eac3to - audio conversion tool [Archive] - Page 12

djmasturbeat

9th March 2011, 20:25

Hi i am a noob to eac3to, so bear with me if this was answered in the 550 pages before :)

i am trying to convert 5.1 FLAC (48k / 24 bit) files to LPCM (or DTSHDMA or True HD), if this is possible.
I have tried to convert to LPCM, wavs and w64 but i am getting errors.

example of log (same errors i get for wavs or w64 or LPCM)

eac3to v3.24
command line: "C:\Users\xxx\Downloads\Application exe files\eac3to\eac3to Version3.24\eac3to" "C:\Disc 8\Doomsday.Unrated\Doomsday.Unrated_track3.flac" "C:\Disc 8\Doomsday.Unrated\Doomsday.Unrated_track3.wavs"
------------------------------------------------------------------------------
FLAC, 5.1 channels, 1:52:51, 24 bits, 4239kbps, 48kHz
Decoding FLAC...
Writing WAVs...
Creating file "C:\Disc 8\Doomsday.Unrated\Doomsday.Unrated_track3.L.wav"...
Creating file "C:\Disc 8\Doomsday.Unrated\Doomsday.Unrated_track3.SR.wav"...
Creating file "C:\Disc 8\Doomsday.Unrated\Doomsday.Unrated_track3.R.wav"...
Creating file "C:\Disc 8\Doomsday.Unrated\Doomsday.Unrated_track3.C.wav"...
Creating file "C:\Disc 8\Doomsday.Unrated\Doomsday.Unrated_track3.LFE.wav"...
Creating file "C:\Disc 8\Doomsday.Unrated\Doomsday.Unrated_track3.SL.wav"...
The frame's data did not match the CRC in the footer. <ERROR>
Aborted at file position 1032060928. <ERROR>

what am i leaving out? is there some header i need to remove or change. not sure why i am getting crc errors. i have extracted the FLAC from an mkv remux .
I could make it easier and just re-rip it from disc to make a backup disc (which is what i am trying to do here), but now i am curious how i go about converting FLAC back to other lossless audio accepted on Bluray discs. Any noob-friendly advice would be appreciated.

EDIT:
I have tried using megui (which in turn uses eac3to) and i still get the same type of errors. Not sure what the issue is, if someone can help me i would appreciate it.
Now i suppose i should try other BD/ HDDVD movies in which i xcoded the audio to FLAC, since so far i have only tried with one remux/ reencode i have made.

TDiTP_

12th March 2011, 21:45

To All.

What about decoding E-AC3 in eac3to? I have a problem (sample is here (http://www.mediafire.com/?2q2a0vrx30i1hm6)).

1. Decode with libav-eac3to:

eac3to v3.24
command line: eac3to 1.ec3 1.lib.wavs -libav
------------------------------------------------------------------------------
E-AC3, 5.1 channels, 0:04:06, 1536kbps, 48kHz, dialnorm: -27dB
Removing E-AC3 dialog normalization...
Decoding with libav/ffmpeg...
Remapping channels...
Reducing depth from 64 to 24 bits...
Writing WAVs...
Creating file "1.lib.L.wav"...
Creating file "1.lib.C.wav"...
Creating file "1.lib.SL.wav"...
Creating file "1.lib.SR.wav"...
Creating file "1.lib.LFE.wav"...
Creating file "1.lib.R.wav"...
eac3to processing took 16 seconds.
Done.

2. Decode with nero-eac3to. I know about bug (http://forum.doom9.org/showthread.php?p=1404212#post1404212) in DRC-off but anyway:

eac3to v3.24
command line: eac3to 1.ec3 1.nero.wavs
------------------------------------------------------------------------------
E-AC3, 5.1 channels, 0:04:06, 1536kbps, 48kHz, dialnorm: -27dB
Disabling DRC for Nero (E-)AC3 decoding...
Removing E-AC3 dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 5.1 channels, 24 bits, 48kHz
Writing WAVs...
Creating file "1.nero.L.wav"...
Creating file "1.nero.R.wav"...
Creating file "1.nero.SR.wav"...
Creating file "1.nero.SL.wav"...
Creating file "1.nero.C.wav"...
Creating file "1.nero.LFE.wav"...
eac3to processing took 19 seconds.
Done.

3. Decode with libav-ffmpeg (build from last MeGui):

D:\test>ffmpeg.exe -i 1.ec3 1.ffmpeg.wav
FFmpeg version SVN-r25870-Sherpya, Copyright (c) 2000-2010 the FFmpeg developers
built on Dec 4 2010 05:23:23 with gcc 4.2.5 20090330 (prerelease) [Sherpya]
libavutil 50.33. 0 / 50.33. 0
libavcore 0.14. 0 / 0.14. 0
libavcodec 52.97. 2 / 52.97. 2
libavformat 52.87. 1 / 52.87. 1
libavdevice 52. 2. 2 / 52. 2. 2
libavfilter 1.67. 0 / 1.67. 0
libswscale 0.12. 0 / 0.12. 0
libpostproc 51. 2. 0 / 51. 2. 0
[eac3 @ 01a7ce80] max_analyze_duration reached
[eac3 @ 01a7ce80] Estimating duration from bitrate, this may be inaccurate
Input #0, eac3, from '1.ec3':
Duration: 00:04:06.12, bitrate: 1536 kb/s
Stream #0.0: Audio: eac3, 48000 Hz, 5.1, s16, 1536 kb/s
Output #0, wav, to '1.ffmpeg.wav':
Metadata:
encoder : Lavf52.87.1
Stream #0.0: Audio: pcm_s16le, 48000 Hz, 5.1, s16, 4608 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Press [q] to stop encoding
size= 138447kB time=246.13 bitrate=4608.0kbits/s
video:0kB audio:138447kB global headers:0kB muxing overhead 0.000048%

Results. Comparison C-channels:
1.
http://thumb.phyrefile.com/t/td/tditp/2011/03/12/300/eac3to-libav.png (http://www.phyrefile.com/image/view/XgshvYoL08UtF4N2)http://thumb.phyrefile.com/t/td/tditp/2011/03/12/300/eac3to-nero.png (http://www.phyrefile.com/image/view/2KMgP7g29ELn9EM0)http://thumb.phyrefile.com/t/td/tditp/2011/03/12/300/ffmpeg.png (http://www.phyrefile.com/image/view/S9JttITT9bx7wQWn)
Comparing libav-eac3to and nero-eac3to, looks like bug in DRC. DRC is only partially ignored: volume increasing still applies, while attenuation is ignored.
But what about libav-ffmpeg?

2.
Spectrums:
http://thumb.phyrefile.com/t/td/tditp/2011/03/12/300/spectrum.png (http://www.phyrefile.com/image/view/g33GdGWPplFOrRIT)
-----------------------------------
Also i used Nero9 and Sonic decoders (graphedit: source->dec->WavDest->FileWriter) but DRC and DN was applied - it was expected, they're Dolby certified. AFAIK, libavcodec - is only one freeware opensource decoder and it doesn't apply DN and DRC.
I've tried ffdshow tryouts [rev3771 Mar 3 2011 with] with graphedit. In this revision ffmpeg was updated: "Libavcodec AC3/E-AC3/DTS decoders now output floating point data.". With this one i've got the same result as with libav-eac3to (-full). byte-in-byte but WAV (eac3to) is bigger WAV (ffdshow) in last 2048 byte.

So, i can't understand why ffmpeg.exe produces a different result and what is right solution for decode E-AC3 ? Unfortunately I have no DD+ Encoder and can't check it myself.

henryho_hk

15th March 2011, 05:45

Can I force eac3to listing all playlists of a bluray disc (it currently excludes short playlists like trailers)?

Vilous

16th March 2011, 03:50

Hope someone can help me with this mystery I've been having. I'm betting it is something obvious.. well obvious to someone.

I've been using RipBot264 for about a year now, converting Blu-Rays to AVCHD's. And for the most part, everything has been going fine but recently, things stopped working.

So, I installed a new copy of RipBot264 to a new location on a mostly empty 3TB harddrive, and uninstalled and reinstalled all the pre-requisites for RipBot to work (AviSynth, Haali, FFDShow, and Java), just to make sure it wasn't some problem with those. After doing all that, I still have the same issue.

Here's what happens. (this is happening now with every bluray I try to rip)

So copy the bluray to my HD, I use RipBot to 'add' one of the M2TS files from the Stream folder. I then select the playlist that I want to use and keep most of the other default settings on that screen.

Now, clearly, since I got this far, eac2to has analyzed the bluray and presented me with the playback options. Looking in the job file I see all of this in the job folder under the ripBot264temp folder.

Anyway, at this point I choose OK, and here's what happens.. it gets stuck Demuxing the stream at Progress 1% -forever-. No log file is seemingly generated, at least nothing with any useful debug info.

Looking at the processor, eac2to.exe drops to 00 CPU usage. And it never goes back up, I have left one job there for days. (btw, I turned OFF Low Priority on RipBot's main screen)

So, in desperation, I tried running the command from Blu-Ray_ExtractStreams.cmd that Ripbot generates from a command line.

Not surprisingly, it gets through the analyze phase with no problems, but when it gets to process phase, it hands right there at 1%. Here's the last few lines of output from eac3to.exe

s08 Extracting subtitle track number 8...
v02 Extracting video track number 2...
a03 Extracting audio track number 3...
v02 Muxing video to Matroska...
a03 Decoding with ArcSoft DTS Decoder...
a03 Reducing depth from 24 to 16 bits...
a03 Writing W64...
a03 Creating file "G:\temp\RipBot264temp\job1\audio_English.dtsma.w6 4"...
process: 1%

And it stays there forever. There is plenty of harddrive space on G. Everything has the right permissions to run. The OS is Windows 7.

If anyone knows what eac3to is doing at this point, or has some clue as to how I can proceed to figure out why it is stuck, I'd be VERY grateful.

Thanks,
-Vilous

TinTime

16th March 2011, 11:20

Can I force eac3to listing all playlists of a bluray disc (it currently excludes short playlists like trailers)?

Short answer - no, because otherwise there would (typically) be more information than can be viewed in a command window.

You'll need to run eac3to against the individual playlists.

tebasuna51

16th March 2011, 11:38

...
So, in desperation, I tried running the command from Blu-Ray_ExtractStreams.cmd that Ripbot generates from a command line....

Please post the Blu-Ray_ExtractStreams.cmd and the full log.

henryho_hk

16th March 2011, 11:53

You'll need to run eac3to against the individual playlists.

Do you mean "eac3to.exe <playlist file full path name>"? Let me try....

TinTime

16th March 2011, 12:14

Do you mean "eac3to.exe <playlist file full path name>"? Let me try....

Yes, that's exactly it.

e.g.

eac3to.exe "F:\BDMV\PLAYLIST\00001.mpls"

You can treat the mpls files in the same way as a stream or m2ts file.

Vilous

16th March 2011, 17:32

Please post the Blu-Ray_ExtractStreams.cmd and the full log.

Here's the eac3to command line from the CMD file:

"G:\Tools\Ripbot264\Tools\eac3to\eac3to.exe" "G:\Working\bd\SALT\" 1) 2: "G:\temp\RipBot264temp\job1\video.mkv" -seekToIFrames 4: "G:\temp\RipBot264temp\job1\audio_English.dtsma.w64" -down16 15: "G:\temp\RipBot264temp\job1\15_subtitles_English_1080.sup" 16: "G:\temp\RipBot264temp\job1\16_subtitles_English_1080.sup" 17: "G:\temp\RipBot264temp\job1\17_subtitles_French_1080.sup" 18: "G:\temp\RipBot264temp\job1\18_subtitles_Spanish_1080.sup" 19: "G:\temp\RipBot264temp\job1\19_subtitles_English_1080.sup" 20: "G:\temp\RipBot264temp\job1\20_subtitles_Spanish_1080.sup" 21: "G:\temp\RipBot264temp\job1\21_subtitles_English_1080.sup" 22: "G:\temp\RipBot264temp\job1\22_subtitles_Spanish_1080.sup" 23: "G:\temp\RipBot264temp\job1\23_subtitles_English_1080.sup" 24: "G:\temp\RipBot264temp\job1\24_subtitles_Spanish_1080.sup" 25: "G:\temp\RipBot264temp\job1\25_subtitles_English_1080.sup" 26: "G:\temp\RipBot264temp\job1\26_subtitles_Spanish_1080.sup" 1: "G:\temp\RipBot264temp\job1\chapters.txt" -progressnumbers -log="G:\temp\RipBot264temp\job1\demuxlog.txt"

Here is the entire output:

M2TS, 2 video tracks, 11 audio tracks, 12 subtitle tracks, 1:44:03, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
5: AC3, English, 2.0 channels, 192kbps, 48kHz
6: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz
7: DTS Master Audio, French, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48kHz)
8: AC3, Spanish, 5.1 channels, 640kbps, 48kHz
9: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz
10: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz
11: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz
12: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz
13: AC3 Surround, English, 2.0 channels, 448kbps, 48kHz
14: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48kHz
15: Subtitle (PGS), English
16: Subtitle (PGS), English
17: Subtitle (PGS), French
18: Subtitle (PGS), Spanish
19: Subtitle (PGS), English
20: Subtitle (PGS), Spanish
21: Subtitle (PGS), English
22: Subtitle (PGS), Spanish
23: Subtitle (PGS), English
24: Subtitle (PGS), Spanish
25: Subtitle (PGS), English
26: Subtitle (PGS), Spanish
Creating file "G:\temp\RipBot264temp\job1\chapters.txt"...
v02 Extracting video track number 2...
s15 Extracting subtitle track number 15...
s23 Extracting subtitle track number 23...
s16 Extracting subtitle track number 16...
s19 Extracting subtitle track number 19...
s26 Extracting subtitle track number 26...
a04 Extracting audio track number 4...
s24 Extracting subtitle track number 24...
s20 Extracting subtitle track number 20...
s17 Extracting subtitle track number 17...
s25 Extracting subtitle track number 25...
s22 Extracting subtitle track number 22...
s18 Extracting subtitle track number 18...
s21 Extracting subtitle track number 21...
v02 Muxing video to Matroska...
a04 Decoding with ArcSoft DTS Decoder...
a04 Reducing depth from 24 to 16 bits...
a04 Writing W64...
a04 Creating file "G:\temp\RipBot264temp\job1\audio_English.dtsma.w64"...
process: 1%

Also of note: As I said, eac3to.exe process stays around just gets 0% CPU forever. (its using a lot of memory while its doing nothing. :))

Also indicative of it actually doing nothing as opposed to doing something really slowly is that the audio_English.dtsma.w64 file stays at a static size of 26,624KB, and the video.mkv file stays at a static size of 2,688 KB

The demuxlog.txt file never gets created.

The file version of eac3to.exe is 3.24.0.0

Please let me know if there is any additional info you'd like.

tebasuna51

16th March 2011, 18:42

...Also of note: As I said, eac3to.exe process stays around just gets 0% CPU forever. (its using a lot of memory while its doing nothing. :))
...
the audio_English.dtsma.w64 file stays at a static size of 26,624KB, and the video.mkv file stays at a static size of 2,688 KB...

Please let me know if there is any additional info you'd like.
I don't know if is a bug or corrupt data.
I can suggest only try to run the .cmd with only one stream once.

And, maybe, the video without the -seekToIFrames parameter.

Vilous

16th March 2011, 21:39

I don't know if is a bug or corrupt data.
I can suggest only try to run the .cmd with only one stream once.

And, maybe, the video without the -seekToIFrames parameter.

Well, as I indicated, this happens with every movie or M2TS file I try this with now. (I get the same results with just a single stream. I haven't tried without the -seekToIFrames yet, I will do that an report back).

And it is just bizzarre, because everything -was- working fine for a long time, and I didn't change anything and STILL get this result after downloading all new software.:confused:

Oh well, thanks for looking.

-Vilous

Vilous

16th March 2011, 21:55

Well, as I indicated, this happens with every movie or M2TS file I try this with now. (I get the same results with just a single stream. I haven't tried without the -seekToIFrames yet, I will do that an report back).

And it is just bizzarre, because everything -was- working fine for a long time, and I didn't change anything and STILL get this result after downloading all new software.:confused:

Oh well, thanks for looking.

-Vilous

FYI. Removing -seekToIFrames had no impact on the hang. Still happened.

-Vilous

setarip_old

16th March 2011, 22:37

@Vilous

Hi!

Is the following just a typo, or possibly part of your problem? (Notice the space between "6" and "4"):
a03 Creating file "G:\temp\RipBot264temp\job1\audio_English.dtsma.w6 4"

tebasuna51

17th March 2011, 01:01

@setarip_old
See the last post, for that I request the CMD.

setarip_old

17th March 2011, 01:39

@tebasuna51

Hi!

I'm sorry, but I don't understand what you're trying to say...

Vilous

17th March 2011, 03:39

@tebasuna51

Hi!

I'm sorry, but I don't understand what you're trying to say...

It was a typo/artifact of a cut'n'paste.

In my second post with the full CMD and its full output, there is no space.

(I think he was trying to say he noticed that too and thus requested the full CMD)

-Vilous

Snowknight26

17th March 2011, 17:49

Out of space on your G drive? Tried changing output formats? What about input formats?

Vilous

17th March 2011, 21:54

Out of space on your G drive? Tried changing output formats? What about input formats?

First, I appreciate all efforts here.

That said, as I mentioned above, G has tons of room (over 2.5 TB free)

I can't really change the input format, the input format is the movie.

I can try to adjust the output format, and report back what happens. I'm not sure how it will help me if that works though.

-Vilous

Snowknight26

17th March 2011, 22:11

I can't really change the input format, the input format is the movie.

Try a different one if you can.

Try to do each extraction separately to see if a certain one stalls eac3to's progress.

Is the drive you're outputting to a RAID? It could be Haali Media Splitter's gdsmux not liking the drive. I've had an issue where eac3to couldn't write more than ~2MB an MKV on one of my RAIDs. The only solution was to switch RAID controllers/recreate the array. Never figured out exactly what caused it but it might be worthwhile to investigate.

Vilous

18th March 2011, 01:04

Try a different one if you can.

Try to do each extraction separately to see if a certain one stalls eac3to's progress.

Is the drive you're outputting to a RAID? It could be Haali Media Splitter's gdsmux not liking the drive. I've had an issue where eac3to couldn't write more than ~2MB an MKV on one of my RAIDs. The only solution was to switch RAID controllers/recreate the array. Never figured out exactly what caused it but it might be worthwhile to investigate.

The drive is not RAID. So that's not the issue.

I'm running some additional tests, the hangs appears to be in the creation of the mkv file. It does not hang when just creating the .w64 file.

So I ran -test and noticed that, despite Haali being installed and recognized, it says MkvToolnix is not installed.

I never installed that tool in the past, is it actually needed?

Snowknight26

18th March 2011, 01:19

No, it's not used for MKV muxing.

Try remuxing the file with gdsmux (in your Haali Media Splitter directory). Chances are it won't work either, in which case it's not an eac3to issue.

Vilous

18th March 2011, 01:42

No, it's not used for MKV muxing.

Try remuxing the file with gdsmux (in your Haali Media Splitter directory). Chances are it won't work either, in which case it's not an eac3to issue.

You are correct. It did not. It got hung up in a very similiar manner.

Any suggestions as to a good forum to seek support for this issue?

Thanks,
-Vilous

Snowknight26

18th March 2011, 20:52

This one, but it's hopeless. I don't think Haali reads the forums anymore so don't expect it to ever be fixed. The only alternate is to output to a different drive.

Qriist

19th March 2011, 01:55

Thank you for this tool. It is very useful.

I request a couple features, however.

1: A command parameter to demux a given file without demuxing the video. "-demux novideo"

2: The ability to output a plain-text list of all produced files from demux or mkv conversion. "-makelist"
Example list inside Movie1.txt:

video > c:\folder\movie1 - 2 - VC-1, 1080p24.vc1
audio > c:\folder\movie1 - 3 - DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz.dtsma
subtitle > c:\folder\movie1 - 4 - Subtitle (PGS), English.sup
subtitle > c:\folder\movie1 - 5 - Subtitle (PGS), Spanish.sup
subtitle > c:\folder\movie1 - 6 - Subtitle (PGS), French.sup

Please and thank you.
Please keep up the good work.

Sparktank

21st March 2011, 00:27

I have been using this tool for awhile. I really like it.
And I try to prowl around this thread for information.
I find enough to get done what I want to do.
I even use the Google search function to try narrow my search.
But some things are just difficult to solve.

For instance...
When I use the -demux command, it doesn't demux to same folder as the source (most cases ".m2ts" files).
Instead it demuxes files to my Documents...

WinXP
"C:\Documents and Settings\(username)\files.*"

Vista
"C:\Users\(username)\files.*"

I have to copy the files to the desired folder.

And when I do use the command to demux to a different folder, it saves the files in the desired folder and the Documents folder, too.

Is this normal behavior? (in both cases)

Edit:
Scenario one: Demuxing without setting parameter for new folder
eac3to v3.24
command line: "D:\Apps\eac3to\eac3to.exe" "E:\Work\Bram Stoker's Dracula (Trailer)\MainMovie\DRACULA_TRAILER\BDMV\STREAM\00112.m2ts" -demux
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 0:01:28, 24p /1.001
1: MPEG2, 1080p24 /1.001 (16:9)
2: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -29dB
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[v01] Creating file "00112 - 1 - MPEG2, 1080p24.m2v"...
[a02] Creating file "00112 - 2 - AC3 Surround, English, 2.0 channels, 192kbps, 48kHz.ac3"...
Video track 1 contains 2119 frames.
eac3to processing took 12 seconds.
Done.

Saved demuxed files to Documents and not the original folder of the .m2ts file.
Log file is present in documents with demuxed files (named after the .m2ts file) & the eac3to folder as well (simply named "log.txt")

Scenario two: Demuxing to a new folder
eac3to v3.24
command line: "D:\Apps\eac3to\eac3to.exe" "E:\Work\Bram Stoker's Dracula (Trailer)\MainMovie\DRACULA_TRAILER\BDMV\STREAM\00112.m2ts" "E:\Work\Bram Stoker's Dracula (Trailer)(eac3to demuxed)\.*" -demux
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 0:01:28, 24p /1.001
1: MPEG2, 1080p24 /1.001 (16:9)
2: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -29dB
[v01] Extracting video track number 1...
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Removing AC3 dialog normalization...
[v01] Creating file "E:\Work\Bram Stoker's Dracula (Trailer)(eac3to demuxed)\ - 1 - MPEG2, 1080p24.m2v"...
[v01] Creating file "00112 - 1 - MPEG2, 1080p24.m2v"...
[a02] Creating file "00112 - 2 - AC3 Surround, English, 2.0 channels, 192kbps, 48kHz.ac3"...
[a02] Creating file "E:\Work\Bram Stoker's Dracula (Trailer)(eac3to demuxed)\ - 2 - AC3 Surround, English, 2.0 channels, 192kbps, 48kHz.ac3"...
Video track 1 contains 2119 frames.
eac3to processing took 21 seconds.
Done.

Saved demuxed files to Documents & the designated folder.
Log files present in the designated folder & the eac3to folder but not the Documents folder.

Midzuki

21st March 2011, 01:53

@ Sparktank:

eac3to is a command-line application, so, by default, it outputs its results to the current working directory — which, by default, is the <opinion>stupid</opinion> user's homepath. :) The "best" :devil: policy is 1) include eac3to's directory in the PATH environment-variable, and 2) open the command-prompt directly in the target-folder (easy with Vista and Seven, requires registry-editing in XP).

Sparktank

21st March 2011, 02:06

1) include eac3to's directory in the PATH environment-variable
and 2) open the command-prompt directly in the target-folder (easy with Vista and Seven, requires registry-editing in XP).

thanks for the info.
I just extracted eac3to a working folder when i first downloaded it.
I'll have a look into these options.

Joniii

26th March 2011, 10:38

Is this something to worry about and why these errors occur exactly (main movie in this Blu-ray is splitted on several m2ts files on the disc)?

[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Creating file "d:\gl.dts"...
[v02] Creating file "d:\gl.h264"...
[a04] Audio overlaps for 9ms at playtime 0:17:14. <WARNING>
[a04] Audio overlaps for 5ms at playtime 0:18:58. <WARNING>
[a04] Audio overlaps for 8ms at playtime 0:32:51. <WARNING>
[a04] Audio overlaps for 6ms at playtime 0:54:09. <WARNING>
[a04] Audio overlaps for 5ms at playtime 0:58:03. <WARNING>
[a04] Audio overlaps for 7ms at playtime 1:11:07. <WARNING>
[a04] Audio overlaps for 5ms at playtime 1:26:02. <WARNING>
[a04] Audio overlaps for 6ms at playtime 1:44:18. <WARNING>
[a04] Audio overlaps for 12ms at playtime 1:52:11. <WARNING>
[a04] Audio overlaps for 6ms at playtime 2:04:42. <WARNING>
[a04] Audio overlaps for 7ms at playtime 2:10:04. <WARNING>
[a04] Audio overlaps for 11ms at playtime 2:15:25. <WARNING>
[a04] Audio overlaps for 8ms at playtime 2:24:38. <WARNING>
[a04] Audio overlaps for 8ms at playtime 2:25:29. <WARNING>
[a04] Audio overlaps for 8ms at playtime 2:31:13. <WARNING>
[a04] Audio overlaps for 8ms at playtime 2:34:06. <WARNING>
[a04] Audio overlaps for 7ms at playtime 2:43:40. <WARNING>
[a04] Starting 2nd pass...
[a04] Realizing DTS gaps...
[a04] Creating file "d:\gl.dts"...
Video track 2 contains 245919 frames.
eac3to processing took 13 minutes, 11 seconds.
Done.

krosswindz

26th March 2011, 19:38

I have question on the following two options of eac3to. Can one use these options to convert audio length to match the fps of the video. Basically if I have a audio from a NTSC DVD (23.976fps) then I can use the -changeTo24.000 option to convert it so that I can use it with video from a Blu-ray (24fps).

-23.976/... define source fps to be "23.976", "24.000", "25.000", ...
-changeTo24.000 change source fps to "23.976", "24.000", "25.000", ...

henryho_hk

27th March 2011, 00:01

Doesn't Blu-ray's "24" also mean 23.976fps?

tebasuna51

27th March 2011, 00:57

I have question on the following two options of eac3to. Can one use these options to convert audio length to match the fps of the video. Basically if I have a audio from a NTSC DVD (23.976fps) then I can use the -changeTo24.000 option to convert it so that I can use it with video from a Blu-ray (24fps).
Yes.
Doesn't Blu-ray's "24" also mean 23.976fps?
No.

sneaker_ger

27th March 2011, 02:52

Well, most of the time "1080p24" actually means 24/1.001 fps in the Blu-Ray world, but tebasuna51 is correct in that Blu-Ray supports both 24/1.000 and 24/1.001 fps.

krosswindz

27th March 2011, 04:09

Yes.

I used the following command line to do the conversion, I find that the runtime of the audio is still the same as the original. Am I doing something wrong or will the audio and video be in sync. The video runtime for the DVD and Blu-ray source are different.

\path\to\eac3to.exe src.ac3 dest.ac3 -448 -23.976 -changeTo24.000 +1000ms -log=log.txt

Sharc

27th March 2011, 08:59

I used the following command line to do the conversion, I find that the runtime of the audio is still the same as the original. Am I doing something wrong or will the audio and video be in sync. The video runtime for the DVD and Blu-ray source are different.

\path\to\eac3to.exe src.ac3 dest.ac3 -448 -23.976 -changeTo24.000 +1000ms -log=log.txt

Try
\path\to\eac3to.exe src.ac3 dest.ac3 -448 -24.000 +1000ms -log=log.txt

tebasuna51

27th March 2011, 12:21

I used the following command line to do the conversion:
eac3to.exe src.ac3 dest.ac3 -448 -23.976 -changeTo24.000 +1000ms
I find that the runtime of the audio is still the same as the original.
Only if the runtime is 0:16:41.000

Am I doing something wrong or will the audio and video be in sync. The video runtime for the DVD and Blu-ray source are different.
We can't know. Put exact runtime or number of frames.

Try
\path\to\eac3to.exe src.ac3 dest.ac3 -448 -24.000 +1000ms -log=log.txt
Here the parameter -24.000 does nothing.

krosswindz

27th March 2011, 17:47

Only if the runtime is 0:16:41.000

We can't know. Put exact runtime or number of frames.

dvd video (23.976fps) runtime is: 10619286 ms
blu-ray video (24fps) runtime is: 10631458 ms

When I used the converted audio while muxing if I add suitable delay it starts of in sync and loses sync progressively.

jj666

27th March 2011, 18:09

You can't normally just take a DVD track and expect it to fit with BD (slowed down or not) - Audition or similar is needed to do a waveform comparison to check for initial delay and any necessary cuts/inserts.

Cheers,

-jj-

Snowknight26

27th March 2011, 18:44

eac3to doesn't seem to handle TrueHD frame changes at the moment. A sample file I have switches from 6 channels to 2 channels within the first few hundred KB, but when it does, eac3to returns "a03 libav Restart header sync incorrect (got 0x14be)" then fails while ffmpeg decodes it properly. Any chance supporting this?

Sharc

27th March 2011, 18:46

....Here the parameter -24.000 does nothing.
Out of curiosity: Isn't the effect similiar to avisynth's AssumeFPS(xx) vs ChangeFPS(yy)?
I mean -24.000 would set the new audio playback speed and change the original duration, while -changeTo24.000 would keep the original duration?

Undead Sega

27th March 2011, 19:27

Thunderbolt8

27th March 2011, 22:56

Sharc

28th March 2011, 00:25

Hi guys, sorry if this may have been brought up before, but I was wondering if it is possible to do either slowdown or speedup whilst preserving the pitch?

For example: '23.796 to 25' pitch preserving/correction? and vice-versa with different framerates?
I am not sure how it goes with eac3to. However, it's certainly possible via avisynth (timestretch) and more easily via BeSweet/BeLight (GUI) for example.

tebasuna51

28th March 2011, 03:03

dvd video (23.976fps) runtime is: 10619286 ms
blu-ray video (24fps) runtime is: 10631458 ms
Then
dvd video have 10619286 x 23.976 = 254608 frames
bd video have 10631458 x 24 = 255155 frames

The bd video have 547 more frames (~23 seconds) than dvd video. Isn't the same video and you need a big edit to use the audio.

krosswindz

28th March 2011, 03:10

Then
dvd video have 10619286 x 23.976 = 254608 frames
bd video have 10631458 x 24 = 255155 frames

The bd video have 547 more frames (~23 seconds) than dvd video. Isn't the same video and you need a big edit to use the audio.

I noticed the difference in runtime, what is weird is the main movie is the same for both Blu-ray and DVD. There is difference in the title and credits of the movie in terms of additional information. I thought adding suitable delay to sync the audio should work. For some reason it just goes out of sync progressively. When I compare the runtimes of the converted audio from the original audio the original audio is ~1.001 * converted audio runtime. I guess I need to see what else is different to fix it.

tebasuna51

28th March 2011, 12:47

Out of curiosity: Isn't the effect similiar to avisynth's AssumeFPS(xx) vs ChangeFPS(yy)?
I mean -24.000 would set the new audio playback speed and change the original duration, while -changeTo24.000 would keep the original duration?
Audio don't have reference to video fps, then a pair of fps like -24.000 -changeTo23.976 only means enlarge the audio duration by (24/(24/1.001)) = 1.001
Some defaults can work, for instance:
-slowdown is equivalent to -25.000 -changeTo23.976
-speedup is equivalent to -23.976 -changeTo25.000
But I like always put the fps pair and don't assume any fps for audio.

