discussed before, somewhere; DPL (II) does not care about LFE without explicitly mixing it in, the downmix matrix only includes full frequency channels.

No guarantee, though.
__

From my previous experience in a DVD production studio (many years ago), I remember:

Movie studio made audio streams with LFE channel should usually not have simply the filtered low frequency part of the whole spectrum which is then probably missing in all other channels. Instead, this channel usually has, as the name describes, additional and optional "Low Frequency Effects", and the full frequency channels would usually still contain low frequencies of movie set recorded audio.

No guarantee, either.

ndjamena

10th January 2014, 23:20

I'm just wondering what the generally accepted pronunciation for EAC3TO is, I tend to just shorten it to Ecto in my head but I mentioned the program in a verbal conversation recently and realised I'm not sure what I said was correct.

Snowknight26

11th January 2014, 00:06

'E-AC3 to' technically, so probably E A C 3 2 as if you were to pronounce each character separately.

ndjamena

11th January 2014, 00:56

That's how I say it when I'm typing it but it's a bit of a mouthful. As far as I'm aware that was the first time I've ever included the program in a conversation, when experts are discussing it do they shorten it to anything in particular or am I a bit of a yokel?

Guest

11th January 2014, 01:33

It's better than "visual studio 2013 express" so be happy. :)

DoctorM

11th January 2014, 01:34

I think it is overdue for a nickname. I agree with Snowknight26's accepted pronunciation.
I do like Ecto though or treating the 3 like an E we could get something that sounds like Eek-Eee-two or Eck-eh-to.

ndjamena

11th January 2014, 02:49

It's better than "visual studio 2013 express" so be happy. :)

Far be it from little old me to argue with you but I'm pretty sure if you had to include VS2013x in an explanation multiple times you'd shorten it to 'Visual Studio', or if multiple versions of VS were mentioned there's 'Express' or '2013 Express'. But to say EAC3TO repeatedly would get kind of wearing, not to mention the fact that Visual Studio kind of flows whereas EAC3TO is full of hard starts and stops and would be completely meaningless to anyone not fully engaged in the audio encoding community. Do people really keep saying E-A-C-3-2 repeatedly or does the program exist in a realm where the name rarely has to pass through human vocal chords?

(by the way, I love the way you really need to know something about video/audio encoding to even be permitted to post on this forum!)

Guest

11th January 2014, 03:07

Let's let it go, it's not useful for this thread. Thank you.

ndjamena

11th January 2014, 03:41

Agreed.

tijgert

12th January 2014, 22:35

Oh my.. hundreds of pages dating back six years or more? How will I ever find what I need?
I hope you'll all excuse me for perhaps asking an already answered question, but the amount of posts here is staggering and just too much to siff through.

My query: I want to convert 7.1 TrueHD to 7.1 DTS-MA or even 5.1 DTS if need be.
I have access to the DTS suite and a bunch of other codec packs (all legit), but am not very knowledgable beyond ye'r basic Handbrake...
Anyone care to take my hand? Pretty please? How about a shove in the right direction, coupled with some strong hints...?

Chumbo

13th January 2014, 02:17

You should be able to output to separate WAV files and then import to your DTS suite, e.g.,eac3to source.thd output.wavs

PowerGamer

15th January 2014, 11:54

I am trying to demux 2D video track from 3D BluRay disc (that disc contains both 2D and 3D movie versions, i.e. there are options "Play 2D" and "Play 3D" in disc menu when I am running TotalMedia Theater).
This is what I am getting when listing disc info with eac3to:
http://pastebin.com/BbJEWyQP
How do I find out which .mpls file refers to 2D version of the movie?

Snowknight26

15th January 2014, 14:57

All of them do due to the nature of the 3D implementation used on that disc (MVC instead of anaglyph).

73ChargerFan

16th January 2014, 06:19

If you run eac3to <filename> 1)
it will give you stream numbers for left, right eyes.
I think you're supposed to use the left eye stream, but I might have that backwards.

Sparktank

16th January 2014, 07:28

It is the Left-Eye stream that's only 2D. It's presented in AVC/H264 while the Right-Eye (the 3D data; difference) is presented in MVC.

For 3D Blu-ray discs, most MPLS will present everything in 3D fashion. As long as everything is transferred to 3D.
Some discs have all the special features and trailers transferred to 3D (AVC/MVC) while some will only have the main movie in 3D (AVC/MVC).

But, generally, if you're selecting by MPLS, you can just run the .MPLS through eac3to to get the information of how it's read.
Reading from a playlist file (MPLS) is different from reading from an actual file (M2TS/MKV/etc).
Through MPLS, you don't get numbers in front of the stream, instead you just get hyphens ("-") and have to count which stream(s) you want to use.
In case of 3D blu-rays, skip over the "right-eye" data and use only the Left-eye data (AVC/H264).

This will give you 2D data, every time.

I believe the only software that gives you a 3D.mkv file (with AVC/MVC) is MakeMKV.
But, even then, all media players will only decode/play the 2D (Left-eye) data unless they are specifically designed for 3D playback (3dat.tv's StereoScopic Player; Bino; etc).

If you're playing back from a Blu-ray player that has file playback file system, it's very unlikely that it supports MVC at all so it will either A) playback as 2D or B) not play at all.

The best thing to do is grab the smallest file possible (like a trailer) and test it with your playback system.

MPC (and other variations), VLC, and most other media players don't support MVC playback.

DragonQ

16th January 2014, 16:00

Can eac3to combine 6 separate mono WAV files into a 5.1 WAV file? I've tried doing it with other software but I haven't found a way to get it working properly. Adobe Audition 3.0 reduces the volume of each track for no apparent reason, Adobe Audition CS6 raises the volume of each track for no reason.

EDIT: Never mind, managed to do with using SoX. :)

LigH

16th January 2014, 16:49

BeSweet could do it as well, with a *.mux file; but it doesn't support W64, so it won't be a choice in case the result would exceed 4 GB.

Nexin

17th January 2014, 12:52

Trying to convert many audio files in a folder mainly same format to output as wav files. Where any typing needed is minimal than is required now with eac3to such as eac3to "input" "output" "input" "output" "input" "output" "input" "output" etc Ways to make this task easier as like other software's do it.

eac3to converting

using cli by dropping a folder that contains music files only.

eac3to "drive\folder path" "drive\folder path" -wav

Or input as a text.txt file each audio file on a separate line.

"drive:\folder path\A audio file.ext"
"drive:\folder path\An audio file.ext"
"drive:\folder path\Again Another audio file.ext"
"drive:\folder path\Another audio file.ext"

eac3to "drive:\folder path\text file to process.txt" "drive:\folder path\" -wav

All of the above be able to convert many source audio files to wav files.
Where bitrate, kHz, channels etc remain exactly the same as the source.
For eac3to to append source file name to each destination file for each file with a replaced .wav extension

Nexin

17th January 2014, 13:29

Audio File/s or other eac3to known format output to filename.txt

I'm also needing a way to generate a report to a filename.txt file. So far I have tried eac3to "drive:\folder path\file name.ext" >"drive:\folder path\file name.txt" where the outputs has non ascii\unicode characters such as many boxes before file details. If could do this using cli for eac3to to parse a folder of many audio files. Need report to have same info as see with eac3to "drive:\folder path\file name.ext"

All of the above stumbles when drop a folder to cli with "HD DVD / Blu-Ray Disc Structure not found" a way to overcome so can work with folder or folders that contain many audio files. eac3to because it is the only software that can clearly see whole dts mlp ma more than any other software's including foobar2000 and mediainfo for audio files. Although sometimes foobar2000 will see 5.1 audio where eac3to will see only 2.0 same again in reverse eac3to will where foobar2000 won't

using latest releases for eac3to and foobar2000 with win xp sp3

Nexin

18th January 2014, 08:13

Has this project ended ? Madshi first post of this thread last edited the start of last year ''Last edited by madshi; 20th January 2013 at 17:44''

Anyone have idea how better to do the two above posts. With eac3to or other tools for the same formats as straight conversions

Sparktank

18th January 2014, 08:30

Has this project ended ?

No, Madshi said he's focusing his energy on madVR for now (until it reaches a 1.0 version) and then will return to this project.

-------
Have you tried a batch file?

(From an example provided with SoX...)

eac3to_batch.bat
@echo off
%~dp1
CD %~dp1
FOR %%A IN (*.dts) DO "C:\eac3to\eac3to.exe" "%%~nxA" "%%~nA.wav" -log="%%~nA_WAV.txt"

This will convert all files with the extension of ".dts" and will also create individual log files of each ".dts" file and use the same name given for each ".dts" file.

I'm not entirely sure how to do an advanced batch file that includes multiple extensions. Someone more crafty can chime in.

In case of Matroska containers use (*.mkv) and add the stream number after the "%%~nxA" or leave out the stream number if it's the first audio stream you wish to convert.

@echo off
%~dp1
CD %~dp1
FOR %%A IN (*.mkv) DO "C:\eac3to\eac3to.exe" "%%~nxA" 2: "%%~nA.wav" -log="%%~nA_WAV.txt"

You can also add in other functions too like -downStereo or other...

@echo off
%~dp1
CD %~dp1
FOR %%A IN (*.mkv) DO "C:\eac3to\eac3to.exe" "%%~nxA" 2: "%%~nA.wav" -downStereo -log="%%~nA_WAV.txt"

I'm too lazy to look up multiple extensions so I just repeat the conversion line but change the extension....
@echo off
%~dp1
CD %~dp1
FOR %%A IN (*.mkv) DO "C:\eac3to\eac3to.exe" "%%~nxA" 2: "%%~nA.wav" -log="%%~nA_WAV.txt"
FOR %%A IN (*.dts) DO "C:\eac3to\eac3to.exe" "%%~nxA" "%%~nA.wav" -log="%%~nA_WAV.txt"
FOR %%A IN (*.ac3) DO "C:\eac3to\eac3to.exe" "%%~nxA" "%%~nA.wav" -log="%%~nA_WAV.txt"

and that usually works for me (I remember doing a small batch back when I had WinXP.)

Nexin

18th January 2014, 08:53

Maybe Madshi when returns to the project may make it better with more options. Maybe things like use surcode as decoder if better than ffmpeg for mlp etc. Ability to see AOB as files and disc with AOB files parse from .ifo or .bup for chapters. DVD parsing, folder parsing for any audio format file that eac3to can work with. As you say using sox maybe eac3to could work with that more, I didn't think of sox until you mentioned it.

Thanks Sparktank your solution on reading seems it may work for any multiple files from folders and otherwise. Also be able to pipe the source file names to destination files which is needed and again for making log files. Sure if someone has other scripts to improve or better yours then I'm ready to try those also :)

ndjamena

19th January 2014, 15:27

@if "%EMode%"=="" Set EMode=OFF
@ECHO %EMode%
if "!"=="" (
SETLOCAL DISABLEDELAYEDEXPANSION
Set "DISLOC=1"
)

Set "EXTLIST=*.mkv *.dts *.ac3 *.mka"
goto :Skip

:Process_File

"d:\eac3to\eac3to.exe" "%~nx1" "%~n1.wav" -log="%~n1_WAV.txt"

goto :eof

:Skip

if [%1]==[] (
Set "Pattern=%EXTLIST%"
CALL :START
)

:ARGS

if [%1]==[] (
Set "Pattern="
Set "EXTLIST="
if NOT "%DISLOC%"=="" (
Set "DISLOC="
ENDLOCAL
)
pause
goto :eof
)
if NOT EXIST %1 (
SHIFT
goto :ARGS
)
if /I "%~f1"=="%CD%" (
Set "Pattern=%EXTLIST%"
CALL :START
SHIFT
goto :ARGS
)
Type NUL
pushd "%~f1" 2> nul
if NOT "%errorlevel%"=="0" (
if /I NOT "%CD%\"=="%~dp1" (
pushd "%~dp1"
Set "DoPop=1"
)
Set Pattern="%~nx1"
) else (
Set "Pattern=%EXTLIST%"
)
CALL :START
if "%DoPop%"=="1" (
popd
Set "DoPop="
)
SHIFT

goto :ARGS

:START

for %%y in (%Pattern%) do (
CALL :Process_File "%%~fy"
)

goto :eof

Nico8583

20th January 2014, 16:47

Hi,
Is it necessary to use an external decoder to convert DTS to AC3 ? Is a better quality with an external decoder ?
I would like to extract DTS core from DTS-HD MA or HR in order to have a DTS stream and convert it to AC3.
Thanks !

Snowknight26

20th January 2014, 18:22

No, eac3to can do that using internal functions.

Nico8583

20th January 2014, 19:10

Ok thanks, so Arcsoft decoder or other are only to convert HD stream (DTS-HD MA or HR or Dolby TrueHD) to AC3 ?
And does eac3to support 5.1, 6.1 and 7.1 audio track with internal function ?

tebasuna51

20th January 2014, 23:08

AC3 don't support 6.1 or 7.1

Nico8583

21st January 2014, 08:57

OK thanks, but does DTS (not HD) support 6.1 or 7.1 ? And does eac3to support DTS 6.1 or 7.1 decoding with internal function ?

nevcairiel

21st January 2014, 09:08

Core DTS can support 6.1 with the DTS-ES extension, although its usually only used if you have no DTS-HD, and eac3to should be able to decode it as 6.1 as well.
I have never seen 7.1 in core DTS.

Nico8583

21st January 2014, 09:31

Thanks a lot :)
If I want to convert DTS-HD to AC3, I'll extract first DTS core then convert it ;) so no need to an external decoder.

kypec

21st January 2014, 10:29

If I want to convert DTS-HD to AC3, I'll extract first DTS core then convert it ;) so no need to an external decoder.

No need to extract DTS core first, eac3to can convert it directly to AC3 on-the-fly if no suitable external DTS-HD decoder is installed.;)

Nico8583

21st January 2014, 11:06

No need to extract DTS core first, eac3to can convert it directly to AC3 on-the-fly if no suitable external DTS-HD decoder is installed.;)
Thanks for the info, so why an external decoder is needed ?

nevcairiel

21st January 2014, 11:14

The external decoder is only needed if you want to decode the full DTS-HD.

jj666

21st January 2014, 13:54

It doesn't make sense to convert the DTS (lossy) core to AC3, the sensible approach would be to convert the full lossless DTS-HD stream.

Cheers,

-jj-

Nico8583

21st January 2014, 14:19

I don't have any idea about size difference between DTS and AC3, if size is similar yes it doesn't make sense. But if there is a consequent difference, it could be util.
So if I want to convert DTS-HD to AC3, I need an external decoder but if I want to convert TrueHD to AC3, no need to an external decoder ?

jj666

21st January 2014, 15:22

I mean from a compression point of view, compressing from DTS to AC3 is just reducing the audio quality (as you are compressing the original source twice). Compressing DTS-HD to AC3 would be a single compression step. You would need the Arcsoft decoder dll installed to facilitate working with DTS-HD.

TrueHD on Blu-ray will always contain an AC3 core which you can extract, there should be no need to encode anything.

Cheers,

-jj-

Nico8583

21st January 2014, 15:50

Thanks :)
I've seen there are several versions of Arcsoft decoder, one support 6.1, another support 7.1 but not a single version to support both. Is it true ?

tebasuna51

22nd January 2014, 11:24

Nope. At least version 1.1.0.0 support 6.1 and 7.1

Nico8583

22nd January 2014, 14:06

Good news, thank you !

Nexin

23rd January 2014, 14:37

Thanks ndjamena for the .bat script will see if can make eac3to work with them for what I need

elmarikon

18th February 2014, 16:14

Cheers!

And thanx for your great tool!

I have two questions about it:

1. is it possible to decode / export lossless dts-hd audio?
I could not find a way to do so...

2. Is there a way to not have the "annoying" .log file?
Any options to suppress the creation of it?

All the best from Dammtor!

SwK

the_weirdo

18th February 2014, 16:47

1. is it possible to decode / export lossless dts-hd audio?
I could not find a way to do so...
Yes. However, for decoding, you'll need some commercial decoders, e.g. ArcSoft DTS decoder.

2. Is there a way to not have the "annoying" .log file?
Any options to suppress the creation of it?

I don't understand why log file is "annoying". Anyway, you can disable log creation by adding -log=NUL to the command line.

filler56789

18th February 2014, 17:33

I don't understand why log file is "annoying".

Perhaps because it fills the HDD with unrequested junk? :)

DarkSpace

18th February 2014, 17:47

I don't understand why log file is "annoying".
Perhaps because it fills the HDD with unrequested junk? :)
I must say, I agree here. I'd much rather enable writing of a log file and have it disabled by default than having to add -log=NUL every time.
After all, I have to specifically request eac3to to not create a file that I didn't ask for, which seems counterintuitive to me. And I must say, so far I have had absolutely no use for log files, because I always took the time to read what eac3to output to the console directly. And of course, I remember what my input command was (if necessary, I re-run it).

Ryushin

9th March 2014, 22:11

Solution found to dsmux problem when going to certain hard drives. Threads found on this problem:
https://www.doom9.org/showthread.php?p=1485220#post1485220
https://www.doom9.org/showthread.php?p=1546026#post1546026

Solution was changing the storage controller driver that the hard drive was connected to. In my case, I was using the AMD AHCI RAID driver. I changed that to Microsoft's AHCI driver and the problem was resolved.

It took a better part of an entire day to finally find a solution to this issue.

I hope this helps someone that might run across this same issue.

beto

11th March 2014, 13:08

Hi, I have a bunch of AC3 files that I have to apply a delay to. I know I can use eac3to or delaycut to apply this delay.

The problem with delaycut is that it does not support batches and I have to apply the delay manually to each file (it works but its cumbersome).

As for eac3to I know that Eac3to and More GUI support batches but for the life of me I cannot figure out how to use it to just apply a delay to my AC3 files. I am on Windows 8.1

Can anyone please advise on how to use the EAC3to GUI to apply this delay? I would prefer not to use the command line directly.

Thanks.

Overdrive80

11th March 2014, 16:55

I dont test it, you try this:

@echo off
Title Add/Remove delay by Overdrive

set /p delay="Insert delay in miliseconds (negative:-200 or positive:+200): "

for %%@ in (*.ac3) do (

"C:\Program Files\MeGUI\tools\eac3to\eac3to.exe" "%%@" "%%~p@\%%~n@_fixed.ac3" %delay%ms -silence
)

pause&exit

EDIT: Tested.

beto

11th March 2014, 18:07

Thank you. I will sure try this batch and see if I can adapt to it. Regards.

LigH

11th March 2014, 19:39

In general, I would recommend using the audio skew value during multiplexing into the container, instead of manipulating the audio stream... but well, AC3 is famous enough to know the bitstreams of silence blocks for many channel configurations.

Motenai Yoda

12th March 2014, 17:34

There is a way to do a speedup/slowdown with pitch correction?
Currently eac3to doesn't permit this...

tebasuna51

12th March 2014, 18:12

There is a way to do a speedup/slowdown with pitch correction?
Currently eac3to doesn't permit this...

You can try AviSynth-TimeStretch (BeHappy) or audio editors (Audacity or others).

Work fine for stereo audio but the are problems preserving the phase between channels with 5.1.

Or you can use TimeFactory (http://www.prosoniq.com/www/TimeFactory_II.html) for $499.00

DoctorM

12th March 2014, 19:17

Thingol

12th March 2014, 20:01

How can I convert 6.1 channels DTS to 5.1 AC3 properly? When I do this, AC3's file duration is getting longer.

Motenai Yoda

12th March 2014, 20:08

BeSweet (probably related to BeHappy) can do the pitch correction. Use the Tempo setting at -4.096 to slow the audio without altering the pitch for PAL to NTSC, and +4.2709376043 for NTSC to PAL.
Sony SoundForge can do this to, but I believe it works in cents... and don't ask me what those numbers are.

BeSweet/BeHappy use Avisynth TimeStretch (based on SoundTouch) for speed change, while MeGui use SSRC to resample/speed and TimeStretch for pitch correction.

Also other sw for do this can be Audacity, Sox, mencoder (ffmpeg seems to have only atempo), ecc...

-4.096 and +4.2709376043 come out from 100*((24/1.001)/25)-100 and 100*(25/(24/1.001))-100

@tebasuna51 so no way neither to have an implementation in future?

tebasuna51

13th March 2014, 10:00

@tebasuna51 so no way neither to have an implementation in future?

madshi want the pitch change because it say than correct the previous change made with NTSC -> PAL conversion.

Maybe this is true with english tracks but not always with other languaje/translation.

BTW you can always 'pipe' eac3to output to sox.

Nico8583

22nd March 2014, 15:50

I'm lost with Arcsoft 1.1.0.0 to 1.1.0.8 versions when I read posts from eac3to or LAV filter topics, or from differents sites :(
None of 1.1.0.0 to 1.1.0.8 versions can decode properly all types of streams (5.1, 6.0, 6.1, 7.1...) ?
I would like to simply convert DTS/DTS-HD to AC3 because one of my device doesn't support DTS decoding (but I don't know if it supports TrueHD AC3 core or if I must extract it)...

Boulder

22nd March 2014, 15:54

AFAIK, the safest bet is to use v1.1.0.0. There are at least two different builds around, I think the CRC comparisons (and instructions which one to use) is buried somewhere in this thread.

frumble

22nd March 2014, 15:58

It's this:
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.
So there are different 1.1.0.0 builds?

Nico8583

22nd March 2014, 16:38

I've searched CRC on the thread without result, if you have more informations about it I'm interested ;)

Edit : I've found this link http://forum.doom9.org/showthread.php?p=1167775#post1167775 but is 113 or 120 the best version to use ?

tebasuna51

22nd March 2014, 21:53

I have this one:

dtsdecoderdll.dll version 1.1.0.0 (from the latest TMT build 120)
MD5 644aa3ade7742079533dcde2abf153e2

and work fine for me also decoding non-standard 7.1

Nico8583

22nd March 2014, 23:15

Thanks :) it's what I have also

sl1pkn07

25th March 2014, 01:49

and about 1.1.0.9? have any problem?

laserfan

25th March 2014, 13:54

and about 1.1.0.9? have any problem?
I wonder about this too--there are no comments on 1.1.0.9 anywhere here that I can find.

There is a new update to TMT, v6.6.1.190--has anyone installed it yet and what version of dtsdecoderdll.dll does it have?

sl1pkn07

25th March 2014, 15:56

TMT 6.6.1.190 is 1.1.0.9

the checkactivate.dll. not found in installation (still need?)

tebasuna51

26th March 2014, 13:11

checkactivate.dll is always needed for eac3to, you can use a old version.