The fps equivalent parameter for audio is samplerate:
48 KHz means 48000 audio samples per second, like 24 fps means 24 video frames per second.

Hi guys, sorry if this may have been brought up before, but I was wondering if it is possible to do either slowdown or speedup whilst preserving the pitch?
Nope with soft included in eac3to

... it's certainly possible via avisynth (timestretch) and more easily via BeSweet/BeLight (GUI) for example.
Yes.
(but with BeSweet try only with stereo files)

tebasuna51

28th March 2011, 12:55

...
There is difference in the title and credits of the movie in terms of additional information...

Then try only with a initial delay, with the original DVD audio or with the stretched (1/1.001) DVD audio.
No problem with end credits the audio can be short than video.

Sharc

28th March 2011, 19:37

Audio don't have reference to video fps, then a pair of fps like -24.000 -changeTo23.976 only means enlarge the audio duration by (24/(24/1.001)) = 1.001
Thanks for clarification.
Does such pair of fps preserve or change the pitch?

Chouonsoku

28th March 2011, 21:03

I'm having an issue with a DTS-HD MA 6.1 audio track from Ghost in the Shell 2: Innocence's Absolute Edition. The disc has a TrueHD 6.1 and a DTS-HD MA 6.1 track, and I can compress the TrueHD to FLAC, extract the DD5.1-EX core and also extract the DTS-ES 6.1 core, but when I compress the DTS-MA HD track to FLAC, it ends up being around 2.5 GB (vs. the DTS-MA HD's 1.8 GB) and the output file is also stretched somewhat. The original runtime of the film is 1hr 39m, however the FLAC of the DTS track reports it's 1hr 53m long, and during playback the audio is there, but it's distorted and stretched. No warnings or errors are reported by eac3to, and again all other tracks playback fine.

eac3to v3.24
command line: eac3to 1) 1:chapters.txt 2:innocence.h264 3: japanese.ac3 -core 3:japanese-truehd.flac 4:japanese.dts -core 4:japanese-dts.flac 7:english.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 3 subtitle tracks, 1:39:06, 24p /1.001
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, Japanese, 6.1 channels, 48kHz
(embedded: AC3 EX, 5.1 channels, 640kbps, 48kHz)
4: DTS Master Audio, Japanese, 6.1 channels, 16 bits, 48kHz
(core: DTS-ES, 6.1 channels, 16 bits, 1509kbps, 48kHz)
5: Subtitle (PGS), Japanese
6: Subtitle (PGS), Japanese
7: Subtitle (PGS), English
Creating file "chapters.txt"...
[a04] Extracting audio track number 4...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a03] Extracting audio track number 3...
[a04] Extracting DTS core...
[s07] Extracting subtitle track number 7...
[a04] Remapping channels...
[a03] Extracting TrueHD stream...
[a03] Extracting audio track number 3...
[a04] Patching bitdepth to 24 bits...
[a03] Decoding with libav/ffmpeg...
[a03] Extracting AC3 stream...
[a04] Decoding with ArcSoft DTS Decoder...
[a03] Remapping channels...
[a04] Encoding FLAC with libFlac...
[a03] Encoding FLAC with libFlac...
[a04] Creating file "japanese.dts"...
[v02] Creating file "innocence.h264"...
[a04] Creating file "japanese-dts.flac"...
[a03] Creating file "japanese-truehd.flac"...
[a03] Creating file "japanese.ac3"...
[s07] Creating file "english.sup"...
[a03] The original audio track has a constant bit depth of 16 bits.
[a04] The original audio track has a constant bit depth of 16 bits.
[a03] Superfluous zero bytes detected, will be stripped in 2nd pass.
[a03] Starting 2nd pass...
[a03] Decoding FLAC...
[a03] Reducing depth from 24 to 16 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "japanese-truehd.flac"...
[a03] The processed audio track has a constant bit depth of 16 bits.
Video track 2 contains 142560 frames.
Subtitle track 7 contains 814 captions.
eac3to processing took 34 minutes, 53 seconds.
Done.

Snowknight26

28th March 2011, 23:31

Most likely an issue with the ArcSoft DTS decoder. What version is it?

sneaker_ger

29th March 2011, 06:46

Thanks for clarification.
Does such pair of fps preserve or change the pitch?

Change.

TDiTP_

3rd April 2011, 04:36

Stupid question. Why eac3to need MkvToolNix?

Mark_A_W

3rd April 2011, 05:17

Hi all

Quick question..

What is the command line to upconvert a 6.1 TrueHD track to 7.1 FLAC (with centre rear doubled to L back and R back)?

There's a trilogy coming out soon (finally..) with 6.1 sound ;)

TDiTP_

3rd April 2011, 06:26

What is the command line to upconvert a 6.1 TrueHD track to 7.1 FLAC (with centre rear doubled to L back and R back)?

-double7

But this command does work incorrect, you need to add -0,1,2,3,5,6,4:
eac3to input output -double7 -0,1,2,3,5,6,4

Chouonsoku

8th April 2011, 00:00

Most likely an issue with the ArcSoft DTS decoder. What version is it?

eac3to (v3.24) is up to date
Nero Audio Decoder (Nero 7) works fine
ArcSoft DTS Decoder (1.1.0.8) works fine
Sonic Audio Decoder (4.3.0.169) works fine
Haali Matroska Muxer (2011-03-03) is up to date
Nero AAC Encoder (1.5.4.0) is up to date
Surcode DTS Encoder doesn't seem to be installed
http://www.surcode.com
MkvToolnix (4.6.0.0, release version) is up to date

Is there a newer version?

Edit: Something I just noticed though, eac3to reports the track as being 16-bit however the bits aren't stripped in the second pass like they are for the TrueHD track. I'm testing decoding with Sonic Audio Decoder now to see if that works, although I'd rather not use it.

Edit 2: Sonic decoded it fine, same exact bitrate as the TrueHD (figured). Plays back fine and has the correct length. Still don't know why ArcSoft is having a problem with it.

TDiTP_

8th April 2011, 08:10

For decode track 4:

4: DTS Master Audio, Japanese, 6.1 channels, 16 bits, 48kHz
(core: DTS-ES, 6.1 channels, 16 bits, 1509kbps, 48kHz)

you need ArcSoft DTS Decoder 1.1.0.0 (you can find it here (http://www.mediafire.com/?4vsw2dydnz1evme)). Versions 1.1.0.7 and 1.1.0.8 can't decode DTS-ES 6.1 properly, it's known bug.

Sonic decoded it fine, same exact bitrate as the TrueHD (figured).

Sonic can't decode DTS-ES 6.1 properly too. BC-channel is empty always.

Mark_A_W

8th April 2011, 10:49

What about 1.1.0.5? Can that do 6.1?

TDiTP_

8th April 2011, 10:56

What about 1.1.0.5? Can that do 6.1?

i don't know. If you give me this version, i can test it.

Thunderbolt8

8th April 2011, 13:54

I dont remember any more, was there something 1.1.0.7 or .8 could do 1.1.0.0 couldnt do? either were one or two things (dtsma 1.0 or 2.0?), or 1.1.0.0 was just as good or better in each situation.

btw. madshi, would it make sense to integrate remuxing with LAVsplitter instead of haali in the future? are there any improvements (or even bugs) or can there be any improvement at all in this area from a technical point of view?

Chouonsoku

8th April 2011, 15:57

Alright, testing with 1.1.0.0 now. Now that you mention it I remember that being an issue but I didn't know which versions it affected.

Edit: It worked. Thanks. :)

DivxBr

14th April 2011, 14:59

Hi,

Don't get me wrong, I really love eac3to, and how it handles all audio formats, convert every audio stream to almost every audio format available. It's more than half job done for BD backup.

But there's one 5min+ stream (indexed by an mpls - playlist) that eac3to just can't handle right, and is getting me crazy to understand what I could be making wrong, looks like eac3to is messing up with timestamps and destroying A/V Sync.

That's what I'm doing:

eac3to 00202.mpls

1) 00202.mpls, 0:05:40
[44+45+46+47+48+49+50+51].m2ts
- Chapters, 8 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz

I First extract the video and the audio tracks.

eac3to v3.24
command line: eac3to 00202.mpls 1) 2: "w:\test\ExtractedVideo.mkv" 3: "w:\test\ExtractedAudio.ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 10 subtitle tracks, 0:05:39, 7.702p
1: Chapters, 8 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
4: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
5: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
7: Subtitle (PGS), English
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), Dutch
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), Portuguese
13: Subtitle (PGS), Danish
14: Subtitle (PGS), Finnish
15: Subtitle (PGS), Norwegian
16: Subtitle (PGS), Swedish
[a03] Extracting audio track number 3...
[a03] Removing AC3 dialog normalization...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[a03] Creating file "w:\test\ExtractedAudio.ac3"...
[v02] Video overlaps for 7 frames at playtime 0:03:00. <WARNING>
[v02] Video overlaps for 7 frames at playtime 0:03:31. <WARNING>
[v02] Video overlaps for 7 frames at playtime 0:04:00. <WARNING>
[v02] The MKV file was created without making use of the gap/overlap information. <WARNING>
[v02] Please check whether audio is in sync. If it is in sync everything is fine. <WARNING>
[v02] Otherwise ask eac3to to repeat the muxing. It will then automatically make <WARNING>
[v02] use of the detailed gap/overlap information. <WARNING>
[a03] Audio overlaps for 120ms at playtime 0:00:40. <WARNING>
[a03] Audio overlaps for 95ms at playtime 0:01:37. <WARNING>
[a03] Audio overlaps for 52ms at playtime 0:01:55. <WARNING>
[a03] Audio overlaps for 96ms at playtime 0:02:27. <WARNING>
[a03] Audio overlaps for 119ms at playtime 0:03:00. <WARNING>
[a03] Audio overlaps for 105ms at playtime 0:03:31. <WARNING>
[a03] Audio overlaps for 91ms at playtime 0:04:00. <WARNING>
[a03] Starting 2nd pass...
[a03] Realizing (E-)AC3 gaps...
[a03] Creating file "w:\test\ExtractedAudio.ac3"...
Added fps value (24 /1.001) to MKV header.
Video track 2 contains 8144 frames.
eac3to processing took 15 seconds.
Done.

So it detected Audio AND Video overlaps. It corrected the audio, and gave me the possibility to correct the video overlap also.

So I do it:

eac3to v3.24
command line: eac3to 00202.mpls 1) 2: "w:\test\ExtractedVideo_gaps.mkv"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 10 subtitle tracks, 0:05:39, 7.702p
1: Chapters, 8 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
4: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
5: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
7: Subtitle (PGS), English
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), Dutch
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), Portuguese
13: Subtitle (PGS), Danish
14: Subtitle (PGS), Finnish
15: Subtitle (PGS), Norwegian
16: Subtitle (PGS), Swedish
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[v02] Video overlaps for 7 frames at playtime 0:03:00. <WARNING>
[v02] Video overlaps for 7 frames at playtime 0:03:31. <WARNING>
[v02] Video overlaps for 7 frames at playtime 0:04:00. <WARNING>
[v02] The MKV file was created without making use of the gap/overlap information. <WARNING>
[v02] Please check whether audio is in sync. If it is in sync everything is fine. <WARNING>
[v02] Otherwise ask eac3to to repeat the muxing. It will then automatically make <WARNING>
[v02] use of the detailed gap/overlap information. <WARNING>
Added fps value (24 /1.001) to MKV header.
Video track 2 contains 8144 frames.
eac3to processing took 12 seconds.
Done.

eac3to v3.24
command line: eac3to 00202.mpls 1) 2: "w:\test\ExtractedVideo_gaps.mkv"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 10 subtitle tracks, 0:05:39, 7.702p
1: Chapters, 8 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
4: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
5: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
7: Subtitle (PGS), English
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), Dutch
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), Portuguese
13: Subtitle (PGS), Danish
14: Subtitle (PGS), Finnish
15: Subtitle (PGS), Norwegian
16: Subtitle (PGS), Swedish
[v02] Video gap description file detected, will be used for muxing...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
Added fps value (24 /1.001) to MKV header.
Video track 2 contains 8144 frames.
eac3to processing took 12 seconds.
Done.

Now, I have 'ExtractedAudio.ac3' (automatically overlap-corrected by eac3to), 'ExtractedVideo.mkv' without overlap-correction and 'ExtractedVideo_gaps.mkv' with overlap-correction.

When I use MKVMERGE to MUX ExtractedVideo.mkv (no overlap corrected) & ExtractedAudio.ac3, the A/V sync at the start of the clip is fine, but at the end of it, the Audio is 250ms earlier.

When I use MKVMERGE to MUX ExtractedVideo_gaps.mkv (overlap corrected) & ExtractedAudio.AC3, the A/V sync at the star of the clip still fine, but at the end of it, the Audio is approx. 650mS late.

Any ideas?

Fernando

laserfan

14th April 2011, 16:29

I First extract the video and the audio tracks.

eac3to v3.24
command line: eac3to 00202.mpls 1) 2: "w:\test\ExtractedVideo.mkv" 3: "w:\test\ExtractedAudio.ac3"

So it detected Audio AND Video overlaps. It corrected the audio, and gave me the possibility to correct the video overlap also.

So I do it:

eac3to v3.24
command line: eac3to 00202.mpls 1) 2: "w:\test\ExtractedVideo_gaps.mkv"

Any ideas?
Did you try repeating (for the 2nd pass) exactly the same command line as initially?

DivxBr

14th April 2011, 16:45

Did you try repeating (for the 2nd pass) exactly the same command line as initially?

Yes, just done it. Result = Audio with a (approx.) +650mS delay at the end of the clip.

TDiTP_

15th April 2011, 00:11

eac3to 00202.mpls

1) 00202.mpls, 0:05:40
[44+45+46+47+48+49+50+51].m2ts
- Chapters, 8 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz

This is a short fragment. Maybe you can upload it to any fileshare?

DivxBr

15th April 2011, 01:01

This is a short fragment. Maybe you can upload it to any fileshare?

My connection isn't rock stable enough to upload almost 500MB, but I could try. You have any suggestion which upload-site accepts uploads so big?

TDiTP_

15th April 2011, 10:24

You have any suggestion which upload-site accepts uploads so big?

AFAIK mediafire.com

DivxBr

15th April 2011, 20:18

I had to separate the files in a multipart rar archieve to obey maximum size limit of the site, so there's the links:

http://www.mediafire.com/?j76bubvfnvnrbkz
http://www.mediafire.com/?n9jw82nek0xlauu
http://www.mediafire.com/?to0zlc8zau5bqu8
http://www.mediafire.com/?0hd1d7up9plb416
http://www.mediafire.com/?v0ixbf7se3zuvm8

TDiTP_

16th April 2011, 23:39

Thanks for sample :)
It's strange but i can't confirm your problem.

My two logs:

eac3to v3.24
command line: eac3to Segment\BDMV\PLAYLIST\00202.mpls 2: video.mkv 3: audio.ac3
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 10 subtitle tracks, 0:05:39, 7.702p
1: Chapters, 8 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
4: AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
5: AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
6: AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
7: Subtitle (PGS)
8: Subtitle (PGS)
9: Subtitle (PGS)
10: Subtitle (PGS)
11: Subtitle (PGS)
12: Subtitle (PGS)
13: Subtitle (PGS)
14: Subtitle (PGS)
15: Subtitle (PGS)
16: Subtitle (PGS)
[a03] Extracting audio track number 3...
[a03] Removing AC3 dialog normalization...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[a03] Creating file "audio.ac3"...
[v02] Video overlaps for 7 frames at playtime 0:03:00. <WARNING>
[v02] Video overlaps for 7 frames at playtime 0:03:31. <WARNING>
[v02] Video overlaps for 7 frames at playtime 0:04:00. <WARNING>
[v02] The MKV file was created without making use of the gap/overlap information. <WARNING>
[v02] Please check whether audio is in sync. If it is in sync everything is fine. <WARNING>
[v02] Otherwise ask eac3to to repeat the muxing. It will then automatically make <WARNING>
[v02] use of the detailed gap/overlap information. <WARNING>
[a03] Audio overlaps for 120ms at playtime 0:00:40. <WARNING>
[a03] Audio overlaps for 95ms at playtime 0:01:37. <WARNING>
[a03] Audio overlaps for 52ms at playtime 0:01:55. <WARNING>
[a03] Audio overlaps for 96ms at playtime 0:02:27. <WARNING>
[a03] Audio overlaps for 119ms at playtime 0:03:00. <WARNING>
[a03] Audio overlaps for 105ms at playtime 0:03:31. <WARNING>
[a03] Audio overlaps for 91ms at playtime 0:04:00. <WARNING>
[a03] Starting 2nd pass...
[a03] Realizing (E-)AC3 gaps...
[a03] Creating file "audio.ac3"...
Added fps value (24 /1.001) to MKV header.
Video track 2 contains 8144 frames.
eac3to processing took 9 seconds.
Done.

eac3to v3.24
command line: eac3to Segment\BDMV\PLAYLIST\00202.mpls 2: video.mkv
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 10 subtitle tracks, 0:05:39, 7.702p
1: Chapters, 8 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
4: AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
5: AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
6: AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
7: Subtitle (PGS)
8: Subtitle (PGS)
9: Subtitle (PGS)
10: Subtitle (PGS)
11: Subtitle (PGS)
12: Subtitle (PGS)
13: Subtitle (PGS)
14: Subtitle (PGS)
15: Subtitle (PGS)
16: Subtitle (PGS)
[v02] Video gap description file detected, will be used for muxing...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
Added fps value (24 /1.001) to MKV header.
Video track 2 contains 8144 frames.
eac3to processing took 7 seconds.
Done.

I used MMG 4.6.0 to mux video and audio in MKV. Everything is
- ok in case of video with corrected overlaps
- not ok in case of video without corrected overlaps.
I used TsMuxeR 1.10.6 to mux video and audio to BDAV, all the same.

PS. I can hear the difference of a ~600ms.

PPS. With this command line

eac3to v3.24
command line: eac3to Segment\BDMV\PLAYLIST\00202.mpls 2: video2.vc1 3: audio2.ac3
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 10 subtitle tracks, 0:05:39, 7.702p
1: Chapters, 8 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
4: AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
5: AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
6: AC3 Surround, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
7: Subtitle (PGS)
8: Subtitle (PGS)
9: Subtitle (PGS)
10: Subtitle (PGS)
11: Subtitle (PGS)
12: Subtitle (PGS)
13: Subtitle (PGS)
14: Subtitle (PGS)
15: Subtitle (PGS)
16: Subtitle (PGS)
[a03] Extracting audio track number 3...
[a03] Removing AC3 dialog normalization...
[v02] Extracting video track number 2...
[v02] Creating file "video2.vc1"...
[a03] Creating file "audio2.ac3"...
[v02] Video overlaps for 7 frames at playtime 0:03:00. <WARNING>
[v02] Video overlaps for 7 frames at playtime 0:03:31. <WARNING>
[v02] Video overlaps for 7 frames at playtime 0:04:00. <WARNING>
[a03] Audio overlaps for 120ms at playtime 0:00:40. <WARNING>
[a03] Audio overlaps for 95ms at playtime 0:01:37. <WARNING>
[a03] Audio overlaps for 52ms at playtime 0:01:55. <WARNING>
[a03] Audio overlaps for 96ms at playtime 0:02:27. <WARNING>
[a03] Audio overlaps for 119ms at playtime 0:03:00. <WARNING>
[a03] Audio overlaps for 105ms at playtime 0:03:31. <WARNING>
[a03] Audio overlaps for 91ms at playtime 0:04:00. <WARNING>
[a03] Starting 2nd pass...
[a03] Realizing (E-)AC3 gaps...
[a03] Creating file "audio2.ac3"...
Video track 2 contains 8144 frames.
eac3to processing took 11 seconds.
Done.

and then mux in MMG or TsMuxeR everything is ok with synchronization too.

DivxBr

17th April 2011, 05:17

I used MMG 4.6.0 to mux video and audio in MKV. Everything is
- ok in case of video with corrected overlaps
- not ok in case of video without corrected overlaps.
I used TsMuxeR 1.10.6 to mux video and audio to BDAV, all the same.

PS. I can hear the difference of a ~600ms.

Not sure if I understood....

I'm getting audio BEFORE video (250mS) on non-corrected overlaps, and getting audio AFTER video (600mS) on overlap-corrected MUX. Is it what you got?

I'd made another test using DGIndexNV demuxing Video&Audio, and the Result = No AV sync glitch.

I'll double check it later, but looks like the demuxed video is identical (to the not corrected for overlap), and demuxed audio is 8 AC3 frames bigger (+256mS) - which could explain the delay I found.

Making some calculations, if I remove 21 video frames (7 x 3 frames shown overlaped by eac3to), I get +256-(21/23.976) = -619mS (!!!) close enough to delay I get when I let the eac3to correct the video stamps.

I'll test the demux/remux VC-1 stream directly like you showed. Thanks!

I think I'll get some interesting info analysing timestamps file generated by dgindexnv too (at least near the 1st point eac3to finds the 7 overlaped video frames, the timecodes make a big jump, and at this point maybe eac3to is losing track of things).

Carpo

17th April 2011, 16:02

Would people suggest using this over TsMuxer, in the mood to do a bit of experimenting :-)

hubblec4

25th April 2011, 11:19

hi

i would install TMT5. is eac3to able to use the codecs from this version of TMT?

PowerGamer

25th April 2011, 13:14

I extracted PGS subtitles from BluRay into a .sup file using eac3to. I muxed that file into .mkv file using mkvmerge. When I try to extract subtitles from created .mkv file using eac3to I get a new .sup file much shorter than the original .sup file. SupRip fails to open that new .sup file. MPC-HC fails to display subtitles after muxing shorter .sup file into new .mkv file. So it seems that .sup file demuxed by eac3to from .mkv is invalid. I wonder if the author of eac3to can look into this issue?

P.S. Also there is another (very small) issue with eac3to. When listing tracks from .mkv eac3to shows no language for tracks that should be labelled as "English" (see http://www.bunkus.org/videotools/mkvtoolnix/faq.html#default_values).

Weirdo

25th April 2011, 16:26

Hi there madshi, is it possible for eac3to to not require Haali Media Splitter, but work with LAV instead?

(edit: Just saw Thunderbolt8 asked the same thing above...)

Vasch the stampede

26th April 2011, 18:45

eac3to v3.24
command line: eac3to "...\4.thd" "...\4.flac"
------------------------------------------------------------------------------
TrueHD, 5.1 channels, 48kHz
Decoding with libav/ffmpeg...
Encoding FLAC with libFlac...
Creating file "...\4.flac"...
[libav] Lossless check failed - expected 5d, calculated 63 <WARNING>
[libav] Lossless check failed - expected d, calculated 3 <WARNING>
[libav] Lossless check failed - expected fe, calculated b1 <WARNING>
[libav] Lossless check failed - expected 8b, calculated 5f <WARNING>
The original audio track has a constant bit depth of 16 bits.
Superfluous zero bytes detected, will be stripped in 2nd pass.
Starting 2nd pass...
Decoding FLAC...
Reducing depth from 24 to 16 bits...
Encoding FLAC with libFlac...
Creating file "...1\4.flac"...
The processed audio track has a constant bit depth of 16 bits.
Done. Should I be worried?

Kurtnoise

26th April 2011, 19:54

Hi there madshi, is it possible for eac3to to not require Haali Media Splitter, but work with LAV instead?

(edit: Just saw Thunderbolt8 asked the same thing above...)
eac3to uses haali's matroska muxer...LAV filters are just splitter/decoders.

An alternative would be to use the solveig mkv muxer or ffmpeg's one.

Kurtnoise

26th April 2011, 19:54

deathlord

28th April 2011, 09:50

Hi

It has been discussed, here and in other places, how to play back 5.1 and 7.1 sources in a common setup. There does not seem to be a satisfactory solution. I have opend a new thread where I would like to discuss this topic in some detail:

http://forum.doom9.org/showthread.php?t=160857

nibus

10th May 2011, 18:16

Is there a way to preserve artist info (ID3 tag) when converting from FLAC to AAC with eac3to?

Here is my command line:
eac3to.exe "file.flac" "file.m4a" -normalize -quality=0.45

I've included NeroAacTag.exe in the same folder but the output files are not tagged.

kypec

10th May 2011, 18:56

I don't think eac3to cares about any meta data in general. I'd suggest to try foobar2000 (http://www.foobar2000.org/) or latest LameXP (http://forum.doom9.org/showthread.php?p=1499797#post1499797) for such simple conversions.

nibus

11th May 2011, 02:18

mini-moose

12th May 2011, 13:27

probably something that's been discussed a lot before but nothing seems to work for me.

Trying to convert an aac file to ac3 with eac3to and I always get "eac3to Getting "Nero Audio Decoder 2" instance failed"

I don't have nero7 installed and I'm not sure I want to install it either or whether it even works on win7 x64. I do have the NeroAAC decoder/encoder pack but am unsure how to make eac3to use it...

hope someone can help.

TinTime

13th May 2011, 12:04

eac3to will not decode aac without Nero's DirectShow decoder (it does work on Win7 64).

mini-moose

13th May 2011, 12:33

eac3to will not decode aac without Nero's DirectShow decoder (it does work on Win7 64).

so I must install nero7 to convert aac with eac3to or can I just get the directshow dll somehow? I'm not a big fan of Nero though I must say I dont think I tried 7 but most of the ones after 6 seemed like bloated crap

Boulder

13th May 2011, 12:42

You can use Nero Lite if you like.

mini-moose

13th May 2011, 13:04

You can use Nero Lite if you like.

installed Nero Micro 7. Seems to work now. thanks.

Richard1485

15th May 2011, 00:32

3: DTS Master Audio, Italian, 1.0 channels, 16 bits, 48kHz
(core: DTS, 1.0 channels, 16 bits, 384kbps, 48kHz)
[a03] Extracting audio track number 3...
[a03] Decoding with DirectShow (Sonic Audio Decoder)...
[a03] The WAV writer didn't receive the format information. <ERROR>
Aborted at file position 19418609664. <ERROR>

Any ideas what is happening here? This happens when I try to demux the mono to wav (or anything else for that matter).

Moondust

19th May 2011, 09:20

Lately I've come across some DTS-HDMA 6.1 tracks which I want to convert to FLAC. The FLAC file however doesn't play right. The length is wrong (2:14:58 instead of 1:57:50) and the sound is distorted in the way that the pitch seems to be way off. Lots of high frequencies and sharpness. Don't know how to describe it properly, but you can hear immediately that it's all wrong. Is this a known bug?

I tried to downmix it to 5.1, but the result is the same - shitty audio.

C:\Program Files (x86)\EAC3TO>eac3to d:\sources\infernalaffairsIII 1) 3:d:\audio
\InfernalAffairsIII.flac -down6
M2TS, 1 video track, 3 audio tracks, 3 subtitle tracks, 1:57:50, 24p
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 (16:9)
3: DTS Master Audio, Chinese, 6.1 channels, 16 bits, 48kHz
(core: DTS-ES, 6.1 channels, 16 bits, 1509kbps, 48kHz)
4: AC3 EX, Chinese, 5.1 channels, 640kbps, 48kHz
5: AC3 EX, Chinese, 5.1 channels, 640kbps, 48kHz
6: Subtitle (PGS), Chinese
7: Subtitle (PGS), Chinese
8: Subtitle (PGS), English
v02 The video framerate is correct, but rather unusual.
a03 Extracting audio track number 3...
a03 Remapping channels...
a03 Decoding with ArcSoft DTS Decoder...
a03 Mixing surround channels...
a03 Encoding FLAC with libFlac...
a03 Creating file "d:\audio\InfernalAffairsIII.flac"...
a03 The original audio track has a constant bit depth of 16 bits.
a03 The processed audio track has a constant bit depth of 16 bits.
Video track 2 contains 169690 frames.
eac3to processing took 12 minutes, 26 seconds.
Done.

Is there a solution?

nautilus7

19th May 2011, 10:33

Use Sonic decoder instead of the Arcsoft one. It's known to cause problems with 6.1 DTS(-HD) tracks.

TDiTP_

19th May 2011, 10:55

Sonic can't decode DTS(-HD) 6.1 properly (blank BACK_CENTER always). You need ArcSoft v. 1.1.0.0.

Moondust

19th May 2011, 12:41

Sonic can't decode DTS(-HD) 6.1 properly (blank BACK_CENTER always). You need ArcSoft v. 1.1.0.0.
I just converted the DTS-HDMA 6.1 track from Infernal Affairs 3 to FLAC 6.1 using Sonic Audio Decoder version 4.3.0.169 and I checked all 6 channels with Audacity. None of them is empty. So it seems that using this version of Sonic Audio Decoder solves the problem I had with Arcsoft decoder version 1.1.0.7. Thanks for the helping hand nautilus7 and TDiTP_.

TDiTP_

19th May 2011, 18:51

Ouu, excuse me. Sonic can't decode properly DTS-HD 6.0, not 6.1.

Thunderbolt8

21st May 2011, 14:04

Ive lost track again, was there actually one thing which 1.1.0.7 or 1.1.0.8 could do 1.1.0.0 couldnt do, or was there no reason to switch to 1.1.0.7/8 at all, when having 1.1.0.0 working?

nevcairiel

21st May 2011, 16:39

1.1.0.7/1.1.0.8 cannot decode 6.0 properly (results in 5.0), and 1.1.0.7 cannot decode Mono.

1.1.0.0 can decode everything, as far as i am aware.

Thunderbolt8

21st May 2011, 19:17

alright, no sense to talk about .7 and .8 any further :p

Greif

27th May 2011, 19:21

Hey madshi, is there any appetite to adding the ability to mux streams together via either Matroska or MP4 (via mp4box)?

forumator

30th May 2011, 02:15

alright, no sense to talk about .7 and .8 any further :p

I wish that were true, I'm a newbie who just wants to rip all my BD audio to .flac, but there's conflicting information in this thread about the limitations of different Arcsoft versions. On page 545 TDiTP_ says ArcSoft 1.1.0.0 can decode only "normal setup" (-logdts: $84b). ArcSoft 1.1.0.7 and 1.1.0.8 decode all configuration But then on the next page we get some disagreement, xkodi says DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 96kHz" and in fact 1.1.0.8 output is the same as 1.1.0.0 Another response from TDiTP_, WAV of Arcsoft 1.1.0.8, Sonic and Nero are identical byte-in-byte (i'm using HxD) to each other and not idential to Arcsoft 1.1.0.0. to which xkodi replies with his own analysis, in very short there is no version of Arcsoft decoder that can handle DTS-MA 7.1 "strange setup" files in bit-perfect way, i agree that there is small improvement regarding "L" and "R" channel with version 1.1.0.8, but still the decoded information is not correct in terms of bit-perfect decode. the only way to get bit-perfect decode from DTS-MA 7.1 "strange setup" files is to use Sonic decoder and convert it to correct 5.1 WAV file as i tried to explain here:

http://forum.doom9.org/showthread.ph...85#post1443085

Of course on the next page TDiTP_ comes back with another refutation in this post: http://forum.doom9.org/showthread.php?p=1472896#post1472896

Needless to say I'm confused about which version of Arcsoft I should be using, because if there's a problem with 1.1.0.0 I'd like to know about it before using it for all my encodes :/

Snowknight26

5th June 2011, 03:20

WAV, 2.0 channels, 0:08:35, 32 bits <float>, 2822kbps, 44.1kHz
Reading WAV...
Reducing depth from 64 to 24 bits...

I'm guessing that's merely a cosmetic issue.

sub24ox7

5th June 2011, 07:18

I have noticed that same 64bits to 24bits as well and figured it was a typo as well

tebasuna51

5th June 2011, 09:25

If you order some operation like resample or slowdown isn't a typo, put your command line.

nibus

6th June 2011, 22:22

Is there a way to add DRC when encoding DTS 5.1 to AAC 5.1 with eac3to? I am encoding my movies for iPod and would like to compress the audio more for better hearing in loud places. I know about -keepdialnorm but I would also like to add medium DRC. I remember besweet could do it, but I don't think besweet works with neroaacenc.

asarian

11th June 2011, 06:08

When I'm trying to extract the mono LPCM track of Solaris Blu-Ray, to LPCM also, the resultant file is speeding up incredibly (presumably because eac3to assumes a minimal of LPCM 2.0 or something?). Dunno what that is, but would be nice if it gets fixed some time.

SUP title traction is still less than stellar, to be honest, with time-offsets often being way out of sync (like everything starts +1 hour later or some such); tsRemuxeR does a better job of it. Would be nice if that ever got fixed as well.

Gleb Egorych

17th June 2011, 15:30

Tried to decode DTS to 16 bit wav using -16 option, but the output was 24 bit. Is it proper behaviour?

Weirdo

17th June 2011, 15:44

Tried to decode DTS to 16 bit wav using -16 option, but the output was 24 bit. Is it proper behaviour?If the original is 24-bit, you need to use -down16 I think.

Gleb Egorych

17th June 2011, 16:04

I understood my mistake: -16 is input option not output. Thanks, Weirdo.

tebasuna51

17th June 2011, 19:33

By default DTS (like all lossy encoder) is always decoded to 24 bits, no matter the bitdepth of original source what is irrelevant.

zeropc

18th June 2011, 18:12

if have a weird problem with eac3to 3.24 and more gui

i upgraded to windows 7 and when i start eac3to with the more gui i get a very weird behavior. i load a blu-ray and everything is fine. but if i select a title from a blu-ray i get stuck in old encode. i download a new copy of both (eac3to and more gui) but the error sticks with it. this also happens when i select 'test external filters'

thiugh, it's not happening when i run eac3to in cmd mode which i don't like... i'm a gui person ;)

is there any way to reset this behavior?

psymed

23rd June 2011, 02:54

Are you supposed to run the eac3to.exe? or do you use Start --> Run command prompt and then type eac3to input. ??

is that eac3to.exe supposed to close itself after double clicking it?

Thunderbolt8

24th June 2011, 01:15

if you dont run command prompt, then yes :p

psymed

24th June 2011, 10:22

if you dont run command prompt, then yes :p

How would I go about starting eac3to? It shows a log when I drag a file to it and shows the files in that 00000 named file, but then it auto closes eac3to?
Is there a way to keep it open?
Or what do I do after that?

kypec

27th June 2011, 08:58

When you are not skilled enough to type full command line yourself then try some GUI for eac3to - there are at least 4 of them, I personally prefer UsEac3to but it's up to you to pick your favorite. See the initial post of this thread for other GUI candidates.

nibus

1st July 2011, 09:49

is there a switch to tell eac3to to not create a log file after a conversion?

TinTime

1st July 2011, 10:29

is there a switch to tell eac3to to not create a log file after a conversion?

Add...

-log=NUL

...to your command line.

nibus

1st July 2011, 13:05

Add...

-log=NUL

...to your command line.

thanks :)

nibus

5th July 2011, 23:29

one more question -

is there a way to pass extra parameters to NeroAACEnc? I need to specify -he in the command line, but eac3to won't recognize it.

tebasuna51

11th July 2011, 13:21

Use -quality=0.30 or less, then the output is -he

nibus

12th July 2011, 09:21

The quality I need is low enough (0.15) that it is defaulting to HEv2 which won't work on my device.

nurbs

12th July 2011, 10:17

You can pipe stuff from eac3to to the Nero encoder.
eac3to input.data stdout.wav | neroaacenc -q 0.15 -he -ignorelength -if - -of output.mp4
Fill in paths were needed for eac3to, input, neroaacenc and output. The important thing is to have eac3to output to stdout and nero input from stdin (-if -). You also need -ignorelength IIRC.

Boulder

12th July 2011, 13:05

Do you happen to know whether qtaacenc supports piping?

nurbs

12th July 2011, 14:26

Works like Nero. You need a "-" (without the quotation marks) where you normally put the input file and most likely also "-ignorelength".

leeperry

14th July 2011, 20:51

simple question please: I get a checksum error on the BD of "The Nightmare Before Xmas":
M2TS, 1 video track, 6 audio tracks, 17 subtitle tracks, 1:16:38, 24p /1.001
1: VC-1, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, 7.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
<..>
[a02] Encoding FLAC with libFlac...
[a02] Creating file "e:\xmas\audio.flac"...
[a02] [libav] Substream 1 parity check failed <WARNING>
[a02] [libav] Substream 1 checksum failed <WARNING>
[a02] [libav] Lossless check failed - expected c0, calculated 6 <WARNING>
Anything I can do :confused:

starmanj

14th July 2011, 21:08

Is there a tool to automate MLP creation using Surcode with eac3to? Like the old Wavewizard did for DTS files?

tebasuna51

14th July 2011, 22:02

simple question please: I get a checksum error on the BD of "The Nightmare Before Xmas":
...
Anything I can do :confused:

Is only a WARNING, the flac is OK?

lostclusters

17th July 2011, 18:22

I have been attempting to back my BluRay LotR EE discs. My Flac encoded audio is distorted and out of sync. I have not had this trouble with any of my other discs. I use EAC3TO and HdBrStreamExtractor as a gui. I have the TMT codecs loaded and my log show them being used. Has anyone else seen this issue and what can be done about it?

tebasuna51

17th July 2011, 19:09

lostclusters

18th July 2011, 01:07

There are two problems with DTS-MA 6.1 (like included in BluRay LotR EE discs).

- You need ArcSoft Decoder v1.1.0.0, other like v1.1.0.7 make wrong 6.1 decodes.

- If you downmix DTS-MA 6.1 to 5.1 with eac3to the surround channeles are wrong mixed.

What's the best way to ensure the correct version of that codec? When the discs are played (w/TMT3) they play fine.

quartex

18th July 2011, 12:34

piranha (REAL 3D + 2D)

I created the iso using anydvd HD, then using clownBD I'm trying to backup the 2D form

There are two tracks

1) 00005.mpls, 00007.m2ts, 1:28:20
- Chapters, 12 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, Italian, multi-channel, 48kHz
- DTS Master Audio, English, multi-channel, 48kHz

2) 00004.mpls, 00005.m2ts, 1:28:20
- Chapters, 12 chapters
- h264/AVC (left eye), 1080p24 /1.001 (16:9)
- h264/AVC (right eye), 1080p24 /1.001 (16:9)
- DTS Master Audio, Italian, multi-channel, 48kHz
- DTS Master Audio, English, multi-channel, 48kHz

so I selected the 00004.mpls and then the left eye stream with DTS Master Audio, Italian and italian subtitles
eac3to is logging plenty of "The source file seems to be damaged - discontinuity" and cannot demux
It doesn't seem a bad rip because I can demux 00005.mpls without any errors.
Any ideas?

phate89

18th July 2011, 13:01

hi..
edit:the first question was stupid.. there is a specific function for delay without using edit...
And a second question.. is there a way to do multiple edits? it's boring to have to create 4 files to do 4 edits..

kalston

18th July 2011, 18:21

There are two problems with DTS-MA 6.1 (like included in BluRay LotR EE discs).

- You need ArcSoft Decoder v1.1.0.0, other like v1.1.0.7 make wrong 6.1 decodes.

- If you downmix DTS-MA 6.1 to 5.1 with eac3to the surround channeles are wrong mixed.

Thanks!
I was wondering why my ripped LotR EE mkvs had this horrible sound. It's working perfectly now.

tebasuna51

18th July 2011, 21:25

lostclusters

19th July 2011, 01:42

Quote:
Originally Posted by tebasuna51
There are two problems with DTS-MA 6.1 (like included in BluRay LotR EE discs).

- You need ArcSoft Decoder v1.1.0.0, other like v1.1.0.7 make wrong 6.1 decodes.

- If you downmix DTS-MA 6.1 to 5.1 with eac3to the surround channeles are wrong mixed.

What's the best way to ensure the correct version of that codec? When the discs are played (w/TMT3) they play fine.

How do I get the version number from the codec?

Midzuki

19th July 2011, 02:07

How do I get the version number from the codec?

Open Windows Explorer, right-click on the file dtsdecoderdll.dll, select "Properties".

lostclusters

19th July 2011, 02:31

Tyvm!

robpdotcom

19th July 2011, 02:40

Which version of TMT included v1.1.0.0?

NanoBot

19th July 2011, 02:48

Hi,

...- If you downmix DTS-MA 6.1 to 5.1 with eac3to the surround channeles are wrong mixed.

since this is exactly what I want do do ( 6.1 DTS-MA -> 5.1 Flac ), could you please give me a hint what's going wrong when downmixing with eac3to and version 1.1.0.0 of the dll ? My idea to circumvent the problem would be to decode the 6.1 DTS-MA to 7 seperate wave-files, to do the downmix to 5.1 manually in a correct way using audacity and then to reencode the wavs to a 5.1 Flac.

Thx in advance, Nanobot

tebasuna51

19th July 2011, 03:37

...since this is exactly what I want do do ( 6.1 DTS-MA -> 5.1 Flac ), could you please give me a hint what's going wrong when downmixing with eac3to and version 1.1.0.0 of the dll ?...

In this thread: http://forum.doom9.org/showthread.php?p=1465610#post1465610

Now eac3to make the downmix 6.1 -> 5.1 with:
SurroundL = Cs + 0.7071xRs
SurroundR = Ls + 0.7071xRs

And must be:
SurroundL = Ls + 0.7071xCs
SurroundR = Rs + 0.7071xCs

Yes, you can decode the 6.1 like wav's and make the mux manually.

NanoBot

20th July 2011, 01:47

Hi,

:thanks: for the link and the information, that is accurately the info I was looking for.

Nevertheless, I would suspect that this downmix matrix for the surround channels might result in levels above 0dB for Ls and Rs ? Or are the levels for Ls, Cs and Rs in a 6.1 DTS-MA file are already limited to values where Ls + 0.7071 x Cs and Rs + 0.7071 x Cs can't exeed a level of 0dB ?

If my concern is true, would it be ok just to normalize those two channels after the downmix, or do you have a recommendation how to avoid this probable problem ?

C.U. Nanobot

tebasuna51

20th July 2011, 02:50

Yes, we always need normalize after a downmix.
eac3to normalize the 6 channels automatically.

musinsky

20th July 2011, 16:21

Hi,
is there any way to set "not standard" stretch ?!
For example, how with eac3to make stretch 25.002 -> 23.976 ?

Thanks.

tebasuna51

20th July 2011, 19:02

Nope (use the search, this question was answered already in this thread).

lostclusters

21st July 2011, 03:51

I put in a search of "6.1" and it told there are no results. Searching is fun with a crippled search function. It said the same thing when attempting to search on DTS-MA 6.1.

Sparktank

21st July 2011, 05:28

Use Google with search functions

dts-ma 6.1 site:forum.doom9.org

Most search engines in forums don't allow you to search things less than 4 characters.
"quotes" are usually required.

Google has a better chance, since it's diverse.
dts-ma will show you dts-hd ma, dtsma, dts ma, etc with Google.
With the forum search, it looks for exactly what you type in.

lostclusters

22nd July 2011, 03:10

That was a very good tip, TYVM!

nibus

26th July 2011, 03:55

quick question:

If I transcode DTS-MA to FLAC, and later decide to transcode that FLAC back to DTS-MA with Surcode, will the end file be basically identical to the first, or will the -core or other information be lost forever?

mastrboy

26th July 2011, 20:43

Is it possible to disable that annoying "Pling" sound when done with a file?
And is it possible to skip the 1-pass wav file and encode directly to in example a m4a aac file or is that wav file needed?

nurbs

26th July 2011, 20:55

Is it possible to disable that annoying "Pling" sound when done with a file?
Rename or delete the file "success.wav" in your eac3to directory.

And is it possible to skip the 1-pass wav file and encode directly to in example a m4a aac file or is that wav file needed?
I haven't tried, but adding "-no2ndpass" to your commandline might do it. The wav is written so that if a second pass is necessary, which isn't always known before the first pass is complete, you don't need to decode the audio again. You might have audiable problems if you use that option.

asarian

2nd August 2011, 11:03

I'm having trouble converting the following stream to LPCM:

DTS-HD Master Audio English 1064 kbps 1.0 / 48 kHz

The resultant LPCM file is playing way too fast. I take it because there's no official LPCM 1.0? In any case, how do I convert this to a normal, playable, LPCM track?

Thanks.

iSeries

2nd August 2011, 22:13

About the eac3to 6.1 DTS-HD downmixing 'bug', if I output to individual WAV files could someone let me know how to mix the back center channel with the surround left/right channels manually/correctly please?

Could I do it with eac3to, eg: eac3to rightsurround.wav+backcenter.wav output.wav

tebasuna51

3rd August 2011, 09:56

Nope, you can use an audio editor (like Audacity or others), a command line tool like sox, or Avisynth (with BeHappy, wavi, ...) to do a mix like:

BL = 0.5858 x SL + 0.4142 x BC
BR = 0.5858 x SR + 0.4142 x BC

The recommended .avs' can be something like:

sl = WavSource("D:\...\SideLeft.wav")
bc = WavSource("D:\...\BackCenter.wav")
MixAudio(sl, bc, 0.5858, 0.4142)
Normalize()

To obtain BL channel, and for BR channel:

sr = WavSource("D:\...\SideRigth.wav")
bc = WavSource("D:\...\BackCenter.wav")
MixAudio(sr, bc, 0.5858, 0.4142)
Normalize()

iSeries

3rd August 2011, 17:06

tebasuna51, thanks! The resulting .wav files seem to be 32 bit float though (from BeHappy), what do I need to add to the scripts to get 24 bit?

Thanks again!

b66pak

3rd August 2011, 19:09

add:

ConvertAudioTo24bit()
_

iSeries

3rd August 2011, 20:03

Thanks guys!

...is it ok to reduce the bit depth of the new .wav's from 32 bit float back to 24 bit with eac3to? or is it more recommended to do this with BeHappy?

iSeries

3rd August 2011, 21:55

Also, do I need to normailize ALL the wav's that will make up the 5.1, or just the two created with BeHappy?

tebasuna51

3rd August 2011, 22:54

Well, nothing is perfect. You can't obtain always the same global audio volume with 7 speakers than with 6.

If you want preserve the balance volume betwen front and back channels maybe you need make this mix:

BL = SL + 0.7071 x BC
BR = SR + 0.7071 x BC

preserve the 32 bit float and check if there are values greater than 0 dB, if yes you need use the normalize only over BL-BR and lose some back volume, if not you don't need the normalize and the balance volume is ok.

iSeries

3rd August 2011, 23:05

Thanks, like this?

sl = WavSource("D:\...\SideLeft.wav")
bc = WavSource("D:\...\BackCenter.wav")
MixAudio(sl, bc, 0.5858, 0.7071)

kypec

4th August 2011, 09:34

Thanks, like this?

sl = WavSource("D:\...\SideLeft.wav")
bc = WavSource("D:\...\BackCenter.wav")
MixAudio(sl, bc, 0.5858, 0.7071)
Almost correct but the SL / SR channels must be factored by 1, not 0.5858.
Replace your last line with this:MixAudio(sl, bc, 1.0, 0.7071)

iSeries

4th August 2011, 13:29

Thanks! Sorry one last question lol, how can I check if clipping has occured? Am I right in thinking that if I open the wav in Audacity and nothing goes beyond 1.0 or -1.0 then there is no clipping?

Also, I don't really understand any of this so a brief explanation on the difference between

MixAudio(sl, bc, 0.5858, 0.4142)

and

MixAudio(sl, bc, 1.0, 0.7071)

...would be greatly appreciated :thanks:

tebasuna51

4th August 2011, 14:58

..., how can I check if clipping has occured? Am I right in thinking that if I open the wav in Audacity and nothing goes beyond 1.0 or -1.0 then there is no clipping?

You can use Audacity -> View -> Show clipping

MixAudio(sl, bc, 0.5858, 0.4142)
make:
BL = 0.5858 x SL + 0.4142 x BC
the audio never can be greater than 1.0 (0 dB), without clip

and

MixAudio(sl, bc, 1.0, 0.7071)
make:
BL = 1.0 x SL + 0.7071 x BC
the audio can be greater than 1.0, then you can have clipping.

iSeries

4th August 2011, 16:26

Many thanks. I used MixAudio(sl, bc, 1.0, 0.7071) for LOTR Fellowship 1/2, and this produced no clipping. The Two Towers 1/2 and Return of the King 1/2 did produce clipping using that so for these I re-did with MixAudio(sl, bc, 0.5858, 0.4142). I'm guessing thats all ok (Audacity shows no clipping for all wavs) :)

Vasch the stampede

4th August 2011, 18:28

eac3to v3.24
command line: eac3to "...\00010.m2ts" 1:"...\1.txt" 2:"...\2.h264" 4:"...\4.wav"
------------------------------------------------------------------------------
M2TS, 1 video track, 6 audio tracks, 18 subtitle tracks, 2:02:34, 24p /1.001
1: Chapters, 21 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: DTS Master Audio, Italian, 6.1 channels, 16 bits, 48kHz
(core: DTS-ES, 6.1 channels, 16 bits, 1509kbps, 48kHz)
5: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
7: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
8: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
9: Subtitle (PGS), English
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), English
14: Subtitle (PGS), English
15: Subtitle (PGS), Italian
16: Subtitle (PGS), Italian
17: Subtitle (PGS), Italian
18: Subtitle (PGS), Italian
19: Subtitle (PGS), Italian
20: Subtitle (PGS), Italian
21: Subtitle (PGS), Danish
22: Subtitle (PGS), Finnish
23: Subtitle (PGS), Modern Greek
24: Subtitle (PGS), Icelandic
25: Subtitle (PGS), Norwegian
26: Subtitle (PGS), Swedish
Creating file "..\1.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Remapping channels...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Writing WAV...
[v02] Creating file "...\2.h264"...
[a04] Creating file "...\4.wav"...
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] Caution: The WAV file is bigger than 4GB. <WARNING>
[a04] Some WAV readers might not be able to handle this file correctly. <WARNING>
Video track 2 contains 176328 frames.
eac3to processing took 56 minutes, 13 seconds.
Done.

Why remapping channels?

tebasuna51

5th August 2011, 11:04

Because the DTS internal channel order is different than standard WAV order.

der_andi

7th August 2011, 01:39

Hi,
I noticed a performance decrease in reading the blu-ray structure compared to an older version of eac3to. Consider these automated commands:

eac3to.exe f:
eac3to.exe f: 1)
eac3to.exe f: 2)
...
eac3to.exe f: 5)

The output of the first line is used by a program in order to determine the number of titles on that disc. Let it be 5, for example. Then it analyses all titles automatically. Version 3.18 can do these commands in around 400 seconds, while the latest version 3.24 needs around 490. I get similar results with all my blu-rays.

Any ideas what's causing this additional time?

Best regards,
Andi

NanoBot

8th August 2011, 10:30

Hi,

not directly an eac3to question, but the reason for this question is the eac3to bug when downmixing 6.1 to 5.1 and the need to decode and downmix manually: I think and hope that you can help me with this question:

I found these channel orders for 5.1 respectively 6.1 flac files:

5.1 FL, FR, C, LFE, SL, SR
6.1 FL, FR, C, LFE, BC, SL, SR

Are these channels orders, which I got from ffdshow as audio audio processor from and searching for them in the internet, correct ?

Thx in advance, NanoBot

tebasuna51

8th August 2011, 10:46

But is possible also:

6.1 FL, FR, C, LFE, BL, BR, BC

but madshi know the problem, is only a bug that can be solved with a new version ...

tooftah

10th August 2011, 22:59

Hello,

can anyone tell me what decoder to use for 2-channel DTS Express tracks? I've tried using Arcsoft 1.1.0.0, 1.1.0.8 and Nero 7 decoders, and they all produce different results. Stereo WAVs from Arcsoft seem to be identical, except for 1 byte difference every few kilobytes (checked using HxD). Nero 7 gives a totally different stream. Which one to choose?

Can the difference be caused by different versions of Windows? I have Arcsoft 1.1.0.8 on a Windows 7 x86_64 PC and Nero 7 + Arcsoft 1.1.0.0 on a Windows XP x86 PC. However they both decode e.g. DTS-HD 5.1 files identically.

Thunderbolt8

10th August 2011, 23:56

afaik dts express is not supported by any of encoder (which eac3to uses)

edit: or I am wrong (guess I am) and its only that mkvmerge doensnt support muxing it.

tooftah

11th August 2011, 01:39

Thunderbolt8,

I was talking about DEcoders, that work with eac3to. I have a Stereo DTS Express track, which I want to decode using eac3to. And my problem is, that all three decoders output different WAVs. However, you are in a way right, because I intend to compress it and then mux to a matroska file.

Somewhere in the middle of this giant thread madshi talked about using Nero decoder on files, which Arcsoft doesn't handle. But all three decoders produce audio without an error, at least the program doesn't show any + I hear sound on playback, not silence or some random noise, etc.

iSeries

12th August 2011, 17:51

Hi,

Having a bit of difficulty downconverting the sound track to 'Faster' to 16-bit. After processing with the -down16 switch, the eac3to log indicates that the DTS-MA track is a constant 20-bit. Is there a way to get eac3to to downconvert 20-bit to 16-bit and not assume 24-bit?

nurbs

12th August 2011, 19:15

Hard to say without seeing the log, but it probably is a 24bit file, it's just that the actual audio data contained in it is only 20bit. I don't think the initial bitdepth has much influence on the speed of the processing, but tebasuna51 will probably know better than me. If your target format isn't lossless you can skip the downconversion to save time.

iSeries

12th August 2011, 20:51

You are right in that it is a 24-bit file and that the actual audio data is 20-bit. I'm asking if there is a way to down-convert to 16-bit from the 20-bit data. As it is, the -down16 switch treats the file as though it has 24-bits of audio data.

tebasuna51

12th August 2011, 23:38

I can't understand your problem, use -down16.

The uncompressed audio data are stored in 2 bytes for 16 bits and in 3 bytes for 24 bits, don't exist a especific 20 bits storage, when eac3to say is 20 bits means than the 4 less significants bits, from the 3 bytes/24 bits, are always 0.
Here down-convert from 24 or 20 bits is the same, forget the less significant byte.

iSeries

13th August 2011, 02:36

So this doesn't apply to 20-bit audio files padded to 24-bit?

http://forum.doom9.org/showthread.php?p=1404212&highlight=iseries#post1404212

http://forum.doom9.org/showthread.php?p=1404320&highlight=iseries#post1404320

tebasuna51

14th August 2011, 02:46

Sorry but I can't understand your question.

nibus

15th August 2011, 10:34

can you demux to a destination directory instead of the eac3to directory?

der_andi

15th August 2011, 12:58

can you demux to a destination directory instead of the eac3to directory?

Yes, you can. Here's an example for demuxing a video stream from a blu-ray:
eac3to.exe F: 1) 2: C:\my\destination\directory\movie.mkv

That's the simplest way. But if you demux several streams at once, you have to type the path for every stream. That's a lot of work.
Instead, you could append the directory of eac3to.exe to the PATH environment variable. Then eac3to works from every directory, e.g. your desired destination directory. And you can specify filenames as you've been doing all along.

nibus

15th August 2011, 12:59

Instead, you could append the directory of eac3to.exe to the PATH environment variable. Then eac3to works from every directory, e.g. your desired destination directory. And you can specify filenames as you've been doing all along.

Cool, I hadn't thought of this. thanks!

TinTime

15th August 2011, 17:40

Yes, you can. Here's an example for demuxing a video stream from a blu-ray:
eac3to.exe F: 1) 2: C:\my\destination\directory\movie.mkv

If you use something like this then eac3to will demux to the specified directory:

eac3to.exe source "C:\my\destination\directory\movie.*"

Weirdo

15th August 2011, 18:12

can you demux to a destination directory instead of the eac3to directory?I'd suggest Another EAC3to GUI Plus for that. Works beautifully.

nibus

16th August 2011, 09:51

If you use something like this then eac3to will demux to the specified directory:

eac3to.exe source "C:\my\destination\directory\movie.*"

Oh nice, that works even better. :)

ramicio

17th August 2011, 01:56

Any time I get movies with 6.1 DTS-HD tracks the resulting file is much longer than what it's supposed to be, and the sound is bad/wrong. This is with the Arcsoft decoder.

nibus

17th August 2011, 07:13

Any time I get movies with 6.1 DTS-HD tracks the resulting file is much longer than what it's supposed to be, and the sound is bad/wrong. This is with the Arcsoft decoder.

You need dtsdecoder.dll version 1.1.0.0 to properly decode 6.1. However 1.1.0.0 does NOT work correctly with 7.1 (* on non-standard channel mapping only), for that you need version 1.1.0.8. When you switch versions you will need to re-register ASAudioHD.ax by entering this into the command prompt:

regsvr32.exe ASAudioHD.ax

Both versions work with regular 5.1, and to my knowledge there is no need for any other versions.

tebasuna51

17th August 2011, 09:30

... However 1.1.0.0 does NOT work correctly with 7.1, for that you need version 1.1.0.8.
Please can you explain for what don't work 1.1.0.0 with 7.1, works fine for me.

nibus

17th August 2011, 11:02

Please can you explain for what don't work 1.1.0.0 with 7.1, works fine for me.

I think I read earlier in this thread that the channel mapping was off... I'll see if I can dig the specific post up.

edit: Here it is-

http://forum.doom9.org/showthread.php?p=1469584#post1469584

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.

Only for non-standard channel mapping.

ramicio

17th August 2011, 13:52

I know we're not supposed to ask where to get these filters, but I'm going to just ask rhetorically...where are we supposed to find all these random older versions of this filter? Is there one version that works perfectly with every possible layout? Someone really needs to get on the ball and get DTS-HD decoding into libavcodec.

tebasuna51

17th August 2011, 14:07

1.1.0.0 work also with non-standard channel mapping without garbage.
Like don't exist channels Ls-Rs in WAV make a remux to BL-BR correct to my opinion.

ramicio

17th August 2011, 15:33

It's almost getting to be more worth the time to just fake-header a DTS-HD file and let the DTS suite decode the files. I really find that having to switch filter versions to be inexcusable (on Arcsoft's part, not madshi, he makes awesome stuff). I have no idea how I would make a 6.1 wav, though. There is no documentation on channel masks for wavavimux.

Midzuki

17th August 2011, 18:14

It's almost getting to be more worth the time to just fake-header a DTS-HD file and let the DTS suite decode the files. I really find that having to switch filter versions to be inexcusable (on Arcsoft's part, not madshi, he makes awesome stuff).

I agree. However I do not have the entire recipe for adding a new header (AND a new footer) to a "naked" :) DTS-HD file — would you mind sharing yours? :confused:

I have no idea how I would make a 6.1 wav, though. There is no documentation on channel masks for wavavimux.

Channel Mask for "normal" 6.1 audio = 319.

CHANNEL NAME --- Decimal Value
¯¯¯¯¯¯¯¯¯¯¯¯ ¯¯¯¯¯¯¯¯¯¯¯¯¯

FRONT_LEFT 1
FRONT_RIGHT 2
FRONT_CENTER 4
LOW_FREQUENCY 8
BACK_LEFT 16
BACK_RIGHT 32

FRONT_LEFT_OF_CENTER 64
FRONT_RIGHT_OF_CENTER 128

BACK_CENTER 256
SIDE_LEFT 512
SIDE_RIGHT 1024

*************************************

TOP_CENTER 2048
TOP_FRONT_LEFT 4096
TOP_FRONT_CENTER 8192
TOP_FRONT_RIGHT 16384
TOP_BACK_LEFT 32768
TOP_BACK_CENTER 65536
TOP_BACK_RIGHT 131072
RESERVED 262144

ramicio

17th August 2011, 18:48

I make a mono wav that contains silence, make it substantially longer than the track we're trying to work with (we'll call this "the useless source") I need to decode, encode a track with the exact specs as the useless source, copy the first 140 bytes in a hex editor, not the last character and what its pattern of character looks like before that character, in the useless source, cut up to the last character of our nice header, and copy INTO the beginning of the useless source. It's very time consuming, but since it's a DTS tool, and all things most likely come from the tool, there can't be anything it can't handle.

Now I see the scheme for the channel mask, it's just sums! I never saw the chart you posted though, so thank you!

ramicio

19th August 2011, 05:00

Would there ever be any chance of taking multiple mono WAV files in to create a single multi-channel file? I can't find any decent and sure-fire way to do this with any other utility. WavAVImux to graphedit with wavdest, you are limited to a 4 GB output. Avisynth and VirtualDub, save WAV, can break the 4 GB barrier, but what about weird channel mappings such as 6.1? The there's wavewizard which I can't even begin to figure out, and have no idea about size limitations or channel mapping.

Midzuki

19th August 2011, 05:56

^ wavi.exe:

WAVI v1.06F - (c) 2k7 Tamas Kurucsai, with lots of help from tebasuna51
Licensed under the terms of the GNU General Public License.

This utility extracts the first uncompressed PCM audio track
from an AVI file and saves it to a WAV file. This is not quite
useful for most AVI files since they usually contain some kind
of compressed audio, but it can come handy when it's needed
to save an audio track from AviSynth.

Usage: WAVI <avi-file> [ <wav-file> [ /R | /X | /M <mask> ] ]

If <avi-file> is a valid AVI file which contains PCM audio,
the audio track will be written to <wav-file> as a WAV file.
If '-' is passed as <wav-file>, the WAV file will be written
to the standard output.
If <wav-file> is not given, only information will be printed
about the audio track.

On success, the exit code will be 0 and the first line printed
to the standard error will look like the following:

Found PCM audio: <c> channels, <r> Hz, <b> bits, <l> seconds.

where <c> is the number of audio channels, <r> is the sampling
rate, <b> is the number of bits per sample and <l> is the length
of the track in seconds.

If the audio track contains floating-point samples, the next line
printed to the standard error will be:
Audio track contains floating-point samples.

If an error occurs, the exit code will be 1 and some useful
error message will be printed to the standard error.

WAV files larger than 4 GB may be created. However,
such WAV files are non-standard and
may not be handled correctly by some players and encoders.
A warning will be printed to the standard error when such a WAV file is created.

WAVI accepts the following options:

/R - Write a raw file of samples without the WAV header.

/X - Write an extended WAV header containing the default
channel mask for multi-channel audio.

/M <mask> - Write an extended WAV header containing the
specified channel mask for multi-channel audio.