Bigmango

28th March 2014, 23:37

I'm lost with Arcsoft 1.1.0.0 to 1.1.0.8 versions when I read posts from eac3to or LAV filter topics, or from differents sites :(
None of 1.1.0.0 to 1.1.0.8 versions can decode properly all types of streams (5.1, 6.0, 6.1, 7.1...) ?

This is a problem with eac3to.

Makemkv doesn't have this problem.

This is what the makemkv developper has to say about this:

This statement is not correct for MakeMKV. I have to dig into details what dtsdecoderdll.dll really is. It is a reference DTS decoder library (written by DTS) with an API layer on top of it (written by Chinese developers). So, all versions of dtsdecoderdll, from 1.0.0.0 to 1.0.0.8 have exactly the same reference decoder library linked in - "VERSION=325,REVISION=27" . The only difference is a Chinese crap on top. Unlike eac3to and others, MakeMKV uses the reference decoder directly as stated on /dtshd/ . All other software, including eac3to, use the Chinese API. Any version of dtsdecoderdll.dll will work equally fine with MakeMKV, even early 1.0.0.0 variant. MakeMKV decodes proper channel layouts for all possible configurations (that one can create with DTS HD master audio suite). I encourage you to check MakeMKV log file - it lists the DTS decoder version, DTS channel layout and output channel mask.

The tests I have done also seem to confirm what he says. The FLAC audio produced by Makemkv doesn't have the channel mapping and audio corruption issues depending on the dtsdecoderdll.dll versions.

So, if you want to make sure you are having a perfect DTS-HD to flac converstion as done by the reference DTS decoder, without needing to worry about the decoder dll version, process your MKV or bluray/iso with Makemkv.

(if you want maximum flac compression with Makemkv set it to 12 in flac.mmcp.xml).

edit: also, with makemkv you only need to copy the "dtsdecoderdll.dll" to the program directory. It doesn't need the other files and doesn't need the checkactivate, etc... no need to register the dll as it reads the DTS decoder contained within the file directly.

DoctorM

29th March 2014, 01:37

Thanks for the info Bigmango. It makes it pretty clear that Eac3to has a new goal in its next update: to make DTS decoding work right since we know it can be done.

filler56789

29th March 2014, 04:56

Another big :thanks: to Bigmango,

BTW, that useful info was originally written @

http://makemkv.com/forum2/viewtopic.php?f=1&t=7201#p30255

tebasuna51

29th March 2014, 17:47

This is a problem with eac3to.

Makemkv doesn't have this problem.
...
So, if you want to make sure you are having a perfect DTS-HD to flac converstion as done by the reference DTS decoder, without needing to worry about the decoder dll version, process your MKV or bluray/iso with Makemkv...

Thanks to show us a new method to decode DTS-HD.

BTW let me say than the conversion of DTS-HD 7.1 strange setup is not perfect because the channels Ls-Rs from DTS are converted to SL-SR channels in flac/wavex, when the channels SL-SR are the DTS channels Lss-Rss from the standard 7.1 setup.

Bigmango

29th March 2014, 17:55

Some other advantages to the way makmkv converts DTS-HDMA:

multiple DTS-HDMA track conversion on the fly, in 1 shot (eac3to can only do 1 at a time)
thanks to the way it directly reads the dts decoder dll, it also works on Linux (and I guess Mac too, although I haven't used the Mac version)
it writes the channel layout in the FLAC tags (as DTS and Dolby do). I'm not sure if this really changes anything as FLAC is supposed to have a standard channel mapping, but it is nice to have nevertheless as we get a FLAC that more resembles the original audio track. Perhaps this is useful for channel number (matrix) down conversion (i.ex 7.1 to 5.1 to 2.0) as found in DTS/Dolby (when the original track used this feature) ?
Better compression. Makemkv uses the ffmpeg flac encoder (max compression level 12), whereas eac3to is using libflac (max compression level 8). In my experience, the ffmpeg level 12 results in better compression for ~98% of the tracks. Some rare tracks will however still compress a little better with eac3to's libflac (perhaps ~2% of the tracks I have done or less).

Advantage of eac3to:

when a 24 bit DTS-HDMA (or TrueHD) only contains 16 bit, it removes the empty bits to output a smaller file. Makemkv doesn't do this yet (the developer said he'll add this feature in the future). So after converting with Makemkv, I often re-extract the resulting FLAC tracks with eac3to to check this. (but in my experience, although this often happened with movies a few years ago, it rarely is the case today. Most new films almost always contain real 24bit audio on their 24bit tracks; especially DTS-HDMA. This is however still less the case with TrueHD, as 24bit audio is a prerequisite for TrueHD it can't contain 16 bit audio; so, some studios convert the 16 bit to 24bit and this results in a file eac3to will save back in 16 bit).

junior_l3oss

20th April 2014, 07:20

i want to use slowdown speeddown and another option?

why dont you add another option?
for example i saw 25300 fps...
i want to change this fps to 23976...
...
is this possible ?

tebasuna51

20th April 2014, 11:23

why dont you add another option?
for example i saw 25300 fps... i want to change this fps to 23976...

is this possible ?

Nope, with eac3to only standard fps can be used (23.976, 24.000 and 25.000).

You can use SSRC with AviSynth to do the same job, but not all conversions can be done, read http://avisynth.nl/index.php/SSRC

torturesauce

22nd April 2014, 01:54

I'm trying some DTS-ES files to FLAC with eac3to. I'm using Arcsoft and everything. The .dts files have been demuxed properly from the DVD (they show up as ES in MediaInfo), but when I'm using the "convert to flac" command, the files, though 6.1, sound distorted and slowed down.

the_weirdo

22nd April 2014, 06:27

r0lZ

22nd April 2014, 09:22

When I convert DTS 5.1 to AC3 with eac3to v3.27 (using the ArcSoft DTS decoder v1.1.0.0), the conversion is extremely slow: around 15 minutes to convert an input file of 1:16:00. Is it normal, or should I try to re-install eac3to or the ArcSoft decoder? Is it something I can do to boost the conversion?

> eac3to.exe Fra.dts Fra.ac3 -640

DTS, 5.1 channels, 1:15:53, 1509kbps, 48kHz
Decoding with ArcSoft DTS Decoder...
Remapping channels...
Encoding AC3 <640kbps> with libAften...
Creating file "Fra.ac3"...
The last DTS frame is incomplete and thus gets skipped.
eac3to processing took 15 minutes, 3 seconds.
Done.

Boulder

22nd April 2014, 09:31

I'd check Task Manager to make sure that the eac3to process is running as fast as it can. Have you tested decoding to WAV, is it as slow?

r0lZ

22nd April 2014, 10:11

Groucho2004

22nd April 2014, 10:18

Thanks.

Nothing wrong in task manager. eac3to uses only one CPU core. It's a pity, but afaik it's normal.

I just did a conversion to WAV. The process took exactly the same time: 15 minutes, 5 seconds. So, it appears that it's the decoding process that is damn slow. I will try to uninstall the ArcSoft decoder, to see if the default DTS decoder (which one is it?) is faster...
eac3to just uses the functions that the decoder/encoder libraries provide (dtsdecoder.dll, libaften, libav, etc.). If these are not multi-threaded, the process won't be multi-threaded. The default DTS decoder is libav and it is a lot faster than Arcsoft's.
Encoding to AC3 with libaften should be multi-threaded (I think).

madshi

22nd April 2014, 10:19

The ArcSoft DTS decoder is pretty slow - and it's single threaded. eac3to itself tries to process multi-threaded, as far as possible. But if the main decoder is single threaded that's the bottleneck and there's not much eac3to can do about it. No need to uninstall ArcSoft, try "-libav", that will use libav for decoding and process quite a bit faster (but also single threaded). But of course the ArcSoft decoder is using DTS reference code, so it might be "better" than libav. Or maybe not, who knows...

Edit: Cross-post, Groucho2004 was faster...

Boulder

22nd April 2014, 10:24

And something positive: Sonic is even slower :D

r0lZ

22nd April 2014, 10:31

Thanks everybody! That confirms that it's Arcsoft the culprit. I'll try with -libav. Thanks for the tip!

I'm not a specialist in audio decoding, but I wonder why ArcSoft is supposed to give better results than Libav. IMO, any decoder should give the same result, regardless of the techniques it uses. It's only when encoding that it is important to use a good encoder. Or am I wrong? What's the benefit of using "DTS reference code"?

Sorry for that newbie questions.

Richard1485

22nd April 2014, 10:43

is there a way to force eac3to to encode AC3 with libav instead of aften?

I second this request. It would be great to be able to do this and to be able to set dialnorm and DRC profile. While I know that some people (justifiably) consider dialnorm obsolete, I still like to use sometimes.

torturesauce

22nd April 2014, 11:06

Which version of dtsdecoderdll.dll you're using? IIRC, version 1.1.0.8 has problem with decoding DTS(-HD) 6.1.

Thank you! Yeah, I think it was 1.1.0.8. I replaced it with 1.1.0.1 and it worked like a charm!

filler56789

22nd April 2014, 11:21

I'm not a specialist in audio decoding, but I wonder why ArcSoft is supposed to give better results than Libav. IMO, any decoder should give the same result, regardless of the techniques it uses. It's only when encoding that it is important to use a good encoder. Or am I wrong? What's the benefit of using "DTS reference code"?

The so-called "reference code" is supposed to return a "bit-exact" decompression of lossy DTS. The bit-"exactness" is necessary for the correct decompression of DTS-HD Master Audio.

r0lZ

22nd April 2014, 12:45

The so-called "reference code" is supposed to return a "bit-exact" decompression of lossy DTS. The bit-"exactness" is necessary for the correct decompression of DTS-HD Master Audio.
I see. So, there should be no difference when decoding a "simple" (no HD) DTS 5.1. Right?

@Richard1485:
I second this request. It would be great to be able to do this and to be able to set dialnorm and DRC profile. While I know that some people (justifiably) consider dialnorm obsolete, I still like to use sometimes.
No need to uninstall ArcSoft, try "-libav", that will use libav for decoding and process quite a bit faster (but also single threaded).
And for the dialnorm, try -keepDialnorm :rolleyes:

Richard1485

22nd April 2014, 14:27

@r0lZ
What are you talking about? The post that you quoted refers to using libav as a decoder, not an encoder.

And for the dialnorm, try -keepDialnorm :rolleyes:

That keeps the dialnorm of an AC-3 track when extracting. It does not give the user the ability to set dialnorm (let alone a DRC profile) when encoding.

r0lZ

22nd April 2014, 14:37

Oops, mismatch! Sorry.

DoctorM

22nd April 2014, 21:23

You could always try Vobdnorm: http://www.coises.com/software/vobdnorm.htm

It requires the container to be VOB or MPG, but you can mux and demux from those formats as an extra step.

torturesauce

23rd April 2014, 01:55

Anyone knows how can I decode DTSWAV CD's that contain DTS-ES 6.1 audio? Can I do it somehow with Arcsoft and eac3to? MediaInfo cannot detect the sixth channel, reading it as regular DTS 5.1 (though it reads DTS-ES files from DVD's normally).

filler56789

23rd April 2014, 02:25

^ Give a try to bsconvert.exe from the AC3Filter Tools package,
and after converting the fake .WAV into a normal DTS file,
you can apply eac3to onto it.

http://www.ac3filter.net/wiki/Download_AC3Filter_tools

torturesauce

23rd April 2014, 10:41

Richard1485

24th April 2014, 12:34

Madshi, could you comment on whether the information in the post quoted here is accurate or not (http://forum.videohelp.com/threads/363884-How-to-extract-7-1-ch-%288%29-DTS-audio-to-each-separate-audio-and-it-names-it?p=2315777&viewfull=1#post2315777)?

filler56789

24th April 2014, 14:18

^ Actually, that's just an "echo" of this post (http://forum.doom9.org/showthread.php?p=1675362#post1675362) by Bigmango.

kabelbrand

24th April 2014, 15:48

I recently switched to eac3to 3.27 and noticed the negative gain values applied when clipping occurs differ quite a lot from those applied by version 3.24

e.g. Downmixing a 5.1 source to Stereo (ProLogic): eac3to beep51.wav stereo.wav -down2

eac3to v3.24 [...] Applying -3,82dB gain...
eac3to v3.27 [...] Applying -9.84dB gain...

A sample source file and full logs can be found here:
https://docs.google.com/file/d/0ByNsn-r9w8BhS2VGRjJ0X1Y3d2M

Any ideas why the gain values differ so much? My apologies if the answer is already somewhere in this gigantic thread.

Groucho2004

24th April 2014, 15:54

e.g. Downmixing a 5.1 source to Stereo (ProLogic): eac3to beep51.wav stereo.wav -down2
Shouldn't that be "-downDpl"?

kabelbrand

24th April 2014, 15:59

Shouldn't that be "-downDpl"?

Yes, this is the 3.24 command line. I also tested with downStereo and downDpl. Please see the Google Docs link for all log files.

Groucho2004

24th April 2014, 16:10

Yes, this is the 3.24 command line. I also tested with downStereo and downDpl. Please see the Google Docs link for all log files.
When you look at the changelog you'll see that there have been changes to the downmix algorithms. However, madshi can probably shed more light on this.

tebasuna51

24th April 2014, 16:54

Madshi, could you comment on whether the information in the post quoted here
Mango wrote:
All versions of dtsdecoder.dll have some issues with some DTS-HDMA tracks (resulting in non-lossless output, sound corruption, etc...) .
dtsdecoder.dll 1.1.0.0 works fine for all tracks (including 6.0/6.1/7.1), but not 7.1 "strange setup" tracks....
is accurate or not?

Like I say many times dtsdecoder.dll 1.1.0.0 with eac3to decode the 7.1 "strange setup" tracks to wav 7.1 correctly, with low volume but with a correct mix.

The same job with Makemkv finish with a file with incorrect channels Ls-Rs from DTS converted to SL-SR channels in flac/wav.

Decode 7.1 "strange setup" with Makemkv (or DTS-HD StreamPlayer) is only recommended to edit the source and recode another time to 7.1 "strange setup".

tebasuna51

24th April 2014, 17:21

I recently switched to eac3to 3.27 and noticed the negative gain values applied when clipping occurs differ quite a lot from those applied by version 3.24
...
Any ideas why the gain values differ so much? My apologies if the answer is already somewhere in this gigantic thread.
There are a new method over 3.24, read in
http://forum.doom9.org/showthread.php?p=1600693#post1600693
Downmix improvements

- Use not normalized matrix to downmix and let the second pass do the normalize if necessary. [OK in 3.25]
Then the negative gain must be greater now.

And the differences between -downDpl and -downStereo, here http://forum.doom9.org/showthread.php?p=1600695#post1600695

Stereo
FL' = FL + 0.7071 x FC + BL + 0.7071 x BC
FR' = FR + 0.7071 x FC + BR + 0.7071 x BC
Dpl
FL' = FL + 0.7071 x FC + 0.8660 x BL + 0.5000 x BR + 0.7071 x BC
FR' = FR + 0.7071 x FC - 0.5000 x BL - 0.8660 x BR - 0.7071 x BC

Richard1485

24th April 2014, 17:30

^^ Thanks for the info, tebasuna, but there is more in the post than just the processing of "strange setup" 7.1 tracks. Is the bit about the Chinese API true?

madshi

24th April 2014, 17:48

It's quite possible that the Chinese API bit is true. But the only practical consequence is that when picking a good dtsdecoder.dll version, you'll still get 100% bit perfect files with eac3to for 99.9% of all DTS-HD Master Audio files, with the only exception being those rare "strange setup" files, which eac3to explicitly warns you about (so you know when you hit those 0.1%). And even those decode "ok" (just too low volume), as explained by tebasuna51.

Groucho2004

24th April 2014, 17:50

Is the bit about the Chinese API true?True or not, the important question would be what the "Chinese API" does. If it's just a layer on top of the API of the decoder dll to make accessing its functions easier, I see no harm. I also wonder how the makemkv dev knows how madshi uses the library.

Richard1485

24th April 2014, 18:01

Thanks for clarifying that, madshi. It just alarmed me a bit because I've always relied on eac3to for decoding. It's a great tool. :-)

madshi

24th April 2014, 18:03

Some users have done DTS-MA bit perfection decoding tests with eac3to years ago and everything was bit perfect, except the "strange setup" tracks.

DarkSpace

24th April 2014, 20:04

About the "Chinese API" part, I'd just like to mention that perfect decoding of all tracks (even 6.0 and 6.1, or other problematic setups) sounds like a good point - if it works, which I have no idea of.
However, another great point is the ability to decode multiple tracks at once. I always found it more than just a bit annoying that I would have to decode one DTS-HD track at a time, then wait until that finishes, and then decode the second one: For one, it takes time, and another reason is that for all other types of audio, I can just type e.g.
eac3to input.m2ts 2:out1.flac 3:out2.flac 4:out3.flac -log=NUL
while with even just 2 DTS-HD tracks, this will fail. A temporary solution is, of course, to have eac3to just decode one track, then the next, and so on, but this sounds like a lot of effort for an inelegant solution, especially when a solution is available that sounds more elegant to me.

For reference, this is just my opinion, and I'm not trying to apply pressure on you or something like that. I'm aware that sometimes, I write stuff that is easily misunderstood :o

Yoshi

2nd May 2014, 14:31

Hello together,

I've been keen on posting here in regard to the great tool eac3to since quite some time but always been successfully discouraged by this registration hassle here. Well, I finally brought myself to do it at the end, so I want come to the point(s):

I almost love eac3to and use it regularly, however only almost since some oddities remain:

1. despite all the fuss which is been made about the so-called "HD" audio formats like TrueHD and DTS-HD MA, some older releases on DVD feature a higher dynamic range / unaltered mix compared to the Blu-rays. Examples would be Terminator 2 (CDS mix only on the first US DVD), Titanic (higher dynamic range on the first THX-certified US-DVD compared to the 3D BD), Fight Club (even higher DR-values for the rear channels on the US DVD compared to the UK Fox BD), Se7en (different mixes).

This is why I sometimes want to extract the DVD's audio and sync it to the Blu-ray picture. With the old, but in it's days similarly great tool "vStrip", this is no problem at all. However, eac3to keeps screwing it up and truncates the beginning of the AC3-file independent of the movie. During extraction it complains about a weird audio delay which could not be fixed, for example:

eac3to v3.27
command line: eac3to.exe "S:\SEVEN_SF\VIDEO_TS\VTS_01_1.VOB"
+"S:\SEVEN_SF\VIDEO_TS\VTS_01_2.VOB"
+"S:\SEVEN_SF\VIDEO_TS\VTS_01_3.VOB"
+"S:\SEVEN_SF\VIDEO_TS\VTS_01_4.VOB"
+"S:\SEVEN_SF\VIDEO_TS\VTS_01_5.VOB"
+"S:\SEVEN_SF\VIDEO_TS\VTS_01_6.VOB" 3: q:\se7en.ac3
------------------------------------------------------------------------------
VOB, 1 video track, 7 audio tracks, 5 subtitle tracks, 2:01:36
1: Joined VOB file
2: MPEG2, 576i50 (4:3)
3: AC3, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB, 7ms
4: DTS-ES, 6.1 channels, 755kbps, 48kHz, 7ms
5: AC3, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB, 7ms
6: AC3, 2.0 channels, 128kbps, 48kHz, dialnorm: -27dB, 7ms
7: AC3, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB, 7ms
8: AC3, 2.0 channels, 128kbps, 48kHz, dialnorm: -27dB, 7ms
9: AC3, 5.1 channels, 384kbps, 48kHz, dialnorm: -27dB, 7ms
10: Subtitle (DVD)
11: Subtitle (DVD)
12: Subtitle (DVD)
13: Subtitle (DVD)
14: Subtitle (DVD)
[a03] Extracting audio track number 3...
[a03] Removing AC3 dialog normalization...
[a03] A remaining delay of +7ms could not be fixed.
[a03] Creating file "q:\se7en.ac3"...
Video track 2 contains 182498 frames.
eac3to processing took 50 seconds.
Done.

2. This refers to eac3to's output above: As far as I am aware, eac3to actually doesn't alter any audio information when only extracting it via selecting .ac3 .dts .thd, etc. as the file name. If so, it is however confusing that the dialog normalization is supposed to be removed when where is nothing to remove in this case.

3. Compared to MakeMKV, eac3to (sometimes) doesn't list all available playlists as reflected by the MPLS files. Is it an intended behaviour of eac3to to only list the main feature(s) when parsing a whole BDMV-structure? Some discs spread the bonus features across several m2ts-files via seamless branching so it would be more convenient to have them listed by eac3to as well instead of having to look for the single parts on your own.

4. This concerns eac3to's method to access Arcsoft's DTS Decoder where I want to quote two statements:

Like I say many times dtsdecoder.dll 1.1.0.0 with eac3to decode the 7.1 "strange setup" tracks to wav 7.1 correctly, with low volume but with a correct mix.

Logically, this is contradiction in terms: If a conversion of a lossless source is reproduced with lower volume, it isn't lossless and thus [b]not correct by definition. I doubt myself that this will really affect the audio quality whatsoever, but actually it is a similar academical thing like keeping the lossless source in the first place. I do this myself (converting it to FLAC), however I'm very well aware that I would most likely fail any blind test between properly encoded AC3 or DTS versus the PCM master.

It's quite possible that the Chinese API bit is true. But the only practical consequence is that when picking a good dtsdecoder.dll version, you'll still get 100% bit perfect files with eac3to for 99.9% of all DTS-HD Master Audio files, with the only exception being those rare "strange setup" files, which eac3to explicitly warns you about (so you know when you hit those 0.1%). And even those decode "ok" (just too low volume), as explained by tebasuna51.

Being on the safe side at a chance of 0.99 something isn't the worst thing for sure. But why the apparent lack of motivation to get it correctly and reach the 100%? If the claims from the developers of MakeMKV are correct, what's their magic to get it done right where eac3to fails?

Besides, how am I supposed to handle those "weird 7.1 setups" then in combination with seamless branching? MakeMKV decodes it properly but screwes up the audio boundaries, at least if you intend to play it back via something like Dune. Eac3to on the other hand handles this correctly, but messes up the audio decoding in conjunction with the Arcsoft decoder. Great.

5. [BUG] In case of missing access rights, the root folder of c: for instance, eac3to doesn't warn about it and simply doesn't create the files.