The default channel masks are:

Mask Chan. MS channels Description
---- ----- ------------------------- ----------------
4 1 FC Mono
3 2 FL FR Stereo
259 3 FL FR BC First Surround
51 4 FL FR BL BR Quadro
55 5 FL FR FC BL BR like Dpl II (without LFE)
63 6 FL FR FC LF BL BR Standard Surround
319 7 FL FR FC LF BL BR BC With back center
255 8 FL FR FC LF BL BR FLC FRC With front center left/right

Some other common channel masks:

Mask Chan. MS channels Description
---- ----- ------------------------- ----------------
7 3 FL FR FC
263 4 FL FR FC BC like Dpl I
271 5 FL FR FC BC LF
59 5 FL FR BL BR LF

but what about weird channel mappings such as 6.1?

Why do you think 6.1 is "weird"? :confused: IMHO, the only "weird" audio channels are the infamous/Sonyc Front-Left-Of-Center and Front-Right-Of-Center.

The there's wavewizard which I can't even begin to figure out, and have no idea about size limitations or channel mapping.

Me too hates wavewizard, you are not alone :D

tebasuna51

19th August 2011, 09:53

WaveWizard works fine for me but don't fill the MaskChannel field (always 0) then you need some other util like:

wavfix
Usage:
WavFix <input.wav> [output.wav] [-ignorelength] [-o #] [-m #] [-i/f #] [-c #] [-s #]

Where:
<input.wav> wav or w64 up to 8 channels, int or float, 8-16-24-32-64 bits,
PCM/WAVE_FORMAT_EXTENSIBLE/W64, any extra chunks, allowed > 4GB.
To use STDIN use - as input filename.

[output.wav] If not present, input.wav is used suffixed with: _fix
The header can be selected, see -o parameter.

[-ignorelength] If present the length in wav header is ignored, useful for wav
> 2/4 GB. Problem: extrachunks at end of file treated as data

[-o #] Output header. 0=RAW, 1=WAV(Default), 2=RF64, 3=W64

[-m #] MaskChannel. Force output WAVE_FORMAT_EXTENSIBLE. 0 to default

If any of the next parameters are present the input is considered RAW:
[-i #] BitsPerSample Integer. Default 16. Valid 8, 16, 24, 32

[-f #] BitsPerSample Float. Valid 32 or 64.

[-c #] NumChannels. Default 2. Valid only 1 to 8.

[-s #] SampleRate. Default 48000 Hz. Any value is allowed.

[EDIT] wavfix.exe included in wav2Util.7z here (https://forum.doom9.org/showthread.php?p=1624209#post1624209)

ramicio

19th August 2011, 14:55

Wavi doesn't work for me once an AVI file is created with wavavimux, even for ones that are under 2 GB. It used to, now it doesn't. It simply says there is audio in there, but it's not PCM. I don't understand how to use wavewizard, and I don't understand what that wavfix thing would even do. I just don't get how more people, or anyone at all has asked about eac3to being able to accept multiple mono wavs as input.

Midzuki

19th August 2011, 18:23

Wavi doesn't work for me once an AVI file is created with wavavimux, even for ones that are under 2 GB. It used to, now it doesn't. It simply says there is audio in there, but it's not PCM.

Thanks for useful the info --- I wasn't aware of that, because, until now, I had used wavi.exe only for .AVS files. Anyway, here is the workaround:

<intermed.avs>
DirectShowSource("audio-only.avi")
</intermed.avs>

wavi intermed.avs final.wav -M ###

Still, I don't know what you intend to do by creating a real multichannel .WAV greater than 4GB :confused: Applications like NeroAacEnc, WavPack, Aften, whatever, will happily encode from the stdin, and commercial compressors, by default, work with mono .WAVs :confused:

ramicio

19th August 2011, 18:27

Because, as previously stated, I have problems with certain channel schemes of DTS-HD Master, and I'm not going to try to hunt down the internet for specific versions of filters that I'll never find. So I must use the DTS suite to decode these weird ones into mono wavs. My end goal is a 5.1 FLAC. So I need to feed eac3to a multichannel WAV to downmix into 5.1 channels.

Midzuki

19th August 2011, 19:18

Ah, I see, at last :o

Hmmm, IMHO there really is no good reason why eac3to doesn't support .AVI as input. :confused:

It's a shame that neither sox, nor wavi, can output .W64 or .RF64 :(

ramicio

19th August 2011, 19:34

That would be a nice step forward if it supported AVI as an input. But a lot of things still have trouble with opening an AVI with only audio.

tebasuna51

19th August 2011, 21:18

... So I must use the DTS suite to decode these weird ones into mono wavs. My end goal is a 5.1 FLAC. So I need to feed eac3to a multichannel WAV to downmix into 5.1 channels.

eac3to can downmix the 7.1 L,R,C,LFE,Lss,Rss,Lsr,Rsr DTS channels to 5.1, but not 7.1 L,R,C,LFE,Ls,Rs,Lsr,Rsr.

How want you downmix the Ls,Rs,Lsr,Rsr to only 2 Surround channels? Please put the coefficients X1, X2 in:

SL = X1 x Ls + X2 x Lss
SR = X1 x Rs + X2 x Rss

I can make an avs file with your mix and can be used by BeHappy (or wavi) to obtain the 5.1 flac

ramicio

19th August 2011, 21:26

I'm fine with the way eac3to handles 7.1 stuff. Never had a problem. It's these pesky 6.1 I come across now and then. I like eac3to because it detects clipping. Would it be possible for avisynth to turn out 6.1?

hello_hello

20th August 2011, 03:19

Does ffdshow decode 6.1 DTS-HD? If so you could open the DTS file via DirectShow, use ffdshow's mixer matrix to mix it to stereo, 5.1ch, or any way you like. From there, any encoder which will encode via DirectShow should be able to do the job.

I sometimes use ffdshow to mix to stereo (for those times when the audio track starts off stereo then switches to 5.1ch, and foobar's internal 5.1ch mixer has a hissy fit), while using foobar2000 to encode to MP3. Assuming ffdshow decodes 6.1ch DTS, I can't think of a reason why you couldn't use ffdshow to mix it to 5.1, then use foobar2000 to encode the output directly to flac.

Or, if you don't want to use ffdshow's mixer, you could leave it disabled and get foobar200 to spit out a multichannel wave file.

You'd need the foobar DirectShow decoder plugin.

ramicio

20th August 2011, 03:38

No, Arcsoft is the only way to decode DTS-HD Master besides the DTS suite, and the rare and impossible to find Sonic decoder. I want eac3to to mix my stuff because it detects clipping, and it also uses the best possible processing. I also don't care for foobar2000 for anything but playback.

hello_hello

20th August 2011, 06:09

No, Arcsoft is the only way to decode DTS-HD Master besides the DTS suite, and the rare and impossible to find Sonic decoder.

Fair enough. I was just looking at ffdshow's settings and noticed the Pass-Through option for DTS HD, so wondered if that meant it could also decode it.

I want eac3to to mix my stuff because it detects clipping, and it also uses the best possible processing. I also don't care for foobar2000 for anything but playback.

I'm sure there's a rational logic behind all that.

tebasuna51

20th August 2011, 10:34

... It's these pesky 6.1 I come across now and then. I like eac3to because it detects clipping. Would it be possible for avisynth to turn out 6.1?
For 6.1 in monowavs you can use:
sl=WavSource("D:\tmp\SL.wav").ConvertAudioToFloat() # Load channels to mix
sr=WavSource("D:\tmp\SR.wav").ConvertAudioToFloat()
bc=WavSource("D:\tmp\BC.wav").ConvertAudioToFloat()
bl=MixAudio(sl, bc, 1.0, 0.7071) # Mix half power of BC into SL
br=MixAudio(sr, bc, 1.0, 0.7071) # Mix half power of BC into SR

fl=WavSource("D:\tmp\FL.wav").ConvertAudioToFloat() # Load the rest of channels
fr=WavSource("D:\tmp\FR.wav").ConvertAudioToFloat()
fc=WavSource("D:\tmp\FC.wav").ConvertAudioToFloat()
lf=WavSource("D:\tmp\lf.wav").ConvertAudioToFloat()

MergeChannels(fl, fr, fc, lf, bl, br) # Merge the 6 channels
Normalize() # Normalize to avoid clip
ConvertAudioTo24bit() # Change to desired precision

wonkey_monkey

20th August 2011, 22:47

From what I've read, eac3to should be able to extract a core AC3 track from a TrueHD file, but I get this:

M:\TMP\bd>eac3to bd.thd bd.ac3
TrueHD, 7.1 channels, 48kHz, dialnorm: -27dB
AC3 encoding doesn't support back channels. Will mix them into the surround.
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Remapping channels...
Mixing surround channels...
Encoding AC3 <640kbps> with libAften...

which looks like it's completely re-encoding. Am I missing an option?

David

ramicio

20th August 2011, 22:55

That file doesn't have an AC3 core, it would be listed.

It would look like this:

d:\>\apps\eac3to\eac3to.exe file.thd
TrueHD/AC3, 7.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)

Along with my previous dilemma, I decided to just have 1.1.0.0 and 1.1.0.7 of the Arcsoft filters and flip-flop them depending on what I need.

And why does a line get added to the bottom of the code boxes?

Midzuki

20th August 2011, 23:29

Well, maybe Lord Mulder :devil: could want to write a "dtshdma2wavs.exe" :D , which would always pick the "best" Arcsoft DLL for each case :)

Anakunda

22nd August 2011, 11:47

Hello any chance to add custom Vorbis encoder (oggenc2) support to eac3to ?

Chumbo

22nd August 2011, 16:53

@madshi,
I was doing some batch extracts of audio from my media files when I noticed the following that I had not seen before.eac3to v3.24
command line: eac3to "file1.mkv" 2: "file1.ac3"
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 0:42:01, 24p /1.001
1: h264/AVC, 720p24 /1.001
2: AC3, ???, 5.1 channels, 448kbps, 48kHz, dialnorm: -25dB
I tried searching this thread but could not really find anything on the ???.

tebasuna51

22nd August 2011, 17:03

...
2: AC3, ???, 5.1 channels, 448kbps, 48kHz, dialnorm: -25dB
Seems a country code not supported by eac3to.
Try with MediaInfo.

tebasuna51

22nd August 2011, 17:07

Hello any chance to add custom Vorbis encoder (oggenc2) support to eac3to ?
You can use the external encoder with:
eac3to input stdout.wav | OggEnc2 -q 3 --ignorelength -o output.ogg -

Chumbo

22nd August 2011, 17:32

Seems a country code not supported by eac3to.
Try with MediaInfo.
I had done that already and didn't notice any difference. I only ask because normally I don't see anything in the ??? position, i.e.,
2: AC3, 5.1 channels, 448kbps, 48kHz, dialnorm: -25dB, 17ms
Aside from items like duration, file size, etc. MediInfo reports identical results for files with the ??? and without.
Audio
ID : 2
Format : AC-3
Format/Info : Audio Coding 3
Mode extension : CM (complete main)
Codec ID : A_AC3
Duration : 42mn 10s
Bit rate mode : Constant
Bit rate : 448 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Compression mode : Lossy
Delay relative to video : 17ms
Stream size : 135 MiB (12%)
Language : English
I don't see any problems caused by this but I was mostly curious as to what the ??? position is supposed to be reporting.

tebasuna51

23rd August 2011, 02:57

...I only ask because normally I don't see anything in the ??? position,...
No?
Maybe your mkv's don't have the language field filled:
eac3to v3.24
command line: "D:\Programa\eac3to\eac3to.exe" "D:\Test\cono3.mkv"
------------------------------------------------------------------------------
MKV, 1 video track, 2 audio tracks, 1 subtitle track, 0:00:22, 25p
1: h264/AVC, English, 720p25
2: DTS, English, 5.1 channels, 24 bits, 755kbps, 48kHz
3: AC3, German, 5.1 channels, 448kbps, 48kHz
4: Subtitle (SRT), Spanish

Chumbo

23rd August 2011, 03:37

No?
Maybe your mkv's don't have the language field filled:
eac3to v3.24
command line: "D:\Programa\eac3to\eac3to.exe" "D:\Test\cono3.mkv"
------------------------------------------------------------------------------
MKV, 1 video track, 2 audio tracks, 1 subtitle track, 0:00:22, 25p
1: h264/AVC, English, 720p25
2: DTS, English, 5.1 channels, 24 bits, 755kbps, 48kHz
3: AC3, German, 5.1 channels, 448kbps, 48kHz
4: Subtitle (SRT), Spanish
MediaInfo seems to have no issues picking up the language as English and I always set the language on the audio stream specifically to eng using mkvtoolnix when creating my MKV files. Like I said, it's not a big deal, I was just curious about the ???.

sneaker_ger

23rd August 2011, 11:13

Mkvmerge (in recent versions) doesn't write the language field if set to "English" and eac3to doesn't recognize that. That's why eac3to doesn't display any language for English tracks if muxed with such a version. I don't know where the "???" comes from. Look at the value of the field "Language" in mkvinfo - maybe you can spot a difference between files with "" (empty) and "???".

elmarikon

24th August 2011, 15:42

Hi guys!

I don't know if this has been asked before, but how can I prevent eac3to from downsampling wavs when I come from a 24bit, 48KHz, 6ch pcm file, trying to extract mono wav files?
I don't know why, but for some reason eac3to from time to time thinks it has to reduce the bit depth to 16bit...

I have tried to define all values, like eac3to in-6ch.pcm out\out.wavs -48000 -24 -6 -little -override (even tried -down24)... still receving 16bit wavs, but I do need 24bit.

Wht can I do, and why s it doing that anyway?

l the best and thanx for the apart from that great tool!

all the best

swk

Snowknight26

24th August 2011, 15:50

Your input file is 16-bit then.

ramicio

24th August 2011, 16:02

Eac3to detects when a 24-bit file has the 8 least significant bits filled with zeroes (which is easily accomplished by taking a 16-bit file and simply saving it as 24-bit) and drops it to 16 bits. Keeping it 24 bits would be pointless and a waste of space. When eac3to detects that the file is between 17 and 23 bits, it will keep it 24.

Anakunda

25th August 2011, 08:03

You can use the external encoder with:
eac3to input stdout.wav | OggEnc2 -q 3 --ignorelength -o output.ogg -

this doesnot work for me :(
"D:\media\eac3to\eac3to.exe" "I:\video\title00 ID3 Lossless.dts" -110ms -normalize -down6 -progressnumbers -log="I:\video\title00 ID3 Lossless_Output_eac3to.txt" | D:\media\OGG\aoTuV\oggenc2.exe -q 5.0 --ignorelength -o "I:\video\title00 ID3 Lossless_Output.ogg" -

Any idea?

I get the error: Please specify the source and dest files first and then the options.ERROR: Input file "(stdin)" is not a supported format

sneaker_ger

25th August 2011, 09:05

You have to specify "stdout.wav" as a fake output file, like in tebasuna51's example. Try:
"D:\media\eac3to\eac3to.exe" "I:\video\title00 ID3 Lossless.dts" stdout.wav -110ms -normalize -down6 -progressnumbers -log="I:\video\title00 ID3 Lossless_Output_eac3to.txt" | D:\media\OGG\aoTuV\oggenc2.exe -q 5.0 --ignorelength -o "I:\video\title00 ID3 Lossless_Output.ogg" -

Anakunda

25th August 2011, 09:11

If I specify fake output file then the output will go to file and not stdout right?

I tried it now:
"D:\media\eac3to\eac3to.exe" "I:\video\title00 ID3 Lossless.dts" $$fake.wav -110ms -normalize -down6 -progressnumbers -log="I:\video\title00 ID3 Lossless_Output_eac3to.txt" | D:\media\OGG\aoTuV\oggenc2.exe -q 5.0 --ignorelength -o "I:\video\title00 ID3 Lossless_Output.ogg" -

and still got error:

ERROR: Input file "(stdin)" is not a supported format

Sorry I missed the name thankyou!!!

sneaker_ger

25th August 2011, 09:18

No. "stdout.wav" is a reserved name which will let eac3to pipe to stdout instead of writing to a file.

laserfan

27th August 2011, 00:19

@tebasuna51 you wrote (a LONG time ago!)
Use the external encoder Aften with eac3to STDOUT:

eac3to 3.dtsma stdout.wav | Aften -b 320 - 3.ac3

But when I used this with a 3ch DTS that MediaInfo says is "Front: L C R", I got an ac3 file that is "Front: L, R and Side: C" which had the center channel coming-out from the surrounds! But next I looked at aften's help and tried this:

eac3to 3ch.dts stdout.wav | aften -b 576 -chconfig 3/0 - 3ch.ac3

and I think it sounds right, and MediaInfo says this output ac3 file is now "Front: L C R" just as the DTS! :)

I just wanted to thank you for having steered me in the right direction (all the way from 2009)!

:thanks:

tebasuna51

27th August 2011, 01:45

You are welcome.

Yes, with 'pipe' sintax you can add all Aften parameters.
To know the all the parameters:

Aften -longhelp

nibus

27th August 2011, 09:23

madshi, do you have any plans to incorporate qtaacenc.exe, similar to neroaacenc.exe? Piping works fine but it would be pretty handy to have a built in switch to use qtaacenc.

Midzuki

27th August 2011, 09:59

How about looking for mkvmerge, etc., in the PATH, not in the Registry :rolleyes:
[C:\]
=>eac3to -test

eac3to (v3.24) is installed
Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed
http://www.nero.com/eng/store-blu-ray.html
CAUTION: You need Nero 7. Nero 8 won't work with eac3to.
ArcSoft DTS Decoder (1.1.0.1) works fine
Sonic Audio Decoder (3.24.0.0) doesn't seem to be installed
Haali Matroska Muxer doesn't seem to be installed
http://haali.net/mkv
Nero AAC Encoder (1.5.4.0) is installed
Surcode DTS Encoder doesn't seem to be installed
http://www.surcode.com
MkvToolnix doesn't seem to be installed
http://www.bunkus.org/videotools/mkvtoolnix

[C:\]
=>

Also, fancy "background colors" in a CLI window are annoying :( Don't assume

1) everybody has their monitors (and eyes) calibrated in the same way as yours;

2) everybody sets cmd.exe to use white text on a black background;

At least Mplayer gives me an option to turn OFF :cool: the stupid "feature":

## Fancy Color Messages
msgcolor=no

tebasuna51

27th August 2011, 12:18

madshi, do you have any plans to incorporate qtaacenc.exe, similar to neroaacenc.exe? Piping works fine but it would be pretty handy to have a built in switch to use qtaacenc.

Is not the same than neroaacenc (standalone encoder), to work with qtaacenc you need install QuickTime.

To incorporate a encoder you need support for all parameters of the encoder.

If 'pipe' work you can use any new/not-common parameters without a new eac3to version.
I use the 'pipe' method also for Aften.

laserfan

27th August 2011, 13:11

You are welcome.

Yes, with 'pipe' sintax you can add all Aften parameters.
To know the all the parameters:

Aften -longhelp
I missed that one too--will have to look at longhelp.

Many thanks T51. :)

EDIT: BTW where is the latest binary? I have two copies presently on my C: drive, both of which say "Date modified 1/19/2008 8:51 AM" and 272,896 bytes, and the -version info says only

Aften: A/52 audio encoder
Version SVN
(c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al.

:confused:

nibus

27th August 2011, 13:15

Is not the same than neroaacenc (standalone encoder), to work with qtaacenc you need install QuickTime.

To incorporate a encoder you need support for all parameters of the encoder.

If 'pipe' work you can use any new/not-common parameters without a new eac3to version.
I use the 'pipe' method also for Aften.

Right, I do understand the difference between them. Maybe it would be easier to incorporate QuickTime AAC from within eac3to.

Of course I have had issues with QuickTime and large multichannel files. It usually quits partway through them.

tebasuna51

28th August 2011, 09:40

EDIT: BTW where is the latest binary?
The latest binaries I know:
http://code.google.com/p/wavtoac3encoder/downloads/list

laserfan

28th August 2011, 13:08

The latest binaries I know:
http://code.google.com/p/wavtoac3encoder/downloads/list
Thanks! I'm gonna try the x64 version.

nurbs

28th August 2011, 13:54

nibus

28th August 2011, 14:54

Is there a way to make eac3to able to decode DTS Express mono tracks?
6: DTS Express, English, 1.0 channels, 16 bits, 96kbps, 48kHz
I have Arcsoft 1.1.0.7 installed and that apparently isn't able to cope with such tracks. Would either upgrading the Arcsoft decoder or installing the Nero Decoder which I still have lying around somewhere fix my problem?
If that wouldn't work is there some other software I can use to decode the track to PCM?

Thanks in advance.

I tried Arcsoft 1.1.0.0, 1.1.0.7, and 1.1.0.8 and none of them properly decoded mono dts-express. I believe 1.1.0.8 would say it was decoding, but the output file was just silent and too short. Nero wouldn't even start it.

xkodi

28th August 2011, 14:57

I tried Arcsoft 1.1.0.0, 1.1.0.7, and 1.1.0.8 and none of them properly decoded mono dts-express. I believe 1.1.0.8 would say it was decoding, but the output file was just silent and too short. Nero wouldn't even start it.

if my memory serves well somewhere in this thread there are samples of mono dts-express files, which i've tried before and the conclusion was that you need to use 'Nero' decoder for such files, because only the 'Nero' decoder can decode them correctly.

nurbs

30th August 2011, 11:58

Alright, I checked the Nero decoder and the results were interesting. The output was a 30 second long file where the audio fades in and out. The content of that file is fine, I can understand what the people are saying, but being only 30 seconds long it is of course nowhere near the 2 hours of the source and so not usable.

I also updated Arcsoft to 1.1.0.8 and tried to decode the track and the result plays way too fast, so now I'll try to stretch the track and see if that produces usable results.

If someone has a recommendation for a software where I can slow down the track by an arbitrary factor I would appreciate it. I'll report back if I find a working solution. :thanks:

nurbs

30th August 2011, 13:08

I took several tries, but I managed to find the right factor I needed to stretch the audio and was able align it to the video correctly. Certainly not a pretty solution, but it works if you invest some time in it. I still hope a better solution comes up soon :)

Thunderbolt8

2nd September 2011, 08:42

I have a problem remuxing the south pacific BD (BBC, german BD, 1080i50, 2xDTS-HD Hi-res) to .mkv (video and english DTS-HD Hi-res track only included)

problem seems to be that either the video track then is too short or one audio track too long.

the .m2ts plays fine for its total running time of 1:06:17 with both audio tracks.

but when I remux to .mkv, the length of the video track suddenly becomes 0:49:19 for the English DTS-HD track and the video track, while the runtime of the German DTS-HD track (just as a comparison) still is 1:06:17.
when remuxed, both seem to be in sync despite their short runtime, but the overall runtime of the remux is still 1:06:17 and playback basically stops (if you try to skip a bit forward, otherwise it continues silently without changing of the last frame and without sound) at 0:49:19 and nothing happens any more.

so how come the runtimes of the video and the english DTS-HD track are different now from that of the german track and the runtime of the .m2ts?

(am using arcsoft 1.1.0.1 but not sure if that matters for demuxing only; LAV audio decoder at playback)

edit: the English track is 16-bit and gets patched to 24-bit when being demuxed (the german track already is 24-bit) so maybe thats why the different runtime? but still, I guess that doesnt change why the video is only 49:19 as .mkv and plays till 1:06:17 just fine inside the .m2ts file.

Snowknight26

3rd September 2011, 00:38

eac3to fails to convert audio data-less WAVs:

eac3to v3.24
command line: eac3to.exe C:\temp\test_audacity.wav C:\temp\test_eac3to.wav
------------------------------------------------------------------------------
WAV, 5.1 channels, 16 bits, 4234kbps, 44.1kHz
Reading WAV...
Original audio track: no audio data.
The WAV writer didn't receive the format information. <ERROR>
Aborted at file position 44. <ERROR>

tebasuna51

3rd September 2011, 11:12

And what is the problem?

Richard1485

3rd September 2011, 11:34

I would like to confirm something. I am transcoding an AC-3 file (direct from the source) to another AC-3 file. Eac3to reports a delay of 48ms. I take it that eac3to applied this delay properly even though the file is being slowed down. Here is the relevant portion of the log. (I know that 48ms is only a little more than a frame and is probably not noticeable, but I'd like to know for sure because of the slowdown and in case I am in the same situation with greater audio delays.)

2: AC3, English, 5.1 channels, 384kbps, 48kHz, dialnorm: -27dB, 48ms
[a02] Decoding with libav/ffmpeg...
[a02] Remapping channels...
[a02] Applying RAW/PCM delay...
[a02] Changing FPS from 25.000 to 23.976...
[a02] Encoding AC3 <448kbps> with libAften...
[a02] Creating file "E:\ST.1.TMP\ST.1.The.Motion.Picture.1080p.h264.dd51.ac3"...
[a02] The last (E-)AC3 frame is incomplete and thus gets skipped. <WARNING>

Thunderbolt8

3rd September 2011, 11:37

And what is the problem?this :p I have a problem remuxing the south pacific BD (BBC, german BD, 1080i50, 2xDTS-HD Hi-res) to .mkv (video and english DTS-HD Hi-res track only included)

problem seems to be that either the video track then is too short or one audio track too long.

the .m2ts plays fine for its total running time of 1:06:17 with both audio tracks.

but when I remux to .mkv, the length of the video track suddenly becomes 0:49:19 for the English DTS-HD track and the video track, while the runtime of the German DTS-HD track (just as a comparison) still is 1:06:17.
when remuxed, both seem to be in sync despite their short runtime, but the overall runtime of the remux is still 1:06:17 and playback basically stops (if you try to skip a bit forward, otherwise it continues silently without changing of the last frame and without sound) at 0:49:19 and nothing happens any more.

so how come the runtimes of the video and the english DTS-HD track are different now from that of the german track and the runtime of the .m2ts?

(am using arcsoft 1.1.0.1 but not sure if that matters for demuxing only; LAV audio decoder at playback)

edit: the English track is 16-bit and gets patched to 24-bit when being demuxed (the german track already is 24-bit) so maybe thats why the different runtime? but still, I guess that doesnt change why the video is only 49:19 as .mkv and plays till 1:06:17 just fine inside the .m2ts file.

tebasuna51

3rd September 2011, 15:06

I would like to confirm something. I am transcoding an AC-3 file (direct from the source) to another AC-3 file. Eac3to reports a delay of 48ms. I take it that eac3to applied this delay properly even though the file is being slowed down. Here is the relevant portion of the log. (I know that 48ms is only a little more than a frame and is probably not noticeable, but I'd like to know for sure because of the slowdown and in case I am in the same situation with greater audio delays.)

When you decode the source the audio delay is applied to the uncompressed audio exactly to 48 ms.
After the slowdown is applied, then you have a exact sync.

BTW, if you want mix the audio to other video maybe the delay is different.

Richard1485

3rd September 2011, 15:14

When you decode the source the audio delay is applied to the uncompressed audio exactly to 48 ms. After the slowdown is applied, then you have a exact sync.

BTW, if you want mix the audio to other video maybe the delay is different.

:thanks: That's what I needed to know. (I'm not muxing it to different video: it's the same video with a slowdown. Basically I'm converting to NTSC.) :)

tebasuna51

3rd September 2011, 15:19

this :p

Sorry but I don't understand how are related the two problems.

"eac3to fails to convert audio data-less WAVs"
A WAV without data is only the header and is nothing to convert.

What is the size of 'test_audacity.wav'?
If is greater than a few bytes maybe you have a corrupt header without a DATA chunk.

"edit: the English track is 16-bit and gets patched to 24-bit when being demuxed"
I can't understand. If you only demux the DTS you can't change anything, only DTS-MA have bitdeph, don't exist other DTS 16bits or 24bits. Please put the log when you extract audios from m2ts.

Thunderbolt8

3rd September 2011, 15:25

they arent, but you asked what the problem is so I pointed to my problem :)

tebasuna51

3rd September 2011, 18:55

Then put the eac3to log extracting the audios.

Thunderbolt8

3rd September 2011, 19:43

eac3to v3.24
command line: eac3to 3) D:\E06.mkv
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 1:06:17, 50i
1: Chapters, 10 chapters
2: VC-1, 1080i50 (16:9)
3: DTS Hi-Res, German, 5.1 channels, 24 bits, 2814kbps, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
4: DTS Hi-Res, English, 5.1 channels, 16 bits, 2046kbps, 48kHz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48kHz)
Creating file "D:\E06 - Chapters.txt"...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a04] Extracting audio track number 4...
[a04] Patching bitdepth to 24 bits...
[v02] Muxing video to Matroska...
[a03] Creating file "D:\E06 - 3 - DTS Hi-Res, German, 5.1 channels, 24 bits, 2814kbps, 48kHz.dtshr"...
[a04] Creating file "D:\E06 - 4 - DTS Hi-Res, English, 5.1 channels, 16 bits, 2046kbps, 48kHz.dtshr"...
Added fps value (25) to MKV header.
Video track 2 contains 73977 frames.
eac3to processing took 5 minutes, 10 seconds.
Done.

doesnt seem to help much though, as there is nothing wrong you could see. all I can say now is the runtime of the video .mkv file is 49:19, same for the german audio track and the english audio track is 1:06:17.

funny thing is when muxing english track and video together, the final file still has a runtime of 1:06:17 (as the english track), but video finishes at 49:19 and still is completely in sync with the audio. still, there is then ~15 minutes of nothing at the end of the file (still last frame and no sound)

tebasuna51

3rd September 2011, 22:37

Yes, don't help too much.
If you extract the 'core' work fine the mkv?
Maybe a problem with DTS-HR and mkv, I never test this.

Thunderbolt8

3rd September 2011, 23:27

its the same. but anyway, that wouldnt explain why the runtime of the video is now shorter as well (as said, the .m2ts play fine till the end of 1:0617)

its strange that eng track has a longer runtime on the one hand, but is as long as the video when both are remuxed together. is there some form of speedup involved?

me7

4th September 2011, 12:03

I installed the trail version of the latest ArcSoft TotalMedia Theater to get access to their DTS-MA decoder, but eac3to fails to find the decoder. Do I need some additional configuration?

73ChargerFan

5th September 2011, 00:05

[code]eac3to v3.24
command line: eac3to 3) D:\E06.mkv
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 1:06:17, 50i
1: Chapters, 10 chapters
2: VC-1, 1080i50 (16:9)

Perhaps try using tsMuxeR to demux the m2ts into elementary streams first. I had difficulty remuxing 1080i60 extras from a bd a year ago, and madshi told me eac3to sometimes didn't work with those sources.

nibus

5th September 2011, 07:07

I installed the trail version of the latest ArcSoft TotalMedia Theater to get access to their DTS-MA decoder, but eac3to fails to find the decoder. Do I need some additional configuration?

try this:

http://forum.doom9.org/showthread.php?p=1398577#post1398577

Thunderbolt8

5th September 2011, 10:53

Perhaps try using tsMuxeR to demux the m2ts into elementary streams first. I had difficulty remuxing 1080i60 extras from a bd a year ago, and madshi told me eac3to sometimes didn't work with those sources.hm I guess I'll keep it as .m2ts then for now. dont like to output as elementary stream for some reason.

me7

5th September 2011, 17:12

try this:

http://forum.doom9.org/showthread.php?p=1398577#post1398577

Doesn't work with version 1.1.0.8 (the one I have). I get the following log output when I try to decode a 6.1 DTS-HD track:

[a03] The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
[a03] The libav DTS decoder doesn't decode the back channels. <WARNING>
[a03] The libav DTS decoder doesn't decode the full DTS-HD information. <WARNING>
[a03] Extracting audio track number 3...
[a03] Extracting DTS core...
[a03] Decoding with libav/ffmpeg...
[a03] The AC3 encoder received a non-supported data format (float, 7, 64, -). <ERROR>
Aborted at file position 1048576. <ERROR>

Since 1.1.0.8 supposedly can't handle 6.1 DTS it could be related that bug.