Despite the points above, I have some other questions I somehow lack understanding correctly which concern eac3to, but are of general nature. I've been through most of this thread, but certainly not every page so I hope this hasn’t been covered yet. At least, I couldn't find a revealing explanation anywhere so far:

1. Seamless branching on Blu-ray: For all I know, when a feature is spread over several M2TS files, there seems to be a mismatch of video and audio frame-lengths which has to be taken into account in order to prevent summed up lack of sync. However, I don't understand why the several M2TS-files can't be considered as one continuous stream instead and besides that, I wonder how a stand-alone player behaves here.

On the example of "Escape Plan", spread over tons of M2TS and faked playlists, I only get a flawless result when using eac3to. With MakeMKV, although considered to be aware of seamless-branching, the resulting MKV has audio glitches on the original M2TS boundaries, at least when playing it back with a Dune 303D.

Eac3to seems to behave differently when it encounters seamless branching depending on the audio format: when dealing with DTS, it states "skipping identical DTS-frames (seamless branching)", but when dealing with AC3, a second pass is initiated. Where can I find an explanation about which format is treat which way and why?

Somewhere, I read about that MakeMKV would take care of the duplicated frames via timecode-adjustments within the Matroska container instead of altering the audio data. Despite the fact that it doesn't seem to work out for me due to the glitches, can anyone give me more information about that from his experience?

A special case seems to be TrueHD in conjunction with seamless branching (e.g. Monsters University). Why and when is it going to be fixed in eac3to?

2. This is regarding gain adjustments in order to prevent overflows. I understand that in most cases, one has to reduce the gain to prevent clipping when downmixing since the channel levels sum up, but I don't get the idea for this need when extracting a AC3- oder DTS-track into individual PCM-tracks. Since eac3to starts a second pass in this case as well while overwriting the first extraction result, I assume that eac3to asks the AC3- or DTS-decoder to output the audio at a lower volume. I am aware that audio levels (intersample peaks, etc.) can slightly change during lossy encoding and decoding again, but I wonder why some files require a gain of -6dB. I would expect that the encoded highest "volume value" within the lossy codecs would be "calibrated" to equal more or less 0dBFS once decoded so I really ask to be enlightened here.

Furthermore, how do AVRs deal with overflows when decoding lossy codecs? They don't have any chance for a second pass, either.

Cheers,
Yoshi

tebasuna51

2nd May 2014, 16:04

1. [BUG]...
However, eac3to keeps screwing it up and truncates the beginning of the AC3-file independent of the movie. During extraction it complains about a weird audio delay which could not be fixed, for example:
...
3: AC3, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB, 7ms
...
[a03] Removing AC3 dialog normalization...
[a03] A remaining delay of +7ms could not be fixed.

Is not a BUG is a feature.
Here eac3to does not truncate the beginning of the AC3 because only can delete/insert AC3 frames (32 ms).
Even if the delay was big than 32 ms can only be silence, or the DVD was wrong created.

2. ... If so, it is however confusing that the dialog normalization is supposed to be removed when where is nothing to remove in this case.
Yes, the dialnorm: -27dB must be corrected to -31dB to avoid 4 dB of useless attenuation.

3. ..eac3to (sometimes) doesn't list all available playlists as reflected by the MPLS files. Is it an intended behaviour of eac3to to only list the main feature(s) when parsing a whole BDMV-structure?
Yes, the features are ordered by duration until a limit (don't remember, maybe 10m.?)

4. This concerns eac3to's method to access Arcsoft's DTS Decoder.
This is a question with different opinions.
I say than MakeMkv convert wrong the 'strange 7.1 setup' to PCM or FLAC.

The 'strange 7.1 setup' can't be converted lossless to other standard format. You can only preserve the track, don't try to convert losless to other format because is not possible.

5. [BUG] In case of missing access rights, the root folder of c: for instance, eac3to doesn't warn about it and simply doesn't create the files.
This is a user problem with the OS.

1. Seamless branching on Blu-ray
Seamless branching is a complex problem and don't exist always a perfect solution (without recode) to extract tracks.
Read http://forum.doom9.org/showthread.php?p=1600694#post1600694

2. This is regarding gain adjustments in order to prevent overflows. I understand that in most cases but I don't get the idea for this need when extracting a AC3- oder DTS-track into individual PCM-tracks.

Yes, the lossy encode and decode can produce overflows than aren't in sources. You can safely include the parameter -no2ndpass when only want decode to PCM tracks because are only artifacts from the lossy process.

Yoshi

3rd May 2014, 16:56

Hello tebasuna51,

thank you for your response. I wanna quote your parts vice-versa:

Is not a BUG is a feature.
Here eac3to does not truncate the beginning of the AC3 because only can delete/insert AC3 frames (32 ms).
Even if the delay was big than 32 ms can only be silence, or the DVD was wrong created.

Well said. :) However, my resulting AC3 files become significantly truncated by eac3to and I’m not talking about milliseconds, but several seconds and almost a minute in one extreme case which keeps confusing me. For instance, in the case of the Titanic DVD, I only hear the end of the THX-trailer at the beginning.

Yes, the dialnorm: -27dB must be corrected to -31dB to avoid 4 dB of useless attenuation.

My point was that there is no dialnorm to be removed at all when saving the stream as AC3 because eac3to doesn’t alter the data but demultiplexes it only. Correct me, if I’m wrong.

I say than MakeMkv convert wrong the 'strange 7.1 setup' to PCM or FLAC.

In which way is the result wrong?

The 'strange 7.1 setup' can't be converted lossless to other standard format. You can only preserve the track, don't try to convert losless to other format because is not possible.

Why is this not possible?

Seamless branching is a complex problem and don't exist always a perfect solution (without recode) to extract tracks.
Read http://forum.doom9.org/showthread.php?p=1600694#post1600694

Thanks, but I’ve to admit that I don’t fully understand your explanation there. Maybe you better refer me to more basic material since I seem to lack basic knowledge. However, from what I do understand, this neither explains why stand-alone players apparently have no problem with seamless branching at all, nor that the fact alone that we face several files, is no reason for sync problems. DVD-Video also distributes the stream over several files where frames might be cut of, but since it’s considered one continuous stream, no issue arises when processing it.

Yes, the lossy encode and decode can produce overflows than aren't in sources. You can safely include the parameter -no2ndpass when only want decode to PCM tracks because are only artifacts from the lossy process.

However, the same would be true for all other lossy formats, be it MP2, MP3, Vorbis, ADPCM, whatever. And I’ve never heard about the requirement of 2nd passes when decoding them.

tebasuna51

4th May 2014, 12:53

- If there are undesired cuts at the begining maybe there are some corrupt data.

- eac3to change the field DialNorm (if is not -31 dB) when extract the AC3 tracks. Use -keepDialnorm if you want.

- The DTS 'strange 7.1 setup' channels don't have a channel layout equivalence in other standard 7.1 formats: wav, flac, aac, ogg, ...
Then can't be converted without remix surround channels.

- Is not the same extract than play. When play the container timecodes can order to the player delay (+ or -) the audio in file changes.
But we can extract part of a frame because the track results corrupted. We can recode but not a exact audio cut at video frames change.

- Each decoder can do differents outputs. Is not normal than a decoder outputs audio with volume over 0 dB.
Maybe you can put a feature request to libav decoder.

This thread is very big, if you want more detailed explanation over some points, already explained in this thread, you can open a new thread.

kabelbrand

5th May 2014, 18:07

There are a new method over 3.24, read in http://forum.doom9.org/showthread.php?p=1600693#post1600693

Then the negative gain must be greater now.

And the differences between -downDpl and -downStereo, here http://forum.doom9.org/showthread.php?p=1600695#post1600695

Thanks Groucho and tebasuna for pointing me in the right direction!

ndjamena

2nd June 2014, 15:40

DarkSpace

2nd June 2014, 16:16

With 16 bit in 24 bit Lossless audio conversion is there any way to skip the first pass if I already know what's inside, or at least do the first pass without encoding and a forced second pass? I was thinking maybe -down16 but then realised that would just turn the 24 bit track with 8 bits of zeros into a 16 bit track with 5 bits of zeros. Or is there any way to just reverse the process, start with 16 bit and if it encounters a 24 bit section restart at 24, if it winds up being 20 bit it can do a third pass and I can sit there feeling stupid.
When you know that you've got a 16-bit audio track, you can use -down16 in combination with the undocumented -dontDither option.
However, when you don't know what's inside, I know of no way to just restart encoding with 24-bit, you'll have to make do with 24-bit and reduce that to 16t-bit if eac3to later detects that the whole track is only 16-bit...

ndjamena

2nd June 2014, 23:52

-dontDither

Cool. If nothing else that'll come in handy with TV Shows.

DarkSpace

3rd June 2014, 00:24

Cool. If nothing else that'll come in handy with TV Shows.
Is it that common for TV shows to use 16-bit audio, but to signal 24-bit audio (or just to use TrueHD with a 16-bit audio source)? I don't own many TV series' BDs, so I'm curious.

Sparktank

3rd June 2014, 02:05

undocumented -dontDither

Is there anywhere that states all the available switches?

The wiki doesn't list "-dontDither", nor does anything when running eac3to.
http://en.wikibooks.org/wiki/Eac3to/How_to_Use#Command_Line_Syntax

cyberbeing

3rd June 2014, 02:52

Is there anywhere that states all the available switches?

Probably something close to the following string dump:
mb
test
lowPriority
progressnumbers
neroaacenc=
demux
skipbytes
no2ndpass
check
fix
shutdown
logPids
logMkv
logDts
analyzeBitdepth
checkPitch
nero
sonic
arcSoft
libav
downmix
down6
down2
downDpl
downStereo
phaseShift
mixlfe
normalize
2pass
seekToIFrames
full
float32
skip
down32
down31
down30
down29
down28
down27
down26
down25
down24
down23
down22
down21
down20
down19
down18
down17
down16
down15
down14
down13
down12
down11
down10
down9
down8
192000
176400
96000
88200
48000
44100
24000
22050
8
7
6
5
4
3
2
1
16
24
fast
steep
r8brain
23.976
24.000
24.975
25.000
29.970
30.000
47.952
48.000
50.000
59.940
60.000
changeTo
speedupTo
slowdownTo
FPS value
quality=0.
quality=0
quality=1
quality=1.0
quality=1.00
blu-ray
bluray
core
silence
loop
edit=?:??:??,*ms
edit=?:??:??.???,*ms
slowdown
speedup
override
dontRepairDts
dontPatchDts
keepFullRange
dontCrop
stripPulldown
keepPulldown
resampleTo
big
little
mono
double7
simple
check16bit
ignorePitch
decodeHdcd
keepDialnorm
dontDither
ms
dB
192
448
640
768
1536
log

Sparktank

3rd June 2014, 11:33

Interesting.

Groucho2004

3rd June 2014, 12:10

Someone already collected all documented and undocumented switches a while ago:
http://en.wikibooks.org/wiki/Eac3to/How_to_Use#Command_Line_Syntax
Unfortunately, this does not include the options of the latest version.

shearerc

18th June 2014, 23:01

ndjamena

20th June 2014, 14:57

Is it that common for TV shows to use 16-bit audio, but to signal 24-bit audio (or just to use TrueHD with a 16-bit audio source)? I don't own many TV series' BDs, so I'm curious.

Not that I'm aware of, actually I asked as a feature request on the MakeMKV forum if the program could give a report at the end as to whether a 24 bit DTS-MA or TrueHD track turned out to be 16 bit or not during FLAC conversion. If that happens EAC3To going through both passes during the re-encode would be a waste of time. But of course Mike doesn't usually implement anything so it's unlikely to happen, which is why I had to find a way to justify getting the answer I got.

torturesauce

5th July 2014, 01:50

I converted a 24/44.1 6-channel DTS file to a FLAC using eac3to with Arcsoft DTS Decoder. The conversion went fine, but there's something weird: I played both files with foobar2000 and checked the waveforms with the Waveform Seekbar addon.

DTS:
http://i.imgur.com/t3Ia29S.jpg

FLAC:
http://i.imgur.com/XtkGNun.jpg

The LFE channel (on the bottom) is louder on the FLAC. Why is that so?

torturesauce

6th July 2014, 06:43

...and a REAL problem. There's a certain DTS-CD whose DTS-ES files I could extract properly with bsconvert, but once I put them in eac3to, it simply refuses to decode them with Arcsoft. It doesn't even produce an error. It just says:

DTS-ES, 6.1 channels, 0:04:58, 1235kbps, 44.1kHz
Remapping channels...
Decoding with ArcSoft DTS Decoder...

And that's it. Other DTS-ES files I've used work perfectly. I use Arcsoft 1.1.0.0.

Music Fan

6th July 2014, 10:39

I asked a question about thd+ac3 in another topic ;
http://forum.doom9.org/showthread.php?p=1685842#post1685842

I copy it here, the eac3to topic is more appropriate ;
is there a trick to mux a True HD track with an ac3 track (both having exactly the same duration) into a single .thd+ac3 file or this interleaved file can only be created from a single True HD track ?

It's just by curiosity, I understand that it's not very useful because if one really need to have another ac3 track (than the True HD track), it can be muxed into m2ts in addition to the .thd+ac3.
The only utility I see would be to avoid to increase the total bitrate (of a Blu-ray for example) if one need an ac3 track with a different mix than the True HD track.

I guess it is possible (but is maybe not available yet but could perhaps be added in eac3to), because just after having created the ac3 track from the True HD, thus just before interleaving them, there are 2 tracks. And I don't see the difference between taking this ac3 stream (existing as a temporary file) and another (already existing) for the interleaving operation.

tebasuna51

6th July 2014, 11:26

The LFE channel (on the bottom) is louder on the FLAC. Why is that so?
Yes, I can confirm than LFE decoded with Arcsoft 1.1.0.0 is 3 dB louder than the LFE channel decoded with Foobar2000 decoder or with the AviSynth NicAudio decoder.

There's a certain DTS-CD whose DTS-ES files I could extract properly with bsconvert, but once I put them in eac3to, it simply refuses to decode them with Arcsoft.
Please upload a sample (dtswav) before extract with bsconvert.

tebasuna51

6th July 2014, 11:38

is there a trick to mux a True HD track with an ac3 track...?
Nope, now you can create a thd+ac3 track with the ac3 created for eac3to with the thd input.

It's just by curiosity, I understand that it's not very useful...
I guess it is possible (but is maybe not available yet but could perhaps be added in eac3to)
Yes, it's not very useful...
I don't know if madshi can add this option.

Music Fan

6th July 2014, 12:15

filler56789

6th July 2014, 13:22

SeeMoreDigital

6th July 2014, 14:41

Ok, and do you know how the interleaved ac3/True HD tracks are created by professional tools for Blu-ray's ?
Because I guess that unlike eac3to, they don't create an ac3 stream from the THD stream, they probably take 2 streams (even when they come from the same mix) and interleave them.It would appear that tebasuna51's UsEac3To v1.0.0 is able to generate a 5.1Ch Dolby Digital stream from a 'vanilla' Dolby TrueHD stream and multiplex/interleave it correctly to provide a Dolby Digital core ;)

Music Fan

6th July 2014, 16:03

I don't need to get the core (eac3to can do it and also TSMuxer IIRC), I'd like to know if an ac3 stream and a THD stream without core can be muxed together to get a .thd+ac3 (which is usually done by re-creating the core from a THD track whose core has been removed).

Boulder

14th July 2014, 10:25

Yes, I can confirm than LFE decoded with Arcsoft 1.1.0.0 is 3 dB louder than the LFE channel decoded with Foobar2000 decoder or with the AviSynth NicAudio decoder.Does that mean that the Sonic decoder is actually the preferred one if you don't have a 7.1ch file to decode?

Frozen Fractals

17th July 2014, 22:15

I am getting an error when extracting a TrueHD stream. Error states that the "lossless check failed." However, the process completes itself regardless of the error and I get 8 individual wave PCM files (corresponding to each channel)...but, I am unsure of the integrity of the resulting wave PCM files. Log posted below.

C:\Program Files (x86)\eac3to>eac3to.exe E:\Audio\Brave\Brave.mka E:\Audio\Brave
\Brave_pcm.wavs
MKA, 1 audio track, 1:33:37
1: TrueHD, English, 7.1 channels, 48kHz
Track 1 is used for destination file "Brave_pcm.wavs".
a01 Extracting audio track number 1...
a01 Decoding with libav/ffmpeg...
a01 Writing WAVs...
a01 Creating file "E:\Audio\Brave\Brave_pcm.SL.wav"...
a01 Creating file "E:\Audio\Brave\Brave_pcm.BR.wav"...
a01 Creating file "E:\Audio\Brave\Brave_pcm.SR.wav"...
a01 Creating file "E:\Audio\Brave\Brave_pcm.L.wav"...
a01 Creating file "E:\Audio\Brave\Brave_pcm.BL.wav"...
a01 Creating file "E:\Audio\Brave\Brave_pcm.C.wav"...
a01 Creating file "E:\Audio\Brave\Brave_pcm.LFE.wav"...
a01 Creating file "E:\Audio\Brave\Brave_pcm.R.wav"...
-[truehd @ 00307360] Lossless check failed - expected 00, calculated 15.
[truehd @ 00307360] Lossless check failed - expected 00, calculated e7.
--[truehd @ 00307360] End of stream indicated.
[truehd @ 00307360] Lossless check failed - expected 00, calculated 91.
-[truehd @ 00307360] Lossless check failed - expected c2, calculated 6b.
-------------------------------------------------------------------[truehd @ 003
07360] Lossless check failed - expected 00, calculated 9a.
---[truehd @ 00307360] Lossless check failed - expected 00, calculated 9f.
-----[truehd @ 00307360] End of stream indicated.
[truehd @ 00307360] Lossless check failed - expected 51, calculated 2a.
[truehd @ 00307360] End of stream indicated.
a01 The original audio track has a constant bit depth of 20 bits.
eac3to processing took 4 minutes, 48 seconds.
Done.

C:\Program Files (x86)\eac3to>

von Suppé

18th July 2014, 08:11

I remember a same occasion from long time ago.
Can you to try to input the TrueHD stream and directly output to multichannel FLAC in eac3to? I know it sounds not logical because eac3to will have to decode the TrueHD stream first anyways and then convert, but I know it helped me once. Couldn't figure out why though.
Not sure if it'll help you in this case.

Music Fan

18th July 2014, 09:54

Hi,
what is the parameter to change the pitch (or not) of the sound when converting 25 to 24 fps (and vice versa), I guess one has the choice (because sometimes the pitch has to change -to adjust it when it was not done during Pal speed up-, sometimes it has not to).
I guess that -slowdown and -speedup change the pitch and -changeTo.. does not but I'm not sure, thanks.

Overdrive80

18th July 2014, 10:11

The conversions of eac3to does not fix pitch, you will need http://avisynth.nl/index.php/SSRC

Music Fan

18th July 2014, 10:28

The conversions of eac3to does not fix pitch, you will need http://avisynth.nl/index.php/SSRC
Thanks, this filter only works with avisynth I guess.
Does it always change the pitch ? I don't see option to avoid it.

In this case, what's the difference between -slowdown (or-speedup) and -changeTo in eac3to ?

Reino

18th July 2014, 15:05

Hi,
what is the parameter to change the pitch (or not) of the sound when converting 25 to 24 fps (and vice versa)I'm not too sure about eac3to, someone else can probably help you with that, but alternatively you can use AviSynth, FFMpeg, or SoX as explained in this post.

sneaker_ger

18th July 2014, 21:30

Does it always change the pitch ?
Yes, eac3to always does.

In this case, what's the difference between -slowdown (or-speedup) and -changeTo in eac3to ?
No difference. (-changeTo is just useful for when the auto-fps-guessing fails)

Music Fan

18th July 2014, 22:50

Ok.

Yes, eac3to always does.
I was talking about SSRC.

Music Fan

18th July 2014, 22:51

I'm not too sure about eac3to, someone else can probably help you with that, but alternatively you can use AviSynth, FFMpeg, or SoX as explained in this post.
Thanks, I will have a look on this.

upyzl

7th August 2014, 04:20

problem for compiling eac3to-ffmpeg...(I want to try latest ffmpeg to see any improvement, especially in TrueHD, as eac3to stagnates for a long time)

I'm followed by ./legal stuff/ffmpeg steps (slightly mod for ac3dec for match latest ffmpeg also, same config)
but when make install here: (tried TDM-GCC 4.7.1-2 / TDM-GCC 4.8.1-3)
common.mak:18: *** unterminated call to function `foreach': missing `)'. Stop.

also tried normal mingw (gcc 4.8.3)
another error
common.mak:152: *** missing separator. Stop.

but I don't find any strange in common.mak...

line 18-19:$(foreach VAR,$(BRIEF), \
$(eval override $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
(I've tried change to one line, it is the same for TDM-GCC)

line 152:$(eval $(RULES))

full:#
# common bits used by all libraries
#

# first so "all" becomes default target
all: all-yes

ifndef SUBDIR

ifndef V
Q = @
ECHO = printf "$(1)\t%s\n" $(2)
BRIEF = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM

MSG = $@
M = @$(call ECHO,$(TAG),$@);
$(foreach VAR,$(BRIEF), \
$(eval override $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
$(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
endif

ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample

# NASM requires -I path terminated with /
IFLAGS := -I. -I$(SRC_PATH)/
CPPFLAGS := $(IFLAGS) $(CPPFLAGS)
CFLAGS += $(ECFLAGS)
CCFLAGS = $(CPPFLAGS) $(CFLAGS)
ASFLAGS := $(CPPFLAGS) $(ASFLAGS)
CXXFLAGS += $(CPPFLAGS) $(CFLAGS)
YASMFLAGS += $(IFLAGS:%=%/) -Pconfig.asm

HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
LDFLAGS := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS)

define COMPILE
$(call $(1)DEP,$(1))
$($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $<
endef

COMPILE_C = $(call COMPILE,CC)
COMPILE_CXX = $(call COMPILE,CXX)
COMPILE_S = $(call COMPILE,AS)
COMPILE_HOSTC = $(call COMPILE,HOSTCC)

%.o: %.c
$(COMPILE_C)

%.o: %.cpp
$(COMPILE_CXX)

%.o: %.m
$(COMPILE_C)

%.s: %.c
$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<

%.o: %.S
$(COMPILE_S)

%_host.o: %.c
$(COMPILE_HOSTC)

%.o: %.rc
$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<

%.i: %.c
$(CC) $(CCFLAGS) $(CC_E) $<

%.h.c:
$(Q)echo '#include "$*.h"' >$@

%.ver: %.v
$(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@

%.c %.h: TAG = GEN

# Dummy rule to stop make trying to rebuild removed or renamed headers
%.h:
@:

# Disable suffix rules. Most of the builtin rules are suffix rules,
# so this saves some time on slow systems.
.SUFFIXES:

# Do not delete intermediate files from chains of implicit rules
$(OBJS):
endif

include $(SRC_PATH)/arch.mak

OBJS += $(OBJS-yes)
SLIBOBJS += $(SLIBOBJS-yes)
FFLIBS := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
TESTPROGS += $(TESTPROGS-yes)

LDLIBS = $(FFLIBS:%=%$(BUILDSUF))
FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)

OBJS := $(sort $(OBJS:%=$(SUBDIR)%))
SLIBOBJS := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
TESTOBJS := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
HOSTOBJS := $(HOSTPROGS:%=$(SUBDIR)%.o)
HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
TOOLS += $(TOOLS-yes)
TOOLOBJS := $(TOOLS:%=tools/%.o)
TOOLS := $(TOOLS:%=tools/%$(EXESUF))
HEADERS += $(HEADERS-yes)

PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))

SRC_DIR := $(SRC_PATH)/lib$(NAME)
ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
HOBJS = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
checkheaders: $(HOBJS)
.SECONDARY: $(HOBJS:.o=.c)

alltools: $(TOOLS)

$(HOSTOBJS): %.o: %.c
$(COMPILE_HOSTC)

$(HOSTPROGS): %$(HOSTEXESUF): %.o
$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)

$(OBJS): | $(sort $(dir $(OBJS)))
$(HOBJS): | $(sort $(dir $(HOBJS)))
$(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
$(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
$(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
$(TOOLOBJS): | tools

OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))

CLEANSUFFIXES = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
DISTCLEANSUFFIXES = *.pc
LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a

define RULES
clean::
$(RM) $(OBJS) $(OBJS:.o=.d)
$(RM) $(HOSTPROGS)
$(RM) $(TOOLS)
endef

$(eval $(RULES))

-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d))

could any one give me a guide...?