Is there any other way to decode 6.1 DTS?

Thunderbolt8

5th September 2011, 17:44

try 1.1.0.1

tebasuna51

5th September 2011, 21:17

Doesn't work with version 1.1.0.8 (the one I have). I get the following log output when I try to decode a 6.1 DTS-HD track:
...

Please put the full log.
We need know the command line.

me7

5th September 2011, 21:44

Doesn't seem related to eac3to specifically, I can't get the decoder to work with Media Player Classic Home Cinema either.
Is this really the only way to access a 6.1 DTS stream?

asarian

5th September 2011, 23:11

Getting the following error trying to convert 6.1 channel DTS-MA to LPCM:

a03 Blu-Ray PCM doesn't support 6.1 channels. Will upconvert to 7.1.

This is, in itself, normal, except that I had a commandline like this:

"eac3to 1) 3: c:\video\temp.pcm -down6"

Since I'm downcoverting to 5.1 anyway, it seems wiser for eac3to to do so directly, instead of upconverting to 7.1 first.

tebasuna51

6th September 2011, 11:01

"eac3to 1) 3: c:\video\temp.pcm -down6"

Yes I can confirm the bug, work with other output but not with pcm. But isn't the problem, the problem is:

One more time, the eac3to -down6 with 6.1 sources don't work fine, the surround channels have a bad mix.

You need:
eac3to input_6.1 output.wavs

and make the mix manually:

SL' = SL + 0.7071 x BC
SR' = SR + 0.7071 x BC

Thunderbolt8

6th September 2011, 12:20

is there actually any list of bugs which eac3to currently has with 3.24? and maybe also a list of known limitations in regards to other tools like arcsoft etc. (apart from those mentioned at 1st page)?
would be nice if someone with an overview on these things could create such a list, I myself always keep losing track of such things.

might be easier then just to refer back to a certain post which can be kept up to date.

asarian

6th September 2011, 12:36

Yes I can confirm the bug, work with other output but not with pcm. But isn't the problem, the problem is:

One more time, the eac3to -down6 with 6.1 sources don't work fine, the surround channels have a bad mix.

You need:
eac3to input_6.1 output.wavs

and make the mix manually:

SL' = SL + 0.7071 x BC
SR' = SR + 0.7071 x BC

So, I guess I was 'lucky' then, in my case of pcm, that it upconverted to 7.1 first, mix-wise? (assuming downconvert from 7.1 goes fine too).

tebasuna51

7th September 2011, 00:15

So, I guess I was 'lucky' then, in my case of pcm, that it upconverted to 7.1 first, mix-wise? (assuming downconvert from 7.1 goes fine too).

Sorry but the upmix to 7.1 is also wrong because make:

BL' = 0.7 x SR
BR' = 0.7 x SR
SL' = BC
SR' = SL

and after, if you use the -down6, obtain:

SL'' = BC + 0.7 x SR
SR'' = SL + 0.7 x SR

tebasuna51

7th September 2011, 00:22

To avoid the bug we can use a fake remap like this:

eac3to input_6.1 output_5.1 -0,1,2,3,5,6,4 -down6

where input_6.1 can be a DTS 6.1 track, and output_5.1 can be ac3, wav, w64 ...

Edit:
This remap is also needed to upmix 6.1 to 7.1 with the -double7 parameter:

eac3to input_6.1 output_7.1 -0,1,2,3,5,6,4 -double7

asarian

7th September 2011, 00:30

To avoid the bug we can use a fake remap like this:

eac3to input_6.1 output_5.1 -0,1,2,3,5,6,4 -down6

where input_6.1 can be a DTS 6.1 track, and output_5.1 can be ac3, wav, w64 ...

Thanks. Appreciate the help and commandline. :)

nibus

8th September 2011, 08:31

Yes I can confirm the bug, work with other output but not with pcm. But isn't the problem, the problem is:

One more time, the eac3to -down6 with 6.1 sources don't work fine, the surround channels have a bad mix.

You need:
eac3to input_6.1 output.wavs

and make the mix manually:

SL' = SL + 0.7071 x BC
SR' = SR + 0.7071 x BC

tebasuna if I pipe eac3to to another encoder (qaac for example) and set it to -normalize (creating an intermediate wav) and -down6, does this mean 6.1 DTS sources will have a wrong surround channel mix? or is it only when using the .pcm extension? Or is it literally every time you downmix 6.1 to 5.1, no matter what decoder and what output?

tebasuna51

8th September 2011, 10:37

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

nibus

8th September 2011, 10:42

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

Thanks!

One more question - I use a batch file for my conversions - would -0,1,2,3,5,6,4 -down6 work for 5.1 sources as well?

edit: Could I use the Sonic Decoder for 6.1 without problems?

tebasuna51

8th September 2011, 13:43

One more question - I use a batch file for my conversions - would -0,1,2,3,5,6,4 -down6 work for 5.1 sources as well?
No, this remap don't work with 5.1

edit: Could I use the Sonic Decoder for 6.1 without problems?
Sorry, I can't test Sonic Decoder.
I read that work fine, but also need the remap if use -down6 because is needed also with wav files.

shh

11th September 2011, 13:50

Since jruggle sais libav is now better than aften (quote: http://forum.doom9.org/showthread.php?p=1522077#post1522077),
how is it possible to use libav with eac3to for AC3-encoding?

nurbs

11th September 2011, 14:22

Either let eac3to save a wav and reencode that with libav or pipe the output of eac3to to libav. You can do the latter by adapting this:
eac3to.exe "C:\Movie" 1) 3:stdout.wav | qaac.exe --tvbr 100 --quality 2 --rate keep --no-optimize -o "output.aac" -
You'll have to adapt that for libav, but the most inportant thing is to tell eac3to to output to stdout and to tell the encoder to read from stdin (which works by using "-" as the input in qtaac).

shh

11th September 2011, 14:49

Thanks!
Is there any chance that madshi sometimes adds a -encodeWithLibav switch?

asarian

11th September 2011, 19:57

To avoid the bug we can use a fake remap like this:

eac3to input_6.1 output_5.1 -0,1,2,3,5,6,4 -down6

where input_6.1 can be a DTS 6.1 track, and output_5.1 can be ac3, wav, w64 ...

Just tried this on a DTS-MA 6.1 source, and eac3to still upconverts to 7.1 first (because I remap to pcm). I guess I should map to wav then first, right? (that obviously doesn't trigger intermediate upscale phase).

tebasuna51

12th September 2011, 01:33

Just tried this on a DTS-MA 6.1 source, and eac3to still upconverts to 7.1 first (because I remap to pcm). I guess I should map to wav then first, right?

Yes, but one question, for what you need pcm instead WAV ot W64?

asarian

12th September 2011, 04:17

Yes, but one question, for what you need pcm instead WAV ot W64?

Thanks for the reply.

I always convert DTS-MA/TrueHD to LPCM, as I use my PS3 as media server, and it can't stream DTS audio; so, for my HD audio needs I require a LPCM track.

Thunderbolt8

13th September 2011, 14:58

how is it possible to slow down video and audio of a 50p input file (the source is 720p50)? fps rates of video and audio get changed to 23.976 fps instead of 48p, and of course the output files have ~double the runtime of the input file and sound is ultra slow.

edit: -ChangeTo47.952 works

skinnyq

13th September 2011, 18:46

would anyone help with me neroencaac? I always get that god damn ERROR: could not open AAC encoder.
Full log:
D:\Fraps\Movies>eac3to\eac3to 1.wav 1.aac
WAV, 2.0 channels, 0:00:47, 16 bits, 6144kbps, 192kHz
Reading WAV...
Encoding AAC <0.50> with NeroAacEnc...
The Nero AAC encoder aborted processing.
NeroAacEnc ERROR: could not open AAC encoder
Aborted at file position 262144.

ramicio

13th September 2011, 18:47

AAC supports 192 khz?

Midzuki

13th September 2011, 19:02

AAC supports 192 khz?

Perhaps — BUT if the encoder/decoder was not designed to deal with that, then the answer is "no", of course -.-

Centuries ago :) , I copied the info shown below from the MultimediaWiki (http://wiki.multimedia.cx/index.php?title=MP4A):

There are 13 supported frequencies:

* 0: 96000 Hz
* 1: 88200 Hz
* 2: 64000 Hz
* 3: 48000 Hz
* 4: 44100 Hz
* 5: 32000 Hz
* 6: 24000 Hz
* 7: 22050 Hz
* 8: 16000 Hz
* 9: 12000 Hz
* 10: 11025 Hz
* 11: 8000 Hz
* 12: 7350 Hz
* 13: Reserved
* 14: Reserved
* 15: frequency is written explicitly ?????

/////////////////////////////////////

Channel Configurations

These are the channel configurations:

* 0: Defined in AOT Specifc Config
* 1: 1 channel: front-center [1.0]
* 2: 2 channels: front-left, front-right [2.0]
* 3: 3 channels: front-center, front-left, front-right [3.0]
* 4: 4 channels: front-center, front-left, front-right, back-center [3/1]

* 5: 5 channels:
front-center, front-left, front-right, back-left, back-right [3/2]

* 6: 6 channels:
front-center, front-left, front-right, back-left,
back-right, LFE-channel [3/2/.1]

* 7: 8 channels:
front-center, front-left, front-right, side-left,
side-right, back-left, back-right, LFE-channel [3/2/2/.1]

*
* 8-15: Reserved

Turtleggjp

16th September 2011, 03:46

asarian

16th September 2011, 04:15

If you play a 7.1 track on a 5.1 speaker setup (or if the studio downmixes 7.1 to 5.1) the 5.1 surround channel pair usually contains a mix of the 7.1 SR/SR and BL/BR channel pairs.

Sat wha?! All my life I've been manually downmixing 7.1 tracks to 5.1, thinking I'd otherwise lose 2 channels on playback; and you're saying that isn't even necessary? That's interesting, if the case.

If so, will a 7.1 LPCM track play as 5.1 (without losing channel info) as well?

Thanks.

nibus

16th September 2011, 12:10

Just ran across this message while processing the Thor Blu Ray:

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.
Decoding with ArcSoft DTS Decoder...
Encoding FLAC with libFlac...

Searching this thread, I found a good explanation by madshi here (http://forum.doom9.org/showthread.php?p=1216107#post1216107), but the last mention of the message was also dated back in 2008. The version of Arcsoft I'm using is quite old (like from around 2008), since I only use it for decoding DTS Master Audio files with EAC3To, so I had no reason to update it.

So before I go getting a new version of Arcsoft, can anyone tell me if they fixed this little issue in later versions? Or worse, have they locked out their filters from third party programs like Nero does/did? Thanks.

Matt

AFIAK 1.1.0.0, 1.1.0.7 and 1.1.0.8 all have the same problem. The only solution is to use the Sonic decoder, but it only supports 5.1 so you will have to downmix if you have a 6.1 or 7.1 source. If you are downmixing from 6.1, you will need to use this in the command line as per tebasuna51's instructions:

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

asarian

16th September 2011, 12:32

AFIAK 1.1.0.0, 1.1.0.7 and 1.1.0.8 all have the same problem.

Personally, I can't say this 'problem' bothers me a lot. In fact, I hardly notice it (and probably never would have if I hadn't known about it); and if I do, then I just tap the volume on my amp twice to up the volume a bit. I mean to say, don't think you're going to get an extremely low volume.

The way it stands, most of my Blu-Rays play at different volume levels.

Thunderbolt8

16th September 2011, 17:19

AFIAK 1.1.0.0, 1.1.0.7 and 1.1.0.8 all have the same problem. The only solution is to use the Sonic decoder, but it only supports 5.1 so you will have to downmix if you have a 6.1 or 7.1 source. If you are downmixing from 6.1, you will need to use this in the command line as per tebasuna51's instructions:xkodi found out that the sonic decoder decoder drops the XX channel extension and produces a bit-perfect correct 5.1 automatically in case of strange setup(!) 7.1 DTS-HD MA tracks which includes all the channel information and nothing gets dropped. so there should be no need to downmix to 5.1

http://forum.doom9.org/showthread.php?p=1443085#post1443085

there was also a post with a comparison of different arcsoft versions somewhere, stating that 1.1.0.1 might be the most favourable version to use, but I cant find it atm

Turtleggjp

16th September 2011, 20:21

Thanks for the tips guys. I played my FLAC version last night and didn't notice much drop in volume compared to the DTS Core file I also extracted. I guess if you want it to be truly lossless, the best thing to do is just bitstream the original DTS-MA soundtrack to a receiver. Since I don't have one of those yet, I'll just keep converting to FLAC.

You guys mentioned version numbers like 1.1.0.0, and I think Arcsoft is up to version 5. I'm guessing you're referring to the version of the DTS decoding filter. How can I check what version mine is? Thanks.

Matt

Thunderbolt8

16th September 2011, 21:08

eac3to -test

asarian

16th September 2011, 23:01

there was also a post with a comparison of different arcsoft versions somewhere, stating that 1.1.0.1 might be the most favourable version to use, but I cant find it atm

Hmm, I seem to be having 1.1.0.0 installed. :( Can someone please tell me (with certainty) whether this version is okay? Or what other version to use then?

Also, are you people now saying Sonic is better? If so, what version?

Thanks.

nibus

17th September 2011, 00:33

Hmm, I seem to be having 1.1.0.0 installed. :( Can someone please tell me (with certainty) whether this version is okay? Or what other version to use then?

Also, are you people now saying Sonic is better? If so, what version?

Thanks.

There are some differing opinions as to which versions decode properly.

Personally I use 1.1.0.0 for 5.1 and 6.1 DTS-MA, and I use 1.1.0.8 for 7.1 DTS-MA (except strange setup). This is because of this post:

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.

However, I believe tebasuna51 has stated that he hasn't had any problems with non-standard 7.1 sources and Arcsoft 1.1.0.0.... So I don't know what to tell you. I use the different versions just to be safe.

For 5.1 content Sonic is identical to Arcsoft AFAIK, but it automatically downmixes 7.1 content to 5.1, and doesn't work on some 6.1 content. For 7.1 Strange setup you will want to use Sonic anyway because like Thunderbolt8 said, Arcsoft doesn't decode it quite right.

The Sonic DTS decoder is very good for DTS, DTS-ES, DTS-96/24, DTS-HD Master Audio and DTS-HD High Resolution tracks. The only problem is that it decodes DTS-HD 7.1 tracks only as 5.1. Most DTS-ES and DTS-HD Master Audio 6.1 tracks are fully decoded as 6.1, though.

asarian

17th September 2011, 00:57

There are some differing opinions as to which versions decode properly.

Personally I use 1.1.0.0 for 5.1 and 6.1 DTS-MA, and I use 1.1.0.8 for 7.1 DTS-MA (except strange setup). This is because of this post:

However, I believe tebasuna51 has stated that he hasn't had any problems with non-standard 7.1 sources and Arcsoft 1.1.0.0.... So I don't know what to tell you. I use the different versions just to be safe.

For 5.1 content Sonic is identical to Arcsoft AFAIK, but it automatically downmixes 7.1 content to 5.1, and doesn't work on some 6.1 content. So if you mix everything down to 5.1 with eac3to, then Sonic should be fine. And in the case of downmixing 6.1 to 5.1 Sonic works properly even without specifying the correct channel map, as stated 4 posts above.

For 7.1 Strange setup you will want to use Sonic anyway because like Thunderbolt8 said, Arcsoft doesn't decode it quite right.

Thanks for the very illuminating clarification! :thanks:

Thunderbolt8

17th September 2011, 04:28

this also http://forum.doom9.org/showthread.php?p=1508578#post1508578 & http://forum.doom9.org/showthread.php?p=1508746#post1508746

And in the case of downmixing 6.1 to 5.1 Sonic works properly even without specifying the correct channel map, as stated 4 posts above.afaik that was for 7.1 DTS-HD MA strange setup and not for 6.1. for 6.1, you'd need to remap channel according to what tebasuna51 said

nibus

17th September 2011, 08:05

this also http://forum.doom9.org/showthread.php?p=1508578#post1508578 & http://forum.doom9.org/showthread.php?p=1508746#post1508746

afaik that was for 7.1 DTS-HD MA strange setup and not for 6.1. for 6.1, you'd need to remap channel according to what tebasuna51 said

ahh, fix'd.

Boulder

17th September 2011, 09:45

Could someone please create a dedicated page for these many different cases for example to the Avisynth wiki?

hdx264

18th September 2011, 06:02

does eac3to decode and reencode the DTS-HD MA track at all when you are just demuxing it or does it just extract a 1 to 1 copy of the track without doing anything to it?

so i guess I'm asking do you need any DTS-HD MA decoders installed if i just want to extract the DTS-HD MA track and not change it at all

nevcairiel

18th September 2011, 08:19

demuxing works fine without any decoder.

mzso

21st September 2011, 13:12

Hi!
I get a "The ArcSoft and Sonic decoders don't seem to work, will use libav instead." message, but I don't know why. Total media theatre is installed and working fine, I even use the very same decoder in combinations with LAV filters to decode lossless when I play movies. Why would this happen?

Midzuki

21st September 2011, 16:20

@ mzso:

Try to include the "relevant" ArcSoft codec folder in the PATH environment variable.

mzso

21st September 2011, 19:03

@ mzso:

Try to include the "relevant" ArcSoft codec folder in the PATH environment variable.

No use. I actually tried to drop the decoder.dll formerly that didn't work either. Also it says "don't seem to work" and not that it doesn't find it. But why doesn't it work? And what "don't seem to work" actually means?

jmonier

21st September 2011, 19:10

I believe that you may have to register the ARCsoft filter. I don't think the ARCsoft installation does this and I know that the LAV Audio filter handles it a different way and doesn't need registration.

I think there's more information in the Wiki (see first post). If not, search the thread.

nibus

21st September 2011, 21:54

No use. I actually tried to drop the decoder.dll formerly that didn't work either. Also it says "don't seem to work" and not that it doesn't find it. But why doesn't it work? And what "don't seem to work" actually means?

1) Copy following files from ArcSoft Total Media Theater 2 to Windows\system32

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

put checkactivate.dll in \system32 folder as well as eac3to folder

2) Register ASAudioHD.ax -

regsvr32.exe ASAudioHD.ax

me7

22nd September 2011, 16:27

1) Copy following files from ArcSoft Total Media Theater 3.0 to Windows\system32

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

put checkactivate.dll in \system32 folder as well as eac3to folder

2) Register ASAudioHD.ax -

regsvr32.exe ASAudioHD.ax

I don't seem to have a checkactivate.dll in my TMT 3 installation:confused:

heerschop

22nd September 2011, 20:54

I don't seem to have a checkactivate.dll in my TMT 3 installation:confused:

That's write. The checkactivate.dll was installed with previous version of TMT. The checkactivate.dll is only needed for
eac3to, TMT 3 doesn't need it to function. Eac3to check's if this file is installed, if not then eac3to will not use TMT.
Get hold of an older version of TMT (I think version 2) and extract the checkactivate.dll

mzso

22nd September 2011, 22:14

What about TMT 5?

asarian

23rd September 2011, 15:42

Is there already a solution to the below error with TrueHD? This is from the Gantz Blu-Ray.

eac3to 00000.m2ts 3: c:\video\temp.pcm

M2TS, 1 video track, 1 audio track, 2 subtitle tracks, 2:10:40, 24p /1.001
1: Chapters, 17 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, Japanese, 5.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
4: Subtitle (PGS), Japanese
a03 Extracting audio track number 3...
a03 Extracting TrueHD stream...
a03 Decoding with libav/ffmpeg...
a03 Swapping endian...
a03 Remapping channels...
a03 libav Substream 0 parity check failed
a03 libav Substream 0 checksum failed
a03 libav Substream 0 length mismatch.
a03 The libav decoder reported error -1 while decoding.
Aborted at file position 1048576.

Thunderbolt8

23rd September 2011, 15:45

try the -nero switch (nero is limited to 5.1 channels, but this is not a problem here)

asarian

23rd September 2011, 16:01

try the -nero switch (nero is limited to 5.1 channels, but this is not a problem here)

Thanks. I don't have Nero 7 any more, though, since my migration to Windows 7. I'll contact their support desk.

Rodeo

25th September 2011, 04:15

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

First of all, thanks for figuring this out!

I have a few questions (mostly just to make sure I understand it correctly).

Assuming I have an MKV source with a DTS-HD Master Audio 6.1 track:

1: h264/AVC, 1920x1080 23.976p
2: DTS Master Audio, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz)

Let's say I want to convert it to FLAC 5.1. I can do it in a single step, like this:

eac3to.exe input.mkv 2: output.flac -0,1,2,3,5,6,4 -down6

Right? And this should give me a good downmix?

And if I also want a stereo downmix, I should do this:

eac3to.exe input.mkv 2: output_stereo.flac -0,1,2,3,5,6,4 -down2

or maybe this:

eac3to.exe input.mkv 2: output_stereo.flac -0,1,2,3,5,6,4 -down6 -down2

but not this:

eac3to.exe input.mkv 2: output_stereo.flac -down2

Right?

Finally, if I reduce the bit depth from 24- to 16-bit, I get the following in my eac3to log:

eac3to v3.24
command line: eac3to.exe input.mkv output.flac -0,1,2,3,5,6,4 -down6 -down16
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 0:06:15, 24p /1.001
1: h264/AVC, 1920x1080 23.976p
2: DTS Master Audio, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz)
Track 2 is used for destination file output.flac.
[a02] Extracting audio track number 2...
[a02] Remapping channels...
[a02] Decoding with ArcSoft DTS Decoder...
[a02] Remapping channels...
[a02] Mixing surround channels...
[a02] Reducing depth from 24 to 16 bits...
[a02] Encoding FLAC with libFlac...
[a02] Creating file output.flac...
[a02] The original audio track has a constant bit depth of 24 bits.
[a02] Processed audio track, L+R+C+SL+SR: constant bit depth of 16 bits.
[a02] Processed audio track, LFE: no audio data.
Video track 1 contains 8997 frames.
eac3to processing took 54 seconds.
Done.

I'm worried about this:

[a02] Processed audio track, LFE: no audio data.

It's not there if I keep the original bit depth:

eac3to v3.24
command line: eac3to.exe input.mkv output.flac -0,1,2,3,5,6,4 -down6
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 0:06:15, 24p /1.001
1: h264/AVC, 1920x1080 23.976p
2: DTS Master Audio, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz)
Track 2 is used for destination file output.flac.
[a02] Extracting audio track number 2...
[a02] Remapping channels...
[a02] Decoding with ArcSoft DTS Decoder...
[a02] Remapping channels...
[a02] Mixing surround channels...
[a02] Encoding FLAC with libFlac...
[a02] Creating file output.flac...
[a02] The original audio track has a constant bit depth of 24 bits.
[a02] The processed audio track has a constant bit depth of 24 bits.
Video track 1 contains 8997 frames.
eac3to processing took 1 minute, 8 seconds.
Done.

Is it a problem or should I just ignore it?

Note: it's only a 6-minute sample - I don't see the message if I convert the 2-hour source to FLAC, even with -down16.
My guess/hope is that maybe the LFE is indeed empty in the sample, but for some reason eac3to doesn't mention it if I don't specify -down16.

tebasuna51

25th September 2011, 09:46

- Yes

- Use this one, the -normalize is recommended when use -down2 :

eac3to.exe input.mkv 2: output_stereo.flac -0,1,2,3,5,6,4 -down2 -normalize

- Don't worry about the LFE message, appears only when the audio is analyzed to change the bitdepth.

Rodeo

25th September 2011, 11:17

Awesome! Thanks.

mzso

25th September 2011, 17:06

That's write. The checkactivate.dll was installed with previous version of TMT. The checkactivate.dll is only needed for
eac3to, TMT 3 doesn't need it to function. Eac3to check's if this file is installed, if not then eac3to will not use TMT.
Get hold of an older version of TMT (I think version 2) and extract the checkactivate.dll

That sucks. It wasn't updated since TMT2?

By the way why does eac3to automatically encode the dts core to flac if it fails to use the arcsoft decoder? Isn't that like universally condemned?

mzso

25th September 2011, 17:18

1) Copy following files from ArcSoft Total Media Theater 2 to Windows\system32

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

put checkactivate.dll in \system32 folder as well as eac3to folder

2) Register ASAudioHD.ax -

regsvr32.exe ASAudioHD.ax

Played around with it a bit. It seems that it works with TMT5 files too. And it turns out only the following are needed:
checkactivate.dll
ASAudioHD.ax
dtsdecoderdll.dll
MagPCMac.dll
MagUIEngine.dll
MagUIInter.dll
Unfortunately though the checkactivate.dll isn't in TMT5 anymore so you still have to get that separately.

rapscallion

26th September 2011, 21:19

I'm extracting to wavs, a Dolby True HD 7.1 track, using eac3to_more and libav/ffmeg TrueHD decoder.

Does libav/ffmpeg remove dialogue normalization by default ?

When I extrac 5.1 DTHD, I use the Nero decoder and notice that the message always says " removing DN" .
No such message when using libav/ffmpeg.

Edit: a quote from ACrowley, post #9353 : "Im wondering that i cant see the "removing Dialog Normalization" Message anymore when decoding A(E)C3,TrueHD ?
I think because the DialNom removal is a standard Operation and latest eac3to wont output that Message everytime? " There never was an answer, altough one responder stated he saw the msg when extracting to ac3.

jruggle

27th September 2011, 14:06

I'm extracting to wavs, a Dolby True HD 7.1 track, using eac3to_more and libav/ffmeg TrueHD decoder.

Does libav/ffmpeg remove dialogue normalization by default ?

short answer, yes. libav does not apply the dialogue normalization to the decoded output. i also suspect eac3to does not have its own mechanism to extract/apply it.

kws53

27th September 2011, 17:17

Demuxed a 96kbps, 24 bit THD stream using eac3to. When converting to DTS, Surcode [1.0.29.0] has indicated that it can't handle the sampling rate.

I used the following to try to convert the THD stream to 48kbps:

>eac3to "00000 - 4 - TrueHD+AC3, English, 5.1 Channels, 96kHz.thd+ac3" "dion.thd+ac3" -resampleTo48000

TrueHD/AC3, 5.1 channels, 96kHz
<embedded: AC3, 5.1 channels, 448kbps, 48kHz>
Creating file "dion.thd+ac3"...

The conversion took 7minutes and essentially copied over the file. No sampling rate conversion was done. The file is still at 96kbps.

Any ideas or help?

Thanks,
Kurt

Thunderbolt8

27th September 2011, 17:28

is the file just named truehd+ac3 or is it really truehd+ac3? afaik eac3to can only demux single thd or single ac3? or has that been changed?

nevcairiel

27th September 2011, 17:51

You can't resample while still keeping it TrueHD. You would need to make it output WAV or FLAC, then it should be able to resample.

kws53

27th September 2011, 21:19

is the file just named truehd+ac3 or is it really truehd+ac3? afaik eac3to can only demux single thd or single ac3? or has that been changed?

This is the name that eac3to gave the file - it is a THD file with an embedded AC3 "core". I've run a separate conversion to AC3 and to FLAC without any problems, so the file format is supported by eac3to.

And according to the WiKi, I should be able to change sampling rate on any audio format that eac3to supports.

My objective is to get it into DTS format. I do not want to lose the fidelity by converting to AC3 first.

Kurt

kws53

27th September 2011, 21:21

You can't resample while still keeping it TrueHD. You would need to make it output WAV or FLAC, then it should be able to resample.

So I can convert to FLAC [already done], then resample down to 48kHz and then use the Surcode codec to convert to DTS?

Sounds like a plan if this is true.

Kurt

kws53

27th September 2011, 21:32

Played around with it a bit. It seems that it works with TMT5 files too. And it turns out only the following are needed:
checkactivate.dll
ASAudioHD.ax
dtsdecoderdll.dll
MagPCMac.dll
MagUIEngine.dll
MagUIInter.dll
Unfortunately though the checkactivate.dll isn't in TMT5 anymore so you still have to get that separately.

How do you "deregister" an older version of codec? I'm trying to upgrade my 1.0.29.0 version to 1.1.10.0 and the original keeps coming back...

I've replaced all of the appropriate files with new versions and run regsvr, but to no avail. I had NO problem installing to a virgin system. By the way, you do not need to put anything in the system32 folder - this is incorrect information.

Kurt

kws53

27th September 2011, 21:34

How do you "deregister" an older version of codec? I'm trying to upgrade my 1.0.29.0 version to 1.1.10.0 and the original keeps coming back...

I've replaced all of the appropriate files with new versions and run regsvr, but to no avail. I had NO problem installing to a virgin system. By the way, you do not need to put anything in the system32 folder - this is incorrect information.

Kurt

Sorry - I'm referring to the Surcode DTS encoder.

Kurt

rapscallion

27th September 2011, 21:36

short answer, yes. libav does not apply the dialogue normalization to the decoded output. i also suspect eac3to does not have its own mechanism to extract/apply it.
Thanks !

nibus

28th September 2011, 00:08

How do you "deregister" an older version of codec? I'm trying to upgrade my 1.0.29.0 version to 1.1.10.0 and the original keeps coming back...

I've replaced all of the appropriate files with new versions and run regsvr, but to no avail. I had NO problem installing to a virgin system. By the way, you do not need to put anything in the system32 folder - this is incorrect information.

Kurt

regsvr32 /u filename.dll

You don't have to put the files in System32 as long as you have them in the system path somewhere.

tebasuna51

28th September 2011, 02:12

Demuxed a 96kbps, 24 bit THD stream using eac3to. When converting to DTS, Surcode [1.0.29.0] has indicated that it can't handle the sampling rate.

I used the following to try to convert the THD stream to 48kbps:

>eac3to "00000 - 4 - TrueHD+AC3, English, 5.1 Channels, 96kHz.thd+ac3" "dion.thd+ac3" -resampleTo48000

Use:
eac3to "00000 - 4 - TrueHD+AC3, English, 5.1 Channels, 96kHz.thd+ac3" "dion.wavs" -resampleTo48000

And the wav's can be used like input for your DTS encoder.

Anakunda

28th September 2011, 22:24

Hello, how can I extrct video stream from m2ts? It's of format MPEG-4 AVC. It can't be to matroska, so extract to bare video track. But what extension to give it? I've tried .m4v and .avc but eac3to failed

This is the video track I need to extract:

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 : 2 frames
Codec ID : 27
Duration : 1h 34mn
Bit rate mode : Variable
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

eac3to "I:\video\BDMV\PLAYLIST\" 1) 2: I:\video\BDMV\bluray.mkv

gives a freeze on 1% of MKV creation (perhaps encryption or some bug)
what else format I can use to extract this track and what output extensions are supported by eac3to? Thank U!

Doom9User123

29th September 2011, 00:43

First, let me just commend the author of this wonderful program for all of the time and effort spent. This program has really simplified my life!