Edit: after
$git config --global core.autocrlf false
and git revert
use make 3.82 instead of 3.81
Now no common.mak error, but get compile error, I'll try later...

Edit2: using Tag n2.2.6 compile successful, but seems entry point avcodec_open changed from avcodec-54.dll to avcodec-55.dll(rename avcodec-54.dll when use), eac3to cannot call it...hope for official eac3to update (54/55 should be avcodec API version?)

RRAH

11th August 2014, 20:48

No matter the source/decoder or encoder, when you downmix 6.1 to 5.1 with eac3to need use:

-0,1,2,3,5,6,4 -down6

Hello,

Is this still required or is the problem solved?

Thanks

tebasuna51

12th August 2014, 02:06

@RRAH
Solved with eac3to v3.27

Music Fan

12th August 2014, 08:39

I converted 1 mpa and 1 ac3 file, both in 48-16 to wav, and I saw this message during conversion : "reducing depth from 64 to 24 bits".
Does it mean there is always a 64 bits step during process ?
Or is it a difficulty for eac3to to interpret the format as 16 bits with some files and thus consider them as 64 bits ?
When I add -down16, it is well converted in 16 bits.
Does is stay in 16 bits when -down16 is added or there is a 64 bits step anyway (16 => 64 => 16) ?

RRAH

12th August 2014, 09:04

@RRAH
Solved with eac3to v3.27

Thanks @Tebasuna :-)

nevcairiel

12th August 2014, 09:17

tebasuna51

12th August 2014, 10:36

I converted 1 mpa and 1 ac3 file, both in 48-16 to wav,...
Lossy compressed audio don't have bitdepth, only lossless compresion can have the precission to restore the original bitdepth.

Lossy decoders (AC3, standard DTS, MP3, MP2, AAC, ...) works internally decoding samples in frequency domain to float samples (32 or 64 bits like eac3to) in time domain (PCM).

Any subsequent conversion is always unnacurate with less precission than 16 bits, eac3to select by default convert to 24 bits int, but you can choice preserve the 64 bits float with -full, convert to 32 bit int with -down32 or convert to 16 bits int with -down16.

Music Fan

12th August 2014, 10:51

Thanks, that's amazing, I never heard of it.
But I guess that 16 bit waves compressed in ac3, aac, mpa ... stay in 16 bit in a certain way ? For players and analyzers (MediaInfo), it's supposed to be 16 bit.
A lossy compressed 24 bit file is bigger than a 16 bit lossy compressed file, which means there is difference in quantification, even if both are better decoded in 64 bits, right ?

edit : I was wrong, players and analyzers do not show the quantification but only the frequency with lossy compressed files, except with some files (for a TS including ac3 @ 448 k, MediaInfo indicates 16 bits).
When you say "preserve the 64 bits", shouldn't you say preserve the "up-quantification" done by eac3to ?
Because if the original wave was in 16 bit, the decompressed file (coming from this wav) should keep its original quantification, otherwise there is big waste of space.

Boulder

12th August 2014, 11:07

Yes, I can confirm than LFE decoded with Arcsoft 1.1.0.0 is 3 dB louder than the LFE channel decoded with Foobar2000 decoder or with the AviSynth NicAudio decoder.I tried to reproduce this earlier but couldn't. I decoded a normal DTS-HD MA 5.1ch file to WAV with both Arcsoft 1.1.0.0 and Sonic and the results were the same.

nixo

12th August 2014, 13:12

I think tebasuna51 and torturesauce were discussing lossy DTS, not DTS-HD.

--
Nikolaj

tebasuna51

12th August 2014, 21:45

...A lossy compressed 24 bit file is bigger than a 16 bit lossy compressed file, which means there is difference in quantification, even if both are better decoded in 64 bits, right?
...
When you say "preserve the 64 bits", shouldn't you say preserve the "up-quantification" done by eac3to ?
Because if the original wave was in 16 bit, the decompressed file (coming from this wav) should keep its original quantification, otherwise there is big waste of space.
The 64 bits float is the better aproach to the original audio and if you want recode to another format you can use it, like eac3to do internally.

There is no sense store a wav file from a lossy format, if you decode to wav is for edit and after recompress, then preserve the best aproach you can manage until you recode to the desired format.

..(for a TS including ac3 @ 448 k, MediaInfo indicates 16 bits).
This is a wrong info from MediaInfo, don't belive all than MediaInfo say. Also for standard DTS most the times show 24 bits and is wrong.

Music Fan

12th August 2014, 21:59

Ok, but how is stored the quantification in lossy formats ? There is no quantification anymore ?
I don't understand this concept easily.

microchip8

12th August 2014, 22:00

This is a wrong info from MediaInfo, don't belive all than MediaInfo say. Also for standard DTS most the times show 24 bits and is wrong.

No, it's not wrong. The DD spec says max is 16 bits. If you want higher, like 24bit, only TrueHD/DD+ support that

madshi

12th August 2014, 22:54

A lossy compressed 24 bit file is bigger than a 16 bit lossy compressed file
No, it's not. AFAIK, the first step of usual lossy encoders like AC3 or DTS is to convert the WAV to frequency domain (using FFT), which results in floating point numbers. The data is then compressed in the frequency domain. So basically the input bitdepth doesn't matter *at all*. Actually, a higher bitdepth input might actually improve the compression efficiency ever so slightly because it could have a lower noise floor and less quantization artifacts. Noise and quantization artifacts make life harder for the encoder.

Decoders do the opposite thing: They decompress the data, which is then still in frequency domain, and then convert back to time domain, which produces floating point data. So a lossy decoder natively outputs 64bit floating point.

bennynihon

12th August 2014, 23:55

bennynihon

13th August 2014, 00:43

Does eac3to have a hard limit of being able to read just 300 playlists from a disc? The Redbox version of the Divergent Blu-ray has hundreds of playlists and despite other tools like MakeMKV detecting them all, eac3to stops at 300). This is a problem, since playlist 810.mpls is the correct one.

UPDATE: Disregard. The .mpls was 810, but I see that eac3to simply enumerates them sequentially

Music Fan

13th August 2014, 10:55

No, it's not. AFAIK, the first step of usual lossy encoders like AC3 or DTS is to convert the WAV to frequency domain (using FFT), which results in floating point numbers. The data is then compressed in the frequency domain. So basically the input bitdepth doesn't matter *at all*. Actually, a higher bitdepth input might actually improve the compression efficiency ever so slightly because it could have a lower noise floor and less quantization artifacts. Noise and quantization artifacts make life harder for the encoder.

Decoders do the opposite thing: They decompress the data, which is then still in frequency domain, and then convert back to time domain, which produces floating point data. So a lossy decoder natively outputs 64bit floating point.
Interesting.
Does it mean there is less differences between dts 16 bits and dts 24 bits than between pcm 16 bits and pcm 24 bits ?
Despite the fact that the input bitdepth doesn't matter at all, is there a way to evaluate the quality (or the corresponding bitdepth) of a lossy compressed file ?

LigH

13th August 2014, 12:12

What do you mean by "dts 16 bits" and "dts 24 bits"?

As just explained, most compressed formats don't save "samples" (the volume at discrete moments of time), but instead a limited sound spectrum over brief durations. The less precision the original audio had, if it was stored as integer samples, the harder this lack of precision can be transformed into a sound spectrum; imagine it like laying a smooth but slightly stiff carpet over a stairway with stairs of different heights, just the same depth. The compressed format saves the curves in the shape of the carpet. You can have the same shape of carpet laying over different kinds of stairs, with more or less distance of the stair edges to the carpet shape...

The "quality" of the encode may represent the distance of the carpet shape to the edges of the original staircase. If you don't have the staircase available anymore, you can't judge how good the carpet once fit it when it was bent over it with more or less pressure.

Decoding means building a new staircase which fits under the carpet. You can do that with more or less precision, but only compared to the carpet shape, not to the original staircase.

Music Fan

13th August 2014, 12:26

Ok, thanks.

What do you mean by "dts 16 bits" and "dts 24 bits"?
dts is supposed to exist in several levels of quality, I guess you have heard of dts 96/24 (while generally dts is supposed to be 48/16 or 48/20), but now I understand that this only concerns the accepted input formats for the dts encoder (however, one can see the dts 96/24 logo on some dvd's).

LigH

13th August 2014, 12:52

In case of DVD Video, dts Coherent Acoustics was primarily supposed to reduce 6-channel 16 bit PCM to the bitrate of mono or stereo 16 bit PCM, because S/P-DIF is not able to submit higher bitrates than the one of 48 kHz 16 bit stereo (1536 kbps) to A/V receivers. Similar goals will have been used for "dts Audio CDs", to stay below the bitrate of 44.1 kHz 16 bit stereo (1411.2 kbps) but support multi-channel audio.

More precise dts variants are possibly extensions to the "core" format.

But in any case, it doesn't mean that 16 or 24 bit samples are stored inside the compressed audio, just that the encoding is precise enough to reproduce up to 16 or 24 bits of precision when decoded to PCM again and compared to the original. I would believe that in "best cases". In complex cases (like hardrock), precision may drop by a few bits, but won't be noticable due to psychoacoustic effects.

Music Fan

13th August 2014, 14:19

Ok.
I'm astonished you consider hardrock as a complex case, I believed it was simpler than classical music which contains a lot of frequencies and harmonics.
Or you mean it is more complex because of its higher dynamic compression (which is often the case in hardrock), which makes instruments more difficult to detach from each other ?

LigH

13th August 2014, 15:13

Classical music is often easier to compress due to a quite restricted spectrum, having mostly clean sounds based on a few dominating tones with harmonics. Most audio compression algorithms (except Opus) can handle simple tones and sounds with only few harmonics quite easily.

In contrast, especially the sound of overdriven electro guitars is much closer to "noise" than to "sound", it is quite hard to predict for common audio compression.

tebasuna51

13th August 2014, 19:37

No, it's not wrong. The DD spec says max is 16 bits. If you want higher, like 24bit, only TrueHD/DD+ support that
I read in DD docs than the samples in frequency domain have a precission equivalent to a bitdepth of 20 bits.

The question here is not the bitdepth (16, 20 or 24) but in the word equivalent, when the user read this info can wrong supose than the decoded output have a precission of 16/24 bits and that is not true, the precission is always less than 16 bits in lossy formats.

Interesting.
Does it mean there is less differences between dts 16 bits and dts 24 bits than between pcm 16 bits and pcm 24 bits ?
Despite the fact that the input bitdepth doesn't matter at all, is there a way to evaluate the quality (or the corresponding bitdepth) of a lossy compressed file ?
The unique measure of quality of a lossy compressed file is the bitrate and the compressor efficiency.

A DTS have in the header a bitdepth info of the original PCM source (not always true, some Surcode versions put always 24 even if the source is 16 bits), of course the compressed file can be better if the source have a better precission, but, like I say before, associate a bitdepth info to a lossy format only mistake the users like you can see in this discussion.

microchip8

13th August 2014, 19:51

I read in DD docs than the samples in frequency domain have a precission equivalent to a bitdepth of 20 bits.

The question here is not the bitdepth (16, 20 or 24) but in the word equivalent, when the user read this info can wrong supose than the decoded output have a precission of 16/24 bits and that is not true, the precission is always less than 16 bits in lossy formats.

From what I understand, lossy format (don't know if all, but at least for MP3) use FP math to build their output values. From this, there is no true or correct bit depth. So, it is virtually impossible for MediaInfo to correctly identify the bit depth, but it can only estimate it

SeeMoreDigital

13th August 2014, 20:36

dts is supposed to exist in several levels of quality, I guess you have heard of dts 96/24 (while generally dts is supposed to be 48/16 or 48/20), but now I understand that this only concerns the accepted input formats for the dts encoder (however, one can see the dts 96/24 logo on some dvd's).And when you play DTS 96/24 sources through a (supporting) surround sound amplifier you'll even see a dedicated 'DTS 96/24' icon light up... It's meaningless marketing pap!

Mathematically a 96/24 DTS file size should be much larger than a 48/16 DTS file size... But they're the same!

bennynihon

13th August 2014, 22:45

How does one show the stdout when piping the output to a second audio encoder? I do something like this:

eac3to I: 1) 2: video.mkv 3: audio.ac3 3: stdout.wav -downDpl | D:\Apps\qaac\qaac -V 32 --ignorelength -o audio.m4a -

But that prevents me from seeing the progress of eac3to.

I know tee in Unix/Linux does something similar, but usually it's used to log the stdout and still see it as it progresses on the screen. Tried using the GNUWin32 version of tee and for some reason it doesn't work, even though from what I read this should both display stdout and pipe it to the next command (con being the Windows equivalent of /dev/tty)

eac3to title.mkv 1: video.mkv 2: audio.ac3 2: stdout.wav -downDpl | tee con | D:\Apps\qaac\qaac -V 32 --ignorelength -o audio.m4a -

How can I display progress while piping stdout to another encoder?

LigH

14th August 2014, 08:06

There are three common ways to create text output on the console:

a) STDOUT = file descriptor 1, redirectable via '>' and '|'
b) STDERR = file descriptor 2, redirectable via '2>'
c) direct video memory access, not redirectable (for those who used Turbo Pascal: when using unit CRT)

I wouldn't know without testing which output technique eac3to prefers for its usual progress reports while writing to a file. If it is STDOUT, then it will possibly suppress the progress output completely while piping?

jpsdr

14th August 2014, 08:52

I think eac3to is in Delphi. It's not Turbo Pascal, but...:rolleyes:

Music Fan

14th August 2014, 10:53

I read in DD docs than the samples in frequency domain have a precission equivalent to a bitdepth of 20 bits.
That seems contradictory with this ;
the precission is always less than 16 bits in lossy formats.
the compressed file can be better if the source have a better precission
Does it mean the DD docs is wrong ?
And thus, why encode from 24 bits sources ?

The unique measure of quality of a lossy compressed file is the bitrate and the compressor efficiency.
And sample frequency, right ?

tebasuna51

14th August 2014, 21:31

Does it mean the DD docs is wrong?
For what? The samples in frequency domain are stored with a precission equivalent but when recover the samples in time domain don't exist the precission of the source. For that is lossy compression and not lossless.

And thus, why encode from 24 bits sources?
I say because the samples in frequency domain are better calculated with a source more precisse.

And sample frequency, right?
Right, but when there aren't enough bitrate to store all frequency samples the high frequencies are sacrified.

Thunderbolt8

15th August 2014, 04:13

whats the correct file format to choose when demuxing DVB subtitles? I tried .srt and .sup, but both gave me errors.

Music Fan

15th August 2014, 12:41

I'm not sure eac3to handle DVB-SUB (bitmap based format) but Subtitle Edit does (it needs the TS file).
If you are talking about teletext subtitles format, TSDoctor, CCExtractor and maybe VideoRedo and ProjectX can convert it in srt (they also need the TS file).

Music Fan

15th August 2014, 12:43

I say because the samples in frequency domain are better calculated with a source more precisse.
I had understood but it seems contradictory with the fact that the precision is less than 16 bits in lossy formats.

Overdrive80

15th August 2014, 23:09

I had understood but it seems contradictory with the fact that the precision is less than 16 bits in lossy formats.

If you original source is 24 bit depth, the encoder will can calculate better because precision is greater than 16 bit depth source.

This and you say is contradictory, but not tebasuna51 says.

Music Fan

15th August 2014, 23:23

What is contradictory in my message ? I just asked questions.:confused:

the samples in frequency domain have a precission equivalent to a bitdepth of 20 bits.
the precission is always less than 16 bits in lossy formats
Logical for you ?:scared:

nevcairiel

16th August 2014, 08:20

The concept of bit depth just doesn't really translate to the frequency domain transforms used by the lossy codecs. Trying to speak about it will always sound contradictory since it just doesn't fit.

tebasuna51

16th August 2014, 08:29

When a lossy encoder make the change between time domain to frequency domain not all components fit in the bitrate assigned and discard the less important (for low volume or high frequency), for that you can't recover the original precission.

For example the DTS-MA make the 'core' like lossy standard DTS with a CBR bitrate and all components than not fit are stored in a subframe VBR, now we can recover the original precission.

foxyshadis

20th August 2014, 23:13

Here's why input bit depth matters. Obviously this is very artificial to make a point, but the higher the bitdepth, the less noise gets sprayed into the rest of the spectrum, which allows the encoder to encode the sound more accurately in less bits.

(One thing I have to comment on, everyone is assuming that all codecs use float internally, but most don't. Quite a few use DCTs/MDCTs optimized specifically for the bit-depth they internally operate at, which is usually 16-bit or 24-bit, though a few implementations can optionally use float. By the same measure, the specs usually only specify output at a certain bit-depth with a certain internal bit-depth; decoders going beyond that can't be "compliant" even if they can be theoretically slightly higher quality. Personally, I feel that the need for optimized audio algorithms is past, when I can do 100x realtime transcoding, and an all-float chain is important, even if it's not perfectly to-spec.)

Music Fan

21st August 2014, 10:16

Thanks for this explanation.
You didn't write whats is the difference between these 2 images, I guess the source of the first is 24 bit and 16 for the second, right ?

LigH

21st August 2014, 10:39

It doesn't even matter much how many bits of resolution there is exactly in these waveforms. May it be only 8 bit on the right side. But what you see is the bad side effect of "quantization". If samples have too little resolution, the conversion to a frequency spectrum will have artefacts. Backwards the same problem: If the frequency spectrum parameters have too low precision, the reconstructed waveform will have artefacts. Lossy compression is mostly about reducing the precision of the frequency spectrum, though, so there is much effort to try to make the resulting artefacts unobvious, or at least not so annoying...

madshi

21st August 2014, 10:53

The point foxyshadis is making is that the higher the source quality is the easier it is for the lossy encoder to produce a (reasonably) good quality encode with a (relatively) low bitrate. The situation is very different when comparing lossy and lossless encoders: With lossy encoders it's a good idea to feed the encoder the best quality you have. With lossless encoders it's the opposite situation: There higher bitdepths cost additional bitrate. The reason for that is the lossless encoders are not allowed to throw any data away. They have to keep every bit, even noise. And higher bitdepth signals have somewhat random values in the very lowest bits, which are hard to perfectly remember. A lossy encoder can simply throw those away, or encode a signal which only mostly resembles the original. A lossless encoder is not allowed to do such things.

Summary: Feeding a higher quality source into the encoder always improves the overall audio quality, for both lossy and lossless encoders. However, for lossy encoders a higher quality source actually improves the compression efficiency. For lossless encoders a higher quality sources increases the file size instead.

foxyshadis

21st August 2014, 22:05

Whoops, I should have labeled them; the first is 16-bit and the second 8-bit. Real-life noise is more complicated, and obviously float vs 24 vs 16 is much smaller than 16 vs 8, but this was mostly to make the differences visible. Lower bit-depth always leads to spectral leakage. Of course spectral noise (and regular noise) from lossy compression is even worse, but you may as well not compound the problem when transcoding if given the chance.

mville

24th August 2014, 12:11

I've written a vb class to run the eac3to.exe.

In this class I have redirected the error output, but what I get is a truncated string of the full error.

Is there any error documentation/examples available to help me troubleshoot the problem?

LigH

25th August 2014, 10:21

As far as I remember, eac3to has a strange technique of formatting the text output, using many backspace chars instead of a carriage return. Have a look at the output redirected to a text file in a CLI window, using a hex editor or text viewer supporting non-printable characters.

NikosD

25th August 2014, 13:56

Is there a way to avoid the automatic "Patching bitdepth to 24 bits..." procedure when I use the -core option extracting DTS core from DTS-HD ?

Because sometimes DTS-HD is 16bits and I see this phrase during processing.

Does this conversion lead to better, worse or equal results compared to leaving 16bit in DTS-HD stream ?

Thanks!

nevcairiel

25th August 2014, 15:46

The core is lossy audio, it has no bitdepth, even if the lossless DTS-HD was 16-bit.
Lossy audio should be decoded to an as high bitdepth as the decoder will give you, and thats with ffmpeg 32-bit floating point, or with the ArcSoft decoder 24-bit integer.

NikosD

25th August 2014, 18:41

mville

25th August 2014, 22:45

As far as I remember, eac3to has a strange technique of formatting the text output, using many backspace chars instead of a carriage return.

Is there any documentation for eac3to and error handling?

foxyshadis

26th August 2014, 02:23

But I don't have ArcSoft DTS-HD decoder installed or any other DTS-HD decoder.

Also, I think that the -core process doesn't involve any decoding of DTS-HD stream.

It just extracts the DTS core from DTS-HD without decoding.

If you're going to throw away the HD correction, let the core be 24-bit. It's only labeled 16-bit because the lossless correction is 16-bit, it'll sound fine at 24-bit.

NikosD

26th August 2014, 05:04

OK.