Does anyone know if it's possible to use this program to extract audio/video streams w/o producing a log file in the process?

nibus

29th September 2011, 02:13

looney

29th September 2011, 17:22

How to keep decoded wav that eac3to temporarily produce in dts transcoding 1536k->768k (libav) and after manually encode it in dts? Original audio track is 5.0 only and mixed-up channels and Surcode doesnt support that input (at least i dont know how), and decoded wavs are automatically deleted by eac3to when Surcode fails to encode it. I would rather convert it manually in Surcode.

tebasuna51

29th September 2011, 21:30

eac3to input.dts output.wavs

me7

30th September 2011, 14:42

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

What about 7.1 sources? Do I need to remap the channels as well?

tebasuna51

30th September 2011, 17:46

What about 7.1 sources? Do I need to remap the channels as well?
Nope, downmix 7.1 -> 5.1 work fine.

rapscallion

30th September 2011, 19:16

That raises another questiom. I notice when I extract Dolby THD 7.1 track to wavs, that there is a msg displayed in the dialog box "remapping channels".

Why does it do that and will the resulting wavs be incorrect when I enter them into the DTS encoder ?

tebasuna51

1st October 2011, 01:37

The remap is between the internal channel order (different in each format: ac3, dts, ogg, aac, ...) and the WAV order.
Don't worry about the message.

rapscallion

1st October 2011, 04:29

Thanks tebasuna, I was interpreting as remapping their positions and it kind of threw me even though I've seen the msg before and never paid attention to it.

But today, all of a sudden, I had an epithany "huh" ?

7ekno

1st October 2011, 07:17

Hi all :)

Just after a quick clarification!

With this command:
eac3to -input.dtshd -output.ac3 -down6 -down16 -resampleTo48000 -640 -normalize -libav

Does the "-libav" on the end force decoding of the DTS-MA via "libav", or does it force the encoding of "output.ac3" to be done via "libav" ...

Just trying to get the best quality AC3 as possible and I believe the AC3 encoder in libav is much more actively and recently updated ...

Would it be better to start with an MKV, extract the audio and specify AC3 and "-libav" in the extraction (will this force encoding via libav since MKV can't be decoded via libav)? Like:

eac3to -input.mkv -output.ac3 -down6 -down16 -resampleTo48000 -640 -normalize -libav

Thanks for anybody that can help,
7ek

the_weirdo

1st October 2011, 08:58

With this command:
eac3to -input.dtshd -output.ac3 -down6 -down16 -resampleTo48000 -640 -normalize -libav

Does the "-libav" on the end force decoding of the DTS-MA via "libav", or does it force the encoding of "output.ac3" to be done via "libav" ...
Decoding. However, libav doesn't support decoding DTS-MA at all (at least for now), it just fallback to DTS core.

Just trying to get the best quality AC3 as possible and I believe the AC3 encoder in libav is much more actively and recently updated ...
eac3to currently doesn't support encoding AC-3 via libav directly. Even it does, libav came with eac3to is old. You can, however, pipe decoded data from eac3to to Libav/FFmpeg executable binaries, for example:
eac3to -input.dtshd stdout.wav -down6 -down16 -resampleTo48000 -normalize | ffmpeg -i pipe:0 -ab 640k output.ac3

tebasuna51

1st October 2011, 09:24

eac3to -input.dtshd -output.ac3 -down6 -down16 -resampleTo48000 -640 -normalize (-libav)

Just trying to get the best quality AC3 as possible ...
Then, for what -down16?

7ekno

1st October 2011, 11:25

Thanks guys,

Was mislead from the Wiki entry (http://en.wikibooks.org/wiki/Eac3to/How_to_Use):

Convert a DTS track to a AC3 one, using libav encoder:
eac3to.exe input.dts output.ac3 -libav

Great suggestion with the pipe ;)

As for 16 bit, I was under the impression that was standard for AC3 encoder, love to be corrected if that is indeed wrong :)

7ek

tebasuna51

1st October 2011, 15:14

As for 16 bit, I was under the impression that was standard for AC3 encoder, love to be corrected if that is indeed wrong

Yes is wrong, don't limit the precission at all.

Rodeo

4th October 2011, 02:06

So, I was making sure that channel assignments and/or downmixing worked well, and tested eac3to on a bunch of channel checks I had available.

The FFmpeg TrueHD decoder was swapping the side and back surround channels. That was fixed recently: http://git.videolan.org/?p=ffmpeg.git;a=commitdiff;h=6daf513

Given that eac3to 3.24 is much older than that fix, it should have been affected by the bug - but to my surprise, it wasn't.

I used a TrueHD 7.1 channel check available on demo-world.eu

I tested it this way:

eac3to.exe input.mkv 2: output.wavs

and all the channels seemed to be in the correct file, in particular:

output.SL.wav -> left side surround
output.SR.wav -> right side surround
output.BL.wav -> left back surround
output.BR.wav -> right back surround

So I tested it on a DTS-HD MA 7.1 channel check from the Disney World of Wonder Blu-ray, and got incorrect results:

output.SL.wav -> left back surround
output.SR.wav -> right back surround
output.BL.wav -> left side surround
output.BR.wav -> right side surround

eac3to.exe -test

eac3to (v3.24) is up to date
Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed
http://www.nero.com/eng/store-blu-ray.html
CAUTION: You need Nero 7. Nero 8 won't work with eac3to.
ArcSoft DTS Decoder (1.1.0.0) works fine
Sonic Audio Decoder (3.24.0.0) doesn't seem to be installed
Haali Matroska Muxer (2011-03-03) is up to date
Nero AAC Encoder could not be located
http://www.nero.com/eng/nero-aac-codec.html
Copy NeroAacEnc.exe to the eac3to or to the Windows folder.
Surcode DTS Encoder doesn't seem to be installed
http://www.surcode.com
MkvToolnix (5.0.0.0, release version) is up to date

There are a couple possibilities:

1)
- the FFmpeg TrueHD decoder is patched in eac3to to fix the channel swap bug, and either:
- - a. the Arcsoft DTS decoder is swapping channels
- - b. eac3to is remapping channels incorrectly from DTS-HD MA 7.1 sources

2)
- the side and back surround channels are swapped internally in eac3to, affecting all 7.1 sources
- the FFmpeg and eac3to bugs cancel each other out

In the latter case, remapping should always be used when dealing with 7.1 sources, whether downmixing or not:

-0,1,2,3,6,7,4,5

…except when using the FFmpeg decoder and the source is TrueHD 7.1.

Here are the WAV files output by eac3to:

no remapping: http://www.mediafire.com/?3r35vr7gpg3v008

-0,1,2,3,6,7,4,5: http://www.mediafire.com/?68q021d69doy8va

wlee

4th October 2011, 06:19

I have a TrueHD track without AC3 core, it's not recognized by tsMuxer, is there a way to convert to TrueHD+AC3 track? so that I can mux into m2ts container.

Thanks.

tebasuna51

4th October 2011, 11:52

I have a TrueHD track without AC3 core, it's not recognized by tsMuxer, is there a way to convert to TrueHD+AC3 track? so that I can mux into m2ts container.

Thanks.

eac3to input.thd output.thd+ac3

tebasuna51

4th October 2011, 12:47

...
There are a couple possibilities:

...

In the latter case, remapping should always be used when dealing with 7.1 sources, whether downmixing or not

I don't think so. For me work fine, maybe other possibility:

3) Your test channel is wrong. Please put a exact link (or upload) to the test channel.

Please test this test-channel I created now with DTS HD Encoder Suite, here:
ss = Side
sr = Back

http://www.mediafire.com/?sk7mud5h28ol2m4

Thunderbolt8

4th October 2011, 13:05

eac3to input.thd output.thd+ac3this would create a ac3 track from the .thd track though. is it also possible to use the orginal interweaved and demuxed ac3 track as input file for this process, together with the thd track?

nurbs

4th October 2011, 13:26

As far as I know it would only create an AC3 track from THD if there is no interleaved AC3 in the track to begin with. If there is it should just demux.

Thunderbolt8

4th October 2011, 16:22

but if I demux, dont get thd and ac3 seperated automatically? that is what I would want to avoid.

Rodeo

4th October 2011, 17:20

I don't think so. For me work fine, maybe other possibility:

3) Your test channel is wrong. Please put a exact link (or upload) to the test channel.

Please test this test-channel I created now with DTS HD Encoder Suite, here:
ss = Side
sr = Back

http://www.mediafire.com/?sk7mud5h28ol2m4

Looks like we're getting different results:

eac3to.exe test-chan.dtshd channel.wavs

http://www.mediafire.com/?q9x6ovpz726waju

channel.SL.wav -> "beep… back left"
channel.SR.wav -> "beep… back right"
channel.BL.wav -> "beep… side left"
channel.BR.wav -> "beep… side right"

b66pak

4th October 2011, 17:35

@tebasuna51 i can confirm that SL/SR are swapped with BL/BR...

eac3to.exe test-chan.dtshd channel.wavs

eac3to 3.24
ArcSoft DTS Decoder (1.1.0.0) works fine
_

tebasuna51

4th October 2011, 21:40

I can't understand, here with:

eac3to (v3.24) is up to date
ArcSoft DTS Decoder (1.1.0.0) works fine

eac3to.exe test-chan.dtshd channel.wavs

25/07/2008 15:04 462.925 ASAudioHD.ax v1.3.2.60
15/04/2008 19:40 106.496 checkactivate.dll v1.0.0.2
17/01/2008 19:37 925.696 DtsDec.dll v1.0.2.2
25/04/2008 11:50 917.504 dtsdecoderdll.dll v1.1.0.0
02/11/2006 16:35 35.584 MagCore.dll v1.0.0.129
02/11/2006 16:28 60.160 MagPCMac.dll v1.0.0.129
02/11/2006 16:29 158.464 MagUIEngine.dll v1.0.0.129
02/11/2006 16:30 88.832 MagUIInter.dll v1.0.0.129

the channels aren't swapped.

Edit: I remember now what may be the problem (suffered by myself), check the dtsdecoderdll.dll date:
http://forum.doom9.org/showthread.php?p=1266679#post1266679

Rodeo

4th October 2011, 23:29

Hmm - I got my Arcsoft decoder from a very recent TMT5 trial, and wondered how I could have ended up with version 1.1.0.0 (which seemed old) of the DTS decoder.

It looks like eac3to.exe isn't picking up TMT5's decoder, but an old version of dtsdecoder.dll (from 21/04/2008) which found its way to my C:\Windows\SysWOW64 folder somehow…

:confused:

So I guess it's 1) a. then:

1)
- the FFmpeg TrueHD decoder is patched in eac3to to fix the channel swap bug, and either:
- - a. the Arcsoft DTS decoder (1.1.0.0 from 21/04/2008) is swapping channels

wlee

5th October 2011, 03:54

this would create a ac3 track from the .thd track though. is it also possible to use the orginal interweaved and demuxed ac3 track as input file for this process, together with the thd track?

If not possible to convert thd to thd+ac3, is it possible to convert thd to dts-hd ma track without any quality loss? tsMuxer should have no problem with dts-hd ma track.

b66pak

5th October 2011, 16:52

i can confirm that dtsdecoderdll.dll v1.1.0.0 (25 April 2008) is OK...

md5:
644aa3ade7742079533dcde2abf153e2 *dtsdecoderdll.dll
_

rc71

7th October 2011, 21:10

So, I was making sure that channel assignments and/or downmixing worked well, and tested eac3to on a bunch of channel checks I had available.

...................................

-0,1,2,3,6,7,4,5: http://www.mediafire.com/?68q021d69doy8va

I've had this issue myself. I was using Ripbot, which uses eac3to, when the problem arose. No matter what I did I couldn't get the mappings correct so I used MeGUI and set it to NeroAAC to convert the DTS to AAC HE and no problems.

ramicio

13th October 2011, 03:45

Is there any thought into being able to enter a dB gain in a precision other than an integer? The program can easily do it. If it detects clipping it applies a gain with a decimal, and it takes to below peak, so it's not just to get it to 0 dB.

asarian

13th October 2011, 06:57

There are some differing opinions as to which versions decode properly.

Personally I use 1.1.0.0 for 5.1 and 6.1 DTS-MA, and I use 1.1.0.8 for 7.1 DTS-MA (except strange setup).

I'm planning to install Arcsoft Total Media Theatre 5.0.1.114 (because it can play back Blu-Ray from folder). Is there a way I can have eac3to keep using the old 1.1.0.0 decoder? (which I had put in the eac3to directory itself, and registered manually). Or will the newest decoder work too?

rc71

14th October 2011, 18:08

This file was a pain in the ass to get. So here is the dtsdecoderdll.dll 25/04/2008 04:50 v1.1.0.0. If there is a 10:50 am file then well... .s@#$@#$ 79oif@$#&(*@ #$

~link removed by moderator due to possible license violation~

nibus

14th October 2011, 22:23

Just came across a stream that eac3to lists as:

5: TrueHD/AC3, English, 5.1 channels, 48kHz, dialnorm: -27dB, 83ms

Does this mean it is delayed 83ms? So when I transcode I need to apply +83ms to the output?

asarian

18th October 2011, 19:14

Still having major trouble with TrueHD audio (see below). Libav can't do it, Nero (the often suggested alternative) says it can't do it either, and Sonic can't even be bothered to try (as per the documentation).

This concerns the Gantz Blu-ray (Japanese Audio). Is there anything else I can try?!

eac3to 00000.m2ts 2: temp.pcm
M2TS, 1 video track, 1 audio track, 2 subtitle tracks, 2:10:40, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, 5.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
3: Subtitle (PGS)
4: Subtitle (PGS)
a02 Extracting audio track number 2...
a02 Extracting TrueHD stream...
a02 Decoding with libav/ffmpeg...
a02 Swapping endian...
a02 Remapping channels...
a02 libav Substream 0 parity check failed
a02 libav Substream 0 checksum failed
a02 libav Substream 0 length mismatch.
a02 The libav decoder reported error -1 while decoding.
Aborted at file position 1048576.

eac3to 00000.m2ts 2: temp.pcm -nero
M2TS, 1 video track, 1 audio track, 2 subtitle tracks, 2:10:40, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, 5.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
3: Subtitle (PGS)
4: Subtitle (PGS)
Disabling DRC for Nero (E-)AC3 decoding...
a02 Extracting audio track number 2...
a02 Extracting TrueHD stream...
a02 Decoding with DirectShow (Nero Audio Decoder 2)...
a02 The DirectShow audio decoder didn't accept the input stream.
Aborted at file position 1048576.

eac3to 00000.m2ts 2: temp.pcm -sonic
M2TS, 1 video track, 1 audio track, 2 subtitle tracks, 2:10:40, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, 5.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
3: Subtitle (PGS)
4: Subtitle (PGS)
a02 Extracting audio track number 2...
a02 Extracting TrueHD stream...
a02 The Sonic Audio Decoder doesn't decode TrueHD properly.
Aborted at file position 1048576.

asarian

18th October 2011, 21:54

try the -nero switch (nero is limited to 5.1 channels, but this is not a problem here)

As you can tell from my above post, that didn't work either.

Really seems an eac3to (libav) problem somehow, as nearly a quarter of all my TrueHD streams have eac3to abort somewhere (whereas they all appear to be pretty normal streams, that play with TMT 5, XBMC, etc., just fine). Never have this issue with DTS-MA.

Thunderbolt8

18th October 2011, 23:05

send a sample to madshi (try first if the same problem occurs with the sample)

asarian

18th October 2011, 23:50

send a sample to madshi (try first if the same problem occurs with the sample)

Ok, I sent him a PM about it and a link to this file:

10sec Gantz sample (http://www.mediafire.com/file/s1rxt8dtltaw6to/gantz_sample.m2ts)

(I didn't see an option for attachments on PM's)

Now I'll just wait. :)

b66pak

19th October 2011, 19:37

@asirian you can use this (demux the thd with eac3to and then decode it with the latest ffmpeg):

eac3to gantz_sample.m2ts gantz_sample_audio.thd

ffmpeg -i gantz_sample_audio.thd -acodec pcm_s24le gantz_sample_audio.thd.wav

get the latest ffmpeg from here (http://ffmpeg.zeranoe.com/builds/win32/static/)...
_

asarian

19th October 2011, 21:25

Thunderbolt8

19th October 2011, 22:54

seems like eac3to needs an ffmpeg update

Midzuki

20th October 2011, 04:06

madshi should set up an autobuilder :devil: for eac3to.zip, duly in sync with the latest ffmpeg of course :D

asarian

20th October 2011, 11:53

madshi should set up an autobuilder :devil: for eac3to.zip, duly in sync with the latest ffmpeg of course :D

Or maybe it's simply time for a new release, compiled against the latest libav/ffmpeg!? :devil: Seriously, I don't have the Windoze environment to compile C sources.

kypec

20th October 2011, 11:56

Or maybe it's simply time for a new release, compiled against the latest libav/ffmpeg!?
It most certainly is! :goodpost:

Kurtnoise

20th October 2011, 13:21

did you try to grab the dlls from the shared build and put them in the eac3to folder ?

mbcd

20th October 2011, 17:03

I thought that eac3to supports plugins, isnt it possible to build a plugin that fixes some problems as channelorder, thd, ... ?

Otherwise, for what could someone use those plugins-interface ?

Spc01

20th October 2011, 21:34

Looks like we're getting different results:

eac3to.exe test-chan.dtshd channel.wavs

http://www.mediafire.com/?q9x6ovpz726waju

channel.SL.wav -> "beep… back left"
channel.SR.wav -> "beep… back right"
channel.BL.wav -> "beep… side left"
channel.BR.wav -> "beep… side right"

I can confirm that, just tested it out on DTS-HD Master Audio 7.1 24bit, 48kHz.

S* = Back
B* = Side

mindbomb

21st October 2011, 04:45

which channel configurations does down2 work on?
5.1 and 7.1 seem to work, but between 2.0 and 5.1, does anything work?

tebasuna51

21st October 2011, 15:18

To 2.0 -> 5.1 read the Sticky: GUIDE LIST: Stereo-to-Surround Conversion Guides (http://forum.doom9.org/forumdisplay.php?s=&forumid=11)

kypec

21st October 2011, 16:00

To 2.0 -> 5.1 read the Sticky: GUIDE LIST: Stereo-to-Surround Conversion Guides (http://forum.doom9.org/forumdisplay.php?s=&forumid=11)
I think he asked about situation of down-mix to stereo from multi-channel sources with non-standard number of channels, such like 2.1 / 3.0 / 4.0 / 5.0.
At least that's how I understood the wording "between 2.0 and 5.1" of his post's context.:confused:

mindbomb

21st October 2011, 23:11

yea, kypec is right.
maybe i should have been more clear.

If you try to use down2 on 3 channel audio, it just doesnt work, is that right?

tebasuna51

22nd October 2011, 03:27

Sorry.
Only work with 5.1, to downmix 3, 4 or 5 channels to stereo use MeGUI.

Asmodian

29th October 2011, 04:47

Is it possible to output 32 bit audio to stdout?

For 24 bit I use:
eac3to.exe in.mp2 stdout.pcm -24

"eac3to.exe in.mp2 stdout.pcm -32" doesn't seem to give 32 bit audio (it isn't in the docs so no surprise) but is there a way?

ramicio

29th October 2011, 05:03

Use the -full switch to get 64-bit floating point. It's either 24-bit integer or 64-bit float, nothing in between. What is even going to accept 32 float as an input? What is the point? It offers the same precision as 24-bit integer if nothing goes over 0dB.

Asmodian

29th October 2011, 07:09

Thanks, it makes a lot more sense now.

It turns out I am just getting 24 bit from the eac3to, there are no more bits to get. (-full gives 24 bit)

I am piping to qaac, I think 24 bit is all that exist for this mp2 source but qaac could take 32bit float if it did exist.

So there is no point.
:thanks:

pandv2

30th October 2011, 22:59

Hello,

I think a found a little bug:

With this command line:

"C:\MasProgramas\eac3to\eac3to.exe" "C:\Temp\VideoSynch\Vsy_Segmento_Aud_000.ac3" "C:\Temp\VideoSynch\Vsy_Segmento_Aud_000_sil001.ac3" -silence -edit=0:00:00.544,1088ms

Eac3to shows a error:

Invalid edit format "edit=0:00:00.544,1088ms

And also with: -edit=0:00:00.544,544ms

but not with: -edit=0:00:00.544,543ms

If the silence duration to insert, is bigger than the insertion position, eac3to trows a error. So:

-edit=0:00:00.789,800 is a error
-edit=0:00:00.789,788 is not

I think is a easy fix (an outdated check in action, I think). Is the source code available?

Thunderbolt8

30th October 2011, 23:07

nope :p

Atlantis

31st October 2011, 03:57

Guys this is a long thread but please excuse me I couldn't read it all.

I have noticed that eac3to can not read a 32bit float wav file created by audition as input. If I convert the same file in audition to 24bit, eac3to can read it. Any ideas?

tebasuna51

31st October 2011, 11:22

I have noticed that eac3to can not read a 32bit float wav file created by audition as input.

I can't reproduce your problem.
Please put a sample (1 second is enough).

far.in.out

31st October 2011, 19:40

Hi. I'm doing
eac3to input.dtshd output.flac -down2 -mixlfe -down16
and getting this in log

Clipping detected, a 2nd pass will be necessary. <WARNING>
...
Applying -6,73dB gain...

So, should I worry about it?
Should I redo it using like -7dB to be safe?
Or is it fine after second pass?
Does it use the source dtshd for 2nd pass or just created flac?

nurbs

31st October 2011, 20:23

So, should I worry about it?
Should I redo it using like -7dB to be safe?
Or is it fine after second pass?
Does it use the source dtshd for 2nd pass or just created flac?
No, no, yes and temporary wav I guess.

Atlantis

1st November 2011, 06:06

Here is a 1 sec audio file 48khz 5.1 32bit float.

eac3to can't read it.

tebasuna51

1st November 2011, 11:18

Here is a 1 sec audio file 48khz 5.1 32bit float.

Your wav file have a wrong header:
File ........: D:\Internet\audio 32bit float.wav
Size ........: 1152068 bytes

---------------------------------------------- Header Info
ChunkID .....: RIFF
RiffLength ..: 1152060
Container ...: WAVE
SubchunkID ..: fmt (Length: 40)
AudioFormat .: 65534 (WAVE_FORMAT_EXTENSIBLE)
NumChannels .: 6
SampleRate ..: 48000
ByteRate ....: 1152000
BlockAlign ..: 24
BitsPerSample: 32
ValidBitsPS .: 0
MaskChannels : 63 (FL FR FC LF BL BR)
SubType .....: 3 (Float)
SubchunkID ..: data (Length: 1152000)
Offset data .: 68
Duration ....: 1 sec., (0h. 0m. 1s.)
------------------------------------------------- End Info

The field ValidBitsPerSample must be 32 instead 0.

You can hexedit the wav file and put at offset 38 the correct value 32 (hex 20) or fix the file with WavFix (http://forum.doom9.org/showthread.php?p=1520399#post1520399):

wavfix "audio 32bit float.wav" -m 0

Add the parameter -ignorelength if your wav file is greater than 4 GB.
After that eac3to can read the file.

Atlantis

1st November 2011, 12:43

What program did you use to see that information? I want to use it.

Also why the header is wrong? I used audition to get that. Also, if I output as 32bit float it doesn't work but if I output as 32bit integer, it works.

(using adobe audition to output. Works means eac3to can read the file)

dream88

1st November 2011, 14:03

Hi can someone please confirm for me what exactly the bug is with different versions of Arcsoft DTS Decoder, ?

Is it 1.1.0.5 or 1.1.0.0 for 6.1/6.0 audio and 1.1.0.8 for anything else ?
Some people claim that 1.1.0.0 is the best for 6.1/6.0 but others claim 1.1.0.5 is best because 1.1.0.0 lacks many bug fixes ?
I found the following pieces of information around doom9 forums but i'm still confused to which version to use to avoid any bug ?

- 1.1.0.0 can decode DTS(-HD) 6.1/6.0 but can't decode non-standard 7.1
- All versions above v1.1.0.0 do not accurately decode 6.1 DTS tracks with eac3to.
- 1.1.0.8 can't decode DTS(-HD) 6.1/6.0 but can decode non-standard 7.1
- Both decode DTS(-HD) 1.0 correctly, unlike 1.1.0.7.
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 DTS is 1.1.0.5. It decodes all configurations, in proper bitdepth, and decoding algorithm didn't change since this version (except 6.0 bug in .7 and .8).

Anakunda

1st November 2011, 20:56

HI,

eac3to question:

does the -down16 switch have impact on quality or samplerate/bit depth on DTS -> AAC conversion? (providing the DTS is 24b/48kHz/6ch)

ramicio

1st November 2011, 20:58

tebasuna51

2nd November 2011, 00:02

Hi can someone please confirm for me what exactly the bug is with different versions of Arcsoft DTS Decoder, ?

- I don't found any bug using 1.1.0.0, also decode non-standard 7.1 without problems.

- lossy DTS don't have bitdepth, then you can't recover the original bitdepth source. The data in header only inform about the source bitdepth but this data is useless for 2 reasons:

1) Some encoders put always 24 bits in this field no mather the source was 16 or 24 bits.

2) The error recovering the lossy encode is greater than the precission difference betwen 16 and 24 bits, then it makes no sense decode to 16 or 24 bassed in source bitdeph, decode always to 24 bits and you obtain always the best approach.

tebasuna51

2nd November 2011, 18:31

What program did you use to see that information? I want to use it.

http://forum.doom9.org/showthread.php?p=1522330#post1522330

Also why the header is wrong?

I don't know. I don't support commercial soft, ask to the vendor.

Atlantis

2nd November 2011, 19:13

tebasuna51, what you said is not correct.

I did another test. You said that it was because of wrong header.

Well as I said, 32 float was not recognized and 32 integer was recognized by eac3to. So I passed a 32 integer to your tool LeeAudBi and the header is the same incorrect way as you say but it works.

audio 32bit integer.wav
ChunkID .....: RIFF
RiffLength ..: 1152060
Container ...: WAVE
SubchunkID ..: fmt (Length: 40)
AudioFormat .: 65534 (WAVE_FORMAT_EXTENSIBLE)
NumChannels .: 6
SampleRate ..: 48000
ByteRate ....: 1152000
BlockAlign ..: 24
BitsPerSample: 32
ValidBitsPS .: 0
MaskChannels : 63 (FL FR FC LF BL BR)
SubType .....: 1 (Integer)
SubchunkID ..: data (Length: 1152000)
Offset data .: 68
Duration ....: 1 sec., (0h. 0m. 1s.)

I attached the file here. Test it with eac3to and it is recognized so that header problem you mentioned is not the problem. There must be another problem within eac3to.

tebasuna51

3rd November 2011, 02:31

Maybe eac3to don't check this field with int samples, because it make the test to know the real bitdepth of the source, with float sample must know the alignement.

But, how do you explain than changing only this field with a hexeditor the float file is read without problems?

You can read about this field here: http://msdn.microsoft.com/en-us/windows/hardware/gg463006
Details about wValidBitsPerSample
The field wValidBitsPerSample is used to explicitly indicate how many bits of precision are present in the signal. Most of the time this value will be equal to wBitsPerSample...

Audacity fill this field correctly, I don't know for what Audition don't make the same.

Traps

4th November 2011, 15:52

Source is 24bit TrueHD track, i want to convert it to 16bit flac or DTS but i'm getting this error

command line: eac3to.exe" "E:\THD.thd" "E:\flac16.flac -down16"
------------------------------------------------------------------------------
TrueHD, 5.1 channels, 48kHz
This audio conversion is not supported. <ERROR>

Anyone knows what am i doing wrong?

ramicio

4th November 2011, 15:54

Try getting rid of the quotes. You don't need them when there are no spaces in file names or paths. The -down16 switch is part of the file name in your command.

Traps

4th November 2011, 15:55

Yeah, i figured it out like 2sec after i posted here, thanks anyway!

pandv2

6th November 2011, 17:27

If the silence duration to insert, is bigger than the insertion position, eac3to trows a error. So:

-edit=0:00:00.789,800 is a error
-edit=0:00:00.789,788 is not

If the source code is not available, can I hope this, and another bugs, get corrected in the future? Or Madish had left this project?.

Now I am using BeSplit to cut sections and eac3to to insert o delete (with delay) in the beginning. This works but BeSplit is not maintaned, and there are also some bugs.

And i post, also, a feature request: Extract a ac3 segment to a new file.

Boulder

7th November 2011, 19:44

How does normalization work when doing a multichannel to multichannel transcode? Is it even useful to apply or will it only mess the balance between the channels?

tebasuna51

7th November 2011, 22:29

-normalize preserve the balance between channels, all channels are amplified by the same value.

twazerty

8th November 2011, 19:19

In the professional world there is a problem regarding the HDV format with 4 audio channels. For example the Sony S270 which I have access to. This is a professional camera which is able to record a 4ch audio stream. When you want to import the footage of the camera to your computer you are unable to access the 4ch audio file inside the M2T file (MPEG-TS). And big names like Vegas, Avid, etc aren't able to see/extract the 4ch audio stream. Only the second stereo stream will be available. On the camera itself you are able to access all 4 audio channels. This is a problem that all camera's have that record in HDV with 4 channels and there isn't a solution from the manufactures. The only way is via SDI but it is very time consuming and HDD hungry (3GB/s).

Applications like MediaInfo and all video players can only access the second audio stream (or a backwards compatible stream but I am not sure of this). This is the mediainfo log:
General
ID : FF
Complete name : J:\HDVTest 4ch\00_0002_2010-11-07_214302.M2T
Format : MPEG-TS
File size : 138 MiB
Duration : 44s 160ms
Start time : UTC 2010-11-07 21:43:02
End time : UTC 2010-11-07 21:43:07
Overall bit rate : 26.2 Mbps
Maximum Overall bit rate : 33.0 Mbps
Encoded date : UTC 2010-11-07 21:43:02

Video
ID : 2064 (0x810)
Menu ID : 100 (0x64)
Format : MPEG Video
Format version : Version 2
Format profile : Main@High 1440
Format settings, BVOP : Yes
Format settings, Matrix : Default
Duration : 44s 320ms
Bit rate mode : Constant
Bit rate : 24.0 Mbps
Nominal bit rate : 25.0 Mbps
Width : 1 440 pixels
Height : 1 080 pixels
Display aspect ratio : 16:9
Frame rate : 25.000 fps
Standard : Component
Resolution : 8 bits
Colorimetry : 4:2:0
Scan type : Interlaced
Scan order : Top Field First
Bits/(Pixel*Frame) : 0.617
Stream size : 127 MiB (92%)

Audio
ID : 2068 (0x814)
Menu ID : 100 (0x64)
Format : MPEG Audio
Format version : Version 1
Format profile : Layer 2
Duration : 44s 208ms
Bit rate mode : Constant
Bit rate : 384 Kbps
Channel(s) : 2 channels
Sampling rate : 48.0 KHz
Video delay : -232ms
Stream size : 2.02 MiB (1%)

But in fact this file contains a 4 channel audio stream. This is the output of eac3to v3.2.4:
eac3to v3.24
command line: "c:\Program Files (x86)\AVCHDCoder\Tools\eac3to\eac3to.exe" "J:\HDVTest 4ch\00_0002_2010-11-07_214302.M2T" -demux
------------------------------------------------------------------------------
TS, 1 video track, 2 audio tracks, 0:00:44, 50i
1: MPEG2, 1440x1080 50i (16:9)
2: E-AC3, unknown parameters
3: MP2, 2.0 channels, 384kbps, 48kHz, -264ms
Bitstream parsing for track 2 failed. <WARNING>
Demuxing this track may still produce correct results - or not. <WARNING>
[a02] Extracting audio track number 2...
[v01] Extracting video track number 1...
[a03] Extracting audio track number 3...
[v01] Creating file "00_0002_2010-11-07_214302 - 1 - MPEG2, 1440x1080 50i.m2v"...
[a03] Applying MPx delay...
[a03] Creating file "00_0002_2010-11-07_214302 - 3 - MP2, 2.0 channels, 384kbps, 48kHz.mp2"...
Video track 1 contains 1107 frames.
eac3to processing took 1 second.
Done.