Let me rephrase my question.

Is the app really doing something during "Patching bitdepth to 24bits..." process ?

If not, why the message ?

If yes, what's the meaning of bitdepth in a lossy (DTS) format and what is technically doing ?

Thanks!

tebasuna51

26th August 2014, 09:01

...
Is the app really doing something during "Patching bitdepth to 24bits..." process ?

Yes, change the fields "Source PCM Resolution" in the DTS frames header from 16 to 24 to remember the users decode to this bitdepth at least.

If yes, what's the meaning of bitdepth in a lossy (DTS) format and what is technically doing ?

Nothing.

NikosD

26th August 2014, 09:20

Thanks!

rhaz

26th August 2014, 10:29

Hello. Anyone can explain what does this message mean "[dca @ 002f71a0] Number of channels changed in DCA decoder (6 -> 2)" I tried converting the .dts to .wavs or any other format and I always get this message before decoding starts, after its finished, I get a shrinked file. Original file is 1h 20 minutes long, a resulting file ends up in 26 minutes long only. Now it doesn't cut off or anything, it shrinks that 1h20min to 26 minutes long file and when you play it its impossible to understand anything, its all mashed up together or something.

I've been using eac3to cmd line for converting formats and I never had this problem before. Anyone knows how do I successfully convert this particular .dts? I'm sure it has to do something with that DCA decoder message, I've never seen one before. Cheers.

LigH

26th August 2014, 13:26

I don't know for certain, but it looks like you got a mix of 5.1 and 2.0 in your audio stream. I wonder if you possibly tried to convert audio streams extracted from VOBs as they are authored on DVD (file mode), not of an extracted linear PGC (IFO mode).

rhaz

26th August 2014, 13:30

It's from BluRay. I've done this million times. I'm doing it right now. And it all works, except this one, I'm stuck with it. What's wrong with this DTS! Arrgh...

filler56789

26th August 2014, 13:53

Need MORE info :logfile:

nevcairiel

26th August 2014, 14:22

Sounds like it doesn't realize that the format changed, and writes the 2 channel content as 6 channel. It would sound horrible and end up shorter (a third).
If you can, cut off a small piece of the start, until the message goes away.

Music Fan

26th August 2014, 16:49

Yes, change the fields "Source PCM Resolution" in the DTS frames header from 16 to 24 to remember the users decode to this bitdepth at least.
Does it really change anything for decoding ? Because I believe most of decoders work in 24 bit for a while and even for 16 bit pcm sources, DAC often make an "upscale" in 24 bit.

And I guess you mean that some DAC do not work in floating point and thus need to know (with the dts header) how to decode the dts (in 16 or 24 bit).

tebasuna51

27th August 2014, 01:48

Does it really change anything for decoding ? Because I believe most of decoders work in 24 bit for a while and even for 16 bit pcm sources, DAC often make an "upscale" in 24 bit.

And I guess you mean that some DAC do not work in floating point and thus need to know (with the dts header) how to decode the dts (in 16 or 24 bit).
This was explained already.
Don't change anything in the decode and the decoders don't need know the bitdepth of source PCM.

NikosD

28th August 2014, 09:26

After using Haali installer for eac3to, I had problems with the Haali splitter which I managed to disable.

I have two questions though:

1) Is there a way to install just the Haali matroska muxer without the rest of the installer ?

Is there a separate installer/ option to do that ?

2) Is there something special in this specific matroska muxer that makes it mandatory for eac3to ?

The full installation is problematic due to Haali splitter.

Thanks!

tebasuna51

28th August 2014, 09:37

Without Haali matroska muxer you can extract video only like elementary streams: .h264, .vc1 or .m2v

NikosD

28th August 2014, 09:40

OK, but it could be done by any matroska muxer, not Haali explicitly.

filler56789

28th August 2014, 10:55

After using Haali installer for eac3to, I had problems with the Haali splitter which I managed to disable.

I have two questions though:

1) Is there a way to install just the Haali matroska muxer without the rest of the installer ?

Unfortunately, Haali's Matroska Muxer is "inside" the files splitter.ax and mkx.dll. And sadly JEEB either is too busy or doesn't have permission to fix the design flaws of the Haali "bundle" :(

Overdrive80

29th August 2014, 15:42

OK, but it could be done by any matroska muxer, not Haali explicitly.

I install CCCP codecs, and after install this keys:

Windows Registry Editor Version 5.00

[HKEY_CURRENT_USER\Software\Haali]

[HKEY_CURRENT_USER\Software\Haali\Matroska Splitter]
"input.fonts"=dword:00000001
"input.linking"=dword:00000001
"ui.trayicon"=dword:00000001

[HKEY_CURRENT_USER\Software\Haali\Video Renderer]
"BT.709"=dword:00000002

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\HaaliMkx]
"StartMenuFolder"="Haali Media Splitter"
"rmp4"="1"
"rOGM"="1"
"rts"="1"
"rm2ts"="1"
"rEXT"="1"
"rFT"="1"
"rWMP"="1"
"Install_Dir"="C:\\Program Files (x86)\\Haali\\MatroskaSplitter"

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\HaaliMkx\Input]
"avi.1"="ffffffff00000000ffffffff,524946460000000041564920,{51A00247-40A8-4845-9F17-7DBFCC9A8783}"
"mp4.1"="00000000ffffffff,0000000066747970,{B3DE7EDC-0CD4-4d07-B1C5-92219CD475CC}"
"mp4.2"="00000000ffffffff,000000006d6f6f76,{B3DE7EDC-0CD4-4d07-B1C5-92219CD475CC}"
"mp4.3"="00000000ffffffff,000000006d646174,{B3DE7EDC-0CD4-4d07-B1C5-92219CD475CC}"
"ogm.1"="ffffffff,4f676753,{DB43B405-43AA-4f01-82D8-D84D47E6019C}"
"ts.1"="ff,47,{B841F346-4835-4de8-AA5E-2E7CD2D4C435}"

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Microsoft\Windows\CurrentVersion\Uninstall\HaaliMkx]
"DisplayName"="Haali Media Splitter"
"UninstallString"="\"C:\\Program Files (x86)\\Haali\\MatroskaSplitter\\uninstall.exe\""
"NoModify"=dword:00000001
"NoRepair"=dword:00000001

[HKEY_LOCAL_MACHINE\SOFTWARE\HaaliMkx]
"rmp4"="1"
"rOGM"="1"
"rts"="1"
"rm2ts"="1"
"WMP"="1"
"rEXT"="1"
"rFT"="1"

[HKEY_LOCAL_MACHINE\SOFTWARE\HaaliMkx\Input]
"avi.1"="ffffffff00000000ffffffff,524946460000000041564920,{51A00247-40A8-4845-9F17-7DBFCC9A8783}"
"mp4.1"="00000000ffffffff,0000000066747970,{B3DE7EDC-0CD4-4d07-B1C5-92219CD475CC}"
"mp4.2"="00000000ffffffff,000000006d6f6f76,{B3DE7EDC-0CD4-4d07-B1C5-92219CD475CC}"
"mp4.3"="00000000ffffffff,000000006d646174,{B3DE7EDC-0CD4-4d07-B1C5-92219CD475CC}"
"ogm.1"="ffffffff,4f676753,{DB43B405-43AA-4f01-82D8-D84D47E6019C}"
"ts.1"="ff,47,{B841F346-4835-4de8-AA5E-2E7CD2D4C435}"

tebasuna51

7th September 2014, 19:28

Last 9 post in this thread moved to http://forum.doom9.org/showthread.php?t=171150
Reason: off topic

asarian

9th September 2014, 04:35

I have this command:

C:\eac3to>eac3to audio.ac3 output1.wav -edit=0:47:10.200,3200ms -silence
MP2, 2.0 channels, 1:34:19, 224kbps, 44.1kHz
Decoding with libav/ffmpeg...
Editing RAW/PCM...
Reducing depth from 64 to 24 bits...
Writing WAV...
Creating file "output1.wav"...
Clipping detected, a 2nd pass will be necessary.
This track is not clean.
The last MPx frame is incomplete and thus gets skipped.
Starting 2nd pass...
Decoding with libav/ffmpeg...
Editing RAW/PCM...
Reducing depth from 64 to 24 bits...
Writing WAV...
Applying -31.9dB gain...
Creating file "output1.wav"...

I was hoping it would *overwrite* 3200ms of silence at 0:47:10.200. Instead, it just inserts 3200ms! How can I do the former?

Secondly, due to small glitch in the stream (the one I'm trying to blank out), eac30 decides to apply a -31.9dB gain on the file, making it as good as mute. How can I prevent that?

Thx.

Music Fan

9th September 2014, 09:04

Try without -silence.

tebasuna51

9th September 2014, 11:07

...
Clipping detected, a 2nd pass will be necessary.
This track is not clean.
...
Starting 2nd pass...
...
Applying -31.9dB gain...
You need two pass:

eac3to audio.ac3 output1.wav -edit=0:47:10.200,-3200ms -no2ndpass

then the -31.9dB gain is not applied and the glitch is deleted.

eac3to output1.wav output2.wav -edit=0:47:10.200,3200ms -silence

now insert the silence at same point.

asarian

9th September 2014, 11:26

You need two pass:

eac3to audio.ac3 output1.wav -edit=0:47:10.200,-3200ms -no2ndpass

then the -31.9dB gain is not applied and the glitch is deleted.

eac3to output1.wav output2.wav -edit=0:47:10.200,3200ms -silence

now insert the silence at same point.

Thank you very much! The -no2ndpass option isn't listed in the Options somehow, but it works nonetheless! :)

asarian

12th September 2014, 19:14

I use Eac3to with nero without installing nero.
Eac3to only needs a couple files and registry settings
Nero Files:
- AdvrCntr2.dll
- NeAudio2.ax
- NeEacDec.dll

Copy these files in a folder of your choice. I have copied them to my eac3to folder.
Use the following command lines to register the dll's.
regsvr32.exe C:\tools\eac3to\Nero\NeAudio2.ax
regsvr32.exe C:\tools\eac3to\Nero\AdvrCntr2.dll

The following registry keys have to be added to the registry to register the nero plugin.

[HKEY_LOCAL_MACHINE\SOFTWARE\Ahead\Installation\Families\Nero 7\Info]
etc.

[HKEY_LOCAL_MACHINE\SOFTWARE\Ahead\Installation\Families\Plugins\Info]
etc.

That all you need to use the nero plugin with eac3to

The above info seems hopelessly outdated (naturally so, of course). For one, Wimdows 7, 64-bit installs in:

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Ahead]

Copying the pertinent registry info from my Vmware box (only box I still have a Nero 7 running), I can't get the DLL's I copied over to work, though. However many ways I try, I keep getting errors about "the DllRegister Server entry point was not found."

Surely, isn't time eac3to found a better way to deal with aac files? (Yes, I still have some old movies that need it).

asarian

12th September 2014, 22:06

^^ Seriously, no one uses eac3to any more? :p There's gotta be someone who's gone the 'Nero 7' route on Windows 7 before me.

Sparktank

12th September 2014, 22:32

^^ Seriously, no one uses eac3to any more?

You literally let zero time pass before anoyne who's done with their busy lives have a look.

I don't use eac3to for AAC. Not for encoding, nor for decoding.

For AAC encoding, I pipe eac3to to qaac.
For AAC decoding, I've never had a use for it so maybe someone else will chime in for AAC decoding.

asarian

13th September 2014, 00:18

It gets stranger by the minute. Now, on my Vware box with Nero 7.10 Premium installed, the eac3to I just installed there as well says Nero isn't installed either. :( Sigh.

asarian

13th September 2014, 00:59

You literally let zero time pass before anoyne who's done with their busy lives have a look.

I don't use eac3to for AAC. Not for encoding, nor for decoding.

For AAC encoding, I pipe eac3to to qaac.
For AAC decoding, I've never had a use for it so maybe someone else will chime in for AAC decoding.

Well, it's solved, kinda. I just gave up going the eac3to route with it, and used a free tool called 'faad' to decode it.

filler56789

13th September 2014, 01:41

Well, it's solved, kinda. I just gave up going the eac3to route with it, and used a free tool called 'faad' to decode it.

You'd better choose ffmpeg, or qaac, or even foobar2000 :p ,
because

http://forum.doom9.org/showthread.php?p=1563950#post1563950

asarian

13th September 2014, 02:36

You'd better choose ffmpeg, or qaac, or even foobar2000 :p ,
because

http://forum.doom9.org/showthread.php?p=1563950#post1563950

Well, that's pretty frelled. :) No wonder I noticed a very minor off-sync issue. Thanks for bringing it to my attention!

If I did the math right, seems I got myself an exra (1/44100)*2624*1000 = 59.5011ms extra delay.

The people who came up with this silly delay should be taken into the yard and be shot! :p

EDIT: Yup, I subtracted the calculated surplus delay, and now it syncs perfectly again. :)

tebasuna51

13th September 2014, 02:48

Faad work fine decoding .aac files (ADTS headers), maybe VFR maniac was talking about .m4a files (AAC in mp4 container)

Recent test:
z.wav file encoded to z.aac with qaac
Decoded with faad to za.wav (delayed 26 ms)
Decoded with foobar to zb.wav (delayed 44 ms)
Decoded with ffmpg to zc.wav (delayed 44 ms)

I can't decode with qaac (I don't know if the sintax is correct):

qaac -D -o zd.wav z.aac
qaac 2.42, CoreAudioToolbox 7.9.8.3
ERROR: Not available input file format

asarian

13th September 2014, 03:01

Faad work fine decoding .aac files (ADTS headers), maybe VFR maniac was talking about .m4a files (AAC in mp4 container)

Recent test:
z.wav file encoded to z.aac with qaac
Decoded with faad to za.wav (delayed 26 ms)
Decoded with foobar to zb.wav (delayed 44 ms)
Decoded with ffmpg to zc.wav (delayed 44 ms)

Hmm, now I'm confused again. :confused: Yes, it was an .aac file (demuxed from a m4v with DGIndexNV). The new delay adjustment seems to work perfectly, though.

Sparktank

13th September 2014, 04:43

I can't decode with qaac (I don't know if the sintax is correct):

qaac -D -o zd.wav z.aac
qaac 2.42, CoreAudioToolbox 7.9.8.3
ERROR: Not available input file format

Syntax is correct.
I used same switches and was able to decode AAC and M4A.

Did not see any info on delay though, with/without --no-delay switch.

DoctorM

13th September 2014, 05:35

The above info seems hopelessly outdated (naturally so, of course). For one, Wimdows 7, 64-bit installs in:

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Ahead]

Copying the pertinent registry info from my Vmware box (only box I still have a Nero 7 running), I can't get the DLL's I copied over to work, though. However many ways I try, I keep getting errors about "the DllRegister Server entry point was not found."

Surely, isn't time eac3to found a better way to deal with aac files? (Yes, I still have some old movies that need it).

I've gotten Nero 7 to work just fine on Win7 x64. If you still want more info, PM me. But reading the other follow-ups I'm considering giving up on it too.

asarian

13th September 2014, 05:48

Syntax is correct.
I used same switches and was able to decode AAC and M4A.

Did not see any info on delay though, with/without --no-delay switch.

Looks like I may need to take a look at qaac too. Having to calculate the delay each time will get tiresome pretty fast, I reckon.

asarian

13th September 2014, 18:53

Quick question. On -down6, with a 7.1 track, shouldn't eac3to downmix using 24-bit when the original track is only 16-bit? I remember eac3to always used 'one-step-up', bit-width wise, on reencoding? (So as to compensate a bit for the resampling).

tebasuna51

14th September 2014, 00:21

The downmix is a lossy transform than use math operations (in float format), then the bitdepth is not preserved.

foxyshadis

19th September 2014, 00:38

The above info seems hopelessly outdated (naturally so, of course). For one, Wimdows 7, 64-bit installs in:

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Ahead]

Copying the pertinent registry info from my Vmware box (only box I still have a Nero 7 running), I can't get the DLL's I copied over to work, though. However many ways I try, I keep getting errors about "the DllRegister Server entry point was not found."

Surely, isn't time eac3to found a better way to deal with aac files? (Yes, I still have some old movies that need it).

You have to also use the 32-bit regsvr32, in C:\Windows\SysWOW64. In a cmd window, cd to that folder and then try registering. (Or use the full path for regsvr32.) I agree it's kind of outdated to rely on specific filters, though.

Snowknight26

19th September 2014, 14:07

eac3to can't parse TrueHD tracks with the Dolby Atmos extension:

http://stfcc.org/misc/dolby_atmos_truehd_sample.m2ts

Dolby has extended the Dolby TrueHD format, used in Blu-ray discs, to allow the format to
carry Dolby Atmos content. Before Dolby Atmos, Dolby TrueHD included lossless support for
channel-based audio, such as 5.1 and 7.1. We have added a fourth substream for Dolby
Atmos sound. This substream represents a losslessly encoded fully object-based mix.
Dolby Atmos in Dolby TrueHD can be transmitted from a Blu-ray player to your AVR in the
same way it is accomplished today. If your AVR supports Dolby Atmos, when the Dolby
TrueHD signal is decoded, the object-based audio and object audio metadata will be
decoded for further processing.
Dolby Atmos audio can be encoded with Dolby TrueHD at multiple sampling rates (including
48 and 96 kHz) and bit depths (16- and 24-bit). Dolby Atmos enabled receivers will also
support legacy Dolby TrueHD bitstreams at multiple sampling rates (including 48, 96, and
192 kHz) and bit depths (16-, 20-, and 24-bit) to provide full backward compatibility with
legacy Blu-ray Disc media and Dolby TrueHD music files.

nevcairiel

19th September 2014, 16:16

Anything based on ffmpeg/libav will not be able to read or parse TrueHD-Atmos streams, as they modified the syntax apparently.

foxyshadis

19th September 2014, 17:33

Looks like there wasn't a bug report for it yet, so I made one and uploaded your sample to their FTP. Not too important yet, since Transformers won't even be out until the 30th.

gillie

20th September 2014, 00:11

eac3to can't parse TrueHD tracks with the Dolby Atmos extension:

http://stfcc.org/misc/dolby_atmos_truehd_sample.m2ts

Just tried to use eac3to to convert TrueHD to DTS on the new Transformers Age of Extinction Bluray and it will not work at all.
Guessing this will become more of an issue as more and more Blurays are released with the new Dolby Atmos extensions embedded in the TrueHD tracks.
Unless anyone has any ideas on how to sort this?

nevcairiel

20th September 2014, 00:28

Unless someone fixes the TrueHD decoder in ffmpeg/libav, AND you get a new eac3to version which uses the fixed decoder, there is nothing anyone can do.

asarian

20th September 2014, 05:10

Just tried to use eac3to to convert TrueHD to DTS on the new Transformers Age of Extinction Bluray and it will not work at all.
Guessing this will become more of an issue as more and more Blurays are released with the new Dolby Atmos extensions embedded in the TrueHD tracks.
Unless anyone has any ideas on how to sort this?

Yeah, this is no good. Just downloaded and tried the sample myself, and it's a no-go.

So, does this mean my Marantz HD Amp won't be able to even play this? (not at home right now to try). If so, I will be real upset. And, if so, all the more reason for eac3to to find a way to decode it, after all, so I can convert it to LPCM.

madshi

20th September 2014, 08:17

Anybody posted this issue to the eac3to bug tracker yet?

asarian

20th September 2014, 17:07

Anybody posted this issue to the eac3to bug tracker yet?

Not sure it qualifies as a bug per se. Since the Atmos TrueHD stream is said to be fully backwards compatible (= can be played on regular players), I can only draw the conclusion, that they put the Atmos 'ceiling' channels into the metadata (which receivers are supposed to ignore; which is why this works, I presume).

So, all we really need is a knowledgeable coder (you?!) who will strip off the metadata, so we have a regular TrueHD stream again, right?

Thunderbolt8

20th September 2014, 17:37

imho it would be preferable being able to decode the entire stream completely.

minhjirachi

20th September 2014, 17:50

When I use the command to extract the Dolby Atmos to mono wavs. I get this error:

eac3to v3.24
command line: eac3to.exe "trans.4.thd+ac3" Vie.wavs
------------------------------------------------------------------------------
The format of the source file could not be detected. <ERROR>

I think maybe eac3to can not recognize that TrueHD (Dolby Atmos type).

nevcairiel

20th September 2014, 17:51

imho it would be preferable being able to decode the entire stream completely.

You can't just "decode" Atmos, thats not how it works, its not comparable to any audio codec you dealt with before. It effectively needs to be rendered based on the configuration of speakers in your room, which means its unlikely to be easily supported by the existing software solutions for a long time to come (first because we have absolutely no idea how the Atmos data is structured, and secondly because we would need a lot of new code and configuration potential to actually be able to render the audio in a 3D space). If you want, consider it "3D Audio", which is fundamentally different than the previous "2D Audio" we used so far.

Regarding getting the original TrueHD stream out, its not that easy, its not just metadata you have to ignore, its properly integrated into the TrueHD codec itself, and will need proper fixing of the decoder.
Its possible that it would work if we had an "official" TrueHD decoder or a specification, but since the ffmpeg/libav TrueHD decoder is practically reverse engineered, it lacks the ability to deal with those streams, for now.

asarian

20th September 2014, 17:59

Regarding getting the original TrueHD stream out, its not that easy, its not just metadata you have to ignore, its properly integrated into the TrueHD codec itself, and will need proper fixing of the decoder.

Then I wonder how current receivers are able to play the Atmos-enhanced TrueHD stream just fine, if the Atmos data were really a part of the main TrueHD stream. There's obviously some 'easily' discardable info, or current receivers would not be able to pull it off cleanly.

nevcairiel

20th September 2014, 18:44

Like I said, if we had the specification of TrueHD, it could tell us what is wrong, but since Dolby is not giving that out, its not that easy.

Thunderbolt8

20th September 2014, 21:10

You can't just "decode" Atmos, thats not how it works, its not comparable to any audio codec you dealt with before. It effectively needs to be rendered based on the configuration of speakers in your room, which means its unlikely to be easily supported by the existing software solutions for a long time to come (first because we have absolutely no idea how the Atmos data is structured, and secondly because we would need a lot of new code and configuration potential to actually be able to render the audio in a 3D space). If you want, consider it "3D Audio", which is fundamentally different than the previous "2D Audio" we used so far.