In this case a second audio stream will pop up. But I don't think it is an E-AC3 audiostream when we take a look at the HDV specifications that can be found on wikipedia: HDV Specifications (http://en.wikipedia.org/wiki/HDV#Specifications). The most important part is this:
MPEG-1 Part 3 AL 2 Stereo (2-channel) at 384 kbit/s (192 kbit/s per channel);
optional MPEG-2 Part 3 AL 2 4-channel at 96 kbit/s per channel.

As you can see 4 channel audio is part of the HDV format. I am not sure if there are actually 2 audiostreams or only 1. There is no way I can check this because the camera itself only outputs the 4 ch and no 2 ch. But the computer will only play a 2 ch. But when you compare the MediaInfo audio information with the specs I expect that there are 2 audio stream just like eac3to detects. It would be very very nice if HDV support can be integrated into eac3to because many professional cameramen are looking for an solution.

Currently eac3to extracts the m2v and 2ch audio stream fine. Only the 4 channel stream isn't detected nor extracted. I uploaded 2 sample to my server for analysis:
http://tools.twanwintjes.nl/uploads/temp/00_0001_2010-11-07_153401.M2T (49 seconds / 152MB) - Right Click --> Save As
http://tools.twanwintjes.nl/uploads/temp/00_0002_2010-11-07_214302.M2T (44 seconds / 137MB) - Right Click --> Save As

In addition to the main problem the camera splits a recording into 4GB chunks. So sync issues are a problem according to the professionals.

Note: I do not own this professional camera. I professional camera guy I know came to me with this problem and he owns a camera. If you are willing to support HDV and you need more material like other resolutions/framerates/interlaced/progressive with only 2ch or combined with a 4ch audio track. We can deliver test footage. We are sure many people are looking for a solution. Companies like Sony do not offer a software solution, also this problem is still available on new camera's.

Additional info (http://www.dvinfo.net/forum/sony-hvr-z7-hvr-s270/259102-big-problem-use-s270-4-channel-audio-hdv-format.html)

Joniii

9th November 2011, 11:54

Why does eac3to and mkvextractgui2 output different file sizes?

h264 track demuxed from mkv:

eac3to 18 341 959 089
mkvextractgui2 18 341 843 800

Chouonsoku

13th November 2011, 05:18

I'm trying to convert a Dolby TrueHD 1.0 track to FLAC, and I keep getting the following error.

eac3to v3.24
command line: eac3to B:\BDMV\STREAM\00001.m2ts 4:japanese.flac
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 2 subtitle tracks, 0:24:13, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, English, 5.1 channels, 48kHz, dialnorm: -25dB
(embedded: AC3, 5.1 channels, 448kbps, 48kHz, dialnorm: -25dB)
3: TrueHD/AC3, English, 2.0 channels, 48kHz, dialnorm: -25dB
(embedded: AC3, 2.0 channels, 192kbps, 48kHz, dialnorm: -25dB)
4: TrueHD/AC3, Japanese, 1.0 channels, 48kHz, dialnorm: -25dB
(embedded: AC3, 1.0 channels, 96kbps, 48kHz, dialnorm: -25dB)
5: Subtitle (PGS), English
6: Subtitle (PGS), English
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] [libav] Substream min channel cannot be greater than max channel. <WARNING>
[a04] The libav decoder reported error -1 while decoding. <ERROR>
Aborted at file position 1048576. <ERROR>

I did some searching and found that this is supposedly caused by an outdated version of ffmpeg included with eac3to. So I grabbed the latest version and ran the command line:

ffmpeg -i japanese.thd -ac 1 -acodec pcm_s24le -f wav japanese.wav

And got the following error:

Input stream #0:0 frame changed from rate:48000 fmt:s32 ch:1 to rate:48000 fmt:s32 ch:2
Assertion ctx->channels == out->ch_count failed at /home/kyle/software/ffmpeg/source/ffmpeg-git/libswresample/audioconvert.c:66

This application has requested the Runtime to terminate it in an unusual way. Please contact the application's support team for more information.

So now I'm not sure what to do. I don't remember having this problem with any other TrueHD 1.0 tracks in the past.

tebasuna51

13th November 2011, 10:24

The message is clear: this track is a mix of 1.0 and 2.0
Try with:
ffmpeg -i japanese.thd -ac 2 -acodec pcm_s24le -f wav japanese.wav

Chouonsoku

13th November 2011, 20:01

The message is clear: this track is a mix of 1.0 and 2.0
Try with:
ffmpeg -i japanese.thd -ac 2 -acodec pcm_s24le -f wav japanese.wavThe message isn't that clear. :P I'll give that a shot though, I had not considered this might be a mixed channel format track.

Edit: That gives me a much longer list of errors, mostly this:

[truehd @ 01B49040] Lossless check failed - expected 65, calculated fa.
[truehd @ 01B49040] Invalid nonrestart_substr.
Error while decoding stream #0:0

tebasuna51

13th November 2011, 23:38

Then seems you have a unsupported or corrupt THD stream.
Try extract the ac3 file.

Chouonsoku

14th November 2011, 08:55

Then seems you have a unsupported or corrupt THD stream.
Try extract the ac3 file.I can get the core audio just fine, it's just the decoding to WAV that seems to be the problem. I can upload the track in question (it's under 90 MB) if it'll help diagnose the problem.

tebasuna51

14th November 2011, 11:23

I don't know the TrueHD specs and I can't help.
Maybe you can open a new thread to explain the ffmpeg problem, eac3to can't work without a libav decoder than work.

mindbomb

15th November 2011, 04:23

what is the proper procedure for encoding 6.1 audio to flac?

-0,1,2,3,5,6,4 -double7 , right?

and this applies to any audio, or just dts input?

tebasuna51

15th November 2011, 11:37

Only if you want 7.1 output.
If you want 6.1 output the remap is not needed.

The remap is needed, with any 6.1 audio, if you use -double7 or -down6 parameters.

mindbomb

15th November 2011, 15:10

thanks tebasuna

so, just to be clear, with the flac encoder, you can't encode 6.1 audio?

You have to either use down6 or double7, right?

tebasuna51

15th November 2011, 18:38

You can encode 6.1 audio to flac. Just I make the test to se if channel order is preserved.

eac3to 6.1.dtshd 6.1.flac
eac3to 6.1.flac 6.1.wav

and the 6.1.wav is bit-identical to

eac3to 6.1.dtshd 6.1.wav

with correct channel order.

sneaker_ger

15th November 2011, 18:42

What about this (http://mod16.org/hurfdurf/?p=184)?

Was this just a decoder problem or is the channel order really not defined?

tebasuna51

16th November 2011, 01:28

Yes, the channel order for 7 channels (6.1) in flac is undefined, but also for 8 channels (7.1), only until 6 channels (5.1) is well defined:
http://flac.sourceforge.net/format.html#format_overview (Channel assignment)
But the channel order is preserved using eac3to, or flac.exe, like encoder/decoder.

For me the 6.1 and 7.1 flac play fine in my directshow system (madflac installed) and Foobar2000.
Maybe other decoders don't work properly, or the source input have a wrong channel order.

mindbomb

16th November 2011, 01:51

What about this (http://mod16.org/hurfdurf/?p=184)?

Was this just a decoder problem or is the channel order really not defined?

I was literally just about to post that

phate89

16th November 2011, 15:30

Something strange happens to me... i used eac3to to edit more than one time a 5.1 track to resync it avoiding recompressions... i tested the ac3 using media player classic that allows to load external audio and the editing i do seems perfect... then i mux the audio into the file (mkv with h264 video) and in some points i edit it slightly loose sync...
the same ac3 outside works well but not inside the mkv... any clue about how is that possible?

sneaker_ger

16th November 2011, 19:43

tebasuna51

16th November 2011, 22:19

Interesting that they explicitly state that those are not defined, instead of defining them.

Yes, this is a feature request, still open, since 2008-04-22:
http://sourceforge.net/tracker/?func=detail&aid=1949155&group_id=13478&atid=363478

sneaker_ger

16th November 2011, 22:34

Too bad, you'd figure this would be a very simple change.

Thunderbolt8

16th November 2011, 23:47

why is it only possible to remove pulldown from mpeg2 tracks via -strippulldown, but not from AVC or VC-1 tracks? according to -keeppulldown description, those codecs can have pulldown as well, so doesnt it seem to be possible to remove it with -strippulldown?

Chouonsoku

18th November 2011, 07:35

I don't know the TrueHD specs and I can't help.
Maybe you can open a new thread to explain the ffmpeg problem, eac3to can't work without a libav decoder than work.

I submitted a bug to ffmpeg. https://ffmpeg.org/trac/ffmpeg/ticket/667

Thunderbolt8

18th November 2011, 22:15

eac3to still cant detect (all?) aac LOAS/LATM streams. any idea when this is supposed to come?

Vasch the stampede

19th November 2011, 23:24

how should I encode an AAC Q = 1?

I tried whith:

-q 1, -q 1.0, -q1, -q1.0, -q 1.00, -q1.00...

None was correct

tebasuna51

20th November 2011, 00:00

Try with -quality=1

Vasch the stampede

20th November 2011, 00:14

Try with -quality=1
Thanks!!!

73ChargerFan

20th November 2011, 01:03

I'm trying to delete the last 8 seconds from an ac3 track.

Input is 1:50:56.700 in duration
I've tried: -edit=1:50:56,-8000ms
Output is 1:50:56.000 in duration

What am I doing wrong?

tebasuna51

20th November 2011, 04:12

Maybe -edit=1:50:48.700,-8000ms

Lincoln Burrows

25th November 2011, 00:53

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

About DTS-HD Master Audio tracks,

DTS-HD decoding can be achieved through:

"Sonic Cinemaster Audio Decoder 4.3" DirectShow filter (commercial software)
ArcSoft TotalMedia Theatre (commercial software)

Both decode bit-perfect. Sonic is limited to 5.1 or 6.1 channels depending on the source track, while ArcSoft can output up to 7.1 channels.

As a result ArcSoft is the default decoder for DTS-HD.Well, I have TotalMedia Theatre from ARCSOFT, last version. However, eac3to is saying...

http://i.imgur.com/qXcIs.png

How do I fix this? The software is already installed, but eac3to for some reason, can't detect the decoder.

And here we have broken links:
http://forum.doom9.org/showthread.php?p=1168721#post1168721

P.S. I am not using a trial version from TMT.

heerschop

25th November 2011, 23:12

For Eac3to to use the arcsoft decoder you need the checkactivate.dll in the arcsoft folder. Rename the attachment checkactivate.txt to checkactivate.dll en copy this file to your arcsoft folder.

It should work now.

Greets

MrVideo

28th November 2011, 05:59

The WAV file that eac3to produces is garbage. The sample file contains a pure sinewave test tone. Not so the output.

I've tried it with and without the -down16 option, but it didn't make a difference.

Sample file: http://vidiot.com/MPEG-LPCM-sample.ts (67,108,864 bytes)

Stream 1 is the MPEG-2 480i video stream, stream 2 is the MPEG stereo audio stream and stream 3 is the stereo LPCM audio stream.

Any help as to what the right options are to get a usable WAV file will be appreciated.

therealjoeblow

3rd December 2011, 02:31

nibus

3rd December 2011, 04:11

You might not have the right version of NeAudio2.ax - I remember having a similar issue until I found a slightly different version of Nero 7. The SFV checksum for mine is: AD81926E

And I don't think you need to register NeEacDec.dll, just the other two.

tebasuna51

3rd December 2011, 12:11

The WAV file that eac3to produces is garbage. The sample file contains a pure sinewave test tone. Not so the output.

Seems your sample is corrupted.
The PCM track is not recognized by tsMuxeR.
How do you make your sample?

I make a test.
1) Decode the mp2 track to wav:
eac3to MPEG-LPCM-sample.ts 2: test.wav -down16

2) Mux the test.wav to MPEG-LPCM-sample.ts like 2nd audio track (the original PCM 2nd audio track was ignored)

3) Extract the 2nd audio PCM track without problems:
eac3to MPEG-LPCM-sample-new.ts 3: test-new.wav

iSeries

6th December 2011, 09:55

Hi,

When dowmixing 5.1 sources to 2.0, is it a good idea to include -mixlfe? Or better to not include?

tebasuna51

6th December 2011, 11:55

TDiTP_

6th December 2011, 12:21

When dowmixing 5.1 sources to 2.0, is it a good idea to include -mixlfe? Or better to not include?
Dolby inc. does not recommen do it:
There are other concerns when adding an LFE signal to the mix. If the LFE is simply redistributed within the other channels of the mix, they will usually be subject to some low-frequency bandpass filtering. This filtering causes phase shifts of the LFE signal. When they are acoustically added within a room, these phase shifts are fairly subtle and often go unnoticed. However, when they are electronically added together with the five main channels in the encoder, they may produce less than desirable results at certain frequencies. For this reason, it is recommended that the LFE signal not be used in a Dolby Pro Logic II downmix unless it contains unique information that is not repeated in any of the five main channels.

iSeries

6th December 2011, 14:53

Thanks guys, will leave it out then. Thought there must be a reason it was left out by default.

Lincoln Burrows

7th December 2011, 19:13

Thanks to this info from rijnton I finally got this up and running. It seems that the detection most definitely has something to do with the newer installs from Arcsoft surrounding the Total Media Extreme product family. (which includes TMT) If using an older version that is TMT only it works fine even when the newer codec versions are installed on top of it.

Here's how to get it working:

1. Uninstall all previous versions and do the cleanup steps listed in this thread. (Registry clean, delete orphaned directories, delete instances of "arcsoft" in the registry)

1.Download and install the old version of the TMT trial that was pulled from Arcsoft's site awhile ago:

http://rapidshare.com/files/155298896/arcsoft_totalmediatheatre_engintro.exe.html

2. Do not run the program at this point - reboot
3. download this patch: http://www.arcsoft.com/downloads/digitaltheatre/arcsoft/totalmediatheatre_2.1.6.105_2.1.6.125_Update_ALL.exe
and install it. Again, do not run program. Note that this step may not be necessary. Rijton reported that he did not install this patch and went right to installing the retail w/o problems.
4. Reboot
5. eac3to should work at this point. If you own TMT or TME retail, go ahead and install now.I followed all those steps (had TMT 5 installed, but it was removed for this). I downloaded this Rapidshare file, installed, then rebooted the machine, Win7 warned me the driver was not compatible and that it would be disabled, TMT asked if I wanted to check for updates, I refused.

I did not installed this "patch" since the link is offline.

Before all that, eac3to told me I didn't had anything from Arcsoft. I tried converting a 7.1 DTS-HD MA track, from the movie "The Sound of Music", just to check if everything was fine. It failed to convert, since Arcsoft is the only one that recognizes that number of channels, as explained here:

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

DTS-HD decoding can be achieved through:

1) "Sonic Cinemaster Audio Decoder 4.3" DirectShow filter (commercial software)

2) ArcSoft TotalMedia Theatre (commercial software)

Both decode bit-perfect. Sonic is limited to 5.1 or 6.1 channels depending on the source track, while ArcSoft can output up to 7.1 channels.

As a result ArcSoft is the default decoder for DTS-HD.Look (and it won't convert to AC3, it will stay in that screen forever):

http://i.imgur.com/Pz6KZ.png

Please, I beg all of you! For God's sake, DO NOT POST HERE LINKS THAT REDIRECT US TO THE DEVELOPER'S WEBSITE, ALWAYS UPLOAD COPIES FROM SUCH FILES ELSEWHERE, AND NEVER A SINGLE COPY LIKE YOU DID IN THIS CASE!

This is another copy from the "arcsoft_totalmediatheatre_engintro.exe", this time stored in multiple servers:

http://migre.me/72ZKQ

http://migre.me/72ZMi

If you upload these things in a single server, sooner or later the file will be deleted for inactivity. If people do not download this thing, it will be deleted for sure. Megaupload is one of servers that does not delete files for that reason (despite what their FAQ says).

As a result, my only alternative is to use a software like SUPER from eRIGHTSOFT to convert this...

Lincoln Burrows

7th December 2011, 20:30

I did it! I solved the whole thing!

Thanks to this thread:
http://forum.doom9.org/showthread.php?t=139946

All I had to do was browse this folder (using DOS prompt - it won't work using the Start Menu):

C:\Program Files\ArcSoft\TotalMedia Theatre\Codec\

Where the file ASAudioHD.ax is located.

After that, all we have to do is execute that command:

regsvr32 ASAudioHD.ax

And then:

http://i.imgur.com/BWqOy.png

For the record, I only did this after reinstalling the last TMT version, and after that, updating the old one from the RS/Migre links.

It looks like this old version is incompatible with Windows 7, I couldn't make it work even using the compatibility mode.

iSeries

8th December 2011, 19:36

- Dolby recommend not include LFE in downmix.

- To play on TV or low cost audio equipment not include LFE.

- To play on a good hi-fi stereo, mix LFE at your risk. The arithmetic mix of LFE can produce artifacts, or not.

Sorry to bring this up again - is the Dolby recommendation to not include the LFE when downmixing to 2.0 specific to Dolby Digital / AC3 or is it relevent to all formats? I.e if downmixing to wav / PCM is it still not recommended?

Also - and off topic for eac3to (sorry!), when watching a movie with 5.1 do audio processors (eg ffdshow or even Windows mixer) ignore the LFE when downmixing to 2.0?

tebasuna51

9th December 2011, 11:51

Sorry to bring this up again - is the Dolby recommendation to not include the LFE when downmixing to 2.0 specific to Dolby Digital / AC3 or is it relevent to all formats? I.e if downmixing to wav / PCM is it still not recommended?
Read the TDiTP_ post.
The problem is how the original LFE was created, not matter the downmix output format.
Also - and off topic for eac3to (sorry!), when watching a movie with 5.1 do audio processors (eg ffdshow or even Windows mixer) ignore the LFE when downmixing to 2.0?
You can configure the downmix at your choice.

pandv2

9th December 2011, 12:51

Another little bug in eac3to, in the hope someday the programmer, or a friend of it, restart the project.

This is the log:

eac3to v3.24
command line: "C:\MasProgramas\eac3to\eac3to.exe" "G:\Sc\H1.mkv" 2:"C:\Temp\VideoSynch\Vsy_Tmp_Audio_1.ac3" -320 -25.000 -changeTo23.976
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 1 subtitle track, 0:22:43, 25p
1: h264/AVC, 1080p24 /1.001 (16:9)
2: MP3, Spanish, 2.0 channels, 320kbps, 48kHz, 1500ms
"Audio en Castellano"
3: Subtitle (ASS), Spanish, "Subtítulos para el audio Castellano"
[v01] The video bitstream framerate field doesn't match the container framerate. <WARNING>
[a02] Extracting audio track number 2...
[a02] Decoding with libav/ffmpeg...
[a02] Applying RAW/PCM delay...
[a02] Encoding AC3 <320kbps> with libAften...
[a02] Clipping detected, a 2nd pass will be necessary. <WARNING>
[a02] Creating file "C:\Temp\VideoSynch\Vsy_Tmp_Audio_1.ac3"...
[a02] Starting 2nd pass...
[a02] Extracting audio track number 2...
[a02] Decoding with libav/ffmpeg...
[a02] Applying RAW/PCM delay...
[a02] Encoding AC3 <320kbps> with libAften...
[a02] Applying -0,21dB gain...
[a02] Creating file "C:\Temp\VideoSynch\Vsy_Tmp_Audio_1.ac3"...
Video track 1 contains 34071 frames.
eac3to processing took 23 seconds.
Done.

You can see I am trying to convert a mp3 track from a mkv video from 25.000 to 23.976. But the video is a 23.976 avc1 encode, converted by the mkv container to a 25.000 play speed.
Eac3to doesn't does the framerate conversion.

The workaround is simple: extract and convert after.

dream88

9th December 2011, 18:58

Does anyone have Nero 7 decoders working on Win7x64, and if so, could you please provide working instructions?

Despite asking some time ago, and having someone post advice, it doesn't work. Trying to run Regsvr32 on AdvrCntr2.dll works fine; running it on NeAudio2.ax fails (regsvr32 hangs and then crashes); and running it on NeEacDec.dll fails, it reports it can't find an entry-point DllRegisterServer.

I have installed a retail version of Nero7Ultra with retail BluRay/HDDVD plugin and no go.

Many thanks
The REAL Joe
Hi i have made an sfx for eac3to with nero 7 , surcode, arcsoft all in one setup only 8mb, without installing nero/surcode etc, not sure if i could post it here though, i made it for 64 bit too.

For nero 7 the only files you need are ( advrcntr2.dll msvcp71.dll msvcr71.dll NeAudio2.ax NeEacDec.dll ) and the correct registry keys from your nero 7.

Windows Registry Editor Version 5.00

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Ahead]

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Ahead\Installation]

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Ahead\Installation\Families]

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Ahead\Installation\Families\Nero 7]

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Ahead\Installation\Families\Nero 7\Info]
"Company"=""
"EulaAccepted"="x"
"MissingFilesState"="0"
"Serial7_xxxxxxxxxxxx"="xxxx-xxxx-xxxx-xxxx-xxxx-xxxx-xxxx"
"User"=""

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Ahead\Installation\Families\Plugins]

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Ahead\Installation\Families\Plugins\Info]
"Serial7_xxxxxxxxxxx"="*xxxx-xxxx-xxxx-xxxx-xxxx-xxxx-xxxx"

For surcode you need the ( surcodedvd.exe Tables folder and lservrc ) and correct registry keys for surcode.

Windows Registry Editor Version 5.00

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Minnetonka Audio Software]
@=""

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Minnetonka Audio Software\SurCode DVD-DTS]
"Home"="C:\\(path to folder)\\Surcode DTS Encoder"

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Minnetonka Audio Software\SurCode DVD-DTS\1.0.10]
@=""

For arcsoft you need ( ASAudioHD.ax checkactivate.dll DtsDec.dll dtsdecoderdll.dll MagCore.dll MagPCMac.dll MagUIEngine.dll MagUIInter.dll msvcp71.dll msvcr71.dll ).

A copy of msvcp71.dll & msvcr71.dll are required in the same folder as nero 7, arcsoft & eac3to folder.

I worked out how everything works and put a package together myself with all these files.

phate89

12th December 2011, 06:04

fano

12th December 2011, 10:10

TDiTP_

12th December 2011, 10:47

hi..there's a way to make it work eac3to with audio that have different channels? for example an ac3 or dts that have the start on 2.0 and the rest of 5.1? because eac3to recognise it as 2.0... and destroy the audio (3 times longer)... to pass ac3s to wav i use besweet that solve the problem but i don't know how to do with dts
For AC3 you can use SplitAC3. Look it here: http://forum.doom9.org/showthread.php?p=1447695#post1447695. I don't know how besweet works, but AFAIR Azid (used in besweet) can't normal decode AC3 with 2 kind of frames.

For DTS you can use LeeAudBi (http://forum.doom9.org/showthread.php?p=1522330#post1522330) to know numbers of "different" frames and then cut them by using delaycut (http://forum.doom9.org/showthread.php?t=162984). Of course you should not forget about the delay.

It's possible to transcode a multichannel FLAC (or whatever AAC, MP3 and so on...) to Dolby True HD o DTS HD MA?
Yes, but you need:
"Dolby Media Producer Suite" (it's only for MAC) for TrueHD
"DTS-HD Master Audio Suite" for DTS HD MA
It's propietary software, not cheap. :)

If not it's feasible to add?
i don't think so.

Sparktank

12th December 2011, 10:54

It's possible to transcode a multichannel FLAC (or whatever AAC, MP3 and so on...) to Dolby True HD o DTS HD MA?
If not it's feasible to add?
It's not important in this phase if a commercial software it's needed ;)

Thanks for the response...

Multichannel FLAC to DTS-HD MA is possible.
With commercial software.

Multichannel AAC... yeah, but it's a lossy source so not really worth the effort.

I don't think Dolby TrueHD is possible yet.

Check the first page for more functions.
Or run this command in eac3to and you'll see what you can do and what is recommended...

path\eac3to.exe -log=Help.txt
(The -log will save the results in a new txt file named Help.
I like to keep it handy for future references.)

Sparktank

12th December 2011, 10:55

Yes, but you need:
"Dolby Media Producer Suite" (it's only for MAC) for TrueHD

:( I remember now... MAC... :(

fano

12th December 2011, 11:24

Multichannel FLAC to DTS-HD MA is possible.
With commercial software.

What's its name? EAC3TO supports it?

Multichannel AAC... yeah, but it's a lossy source so not really worth the effort.

Can be useful indeed, if, for example, your system does not support
Kernel Streaming or, as in my case, it's broken to have bit-perfect-audio... that is as it becomes DTS MA; windows cannot mess whit it and only your HD AV receiver can decode it ;)

I don't think Dolby TrueHD is possible yet.

Well for that you want to do, it's unimportant :D

Thanks for your FAST response :goodpost:

P.S @TDiTP_ not saw your post, thanks to you, too !

phate89

12th December 2011, 13:34

For AC3 you can use SplitAC3. Look it here: http://forum.doom9.org/showthread.php?p=1447695#post1447695. I don't know how besweet works, but AFAIR Azid (used in besweet) can't normal decode AC3 with 2 kind of frames.

For DTS you can use LeeAudBi (http://forum.doom9.org/showthread.php?p=1522330#post1522330) to know numbers of "different" frames and then cut them by using delaycut (http://forum.doom9.org/showthread.php?t=162984). Of course you should not forget about the delay.

i don't think so.

thanks i didn't know them.. i'll try with that... actually besweet create a full 5.1 wav putting the 2 stereo channels in first 2 channel of 5.1 and leave all the other 4 empty... it's a good compromise to allow converting the file without problems

EDIT:very good tools| dotheir work perfectly! who create this tools?

TDiTP_

12th December 2011, 14:06

EDIT:very good tools| dotheir work perfectly! who create this tools?
tebasuna51, as far as i know. if you're talking about LeeAudBi and SplitAC3.

Lincoln Burrows

12th December 2011, 18:03

How do I convert from WAV to MP3? It says this conversion is not supported by eac3to.

http://i.imgur.com/fxL8W.png

kypec

12th December 2011, 18:29

How do I convert from WAV to MP3? It says this conversion is not supported by eac3to.
See this post (http://forum.doom9.org/showthread.php?p=1464996#post1464996) how to pipe output from eac3to into about any CLI encoder. ;)

Lincoln Burrows

12th December 2011, 18:35

See this post (http://forum.doom9.org/showthread.php?p=1464996#post1464996) how to pipe output from eac3to into about any CLI encoder. ;)That command is not working.

eac3to input stdout.wav [-down2 -normalize] | Lame -V 4 - output.mp3

The command line parameter V is unknown.

the_weirdo

12th December 2011, 18:46

That command is not working.

eac3to input stdout.wav [-down2 -normalize] | Lame -V 4 - output.mp3

The command line parameter V is unknown.

Those parameters in "[]" brackets mean they are optional. If you want to downmix from multichannel to stereo then you need to add them. If input is stereo, you'll need only:
eac3to input stdout.wav | Lame -V 4 - output.mp3

Adjust LAME parameters according to your need.

Lincoln Burrows

12th December 2011, 19:02

Brazil2

12th December 2011, 19:12

sneaker_ger

12th December 2011, 19:26

It's still saying "Warning: unsupported audio format". I have no idea why.

In case you want to verify, here's the WAV file:
http://www.mediafire.com/?gzr4x6dzca91f8n

(Don't worry, it's not copyrighted)

Working fine here, both:
eac3to out.wav stdout.wav | lame -V 4 - output.mp3
lame -V 4 out.wav output.mp3

I'm using eac3to 3.24 and lame 3.99.3 (64 bit) from rarewares.org.

the_weirdo

12th December 2011, 19:26

It's still saying "Warning: unsupported audio format". I have no idea why.

In case you want to verify, here's the WAV file:
http://www.mediafire.com/?gzr4x6dzca91f8n

(Don't worry, it's not copyrighted)

If your source is already a PCM WAV then you don't need eac3to to decode it, just use it as input to LAME:
LAME -V 4 out.wav output.mp3

fano

13th December 2011, 09:36

I've another question DTS-HD Master Audio Suite convert directly flac audio or need particular formats?

It needs perhaps a multichannel wav or worst 6 mono wav file?

eac3to has this features?

tebasuna51

13th December 2011, 10:29

DTS-HD Master Audio Suite needs mono wavs and I don't know if is possible to automate the process, at least is not possible with the last eac3to version.

Decode the flac to mono wavs and add them to DTS-HD Master Audio Suite manually.

eac3to input.flac output.wavs

fano

13th December 2011, 13:13

eac3to input.flac output.wavs

Let see if I've understood correctly this give a wav for all channel or (i Hope no) a multichannel wav?

phate89

13th December 2011, 14:11

Let see if I've understood correctly this give a wav for all channel or (i Hope no) a multichannel wav?

it creates a mono wave file for each channel (6 channels, 6 wavs)

fano

13th December 2011, 14:31

OK, thanks :goodpost:

Tonight I'll try and let you know what happened :D

For DTS-HD Master Audio Suite I've first to find a trial and if it's a normal windows app (I fear not as I've seen on site the photos and it seems the same for Wind & Mac... I suspect Java :( ) a little of AutoIt scripting can solve the problem :p

mbcd

13th December 2011, 18:31

You wont find trial of DTSMS, there is no one, and yes, its java-based.

Would be interesting to get a possibility to use it with eac3to ...:sly:

NanoBot

13th December 2011, 22:52

It's possible to transcode a multichannel FLAC (or whatever AAC, MP3 and so on...) to Dolby True HD o DTS HD MA?
If not it's feasible to add? It's not important in this phase if a commercial software it's needed ;)
Thanks for the response...

I cannot follow why anybody would like to convert a free and better compressible codec ( flac ) to an unfree propertiery and less compressible codec ( DTS-MA / TrueHD ) ?

When I am ripping and remuxxing BDs to mkv, I always go the different way converting the lossless audio tracks to flac, because the filesize of the flac-file is 10% - 20% smaller then the original DTS-MA or TrueHD file.

Perhaps it makes the difference that I am using only my PC, which is connected through an HDMI cable to my AV amp, to playback media files. Therefore using flac is the optimal solution for me.

C.U. NanoBot

Midzuki

14th December 2011, 02:31

@ tebasuna51: the Monster Audio :D Encoder Suite accepts up to 4 stereo .WAVs as a valid input

@ fano: the DTSHDMAS stuff requires the MSVC7 Runtimes + the JRE thing; the so-called StreamPlayer requires Asio4All.

Overdrive80

14th December 2011, 17:17

Hi guys, i have a pair questions.