Regarding getting the original TrueHD stream out, its not that easy, its not just metadata you have to ignore, its properly integrated into the TrueHD codec itself, and will need proper fixing of the decoder.
Its possible that it would work if we had an "official" TrueHD decoder or a specification, but since the ffmpeg/libav TrueHD decoder is practically reverse engineered, it lacks the ability to deal with those streams, for now.well whatever the wording is, what I meant is its rather preferable being able to use 100% of thats whats there compared to stripping something off in order to get just to the normal THD track.

Like I said, if we had the specification of TrueHD, it could tell us what is wrong, but since Dolby is not giving that out, its not that easy.maybe the guy knows who told madshi how to deal with the TrueHD track headers back then when there were those THD delay problems because madshi accidentally stripped too much from these tracks.

foxyshadis

21st September 2014, 02:29

Unless someone fixes the TrueHD decoder in ffmpeg/libav, AND you get a new eac3to version which uses the fixed decoder, there is nothing anyone can do.

Patch posted. (https://ffmpeg.org/pipermail/ffmpeg-devel/2014-September/163189.html)

Superb

21st September 2014, 02:56

Patch posted. (https://ffmpeg.org/pipermail/ffmpeg-devel/2014-September/163189.html)Hendrik Leppkes is nevcairiel... ;)

minhjirachi

21st September 2014, 03:28

Patch posted. (https://ffmpeg.org/pipermail/ffmpeg-devel/2014-September/163189.html)

How to use it?

asarian

21st September 2014, 03:36

Patch posted. (https://ffmpeg.org/pipermail/ffmpeg-devel/2014-September/163189.html)

Hmm, should I just get this (http://ffmpeg.zeranoe.com/builds/win32/dev/ffmpeg-20140919-git-33c752b-win32-dev.7z) Dev version? Can't compile on Windows myself.

And, even though I understand it's just temporary, I would *still* like ffmpeg/eac3to to retain the option to merely extract the TrueHD core (akin to extracting an AC3 core from the TrueHD stream), so as to have something playable/usable on my system.

P.S. Thanks for the speedy patch! :)

tebasuna51

21st September 2014, 03:43

... to extracting an AC3 core from the TrueHD stream), so as to have something playable/usable on my system.

The AC3 from the track can be extracted with tsMuxeR, at least with the sample uploaded here.

asarian

21st September 2014, 03:46

The AC3 from the track can be extracted with tsMuxeR, at least with the sample uploaded here.

Yes. What I meant, though, is the ability to extract the TrueHD stream, like we extract the AC3 stream now. :) I have no interest in the AC3 core, of course.

asarian

21st September 2014, 04:36

How to use it?

Yeah, someone please tell us where this alleged patch resides: the 'posted' patch is just a mailing list post.

I tried the latest binary, and it says:

c:\eac3to\ffmpeg -i "test.m2ts" -vn -ac 8 -acodec pcm_s24le -f wav c:\video\temp.wav

[truehd @ 02bb57e0] major sync info header checksum error
Last message repeated 42 times
[NULL @ 02bb57e0] start time for stream 1 is not set in estimate_timings_from_pts
[mpegts @ 02bb9080] Could not find codec parameters for stream 1 (Audio: truehd (AC-3 / 0x332D4341), 0 channels): unspecified sample format

It *does* extract a stream, btw, but, upon conversion (and downmixing) to LPCM, eac3to says:

Mixing surround channels...
Writing WAV...
Creating file "c:\video\finak.wav"...
Original audio track, L+R+C+LFE+SL+SR: constant bit depth of 24 bits
Original audio track, BL+BR: no audio data.
The processed audio track has a constant bit depth of 24 bits.

Superb

21st September 2014, 05:59

Looking at ffmpeg's git log, the patch wasn't merged in yet... So using the latest precompiled binary won't do you any good.

foxyshadis

21st September 2014, 06:02

Hendrik Leppkes is nevcairiel... ;)

Oh, I had no idea. Thank you, nev!

asarian

21st September 2014, 06:02

Looking at ffmpeg's git log, the patch wasn't merged in yet... So using the latest precompiled binary won't do you any good.

Yeah, I noticed (when I found the log); seems latest compile was yesterday's. Thx.

asarian

25th September 2014, 11:06

Looking at ffmpeg's git log, the patch wasn't merged in yet... So using the latest precompiled binary won't do you any good.

FYI, new ffmpeg with fix is up! :)

http://ffmpeg.zeranoe.com/builds/

JimmyBarnes

28th September 2014, 15:02

FYI, new ffmpeg with fix is up! :)

http://ffmpeg.zeranoe.com/builds/

Could someone indicate how is this supposed to be used, especially regarding using eac3to convert the Age of Extinction THD+AC3 stream to DTS or AC3.

TIA

tebasuna51

28th September 2014, 18:11

maxmercy

28th September 2014, 20:27

JimmyBarnes

29th September 2014, 05:47

tebasuna51

29th September 2014, 10:08

Thanks for this, tho what I would really like to do is recode to AC3 640 kbps. Is this possible?

Of course you can recode to AC3 with ffmpeg, but a 7.1 must be downmix to 5.1 and is always better extract the AC3 included in the track with tsMuxeR (like I say in a precedent post).

BTW:

ffmpeg -i 00000.m2ts -map 0:1 -acodec ac3 -ac 6 -ab 640000 recoded.ac3

JimmyBarnes

29th September 2014, 16:08

Of course you can recode to AC3 with ffmpeg, but a 7.1 must be downmix to 5.1 and is always better extract the AC3 included in the track with tsMuxeR (like I say in a precedent post).

BTW:

ffmpeg -i 00000.m2ts -map 0:1 -acodec ac3 -ac 6 -ab 640000 recoded.ac3

Thanks for the above.

Does this actually convert the entire THD+AC3 to AC3 (rather than simply extract the AC3 stream, as you suggest with tsMuxeR)?

asarian

29th September 2014, 17:01

Could someone indicate how is this supposed to be used, especially regarding using eac3to convert the Age of Extinction THD+AC3 stream to DTS or AC3.

TIA

I simply used this:

ffmpeg -i "test.m2ts" -vn -ac 8 -acodec pcm_s24le -f wav c:\video\temp.wav

DoctorM

29th September 2014, 19:44

Thanks for the above.

Does this actually convert the entire THD+AC3 to AC3 (rather than simply extract the AC3 stream, as you suggest with tsMuxeR)?

Re-read what you quoted.

JimmyBarnes

30th September 2014, 02:51

ffmpeg -i 00000.m2ts -map 0:1 -acodec ac3 -ac 6 -ab 640000 recoded.ac3

This worked and the resulting AC3 synced OK with the recoded x264 video :thanks:

minhjirachi

30th September 2014, 11:57

Waiting for an update of eac3to.

Lenmaer

30th September 2014, 12:11

We were promised updates for over a year now, will it finally happen?

JimmyBarnes

30th September 2014, 12:18

Waiting for an update of eac3to.

This would be the ideal solution to the Dolby Atmos issue, I would expect..

But the ffmpeg solution described here can be used meanwhile.

Nico8583

5th October 2014, 00:03

Hi :) could you explain me the difference between 16bits and 24bits DTS ? And the real utility to use -dontPatchDts option or not ? Thanks !

tebasuna51

5th October 2014, 03:27

In the standard (lossy) DTS headers there are a field to remember the bitdepth (precission) of the source PCM used to encode the DTS.
After the encode this info is useless because we can't recover the original precission.

By default eac3to change this field to put always 24 bits to remember than any lossy format (AC3, DTS, ...) must be decoded at last to 24 bits, even if the source PCM was 16 bits, to preserve the max quality.
Then use -dontPatchDts is useless because don't change any relevant info in the stream.

Nico8583

5th October 2014, 10:49

Thanks for your response. So it's useless now to use this option because any player can read a "virrtual" 24bits stream ? Even in a MKV ?

tebasuna51

5th October 2014, 13:31

Players don't use that field because lossy formats don't have bitdepth, must be ignored in standalone DTS or in any container.

Nico8583

5th October 2014, 15:31

Ok thanks for these informations !

RoseM

19th October 2014, 16:47

Hello

Someone could see why the audio of the file below isn't recognized by the program?

http://www.mediafire.com/download/5wndl415r97o84z/09-30_12-25-28_Globo_HD_SP.ts

Thanks
Because this format is not supported by eac3to tool.

Audio #1
ID : 274 (0x112)
Menu ID : 59200 (0xE740)
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : HE-AAC / LC
Muxing mode : LATM
Codec ID : 17
Duration : 1mn 11s
Bit rate mode : Variable
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 KHz / 24.0 KHz
Compression mode : Lossy
Delay relative to video : -400ms
Could add support for this format in a update future .

thanks

Overdrive80

19th October 2014, 19:36

That is not format audio, is muxing mode. The audio format for you code is AAC, and eac3to already support that format.

Music Fan

19th October 2014, 22:48

The message from eac3to about the aac tracks of this file is "parsing failed".
But it works when put in MKA (and probably also MKV), eac3to has maybe problems with aac in TS container.

Could add support for this format in a update future .
Open your TS in MKVMerge, uncheck video if not needed, export in mka, open the mka in eac3to.
Strange thing : the aac 2.0 becomes 3.0 when in mka and the frequency becomes 22.05 KHz ;

Audio
ID : 1
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : LC
Codec ID : A_AAC
Duration : 1mn 8s
Channel(s) : 3 channels
Channel positions : Front: L C R
Sampling rate : 22.05 KHz
Compression mode : Lossy
Default : Yes
Forced : No

edit : I spoke too quickly ; analyzing works but not decoding ;

eac3to v3.27
command line: "C:\Program Files\eac3to\eac3to.exe" "F:\09-30_12-25-28_Globo HD_SP.mka" "F:\09-30_12-25-28_Globo HD_SP.wavs"
------------------------------------------------------------------------------
MKA, 1 audio track, 0:01:09
1: AAC, 3.0 channels, 22kHz
Track 1 is used for destination file "09-30_12-25-28_Globo HD_SP.wavs".
[a01] Extracting audio track number 1...
[a01] Decoding with DirectShow (Nero Audio Decoder 2)...
[a01] Getting "Nero Audio Decoder 2" instance failed. <ERROR>
Aborted at file position 220599. <ERROR>

tebasuna51

20th October 2014, 07:35

RoseM

20th October 2014, 15:07

@RoseM

1) Extract with eac3to (ignore warnings)

2) Use vlc to convert LATM to standard .m4a (see http://forum.doom9.org/showthread.php?p=1453484#post1453484 ), for instance:

vlc.exe -I dummy aud_latm.aac --sout #std{access=file,mux=mp4,dst="aud_nolatm.mp4"} vlc://quit
Okay.'ll Try

There's no mode of eac3to do the demux without having to use other programs.

TMPGEnc makes the conversion but isn't free. Wanted to use RipBot264

'm Half noob in conversions.

Thank you guys for replys

tebasuna51

20th October 2014, 15:25

There's no mode of eac3to do the demux without having to use other programs.

eac3to demux the aac but still in LATM format not recognized by standard decoders.

I only know VLC to change the LATM to a standard AAC format.

DonDan

26th October 2014, 12:55

Hey guys,

I've been testing around for quite a while now trying to achieve the following: AC3 stereo (well fake stereo, originally mono source made to 2 identical stereo chans on DVD) to aac mono in one step piping output from eac3to to neroaac or fdkaac (to make sure it's not an encoder issue).

Whereas both commands (that encode stereo) work fine:

eac3to test.AC3 stdout.wav | neroaacenc -ignorelength -lc -q 0.4 -if - -of neroenc_out.mp4

eac3to test.AC3 stdout.wav | fdkaac --profile 2 --bitrate-mode 2 -o fdkaac_out.mp4 -

I fail as soon as I try to make eac3to output mono.

On the eac3to side I tried with adding -normalize -down16 -mono or just -mono, also tried naming the output both stdout.wav aswell as stdout.wavs.

On the encoder side I also tried all available options like --ignorelength --raw --raw-channels 1 --raw-rate 48000 --raw-format S24L for fdkaac and *well nothing* for neroaac to no extent...

Nero will just refuse to encode, fdkaac at least tries but the best I got so far was a file with noise for the correct duration of the input file.

Ofc when I encode the ac3 to mono.wav in eac3to and in a second step encode that to aac I get a proper result, but I'ld love to find a way to use a pipe to get there.

Anybody got an idea? Or is it just plain impossible?

Muchas Gracias!

tebasuna51

26th October 2014, 18:12

When you decode AC3 2.0 to stdout.wav with the -mono parameter (Left channel only) with:

eac3to test.AC3 stdout.wav -mono | neroaacenc -ignorelength -q 0.4 -if - -of neroenc_out.mp4

NeroAacEnc send the message: "Invalid Channel mask" and finish.
In fact the stdout.wav have the channel mask 3 (Left + Right) like the original AC3 file.

A workaround is use simple wav header instead the default WAVE_FORMAT_EXTENSIBLE header with:

eac3to test.AC3 stdout.wav -mono -simple | neroaacenc -ignorelength -q 0.4 -if - -of neroenc_out.mp4

And now work fine.

DonDan

26th October 2014, 19:31

I guess this is the right time to say: 'That -simple?!?' ;-)

:thanks: :thanks:

jpsdr

4th November 2014, 09:49

Madshi, could you take a look here (http://forum.doom9.org/showthread.php?p=1698609#post1698609) and enlight us ?
Is it something you add on your own, or can it be handled by an option using flac.exe (or libflac) ?

madshi

4th November 2014, 12:33

Is it something you add on your own, or can it be handled by an option using flac.exe (or libflac) ?
What do you mean with "it" exactly?

jpsdr

4th November 2014, 12:58

... You're right indeed... "it" was not clear. It was the "HDCD" and "VALID_BIT" tags. But, i think Groucho2004 provided the answer.
I am right to say it's a custom meta_data tag, finaly exclusive specific to eac3to, and so irrelevant or with no interest for any other software ?

madshi

4th November 2014, 14:37

Well, any other software is free to use it, but yes, it's not an "official" meta data tag.

mini-moose

15th November 2014, 14:02

apologies if this was asked before..

Are there plans to add support for TrueHD Atmos?

tebasuna51

15th November 2014, 18:16

Use 'Search this Thread' (up right of the page) "Atmos"

xerox44

3rd December 2014, 11:27

Can be used to decode the track but can't be used by eac3to until madshi can parse the track and add new libav dll's.

By the moment you can do, for instance:

ffmpeg -i 000000.m2ts -map 0:1 recoded.flac

Is there any news regarding eac3to supporting Dolby Atmos streams? Support, as in, stripping the atmos meta-data, so that we can decode the truehd track into mono-wavs as we could before atmos was added?

I've tried using the latest ffmpeg (the above code), but the best I can get from a truehd/atmos track is the ac3 subtrack. Is there any way to get the truehd track by using ffmpeg in the mean time? I've googled and tried working through the man-pages of ffmpeg, but no luck so far.

Any help would be greatly appreciated, thanks!

LigH

3rd December 2014, 11:56

Lenmaer

3rd December 2014, 12:08

Would be nice if someone would actually be willing to update eac3to. It kinda feels like a dead project now, the developer seems to have abandoned it. >.<

Music Fan

3rd December 2014, 12:15

LAV Filters 0.63.0 and L-SMASH Works (with Libav, r733 by the_weirdo) are now able to ignore Atmos extensions correctly.
Does it allow to keep the True HD track (removing the Atmos extensions) or it can only decode the True HD to wav ?

LigH

3rd December 2014, 12:39

Would be nice if someone would actually be willing to update eac3to. It kinda feels like a dead project now, the developer seems to have abandoned it. >.<

But eac3to is not OpenSource. Noone else can do that.

Does it allow to keep the True HD track (removing the Atmos extensions) or it can only decode the True HD to wav ?

Neither is just a demultiplexer. Both are decoders (as DirectShow filter or as AviSynth source plugin).

Well, I am not sure if LAV Filters could be used as splitter only. And if it would strip Atmos extensions off TrueHD separately, not just in the decoder.

Kurtnoise

3rd December 2014, 13:06

LAV Filters 0.63.0 and L-SMASH Works (with Libav, r733 by the_weirdo) are now able to ignore Atmos extensions correctly. But to support it in ffmpeg too, someone would have to port this patch to the ffmpeg source repository...
Or compile the FFmpeg executable using the LAV FFmpeg branch...;)

LigH

3rd December 2014, 13:10

Ah, OK ... is there a usual source for such binaries? Do zeranoe builds use "legacy" or LAV libs?

Kurtnoise

3rd December 2014, 13:16

Yeah, the branch from LAV Filters > http://git.1f0.de/gitweb?p=ffmpeg.git;a=summary

but you have to compile it by yourself.

nevcairiel

3rd December 2014, 14:19

LAV Filters 0.63.0 and L-SMASH Works (with Libav, r733 by the_weirdo) are now able to ignore Atmos extensions correctly. But to support it in ffmpeg too, someone would have to port this patch to the ffmpeg source repository...

I pushed my TrueHD patch to upstream ffmpeg, it should be in recent zeranoe builds for example.

LigH

3rd December 2014, 14:23

:thanks: – Hope someone with matching material can confirm it.

Now back to eac3to... ;)

Nebudchanezzer

8th December 2014, 14:50

Hi everyone!

I would like to revisit the "strange setup" issue a little bit....

I know this has been up before, but I would like to look at it some more.
If there isn't a possibility to "inject" that it is normal 7.1 setup into the header when decoding.

I have tried encoding a small audiofile into both "strange setup" and normal setup, and the encoded data is the same (using a hexeditor for comparison) only the header differs.

I have also tried altering the header of the "strange setup" with the hex-editor to see if I could do what is asked here my self, but I had no luck.
Doesn't matter what I change in the header really, the track decodes as "strange setup" whatever....

I just don't know what it is in the header that tells the decoder that it is "strange setup".

EDIT: Also to add, someone mentioned earlier that this might not be possible due to the internal error calculations of DTS-HD, but I have actually edited the header without any problems to decode the track afterwards (only it still decodes as "strange setup").
From the little I understand from the published documents regarding DTS-HD MA, I find the error calculations depends of the acutal header-size and not some sort of hash of it.

So I would really like if someone could tell me how to read the header, or what specifically tell the decoder it is "strange setup".
I would be happy to share what I have with anyone interested more in this.

73ChargerFan

16th December 2014, 06:20

Link in first post to Haali is bad.

Sparktank

16th December 2014, 07:13

Link in first post to Haali is bad.

This Haali link works:
http://haali.su/mkv/

Infineon

18th December 2014, 20:08

I get this error when trying to convert DTS-HD music tracks that have been demuxed from a Blu-Ray. I'm using v3.27. The first track is identified and converts to FLAC perfectly, after that for all subsequent tracks I get this error. I can convert TrueHD to FLAC with no issues. Can anyone suggest what might be causing this?

microchip8

18th December 2014, 20:11

Music Fan

18th December 2014, 20:13

Look at the name of his post : "The format of the source file could not be detected".

microchip8

18th December 2014, 23:16

Look at the name of his post : "The format of the source file could not be detected".

Thanks. However, this still doesn't bring us anywhere. No info on content he used (no sample either)

Infineon

19th December 2014, 09:27

Thanks. However, this still doesn't bring us anywhere. No info on content he used (no sample either)

The source media is an audio only Blu-Ray music disc containing 2Ch PCM 96/24 and 6Ch DTS-HD 96/24. I use AnyDVD-HD to decrypt and save as ISO image (mainly so I have a backup of my media that can't get scratched). I then demux the DTS HD tracks from the image using DVD Audio Extractor. I get the same problems if I demux direct from the disc.

I'm open to suggestions of a better demuxing app that I could try? I am seeing this problem with DTS-HD on other blu-rays. The ISO image and demuxed DTS-HD files both play back fine on my PC. The weird thing is some tracks convert to FLAC in eac3to, some don't. On a previous disk the first two tracks worked, then the rest weren't recognised.

If you need it I could make the a copy of a working and non working track available for comparison but pls msg me in private to set this up.

Music Fan

19th December 2014, 09:53

You can try eac3to or TSmuxer to demux audio and video.

Infineon

19th December 2014, 10:53

You can try eac3to or TSmuxer to demux audio and video.

Ok, with eac3to am I right in thinking that I must use the raw blu-ray m2ts files to demux from? The problem I found was it creates a single output FLAC file. Perhaps I'm being a noob, is there a way to demux into individual DTS-HD files or just demux and convert straight to individual FLACs? Thanks.

Music Fan

19th December 2014, 15:02

You can do what you want with eac3to ; extract a dts track as is, convert it in a 5.1 wav, 6 mono waves, flac ...
The first thing to do is to analyze the m2ts file to see the numbers of the tracks, with this very simple command ;
"C:\Program Files (x86)\eac3to\eac3to.exe" "H:\my video.m2ts"
eac3to will show tracks with a number. If the dts track you need is number 2, the command to extract it looks like ;
"C:\Program Files (x86)\eac3to\eac3to.exe" "H:\my video.m2ts" 2: "M:\sound of my video.dts"
It's not only the core, it contains the DTS HD and the core.

To convert in Flac, just change the extension ;
"C:\Program Files (x86)\eac3to\eac3to.exe" "H:\my video.m2ts" 2: "M:\sound of my video.flac"

But you need somme dll (from ArcSoft TotalMedia Theatre) to decode Dts HD ;
http://forum.videohelp.com/threads/352153-DTS-HD-MA-7-1-to-WAVS-%288-channels%29?p=2210419&viewfull=1#post2210419

Sparktank

19th December 2014, 23:55

single output FLAC file...to individual FLACs

Sounds like you want a track list rip as if it were a CD with a CUE.
DVDAudio Extractor can rip by "chapter". But I often find that different versions of DVDAE produce different results than eac3to.

You can remux with MKVtoolnix to split by timecode or framenumbers.
You have to input this list manually.
You can use the disc chapters.

If one track segue's into the next track, and you don't like the disc chapter, you'll have to edit yourself in something like Audacity.

eac3to doesn't do splitting by chapter (automated).
Best thing to do is split into files with MKVtoolnix for remuxing and then use eac3to for each split remux.

Infineon

20th December 2014, 10:35

Infineon

21st December 2014, 01:30

Sounds like you want a track list rip as if it were a CD with a CUE.
DVDAudio Extractor can rip by "chapter". But I often find that different versions of DVDAE produce different results than eac3to.

You can remux with MKVtoolnix to split by timecode or framenumbers.
You have to input this list manually.
You can use the disc chapters.

If one track segue's into the next track, and you don't like the disc chapter, you'll have to edit yourself in something like Audacity.

eac3to doesn't do splitting by chapter (automated).
Best thing to do is split into files with MKVtoolnix for remuxing and then use eac3to for each split remux.

Perfect, MKVToolNix did the job. DVD-AE has an option to create a DTS-CD plus Cue sheet so I just used this to generate the cue quickly from the ISO. Then loaded the cue into MKVMerge to split the chapters and demux the M2TS stream to MKA. eac3to was then able to correctly identify the DTS-HD in the MKAs and convert them either to FLAC or .dts files. Discovered something new as well, using the MKA container JRiver appears to be playing back the full DTS-HD bitrate now (8000Kbps) whereas the .dts files just play the core (1500Kbps). Not sure why that is. eac3to identifies them as 96/24 and they're just as large as the MKAs. Anyway, thanks for your help!

XadoX

22nd December 2014, 08:52

Is there a know problem, with discs containing Dolby Atmos tracks?

Infineon

22nd December 2014, 09:26

Sorry I've been ripping from music blu-rays not movies so couldn't tell you. I suspect the problem I've encountered is with DVD Audio Extractors demux of Dts-hd.

LigH

22nd December 2014, 09:34

Did you really search this thread for "Atmos" before asking this, XadoX?

XadoX

22nd December 2014, 14:44

Did you really search this thread for "Atmos" before asking this, XadoX?
No I don't. But now I did.

My problem is not relatet in playing those files. I'm having problmes while demuxing a disc containing Dolby Atmos.

Boulder

27th December 2014, 16:40

(Does anyone have a new libFLAC.dll to use with eac3to? FLAC version 1.3.1 has some improvements to encoding efficiency and speed, which is always a good thing.)

EDIT: found this thread after some more searching: http://forum.doom9.org/showthread.php?t=171480

Lenmaer

27th December 2014, 16:45

Anakunda

31st December 2014, 14:10

HI!
what output format should I use to extract MPEG-2 video stream from Bluray? (without any conversion)
I tried .mpeg .mpg .vob .m2ts .mts, everything fail. Does even eac3to support extracting MPEG-2 video from bluray?

LigH

31st December 2014, 14:20

In general, eac3to is an audio converter. It will probably not process video streams.

Boulder

31st December 2014, 14:24

Won't it demux it to a Matroska container?

Anakunda

31st December 2014, 14:33

In general, eac3to is an audio converter. It will probably not process video streams.
It does. If they're organized into Bluray structure. For DVDs you're right, however I'm demuxing Bluray.

Won't it demux it to a Matroska container?
MakeMKV can do this but I'd prefer to use eac3to to demux into bare streams (ie. separate video, separate audio track etc., this is the best for further processing)

r0lZ

31st December 2014, 14:43

Use TsMuxeR. It can demux everything from a blu-ray. However, it can't properly demux the video streams from a multi-angle MPLS.

Boulder

31st December 2014, 14:55

I also recommend TsMuxeR, but if you insist on using eac3to, what is the error that you have? eac3to does support muxing MPEG-2 to mkv.

Anakunda

31st December 2014, 15:29

I also recommend TsMuxeR, but if you insist on using eac3to, what is the error that you have? eac3to does support muxing MPEG-2 to mkv.

It's the error

H:\video\(workset)\Pulp fiction>eac3to 2) 1:bonus1\source.mpeg
M2TS, 1 video track, 1 audio track, 1 subtitle track, 0:55:26, 50i
1: MPEG2, 576i50 (4:3)
2: AC3, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
3: Subtitle (PGS), Czech
This video conversion is not supported.

Boulder

31st December 2014, 15:33

Try "eac3to 2) 1: bonus1\source.mkv" (note the space after "1:")

Anakunda

31st December 2014, 15:39

Yes eac3to 2) 1:bonus1.mkv works but I was hoping that raw demuxing like on Bluray formats will work too. Issue closed, thanks to everybody.

madshi

31st December 2014, 16:26

Use ".m2v".

Anakunda

31st December 2014, 18:59

Use ".m2v".

.m2v is accepted by eac3to. Thanks madshi.
Btw your renderer has problem to render properly subtitles muxed in MKV (not so with subtitles placed aside the mkv)

Sparktank

1st January 2015, 02:28

Btw your renderer has problem

That should be posted in the MadVR thread, along with a sample (and check the bugtracker).

73ChargerFan

1st January 2015, 09:34

When you need to know what extension to use for a particular stream, try the -demux option then cancel after 5 seconds, and look at the files it created.

whitey1977

2nd January 2015, 12:15

I recently had 2 drives fail on my RAID5 array so I'm now having to re-rip all my movies. In my collection I have a few HD-DVDs which I converted to blu-rays for easier playback. Since it was over 6 years ago when I originally did this I can't quite remember how I did it.

I've used eac3to to demux amd convert the DD+ tracks and then TSMuxer to combine them into blu-ray structure and this has all worked great, the problem is with the HD-DVDs that have TrueHD audio tracks. These discs show no streams when loaded into eac3to so I can't select them for demux. I'm sure this is the method I used 6 years ago and it may simply be that I'm using a newer version of eac3to that has a bug or no longer supports TrueHD from HD-DVD.

Can anyone help?

Music Fan

2nd January 2015, 12:31

Try HdBrStreamExtractor (GUI for eac3to), it will perhaps detect the True HD track. Or try an old eac3to version.
For the DD+ tracks, I'm not sure you have to make conversion, my Oppo BDP-83 plays DD+ ripped from HD-dvd.

LigH

2nd January 2015, 12:41

One source for older eac3to versions should probably be the VideoHelp archive (http://www.videohelp.com/tools/eac3to/old-versions#download).

Hitesh12

2nd January 2015, 12:44

Hey guys I'm very new here so please go easy :scared:

So this what I'm trying to do -

Convert a .mkv file with 5.1 DTS audio -
to a .mkv file with 2 audio tracks -
Track no.1 - Encoded flac audio
Track no.2 - Original 5.1 DTS track

Is this possible with a single command (what I mean is that everything occurs in one go; I don't have to extract audio then convert it then mux it; I hope it is clear) ?

If yes, then what should be the commands ?

OT : Just read that post mentioning new FLAC, so I was wondering if there are any other new versions of the default encoders that I should update ?

Boulder

2nd January 2015, 13:14

You can call the same track twice in the command line so eac3to file.mkv 1: audio.dts 1: audio.flac should do (replace 1: by whatever the reported track is). I don't recall if you need to specify the playlist with mkv files.

tebasuna51

2nd January 2015, 13:17

Is this possible with a single command (what I mean is that everything occurs in one go; I don't have to extract audio then convert it then mux it; I hope it is clear) ?
Nope.
You don't need extract the dts to convert to flac, but you need mux the flac to the mkv with MkvMerge.

OT : Just read that post mentioning new FLAC, so I was wondering if there are any other new versions of the default encoders that I should update ?

The other encoder (libAften.dll) don't have a new version.

Hitesh12

2nd January 2015, 13:49

Hitesh12

2nd January 2015, 13:51

Nope.
You don't need extract the dts to convert to flac, but you need mux the flac to the mkv with MkvMerge.

The output file cannot be in .mkv container ?

Boulder

2nd January 2015, 18:05

eac3to cannot mux different files to one container.

Hitesh12

2nd January 2015, 18:16

eac3to cannot mux different files to one container.

Thanks. It makes sense now

Another query -
Is using Nero 7 AAC codec to convert DTS audio into stereo AAC the most efficient way ?
I mean Nero 7 is old, it might be using old version of AAC ?

Boulder

2nd January 2015, 18:20

Hitesh12

2nd January 2015, 18:23

It's not the most efficient way (if you mean quality-wise). I personally use Apple's AAC codec for all my stuff, you can search for 'qaac' if you're interested. There's a big thread at hydrogenaud.io but probably something here as well.

So how do I use qaac with eac3to ?
EDIT : Also is there a list of AAC encodes with their comparisions ? nvm got it

Music Fan

2nd January 2015, 18:27

eac3to cannot mux different files to one container.
Strange, because I guess it can be used to mux to mkv audio and video coming from the same file (m2ts for example), and that's what Hitesh12 wanna do.

Boulder

2nd January 2015, 18:28

So how do I use qaac with eac3to ?
You need to use piping. The search should help you with that.

Boulder

2nd January 2015, 18:29

Strange, because I guess it can be used to mux to mkv audio and video coming from the same file (m2ts for example), and that's what Hitesh12 wanna do.I think it can only work if you remux the whole Blu-ray folder and thus cannot choose what you mux or re-encode and then mux. At least that's what the example in the first post by madshi shows.

Hitesh12

2nd January 2015, 18:31

Strange, because I guess it can be used to mux to mkv audio and video coming from the same file (m2ts for example), and that's what Hitesh12 wanna do.

As mentioned by boulder, I'm using this currently -

eac3to input.mkv 2: audio.dts 2: audio.flac

Is there some other command to mux these ?

Hitesh12

2nd January 2015, 18:38

You need to use piping. The search should help you with that.

Thanks !

Another query (:p) -

AFAIK there are different DTS codecs (DTS, DTS-HD I think)
So when I select .dts as output format, which DTS is it referring to ?

Also how do I check which DTS codec does the file has ? I don't think mediainfo differentiates between different DTS codecs

Boulder

2nd January 2015, 18:41

If you don't have a DTS encoder installed and you don't specify -core, you get DTS-HD if the original track is such. Otherwise you get a regular DTS track. I don't know if there is a simple way to determine which is which other than using eac3to ;)

Hitesh12

2nd January 2015, 18:48

Hitesh12

2nd January 2015, 19:05

Hitesh12

2nd January 2015, 19:16

I tried/searched but couldn't understand this piping along with all those commands.

Is there any guide (like this (http://en.wikibooks.org/wiki/Eac3to/How_to_Use) for eac3to) ?

Boulder

2nd January 2015, 19:22

It seems you first need to learn the basics of command line stuff. You might be better off using some GUI for eac3to such as the one here: http://forum.doom9.org/showthread.php?t=145574

Music Fan

2nd January 2015, 19:43

This is a very odd question but how do I access file in some other directory ?
Currently I'm moving files which I want to encode in the eac3to directory (Only then this works - eac3to input.mkv output.whatever) , which is not the correct way :p
Type the complete path between quotation marks if there are spaces in the path, for example ;
"C:\Program Files (x86)\eac3to\eac3to.exe" "H:\my video.m2ts" 2: "P:\my video sound.ac3"

Music Fan

2nd January 2015, 19:56

I tried/searched but couldn't understand this piping along with all those commands.

Is there any guide (like this (http://en.wikibooks.org/wiki/Eac3to/How_to_Use) for eac3to) ?
This works (always type stdout.wav without path) ;
"C:\Program Files (x86)\eac3to\eac3to.exe" "H:\my audio.ac3" stdout.wav | "C:\Program Files (x86)\neroAacEnc\neroAacEnc.exe" -q 0.4 -if - -of "P:\my audio.mp4"
You will get aac in mp4 container.
-q is for quality, 1 is the maximum.

Other example with extraction from m2ts and downmix ;
"C:\Program Files (x86)\eac3to\eac3to.exe" "H:\my file ac3 5.1.m2ts" 2: stdout.wav -downStereo | "C:\Program Files (x86)\neroAacEnc\neroAacEnc.exe" -q 0.65 -if - -of "P:\my file (only sound) aac 2.0.mp4"

agogley

5th January 2015, 04:24

I'm having a problem with eac3to. I can't seem to extract the DTS-HD master audio from a BD. I can rip the BD then make an MKV then extract the audio but I can't seem to do it from my original source. I keep getting an error that says eac3to can't open the source file. eac3to gets to the end of it's 3rd Pass before giving me this error in the log.

What are some things I can do to troubleshoot? I've been using eac3to successfully for many years until recently. Recent rips of Stargate and Divergent are giving me trouble. Before that, no problems.

Music Fan

5th January 2015, 18:20

Did you try to extract first the DTS-HD track with TSMuxer, then to open it in eac3to ? But it's only useful if you wanna re-encode the sound, otherwise use MKVMerge.

agogley

6th January 2015, 03:51

Did you try to extract first the DTS-HD track with TSMuxer, then to open it in eac3to ? But it's only useful if you wanna re-encode the sound, otherwise use MKVMerge.

Once I know the proper playlist, I was able to extract the DTS-HD track (aqnd video) with TSMuxer. I am also able to use MakeMKV to make an MKV, then Clown_BD or tsMuxeR to build BD structure again (or ISO).

For whatever reason, eac3to will not read the source file but everything else seems to work fine.

How would I "open it" with eac3to?

I get a message saying that "Sony Screenpass, use mpls 00267?" I choose now, but I wonder if that has something to do with it?

Arm3nian

6th January 2015, 09:20

I'm trying to extract an aac audio file from an mp4. I don't think eac3to supports mp4 input so I used mkvmerge to extract only the audio as an mka. I'm guessing that process does no encoding. Then I used eac3to to extract the aac from the mka, but the bitrate observed by foobar and mpcbe (lav) was a bit higher, no pun intended :)
I only have the Nero aac encoder installed, not the decoder so I'm not sure why it works to begin with. Why is the bitrate slightly higher when coming out of eac3to using just input and output commands? The reported bitrate of the mp4 audio source file is 126kbps, the bitrate of the mka is 127kbps, and the bitrate of the aac is 128kbps... but the real time bitrate of the source and the mka is the same, the aac out of eac3to is 2kbps higher at some times.
Just for experimental purposes, I want a bit perfect adts aac file.

Music Fan

6th January 2015, 10:03

To demux aac from mp4, you can also try YAMB (gui for MP4Box.exe).

Music Fan

6th January 2015, 10:06

How would I "open it" with eac3to?
You can try this (open the folder, not a file) ;
http://en.wikibooks.org/wiki/Eac3to/How_to_Use#Using_complete_disc_structure_as_input

tebasuna51

6th January 2015, 13:37

...Why is the bitrate slightly higher when coming out of eac3to using just input and output commands? The reported bitrate of the mp4 audio source file is 126kbps, the bitrate of the mka is 127kbps, and the bitrate of the aac is 128kbps...
I encode a WAV (44.1 KHz, 165.000 seconds) to .M4A with NeroAacEnc.
With MkvMerge I obtained a .MKA.
I extracted the ADTS .AAC (the same with eac3to or MkvExtract).
You can be sure than the audio samples are the same in the 3 files, but if you include the headers to calculate the bitrate (Filesize*8/Duration):

FileSize Bitrate Foobar2000
-------- ------- ----------
M4A 3120294 151.287 150 Kb/s
MKA 3113283 150.947 151 Kb/s
AAC 3138214 152.156 ?

Some containers, like MP4 (M4A) can include, in header, the size of RAW AAC without headers.
Then MediaInfo show for the AAC stream in M4A the bitrate 149.680

Like you can see the little differences between bitrates is only the method to calculate (with/without headers).
BTW the standard method in audio files is include the headers overload (Filesize*8/Duration).

agogley

9th January 2015, 16:10

My eac3to logs are a mess. I don't understand why I'm getting all these messages. I'm going to post some logs to see if I can't get somebody to help me out.

The ones that I named - failed are from a process where eac3to will not finish the job. It simply fails. The three logs named normally completed the job. But there is still a bunch of errors, etc.

agogley

9th January 2015, 16:11

Here are the last two.

Asmodian

10th January 2015, 02:56

I don't see anything out of the ordinary in the non-fail logs, which message is worrying you?

agogley

10th January 2015, 03:09

I don't see anything out of the ordinary in the non-fail logs, which message is worrying you?

a03] The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
[a03] The libav DTS decoder doesn't decode the full DTS-HD information. <WARNING>
[a03] The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
[a03] The libav DTS decoder doesn't decode the full DTS-HD information. <WARNING>

a03] The original audio track has a constant bit depth of 64 bits.
[a03] The processed audio track has a constant bit depth of 24 bits.

Music Fan

10th January 2015, 10:12

agogley

10th January 2015, 20:25

Are you sure you installed correctly the ArcSoft decoder ?
Look at this post, you have to register ASAudioHD.ax ;
http://forum.videohelp.com/threads/352153-DTS-HD-MA-7-1-to-WAVS-%288-channels%29?p=2210419&viewfull=1#post2210419

I tried the regsvr fix. I tried to do regSvr32 but I get an error message that the DLL was loaded but the entry point DLLRegisterServer was not found.

I also tried to uninstall and reinstall Arcsoft TMT 5.whatever and that didn't clear it up any.

Asmodian

11th January 2015, 01:49

a03] The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
[a03] The libav DTS decoder doesn't decode the full DTS-HD information. <WARNING>
[a03] The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
[a03] The libav DTS decoder doesn't decode the full DTS-HD information. <WARNING>

a03] The original audio track has a constant bit depth of 64 bits.
[a03] The processed audio track has a constant bit depth of 24 bits.

Oh, I assumed that was expected as that simply means you don't have the ArcSoft decoder installed.

Edit: Also converting the 64 bit to 24 bit will not result in a quality loss, that is normal and how it is supposed to work.

I tried the regsvr fix. I tried to do regSvr32 but I get an error message that the DLL was loaded but the entry point DLLRegisterServer was not found.

I also tried to uninstall and reinstall Arcsoft TMT 5.whatever and that didn't clear it up any.

You do regsvr32 on the .ax file. You shouldn't get any messages about DLLs. Simply follow the instructions given in Music Fan's link.

Music Fan

11th January 2015, 10:00

I tried to do regSvr32 but I get an error message that the DLL was loaded but the entry point DLLRegisterServer was not found.
Re-try eac3to anyway because sometimes there are error messages about dll while they actually work.

Boulder

11th January 2015, 11:27

Does TMT5 files even work with eac3to? I thought that v2 is the only one that works.

JohnTucker

13th January 2015, 09:10

I want to use eac3to to batch demuxing multiple m2ts files. Those m2ts files have multiple audio and subtitle streams but I am interested only in the English ones (the first ones). Problem is that some of them have the first pgs (sup) file as number 5, others as number 8.

So how can I tell eac3to to demux the first h264, ac3 and pgs streams that it finds?

bmcelvan

15th January 2015, 15:45

Is there a way to set the compression value when outputting FLAC with eac3to? I've looked at all the options in cmd line with eac3to and haven't seen any.

Does it output the "most" compression by default (which would render my question moot)?

Lastly, eac3to uses libFLAC for it's decoding/encoding, correct? Is there a way to update this to the newest version 1.3.1 or is it even necessary?

Thanks

Music Fan

15th January 2015, 16:55

Is there a way to update this to the newest version 1.3.1
Yes ;
http://forum.doom9.org/showthread.php?p=1701272#post1701272

bmcelvan

15th January 2015, 21:27

Yes ;
http://forum.doom9.org/showthread.php?p=1701272#post1701272

Thank you

r0lZ

18th January 2015, 12:55

When eac3to cannot process a command, it outputs an error message somehow. For example, typing eac3to test.dts test.ac3 when test.dts doesn't exist produces this in the command prompt window: Source file "test.dts" not found. So far so good.
But I need to grab the output of eac3to to process it. Therefore, I need to redirect the error message to a file. If I redirect stdout to a file with eac3to test.dts test.ac3 > log.txt, the error message is still visible in the command prompt window, and is not present in the log file. Therefore, I must assume that the error message is printed to stderr, as it should.
But if I redirect stderr to a file with eac3to test.dts test.ac3 2> err.txt, the error message is not printed in the console any more, but the err.txt file is empty! :confused:
Redirecting stdout and stderr in two different files with eac3to test.dts test.ac3 > log.txt 2> err.txt or to the same file with eac3to test.dts test.ac3 1>log.txt 2>&1 doesn't work either.
In all cases, I can't get the error message. It seems to be destroyed as soon as it is redirected to a file!
I know that eac3to uses sequences of backspace and space characters to format its output, but I suppose that these characters should not destroy the messages when they are redirected to a file. I don't understand what's happening.

Is it possible to redirect the error messages to a file handle?
Is it an undocumented option similar to -progressnumbers to force eac3to to NOT use its sequences of control characters to format its output?

Thanks in advance for any pointer.

Overdrive80

18th January 2015, 14:25

Have you tried eac3to test.dts test.ac3 2>&1 >log.txt??

r0lZ

18th January 2015, 14:39

Yes. If you use that syntax, the error message is still printed in the command prompt window, and the log file contains only backspace and space characters.

The syntax I've described above eac3to test.dts test.ac3 1>log.txt 2>&1 is the way recommended by M$ to redirect stderr and stdout to the same file, and it works somewhat better: nothing is printed to the command prompt window. But the content of the log file is exactly identical than with your syntax. The message is lost in the two cases, and only the control characters are kept in the log.

AYColumbia

18th January 2015, 16:44

Why don't you just use the -log switch?

r0lZ

18th January 2015, 17:59

Why don't you just use the -log switch?

E:\>eac3to test.dts test.ac3 -log
Command line parameter "log" is unknown.

LigH

18th January 2015, 19:58

Maybe try LoRd_MuldeR's Logging Utility (https://sourceforge.net/projects/muldersoft/files/Logging%20Utility/), it can log both stdout and stderr to the same file (and also does *not* confuse text editors with a bunch of "delete" characters = ^H, like the normal eac3to help output does).

r0lZ

18th January 2015, 20:39

Thanks for the suggestion, but that doesn't work. Again, the error message is not in the output of LoggingUtil.exe, regardless of the syntax and parameters used.

It seems that eac3to "knows" that its output is captured, and it modifies its formatting accordingly. I guess it suppress the error message by mistake (bug).

LigH

18th January 2015, 20:46

So, for example, which command line did you use with the logger?

There is a third method to output text: Manipulating the console output in the "graphic memory". Whatever that is in Windows NT... But that's quite unusual if a programmer once decides to use file output methods.

heerschop

18th January 2015, 21:01

E:\>eac3to test.dts test.ac3 -log
Command line parameter "log" is unknown.

Try this command line:

e:\>eac3to test.dts test.ac3 -log="test.txt"

This writes the error in the log file:

eac3to v3.27
command line: C:\Tools\eac3to\eac3to.exe test.dts test.ac3 -log="test.txt"
------------------------------------------------------------------------------
Source file "test.dts" not found. <ERROR>

greetz

r0lZ

19th January 2015, 01:13

Yeah, -log works with the syntax above. Thanks heerschop.
Why that option and -progressnumbers are not documented is a mystery. But at least they exist.
Now, I would like to redirect the log to STDOUT instead of to a physical file, so that I can parse the output while eac3to is running. How can I do that with the -log option? I've tried -log=STDOUT and -log=-, but each time, a file is written to disc.

LigH

19th January 2015, 06:21

I wonder if -log="CON:" (with or without colon) may work. The "CON"sole was the combined device for stdin and stdout in good old MS-DOS times.

r0lZ

19th January 2015, 12:00

Well, indeed, you are right. Using CON: (with and without the quotes, and with and without the colon) seems to work. Indeed, no file is written to disc, and the message appears on screen. But... the original behaviour is back! The error message appears again with a red background, there are again a lot of BS and Space characters printed to STDOUT. More importantly, the error message disappears again if STDERR is redirected. In fact, it seems that the -log argument is totally ignored when CON: is used.

I think now that there is absolutely NO WAY to grab the error messages when eac3to is launched from another program. It's a very serious bug, and I suppose that we have to live with it. Pity!

@LigH: I'm sure that the error message is not simply hidden because a text editor is confused by the BS characters. I have used an hex editor to examine the outputs. Also, I can already parse the output of eac3to when it prints the stream information, for example. It is theoretically sufficient to remove all BS and tailing spaces to get a "normal" output. It's only the STDERR output that is wrong and cannot be processed properly.

Thanks to all of you.

Kurtnoise

19th January 2015, 12:17

Anyway, using -log has a serious drawback. I have no way to distinguish the error message (if any) from the other messages. There is no leading string such as "ERROR:" to identify the line containing the error in the log.
This is not true...as soon as an error occurs, you have the <ERROR> string that appears.

r0lZ

19th January 2015, 12:27

OK, I think I have a working solution.
In fact, if you use the -log option, the log file is written to disc only at the end of the process (or when the process is aborted due to an error). But STDOUT still contains the normal messages (including the progress numbers when --progressnumbers is specified, as well as all that nasty BS and Spaces, that can be stripped out). Therefore, it is still possible to display the messages and update a progress bar while eac3to is working. Only the error messages are missing. But if the process returns a non-zero return code, you can then open the log file that has been written to disc, and search for the lines ending with "<ERROR>". They are the lines that are theoretically printed to stdout, but that are unavailable when STDOUT is redirected. It's a somewhat convoluted method, but it works.
Thanks again to everybody.

[EDIT] Kurtnoise has posted his message when I was writing mine and I was editing my previous message. Yes, <ERROR> is printed in the log at the end of the lines containing the error messages. I haven't noticed that when I did my first tests.

stax76

21st January 2015, 23:16

This and hoping for a fix is all we can do, I could read from a simple colored test app so it's possible.

frenshprince

24th January 2015, 19:32

Hello,

Coule you tell mi if its possible to use eac3to to convert a bluray playlist (mpls) with multiple m2ts to one m2ts ?

Of course I could use tsmuxer to do that, but I don't know why, it doen't work well' and the length of the m2ts is doubled.

Thanks

Music Fan

24th January 2015, 22:35

Coule you tell mi if its possible to use eac3to to convert a bluray playlist (mpls) with multiple m2ts to one m2ts ?
Previous page ;
http://forum.doom9.org/showthread.php?p=1704811#post1704811

tebasuna51

25th January 2015, 00:40

eac3to can extract the video to one mkv (or one video stream .h264/.vc1), can extract/recode the audio streams and extract subs.

But can't mux the streams to m2ts.

frenshprince

25th January 2015, 02:50

That's what I thought.

Thank you for your help.

stinman

25th January 2015, 04:27

Wil this program ever be able to decode TrueHD with Atmos or at least still decode TrueHD and disregard the atmos data like Lav Filters do? Or is there any way to get eac3to use the Lav Filters kind of like it will use the ArcSoft Decoders? I tried to put the Lav Filters in the folder with eac3to 3.27 and the HDDVD Bluray Stream Extracter to see if it would work. The only way to to extract a TrueHD atmos stream now is to run a MKV thru MakeMKV and out put the TrueHD atmos to flac, then the flac can be converted to wavs. I can't code like you guys can, if I could I would try and make eac3to use the new lav filters.:confused: ;)

Music Fan

25th January 2015, 12:09

That's what I thought.
You can use eac3to anyway to demux streams then use TSMuxer to remux in ts (or m2ts).

tebasuna51

25th January 2015, 12:36

The only way to to extract a TrueHD atmos stream now is to run a MKV thru MakeMKV...

You can use also ffmpeg, search in this thread. Or better in ffmpeg syntax (http://forum.doom9.org/showthread.php?p=1695623#post1695623) thread.

The problem was already reported to madshi.

dade49

26th January 2015, 05:35

The only way to to extract a TrueHD atmos stream now is to run a MKV thru MakeMKV and out put the TrueHD atmos to flac, then the flac can be converted to wavs.
This is much simpler. Extract the TrueHD Atmos stream to a multi-channel wav via ffmpeg, and then run that wav through eac3to to dts or whatever.
ffmpeg -i 00000.m2ts -map 0:1 extract.wav
eac3to extract.wav extract.dts

tebasuna51

26th January 2015, 14:02

Post about ffmpeg moved to ffmpeg syntax (http://forum.doom9.org/showthread.php?p=1707195#post1707195) thread.

Please don't use this thread for ffmpeg discussions.

Arm3nian

31st January 2015, 05:12

I'm experimenting with the dolby headphone wrapper plugin that's available with foobar, so I'm remapping channels with eac3to to see the effects. My source file is the song from the end of the first hobbit movie (7.1 channel dts-hd ma). I downmix to 5.1 and then analyze the channels with the foobar oscilloscope. The default output of eac3to is L,R,C,LFE,SL,SR, and is verified by the waveforms. The loudest is the center, followed by the left and right, then the two surrounds, and the LFE is near non existent, but is there.

http://imgur.com/DeIVxyd

I did the same thing but remapped the channels to L,R,SL,SR,LFE,C. As can be seen in the oscilloscope, the center is still the loudest (6th waveform), followed by the left and right (first and second waveforms), but the LFE somehow has more activity now and is louder (5th waveform). The SL and SR are closely related in activity and amplitude to each other, just like before remapping, but have lower activity and amplitude than the LFE... and also have lower amplitude than before the remap.

http://imgur.com/mTc9GgA

What's going on? Does eac3to apply different mix levels while disregarding which channel is what and going just based on sequential order, or is foobar doing something.

tebasuna51

31st January 2015, 11:34

...I did the same thing but remapped the channels to L,R,SL,SR,LFE,C.

Please put the log or at least the command line used to do this.

Does eac3to apply different mix levels while disregarding which channel is what and going just based on sequential order,

Of course, the channel order define which channel is what.

or is foobar doing something.

I don't know what you do with foobar.

stax76

31st January 2015, 13:26

Hi everybody,

I've improved StaxRip's eac3to support quite much lately, maybe a few people here have some interest to look at the latest release:

http://forum.doom9.org/showthread.php?p=1707791#post1707791

http://s17.postimg.org/vls7dqkaz/Unbenannt4.png (http://postimg.org/image/vls7dqkaz/)

http://s7.postimg.org/f62v5hs07/Unbenannt.jpg (http://postimg.org/image/f62v5hs07/)

Arm3nian

1st February 2015, 00:51

Please put the log or at least the command line used to do this.
Of course, the channel order define which channel is what.

14612
So if the channels are not in the default order when remapped then they get mixed at wrong levels?

tebasuna51

1st February 2015, 19:17

eac3to input71.dt output.wav -down6
finish with:
FL' = FL (0)
FR' = FR (1)
FC' = FC (2)
LF' = LF (3)
SL' = BL + SL (4 + 6)
SR' = BR + SR (5 + 7)

But with remap: eac3to input71.dt output.wav -down6 -0,1,4,5,3,2
finish with:
FL' = FL (0)
FR' = FR (1)
FC' = BL (4)
LF' = BR (5)
SL' = LF + SL (3 + 6)
SR' = FC + SR (2 + 7)

Seems first remap the first 6 channel and after mix the Side channels .

Maybe you need 2 pass, first to only -down6 and after remap with -0,1,4,5,3,2

Arm3nian

6th February 2015, 03:05

kevmitch

11th February 2015, 13:26

This seems to be a semi-popular topic, but I don't think my specifc question has previously been answered.

With lord of the rings 6.1 dts-hd-ma audio

C:\> eac3to lotr.mkv
MKV, 1 video track, 5 audio tracks, 5 subtitle tracks, 2:07:41, 24p /1.001
1: h264/AVC, English, 1080p24 /1.001 (16:9)
2: DTS Master Audio, English, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz)
"Surround 6.1"
...

I decode it to flac (arcsoft 1.1.0.0)

C:\> eac3to lotr.mkv 1:lord_of_the_rings.flac

What order will the tracks be in the flac file for example if I open it in audacity?

I have verified that the flac is bit-for-bit identical (including channel order) to the makemkv decode results. This post (http://www.makemkv.com/forum2/viewtopic.php?f=4&t=5710#p23881) on the makemkv forum says that the channel order should therefore be

L R C LFE Ls Rs Cs

However, if I produce my own 7 channel flac file (with libsoundfile/audiolab), I find that downmixing it with eac3to

eac3to my_7_channel.flac my_7_channel.stereo.flac -downstereo

emperically uses the following matrix

[[ 1. 0. 0.7071 0. 0.7071 1. 0. ]
[ 0. 1. 0.7071 0. 0.7071 0. 1. ]]

which suggests that it is interpeting my file to have the order

L R C LFE Cs Ls Rs

Is this discrepancy due to differing channel mask between my file and that created by eac3to? Or is my first conclusion incorrect that eac3to places the centre surround channel last in the flac when decoding from dts-hd-ma?

nevcairiel

11th February 2015, 13:50

The "official" FLAC 6.1 order is L R C LFE Cb Ls Rs .. unfortunately 6.1 is annoying and both orders are commonly used.
See here: http://xiph.org/flac/format.html#frame_header (scroll down to <4> channel assignment)

7 channels: front left, front right, front center, LFE, back center, side left, side right

The MakeMKV post you referenced is old, since FLAC did add official 7 and 8 channel orders now.
However, it may be nice and write metadata into the FLAC headers to indicate the channel order, which eac3to should hopefully read.

tebasuna51

11th February 2015, 19:06

This post (http://www.makemkv.com/forum2/viewtopic.php?f=4&t=5710#p23881) on the makemkv forum says that the channel order should therefore be

L R C LFE Ls Rs Cs

And the info in the post is wrong.

I tested MakeMKV decoding to wav a DTS-MA 6.1 and output the standard 6.1 channel order (the same than eac3to output):

FL FR FC LF BC SL SR

where
BC = Back Center
SL = Side Left
SR = Side Rigth

Talking about channel order please use always the channel names and order defined here (http://www-mmsp.ece.mcgill.ca/documents/AudioFormats/WAVE/Docs/multichaudP.pdf) to avoid confusions.

The channel names and order inside a encoded format can be differents but, to transcode, we needs always decode to standard WAV order and channels, to allow the new encoder convert the audio to their internal order and channels.

Standard order for 7.1 is (like I put in my precedent post):

FL FR FC LF BL BR SL SR

Thunderbolt8

13th February 2015, 17:17

regarding 2.1 DTS-HD MA tracks: Does the arcsoft bug (LFE channel cannot be decoded) still exist in each arcsoft version? Or is there a version which is not affected by the bug?

when I play such a track with MPC-HC & LAV Audio with arcsoft capabilities, I wont hear the LFE channel then? what happens when I transform the 2.1 track with eac3to, using the arcsoft decoder, e.g. into wavs. will the LFE channel then be processed incorrectly? but when I just demux the track nothing happens?

stax76

13th February 2015, 18:07

M2TS, 1 video track, 2 audio tracks, 5 subtitle tracks, 24p /1.001
1: Chapters, 24 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 Surround, English, 2.0 channels, 320kbps, 48kHz
5: Subtitle (PGS), English
6: Subtitle (PGS), Swedish
7: Subtitle (PGS), Norwegian
8: Subtitle (PGS), Danish
9: Subtitle (PGS), Finnish
a03 The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
a03 The libav DTS decoder doesn't decode the full DTS-HD information.
Creating file "V:\Blurays and DVDs\Saving Mr Banks\Saving Mr Banks_Chapters.txt"...
v02 Extracting video track number 2...
a03 Extracting audio track number 3...
a03 Extracting DTS core...
a03 Decoding with libav/ffmpeg...
a03 Reducing depth from 64 to 32 bits...
a03 Encoding AAC <0.50> with NeroAacEnc...
v02 Creating file "V:\Blurays and DVDs\Saving Mr Banks\Saving Mr Banks.h264"...
a03 0:21:12 The source file seems to be damaged (sync byte missing).
v02 0:21:13 The source file seems to be damaged (sync byte missing).
v02 0:21:13 The source file seems to be damaged (discontinuity).
a03 0:21:12 The source file seems to be damaged (discontinuity).
a03

Can somebody tell me what could be the reason for this error?

Currently it's not possible that GUIs can read colored eac3to output, because of this I don't have the full error message.

Snowknight26

13th February 2015, 19:45

The reason is probably that "the source file seems to be damaged."

tebasuna51

13th February 2015, 22:13

regarding 2.1 DTS-HD MA tracks: Does the arcsoft bug (LFE channel cannot be decoded) still exist in each arcsoft version? Or is there a version which is not affected by the bug?

Using dtsdecoderdll v1.1.0.0 the LFE is decoded without problems.

GCRaistlin

14th February 2015, 22:21

Does the DRC bug (http://forum.doom9.org/showthread.php?p=1404212#post1404212) still exist in 3.27?

tebasuna51

14th February 2015, 23:06

Does the DRC bug (http://forum.doom9.org/showthread.php?p=1404212#post1404212) still exist in 3.27?

Seems you talk about the NERO DRC bug (http://forum.doom9.org/showthread.php?p=1399113#post1399113)

Now the default decoder for AC3 is libav instead Nero for that problem.

GCRaistlin

14th February 2015, 23:20

tebasuna51, using libav leads to another issue (http://forum.doom9.org/showthread.php?p=1508936#post1508936), doesn't it?

nevcairiel

15th February 2015, 01:19

tebasuna51, using libav leads to another issue (http://forum.doom9.org/showthread.php?p=1508936#post1508936), doesn't it?

Those issues have long been fixed.

r0lZ

15th February 2015, 11:22

Another question about the Arcsoft DTS decoder. Is it better to use it when decoding "regular" 5.1 DTS tracks? Or is it useful only for the recent 7.1 evolutions? AFAIK, a decoder that is able to decode a specific format (without bugs) should give exactly the same result than another decoder, and therefore there is no advantage to use the slow Arcsoft decoder when decoding a DTS tracks that the libav decoder can also decode properly. Am I right?

tebasuna51

15th February 2015, 13:08

tebasuna51, using libav leads to another issue (http://forum.doom9.org/showthread.php?p=1508936#post1508936), doesn't it?
What is the problem?

Of course there are very light differences between decoders but jruggle (ffmpeg developer in AC3 area) minimize the problem.
eac3to uses libav to decode AC3 to 64 bits float (the best resolution), and libav was actualized in eac3to v3.27 (the test was made with 3.24).
I'm trust in the AC3 expert (jruggle) about that.

BTW, if you prefer use a class "A" free decoder liba52 you can use BeHappy with NicAudio to decode AC3.

tebasuna51

15th February 2015, 13:33

Another question about the Arcsoft DTS decoder....is it useful only for the recent 7.1 evolutions?

libav can't decode DTS-HD, only the 'core' is decoded, then can't be used to obtain 7.1 output and lose info from DTS-HR or DTS-MA (lossless).

... Is it better to use it when decoding "regular" 5.1 DTS tracks?... AFAIK, a decoder that is able to decode a specific format (without bugs) should give exactly the same result than another decoder, and therefore there is no advantage to use the slow Arcsoft decoder when decoding a DTS tracks that the libav decoder can also decode properly. Am I right?

Lossy decoders not always output exactly the same result.
There are light differences (like with AC3 in precedents posts).
ArcSoft is a "certified" decoder, libav a free decoder.
It's your choice.

r0lZ

15th February 2015, 16:45

OK, thanks for the information.

Thunderbolt8

18th February 2015, 17:12

got an error message:

X:\file.m2ts>eac3to 1) 2: X:\file.mkv
M2TS, 1 video track, 11 audio tracks, 10 subtitle tracks, 1:48:42, 24p /1.001
1: Chapters, 37 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
...
v02 Extracting video track number 2...
v02 The h264 muxer doesn't support this stream type yet.
v02 Please send a 20MB sample to dear@madshi.net
v02 Muxing video to Matroska...
Aborted at file position 1048576.

sample can be found here https://www.sendspace.com/file/f7n75i (its a file from 2013)

bmcelvan

25th February 2015, 17:11

I am having a problem trying trying to decode a TrueHD to anything, wavs, w64, wav or flac. I get the following log:

bitstream parsing for track 2 failed
demuxing this track may still produce correct results - or not.
This audio conversion is not supported.

The eac3to info says:
2: TrueHD, English, 7.1 channels, 48KHz

The mediainfo says:
Audio #1
ID : 2
Format : TrueHD
Codec ID : A_TRUEHD
Duration : 2h 2mn
Bit rate mode : Variable
Maximum bit rate : 7 719 Kbps
Channel(s) : 8 channels
Channel positions : Front: L C R, Side: L R, Back: L R, LFE
Sampling rate : 48.0 KHz
Compression mode : Lossless
Language : English
Default : Yes
Forced : No

I have demuxed using mkvtoolnix and changed the extension to .ac3, .thd+ac3 and .truehd and when I try to look at it with eac3to it just says The format of the source file could not be detected.

tsmuxer won't recognize it and vidcoder (handbrake) doesn't even acknowledge it is there.

Could this be some new form of protection from Dolby. Do I need to update a codec or something.

Could it by Dolby atmos - is there a decoder for that?

bmcelvan

25th February 2015, 18:09

*Edit: My apologies, due to bad searching I didn't find the posts about using ffmpeg.

Is there a way to determine if the above track I was talking about is Dolby Atmos. Is that what the A in A_TrueHD stands for? If Yes, then:

I just want to be certain. If I have an .mkv file or a Bluray folder with a 5.1 or 7.1 TrueHD (atmos) track and use the following command, it will output the lossless TrueHD extracted audio in a flac file with proper channel mapping?

ffmpeg -i "my file or folder" -map 0:1 "extracted_audio.flac"

ndjamena

25th February 2015, 18:29

(The "A" is for audio)

Sparktank

26th February 2015, 03:16

Is there a way to determine if the above track I was talking about is Dolby Atmos.

The case of the movie should tell you if it has Dolby Atmos.

tebasuna51

26th February 2015, 10:08

...
ffmpeg -i "my file or folder" -map 0:1 "extracted_audio.flac"

That must work for the second track of a input file (tested .m2ts or .mkv).
I don't know what happens with a folder input.

For questions about ffmpeg syntax please use this thread:
ffmpeg syntax (http://forum.doom9.org/showthread.php?t=171251)

eddman

27th February 2015, 01:26

LigH

27th February 2015, 09:05

You can certainly convert DVD Video compatible dts or Dolby Digital audio with other tools, like BeHappy or MeGUI, maybe TEncoder. HD audio from Blu-rays is probably not widely supported though.

tebasuna51

27th February 2015, 12:00

I tried replacing the avcodec-54.dll and avutil-52.dll files with newer versions from the latest version of Zeranoe FFmpeg, but eac3to gives me an error that the dll is missing.

What am I doing wrong?
The dll's for eac3to aren't the standard dll's, must be compiled for eac3to by madshi.

As a side question; is there any other tool/method for converting DTS or dolby audio besides eac3to?
The answer is in your first question: ffmpeg

Try TAudioConverter (http://forum.doom9.org/showthread.php?t=165577) (ffmpeg)
Or AviSynth based BeHappy/MeGUI like LigH say.

shh

27th February 2015, 14:02

Hello!
Config:
C:\TMP>C:\eac3to\eac3to.exe -test
eac3to (v3.27) is up to date
Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed
...
ArcSoft DTS Decoder (1.1.0.8) works fine
Sonic Audio Decoder (3.27.0.0) doesn't seem to be installed

I got some problems with a DTS-HD MA Mono file:
10MB snippet: http://www.file-upload.net/download-10353765/test.dtsma.html
C:\TMP>C:\eac3to\eac3to.exe test.dtsma
DTS Master Audio, 1.0 channels, 24 bits, 48kHz
(core: DTS, 1.0 channels, 24 bits, 768kbps, 48kHz)
Transcoding to WAV / FLAC / ... results in a sped up audio file.
Only way to handle this is extracting the DTS-core and convert the DTS afterwords.
However, MPC-HC is able to play the original .dtsma file. So the ArcSoft DTS Decoder does work for this file?
Any hints on what to do?

Music Fan

27th February 2015, 15:18

You should maybe try ArcSoft DTS Decoder v1.1.0.0 instead of 1.1.0.8 for this file.

shh

27th February 2015, 22:43

v1.1.0.0 workend. :eek:
Thanks a lot!

73ChargerFan

28th February 2015, 20:59

This is much simpler. Extract the TrueHD Atmos stream to a multi-channel wav via ffmpeg, and then run that wav through eac3to to dts or whatever.
ffmpeg -i 00000.m2ts -map 0:1 extract.wav
eac3to extract.wav extract.dts

Madshi, any ETA when you'll be able update to discard the Atmos extensions? I'm finding it in more and more BDs.

Thanks.

Batman007

1st March 2015, 16:18

Guys
I'm trying to slow down an audio from 25.000 FPS to 23.976 FPS through ea3to
I put this code into cmd
C:\eac3to\ea3to.exe "D:\MeGUI 2418\MeGUI output\AVSEQ02 Tc0 L2 2ch 48 224 DELAY 0ms.mp3" "101 D.mp3" -changeTo23.976

But I'm getting this error :
http://i.imgur.com/kbhusvs.jpg

Please tell me what mistake I'm making or what I'm missing

Thank you ......

LigH

1st March 2015, 16:24

ea>c<3to.exe

r0lZ

1st March 2015, 16:25

Try:
eac3to.exe "D:\MeGUI 2418\MeGUI output\AVSEQ02 Tc0 L2 2ch 48 224 DELAY 0ms.mp3" "101 D.mp3" -slowdown

-slowdown convert 25.000 and 24.000 content to 23.976 fps

xxx666yyy777

1st March 2015, 16:55

Guys
I'm trying to slow down an audio from 25.000 FPS to 23.976 FPS through ea3to
I put this code into cmd

But I'm getting this error :
http://i.imgur.com/kbhusvs.jpg

Please tell me what mistake I'm making or what I'm missing

Thank you ......

BTW - you can usually use the "Tab" button to auto-complete directories and filenames in the MS-DOS window...