In main post, for solving issue with arcsoft says that is necessary fix environment path. However, i had installed and fixed path C:\Program Files\Common Files\ArcSoft\Bin but in this path dont exists nothing. Maybe, should I add the arcsoft codec directory, or not??

I have installed nero 7, and registered plugin too, but not detected its. Any solution??

Thanks

EDIT: Solve problem with ArcSoft using regsvr32. ^^

EDIT2: http://www.cnpdb.com/down/2006/d1270.shtml Sonic 4.3. It isnt ilegal download or cracked.

chainring

14th December 2011, 19:01

I cannot follow why anybody would like to convert a free and better compressible codec ( flac ) to an unfree propertiery and less compressible codec ( DTS-MA / TrueHD ) ?Bitstreaming the core portion over SPDIF, perhaps?

NanoBot

14th December 2011, 23:56

Maybe, but in this case reencoding from flac to DTS-MA oder TrueHD would not be necessary, reencoding the DTS core or AC3 core would be sufficient. And indeed, those files would be smaller than using flac. So if anybody uses a standalone media player and / or an AV-Amp, which supports only spdif, I can follow this way.

Nevertheless, when using a PC for playback, the reencoding to the ac3 / dts core could also be done on-the-fly using ffdshow or DTS interactive, if the mobo supports it. I used the second method before I got my new AV amp with HDMI-inputs. My old AV amp only had the spdif input for multichannel sound formats, it does not even had an analog 5.1 RCA input.

But the main reason for my interjection was to state my point of view that the usage of codecs with a well known brand name in many cases is inferior to the usage of a patent and licence free codec like flac. In other words, I cannot see why DTS-MA and / or TrueHD has ever been used by the industrie instead of flac, which would have saved the manufacters the license fees.

C.U. NanoBot

Overdrive80

14th December 2011, 23:59

EDIT: Solved with nero micro/lite

fano

15th December 2011, 13:06

I cannot follow why anybody would like to convert a free and better compressible codec ( flac ) to an unfree propertiery and less compressible codec ( DTS-MA / TrueHD ) ?

Well the problem is that my Mediacenter is used as a PC and, you know, when you've an opened application she likes to do "click" or play a sound when ypu're playing a movie in DTS HD it's not pleasent as you know imagine :mad:

The idea to convert all (MP3, AAC, AC3 and so on) in one of the commercial lossless in no in a open source amatorial (but of better quality, I know :rolleyes:) it's to have the guarentee Windows (as you send the bitstream to your receiver) and it's shitty apps don't mess with MY audio :eek:

Yes I know you'd use Kernel Streaming / Wasapi but if they don't work?
I've to hear the clicks... no I've to encode in a lossless commercial codec and send via HDMI the bitstream to the receiver... as I've to do now via SPDIF with AC3 ;)

Right?
So... let's do it :thanks:

NanoBot

15th December 2011, 16:48

fano

15th December 2011, 20:48

Ok, now I can understand the reason why you want to convert to TrueHD or DTS-MA. I don't have the first problem at all, because I am using a PC monitor and a HDTV in Nvidias Dualview setup. That makes it possible to use the onboard realtek soundchip as default soundcard, while only the sound of the media files is played back through the HDMI "soundcard". So I never hear the clicks, alert sounds or something like that through the AV amp, they are only heard through the desktop stereo speakers sideways the PC monitor.

In your opinion I could do the same?
That is:

SPDIF output as default sound card for PCist thing (Browsing, Windows Click&Piip, etc...) connected to an optical receiver input port
HDMI as audio card for Mediaportal, foobar, VLC, and so on... connected to the same reeciver using an HDMI port...that is a port dedicated for MediaCenter thing?

And for the second reason, obviously I am not an audiophile, since the quality of 6 channel LPCM send through HDMI to the AV amp without Wasapi / kernel streaming sounds good enough for me. I simply would not notice if the sound would be slightly modified e.g. by the windows mixer. E.g. I do not even notice any quality difference between 16bit / 48kHz and 24bit / 96kHz, my ears are to old for that *g*

Maybe I, too, can't hear Windows' bugged audio mixer messing (but I can hear the click and piiip), but call me mad I KNOW IT CAN do this and I don't see the movie anymore as I'd say: bello paciarotto (tranquil, serene understood?), I try to hear the clicks instead of try to understand the story...

A simpatic example (repeating all nights, so no such humorous for me anymore :rolleyes: )
Actor1: <<Jack is dead>>
Actor2: <<Poor Jack! Such a great man!>>
fano(ME): <<Who's Jack :confused: ? ... and when (s)he died? Ohh piece of Sh1t, it clicked :mad::mad::mad: !

Hoping my receiver came in the next week to do these test suggested by you :cool:

Thanks for your help :D

TDiTP_

16th December 2011, 08:25

i've tested decoding of TrueHD 7.1-layouts by eac3to and ffmpeg. May be it will be usefull. So..

Eight test wavs were encoding to THD by "Dolby Media Producer Suite" to seven available 7.1-layouts. THDs were decoding by:
- eac3to: eac3to input.thd output.wavs -libav
- ffmpeg (git-7d531e8 32-bit Static (http://ffmpeg.zeranoe.com/builds/win32/static/), 2011.12.12): ffmpeg -i input.thd -acodec pcm_s24le output.wav

Some channels were swapped for some configurations. L,C,R and LFE channels stayed in the correct positions always, therefore i would not write about them.
Here (http://img-fotki.yandex.ru/get/4422/79369042.0/0_77053_89fd1c63_XL.jpg) Dolby's channel names.

All samples (input wavs and output THDs with logs of encoding) you can find here (http://www.mediafire.com/?2tyhc55zt14x4yu).

=============================================
Results:

0).
http://img-fotki.yandex.ru/get/5314/79369042.0/0_77046_f63abb03_XS.jpg (http://fotki.yandex.ru/users/sir-trebushat/view/487494/)
output (eac3to) = output (ffmpeg) byte-in-byte. MaskChannels : 1599 (FL FR FC LF BL BR SL SR)

input eac3to ffmpeg

Lrs -> BL BL
Rrs -> BR BR
Ls -> SL SL
Rs -> SR SR

Properly decoding :)

1).
http://img-fotki.yandex.ru/get/4527/79369042.0/0_77047_439a0304_XS.jpg (http://fotki.yandex.ru/users/sir-trebushat/view/487495/)
MaskChannels (eac3to/libav) : 1599 (FL FR FC LF BL BR SL SR)
MaskChannels (ffmpeg) : 1743 (FL FR FC LF FLC FRC SL SR)

input eac3to ffmpeg

Lc -> BL SL
Rc -> BR SR
Ls -> SL LC
Rs -> SR RC

No one decodes properly: eac3to uses incorrect mask, ffmpeg swaps the channels.

2).
http://img-fotki.yandex.ru/get/5822/79369042.0/0_77048_5fad514c_XS.jpg (http://fotki.yandex.ru/users/sir-trebushat/view/487496/)
MaskChannels (eac3to/libav) : 1599 (FL FR FC LF BL BR SL SR)
MaskChannels (ffmpeg) : 22031 (FL FR FC LF SL SR TFL TFR)

input eac3to ffmpeg

Ls -> SL SL
Lvh -> BL TL
Rs -> SR SR
Rvh -> BR TR

eac3to uses incorrect mask, ffmpeg decodes absolutely properly.

3).
http://img-fotki.yandex.ru/get/5822/79369042.0/0_77049_317e27c0_XS.jpg (http://fotki.yandex.ru/users/sir-trebushat/view/487497/)
MaskChannels (eac3to/libav) : 1599 (FL FR FC LF BL BR SL SR)
MaskChannels (ffmpeg) : 1743 (FL FR FC LF FLC FRC SL SR)

input eac3to ffmpeg

Rw -> BR SR
Lw -> BL SL
Rs -> SR RC
Ls -> SL LC

eac3to uses incorrect mask. ffmpeg uses correct mask (or not?) but swaps the channels.

4).
http://img-fotki.yandex.ru/get/4714/79369042.0/0_7704a_53d6d052_XS.jpg (http://fotki.yandex.ru/users/sir-trebushat/view/487498/)
MaskChannels (eac3to/libav) : 11791 (FL FR FC LF SL SR TC TFC)
MaskChannels (ffmpeg) : 11791 (FL FR FC LF SL SR TC TFC)

input eac3to ffmpeg

Cvh -> TFC TFC
Ls -> SL SL
Rs -> SR SR
Ts -> TC TC

eac3to and ffmpeg decodes absolutely properly.

5).
http://img-fotki.yandex.ru/get/4527/79369042.0/0_7704b_fc87784c_XS.jpg (http://fotki.yandex.ru/users/sir-trebushat/view/487499/)
MaskChannels (eac3to/libav) : 3855 (FL FR FC LF BC SL SR TC)
MaskChannels (ffmpeg) : 3855 (FL FR FC LF BC SL SR TC)

input eac3to ffmpeg

Ls -> BC BC
Cs -> SR SR
Rs -> SL SL
Ts -> TC TC

eac3to and ffmpeg uses correct mask but swaps the channels.

6).
http://img-fotki.yandex.ru/get/4526/79369042.0/0_7704c_1260b30a_XS.jpg (http://fotki.yandex.ru/users/sir-trebushat/view/487500/)
MaskChannels (eac3to/libav) : 9999 (FL FR FC LF BC SL SR TFC)
MaskChannels (ffmpeg-last) : 9999 (FL FR FC LF BC SL SR TFC)

input eac3to ffmpeg

Rs -> SL SL
Ls -> BC BC
Cs -> SR SR
Cvh -> TFC TFC

eac3to and ffmpeg uses correct mask but swaps the channels.

=============================================
Conclusions:

eac3to.exe can properly decode only configurations 0 and 4.
ffmpeg.exe can properly decode only configurations 0, 2 and 4.

UPD. Lav Audio 0.42 has exactly the same problems as ffmpeg (all dump files identical byte-in-byte).

UPD2 (20.12.2011). ffmpeg was patched, now all samples is decoded correctly: 7.1 from this comparison and all three 6.1-layouts (there were problems too).

fano

16th December 2011, 22:48

.TDiTP_ thanks for your great work :thanks:

As it seems my receiver will be here the next week all try to do something next Sunday after this Sunday :D

I've to a lot of test :eek:
When I'll use it for something useful?
It's possile to see a grteat film a "I Pirati Dei Caraibi" (The Pirates of the Caribbean) and say the good a/v quality?

A propose who's Jack Sparrow :p ?

nevcairiel

16th December 2011, 23:08

UPD. Lav Audio 0.42 has exactly the same problems as ffmpeg (all dump files identical byte-in-byte).

I fixed cases 1 and 3 in LAV Audio, and i guess the changes will find their way into normal ffmpeg as well "soon".

fano

18th December 2011, 13:25

A curiosity (partially OT, maybe) but I've a Walt Disney's Fantasia Blu Ray legally ripped on my PC and again I've not an HD receiver, Mediaportal reads the .m2ts and shows the first audio track as DTS-HD MA (and I play, for now, as an AC3 as my old receiver has not DTS, too)... the strange thing is that if I change track to "Italian" I've the impression to see no real change (when the man present the tracks) on audio no interruption :confused:

It's possible Disney does the right thing? The "SD" track are only for the talked part but then after when the song plays DTS-MA international track automatically plays?
Or it's a Mediaportal bug?

This the MediaInfo Report:

General
ID : 0
Complete name : *:\*********************\BDMV\STREAM\00125.m2ts
Format : BDAV
Format/Info : Blu-ray Video
File size : 35.1 GiB
Duration : 2h 4mn
Overall bit rate : 40.3 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
Duration : 2h 4mn
Bit rate mode : Variable
Maximum bit rate : 27.5 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
Color primaries : BT.709-5, BT.1361, IEC 61966-2-4, SMPTE RP177
Transfer characteristics : BT.709-5, BT.1361
Matrix coefficients : BT.709-5, BT.1361, IEC 61966-2-4 709, SMPTE RP177

Audio #1
ID : 4352 (0x1100)
Menu ID : 1 (0x1)
Format : DTS
Format/Info : Digital Theater Systems
Format profile : MA
Muxing mode : Stream extension
Duration : 2h 4mn
Bit rate mode : Variable
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

Audio #2
ID : 4353 (0x1101)
Menu ID : 1 (0x1)
Format : DTS
Format/Info : Digital Theater Systems
Format profile : MA
Muxing mode : Stream extension
Duration : 2h 4mn
Bit rate mode : Variable
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

Audio #3
ID : 4354 (0x1102)
Menu ID : 1 (0x1)
Format : DTS
Format/Info : Digital Theater Systems
Duration : 2h 4mn
Bit rate mode : Constant
Bit rate : 1 510 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 24 bits
Stream size : 1.31 GiB (4%)

Audio #4
ID : 4355 (0x1103)
Menu ID : 1 (0x1)
Format : AC-3
Format/Info : Audio Coding 3
Mode extension : CM (complete main)
Duration : 2h 4mn
Bit rate mode : Constant
Bit rate : 640 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Stream size : 570 MiB (2%)

Audio #5
ID : 4356 (0x1104)
Menu ID : 1 (0x1)
Format : AC-3
Format/Info : Audio Coding 3
Mode extension : CM (complete main)
Duration : 2h 4mn
Bit rate mode : Constant
Bit rate : 640 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Stream size : 570 MiB (2%)

Audio #6
ID : 4357 (0x1105)
Menu ID : 1 (0x1)
Format : AC-3
Format/Info : Audio Coding 3
Mode extension : CM (complete main)
Duration : 2h 4mn
Bit rate mode : Constant
Bit rate : 640 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Stream size : 570 MiB (2%)

CUT THERE ARE 6 other audio Tracks
Text #1
ID : 4608 (0x1200)
Menu ID : 1 (0x1)
Format : PGS

Text #2
ID : 4609 (0x1201)
Menu ID : 1 (0x1)
Format : PGS

Text #3
ID : 4610 (0x1202)
Menu ID : 1 (0x1)
Format : PGS

CUT THERE ARE 6 other text Tracks

In particular what's a "DTS MA Stream extension" ?
If it is done in this way as I suppose (the secondary tracks contains only talks, the audio part is DTS-MA Stream extension) eac3 can create an unique DTS-HD Ma track with the Italian audio (or Spanish or Turkish or whatever) and the DTS-HD MA audio track for the musical part?
What do you think ;)?

tebasuna51

18th December 2011, 18:28

A curiosity (partially OT, maybe)
Yes is OT.
Please open new threads and let in peace this one.
The "SD" track are only for the talked part but then after when the song plays DTS-MA international track automatically plays?
No

eac3 can create an unique DTS-HD Ma track with the Italian audio (or Spanish or Turkish or whatever) and the DTS-HD MA audio track for the musical part?
No

kws53

20th December 2011, 02:01

This has been reported by other folks [Vilous] on other forums. In my case, I've tracked down the problem to a drive that was not formatted by the computer with EAC3TO installed.

I have identical WD 2TB drives. I loaded one drive with ripped HD-DVD [HVDVD_TS] and BluRay [BDMV] files. This drive was formatted using my laptop [attached via USB and external power supply].

While I can demux the files with no problems to VC1 [after attaching the drive to my desktop via USB], when muxing the VC1 to MKV, the process freezes at 2688K [always at 2688 - I've tried four different movies, both BluRay and HD-DVD sourced]. Yet when I move the VC1 file to another identical WD 2TB drive [formatted on my desktop], EAC3TO has no problem with the muxing.

I do not feel like moving all of these files to a locally formatted drive, and then muxing, since there must be an solution for this that can be quickly implemented. It seems that there is an underlying header on the drive that EAC3TO is reading which stops the process.

In case folks think that the USB is the problem, I've used the USB for muxing VC1 files with no problems on locally formatted drives already. The only variable that is different is the drive formatting.

Thanks in advance.

Kurt

kws53

20th December 2011, 14:01

As a follow-up, it appears that "writing" an mkv file to a foreign [not formatted by your computer] drive is the problem. Something in the header that is not being accepted?

Kurt

Snowknight26

20th December 2011, 15:57

It (not being able to mux MKVs to certain drives) is a known issue with the Haali Matroska Muxer (dsmux). Don't hope for it every being fixed.

Abradoks

21st December 2011, 15:51

When remuxing wav > 4 GB to w64 eac3to sets duration of w64 file to the incorrect value found in original wav file (instead of adjusting it according to file size). It will result in truncation of file if eac3to tries to do a second pass.

kws53

21st December 2011, 16:26

It (not being able to mux MKVs to certain drives) is a known issue with the Haali Matroska Muxer (dsmux). Don't hope for it every being fixed.
Eac3to does not display the requisite line "Added fps value to MKV header" which indicates that it [or haali] locks up attempting to do this phase. Presumably there is some drive check being done which fails - and no error message. Any ideas on a work around?

Kurt

Snowknight26

21st December 2011, 17:00

Mux to a different drive or use mkvmerge.

doom-nine

5th January 2012, 04:41

Snowknight26

5th January 2012, 04:43

Not with this program. Try WaveWizard, Audacity, etc.

ramicio

5th January 2012, 04:53

doom-nine

5th January 2012, 07:58

I really don't get why eac3to can't do this yet. It seems to be finalized and never to be updated again. One could just specify the input using the ".wavs" extension, specify the number of channels, or just use a proprietary format where you just enter the list of files into a text file, and have a switch to specify a channel mask.
Thanks. I tried to name the wav files into 1.wav, 2.wav..... also a1.wav, a2.wav.... But all failed. eac3to cannot specify the input files.

Not with this program. Try WaveWizard, Audacity, etc.
Well, I think you're right. eac3to cannot encode multi wav files into lpcm. Thanks.

Thunderbolt8

5th January 2012, 11:16

I really don't get why eac3to can't do this yet. It seems to be finalized and never to be updated again.
it would be nice if madshi would bring out an update/bugfix again ;)

ramicio

5th January 2012, 14:25

A nice feature to also add would be to allow non-integers for gain. Maybe a maximum of 2 decimal points (to the one-hundredths). At least tenths is necessary.

Superb

11th January 2012, 10:42

How do i demux a VobSub subtitle track from a mkv file?

Tried:
eac3to v3.24
command line: d:\eac3to\eac3to.exe c:\new\title00.mkv 10: subtitle.sub
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 19 subtitle tracks, 2:11:40, 50i
1: MPEG2, 576i50 (4:3)
2: AC3, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
"3/2+1"
3: Subtitle (VobSub)
4: Subtitle (VobSub), French
5: Subtitle (VobSub), Spanish
6: Subtitle (VobSub), Dutch
7: Subtitle (VobSub), Swedish
8: Subtitle (VobSub), Finnish
9: Subtitle (VobSub), Norwegian
10: Subtitle (VobSub), Danish
11: Subtitle (VobSub), Portuguese
12: Subtitle (VobSub), Modern Greek
13: Subtitle (VobSub), Hungarian
14: Subtitle (VobSub), iw
15: Subtitle (VobSub), Slovenian
16: Subtitle (VobSub), Croatian
17: Subtitle (VobSub), Bulgarian
18: Subtitle (VobSub)
19: Subtitle (VobSub), French
20: Subtitle (VobSub), Spanish
21: Subtitle (VobSub), Dutch
Bitstream parsing for tracks 3-21 failed. <WARNING>
Demuxing these tracks may still produce correct results - or not. <WARNING>
This subtitle conversion is not supported. <ERROR>
-demux didn't work as well (demuxed only the video and audio)...

geminigod

11th January 2012, 15:35

Thanks for the excellent software. Now I got a question about encoding.
As the introduction says, eac3to can encode audios to (L)PCM. I have 6 pcm wav files that represent 5.1 channels. Then how can I encode these 6 wav files into 1 LPCM file? I cannot find any instruction about how to do that.
Can anyone provide any instruction? Thanks.

It is a shame there is no command line option to add channels based on SMPTE order. Considering the program is able to do this in the background when converting between various formats, I can't imagine it would be too hard to program...

tebasuna51

11th January 2012, 22:47

How do i demux a VobSub subtitle track from a mkv file?

eac3to don't support extract VobSub subtitle from a mkv.
Use MkvExtract

geminigod

13th January 2012, 00:10

This is mildly off topic, but does anyone know of a program that allows one to make ac3 metadata changes, specifically for dialogue normalization and dynamic range compression? eac3to can remove the metadata, but can't change it. I have a file that I would like to change the DRC from film light to film standard. The dialogue is just too quiet in it.

TDiTP_

13th January 2012, 00:52

Abradoks

13th January 2012, 13:11

You can't change DRC without reencoding
It isn't simple, but still possible. jruggle proposed (http://forum.doom9.org/showthread.php?t=143416) such tool some time ago.

geminigod

14th January 2012, 06:20

tebasuna51

14th January 2012, 10:19

It is definitely theoretically possible, but I guess nobody has ever cared enough to delve into it? I don't think DRC was ever a well enough understood topic by the general public or well enough supported with commercial software and hardware. Now it is a dying piece of tech anyway, so maybe it doesn't make sense for somebody to work on.

I don't think so.

Play with DRC enabled lose the original quality of audio.

Was a help for old amplifiers/receivers without the actual process capacity than can replace DRC with Night Mode for all audio formats, not only AC3.

BTW, if you play with PC you can use Ac3Filter with a dynamic DRC function, or you can recode the audio with BeLight with a Boost function.

rapscallion

14th January 2012, 20:33

A little Ot but hopefully you folks can answer a "delay" question for me. (first time I've seen a delay on a BD)

I have a retail dl bd movie (Home Alone) that I want to re encode down to a 25 gb bd.
MediaInfo shows that there is a 6 ms video delay.

I extracted the main *.m2ts (32 GB) via Tsmuxer and then the DTS-HD track via eac3to.

Eac3to log states: "applying DTS delay" and when finished "A remaining delay of +2ms could not be fixed"

After I run the *.m2ts through Megui and remux with the DTS-HD track, via Tsmuxer, the question is: What delay, if any do I apply ?

tebasuna51

14th January 2012, 21:48

And wath is the delay than show eac3to?

rapscallion

14th January 2012, 21:55

And wath is the delay than show eac3to?
Oh, yes, sorry forgot that . This is the track before eac3to processing:

DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz, -8ms
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)

That delay is 2ms different than reported by MediaInfo (-6)

This is what is confusing the issue for me, when extracting: "Applying DTS delay...a remaining delay of +2ms could not be fixed"
What exactly does that mean ? :confused:

tebasuna51

15th January 2012, 01:27

"Applying DTS delay...a remaining delay of +2ms could not be fixed"
What exactly does that mean ? :confused:

eac3to (or Delaycut) can only add/cut DTS frames, and each frame of 5.1 channels, 1509kbps, 48kHz have 10.6 ms.

With your sample eac3to add a silence frame then now the delay is:

10 - 8 = +2

But don't worry, like a video frame is 41.7 ms long this delay is unnoticeable.

But the delay is:

MediaInfo shows that there is a 6 sec video delay.

or 6 ms?

rapscallion

15th January 2012, 18:13

Thank you very much for the explanation ! eac3to really is an amazing program.

I've edited the post to reflect 6 ms, which was what I meant. 6 sec would be some delay !!

geminigod

15th January 2012, 23:58

I don't think so.

Play with DRC enabled lose the original quality of audio.

Was a help for old amplifiers/receivers without the actual process capacity than can replace DRC with Night Mode for all audio formats, not only AC3.

BTW, if you play with PC you can use Ac3Filter with a dynamic DRC function, or you can recode the audio with BeLight with a Boost function.

I wish my receiver had a "night mode". Even some of the new receivers out there are retarded about the signal processing options. My father just bought a new Denon receiver that sucks. There are a bazillion useless ways to screw with the EQ, all of which make the sound worse, and no way to process dynamic range.

At any rate, no worries. I just have an ac3 audio track that I have since deleted the source material for. I encoded it with a Film Light DRC setting and decided that Film Standard would be preferrable. It isn't worth transcoding to change it though.

An IP BreAKDoWN

16th January 2012, 07:58

Are mono audio files still not supported? I'm trying to rip a mono audio file that is in thd to a dts. Does anyone know how to do this if eac3to can't?

kalston

17th January 2012, 11:55

I converted a mono DTS file to FLAC without any problem.

An IP BreAKDoWN

17th January 2012, 18:06

How about a mono THD to DTS (FLAC, or WAV for that matter)?

geminigod

19th January 2012, 10:03

Are mono audio files still not supported? I'm trying to rip a mono audio file that is in thd to a dts. Does anyone know how to do this if eac3to can't?

I can't remember the specifics of whether surecode supports mono dts encoding or not, assuming you even have this $250 codec setup in your eac3to. Try converting to wav first and then to dts. See if it works then.

More likely your eac3to is not setup to properly decode or encode dts. Type eac3to -test at the command prompt and see what it says.

I could ask why you are trying to make a mono dts file... :scared: Stereo acoustics will sound much better, even if you are just duplicating the mono track.

If you are determined to work with this challenging format (due to commercial copyright issues), you can play around with this brand new experimental open source dts codec that folks here have been working on.

http://forum.doom9.org/showthread.php?p=1552620#post1552620

kws53

19th January 2012, 18:11

I can't get eac3to to downconvert a 3 channel [L/R/C] DTS audio stream. Using -down2 switch does not solve it. The output is still 3 channels.

Kurt

An IP BreAKDoWN

19th January 2012, 20:17

I can't remember the specifics of whether surecode supports mono dts encoding or not, assuming you even have this $250 codec setup in your eac3to. Try converting to wav first and then to dts. See if it works then.

More likely your eac3to is not setup to properly decode or encode dts. Type eac3to -test at the command prompt and see what it says.

I could ask why you are trying to make a mono dts file... :scared: Stereo acoustics will sound much better, even if you are just duplicating the mono track.

If you are determined to work with this challenging format (due to commercial copyright issues), you can play around with this brand new experimental open source dts codec that folks here have been working on.

http://forum.doom9.org/showthread.php?p=1552620#post1552620

Yes I have Surecode, it set me back but I have it installed. The reason I'm trying to convert to mono is because the source is mono in THD format.

Here is that command you told me to use:

eac3to v3.24
command line: eac3to -test
------------------------------------------------------------------------------
eac3to (v3.24) is up to date
Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed
http://www.nero.com/eng/store-blu-ray.html
CAUTION: You need Nero 7. Nero 8 won't work with eac3to.
ArcSoft DTS Decoder (1.1.0.1) works fine
Sonic Audio Decoder (3.24.0.0) doesn't seem to be installed
Haali Matroska Muxer (2011-09-08) is up to date
Nero AAC Encoder could not be located
http://www.nero.com/eng/nero-aac-codec.html
Copy NeroAacEnc.exe to the eac3to or to the Windows folder.
Surcode DTS Encoder (1.0.29.0) is installed
MkvToolnix (2.5.3.0, release version) is installed
There's a new release version (5.2.1.0) available
http://www.bunkus.org/videotools/mkvtoolnix
There's a new beta version (5.2.1.0, 2012-01-13) available
http://www.bunkus.org/videotools/mkvtoolnix/win32/pre

And here is the error I am having when I try to rip it to .dts

eac3to v3.24
command line: eac3to 00001.m2ts 4: japanese-1.0.dts
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 2 subtitle tracks, 0:24:13, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, 5.1 channels, 48kHz, dialnorm: -25dB
(embedded: AC3, 5.1 channels, 448kbps, 48kHz, dialnorm: -25dB)
3: TrueHD/AC3, 2.0 channels, 48kHz, dialnorm: -25dB
(embedded: AC3, 2.0 channels, 192kbps, 48kHz, dialnorm: -25dB)
4: TrueHD/AC3, 1.0 channels, 48kHz, dialnorm: -25dB
(embedded: AC3, 1.0 channels, 96kbps, 48kHz, dialnorm: -25dB)
5: Subtitle (PGS)
6: Subtitle (PGS)
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] [libav] Substream min channel cannot be greater than max channel. <WARNING>
[a04] The libav decoder reported error -1 while decoding. <ERROR>
Aborted at file position 1048576. <ERROR>

Here is the error message when I rip it to .wav

eac3to v3.24
command line: eac3to 00001.m2ts 4: japanese-1.0.wav
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 2 subtitle tracks, 0:24:13, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, 5.1 channels, 48kHz, dialnorm: -25dB
(embedded: AC3, 5.1 channels, 448kbps, 48kHz, dialnorm: -25dB)
3: TrueHD/AC3, 2.0 channels, 48kHz, dialnorm: -25dB
(embedded: AC3, 2.0 channels, 192kbps, 48kHz, dialnorm: -25dB)
4: TrueHD/AC3, 1.0 channels, 48kHz, dialnorm: -25dB
(embedded: AC3, 1.0 channels, 96kbps, 48kHz, dialnorm: -25dB)
5: Subtitle (PGS)
6: Subtitle (PGS)
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] [libav] Substream min channel cannot be greater than max channel. <WARNING>
[a04] The libav decoder reported error -1 while decoding. <ERROR>
Aborted at file position 1048576. <ERROR>

And here is the error message when I rip to .wavs

eac3to v3.24
command line: eac3to 00001.m2ts 4: japanese-1.0.wavs
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 2 subtitle tracks, 0:24:13, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: TrueHD/AC3, 5.1 channels, 48kHz, dialnorm: -25dB
(embedded: AC3, 5.1 channels, 448kbps, 48kHz, dialnorm: -25dB)
3: TrueHD/AC3, 2.0 channels, 48kHz, dialnorm: -25dB
(embedded: AC3, 2.0 channels, 192kbps, 48kHz, dialnorm: -25dB)
4: TrueHD/AC3, 1.0 channels, 48kHz, dialnorm: -25dB
(embedded: AC3, 1.0 channels, 96kbps, 48kHz, dialnorm: -25dB)
5: Subtitle (PGS)
6: Subtitle (PGS)
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] [libav] Substream min channel cannot be greater than max channel. <WARNING>
[a04] The libav decoder reported error -1 while decoding. <ERROR>
Aborted at file position 1048576. <ERROR>

I would like to try the experimental version, however I can't get the wav so I'm stuck till I can get the wav. :(

b66pak

19th January 2012, 20:27

first you need to demux the thd from the m2ts:

eac3to 00001.m2ts 4:japanese.thd

get the latest ffmpeg (http://ffmpeg.zeranoe.com/builds/)...and use this:

ffmpeg -i japanese.thd japanese.wav
_

An IP BreAKDoWN

20th January 2012, 07:36

@b66pak Thanks that worked. Except the eac3to complained about it being mono on the conversion from wav to dts, however I used the SureCode GUI and that worked! Thanks again!

TDiTP_

20th January 2012, 08:50

ffmpeg -i japanese.thd japanese.wav
As far as i remember, with this command line you'll be obtain 16-bit PCM always, even if TrueHD is 24-bit.
You need:
ffmpeg -i input.thd -acodec pcm_s24le output.wav
and then, if you need it, you can check the true bitdepth in audioeditor or by -analyzebitdepth in eac3to:
eac3to input.wav -analyzebitdepth

An IP BreAKDoWN

20th January 2012, 11:40

As far as i remember, with this command line you'll be obtain 16-bit PCM always, even if TrueHD is 24-bit.
You need:
ffmpeg -i input.thd -acodec pcm_s24le output.wav
and then, if you need it, you can check the true bitdepth in audioeditor or by -analyzebitdepth in eac3to:
eac3to input.wav -analyzebitdepth
Thats very useful thanks!

One question though, how can I find the bit depth of the THD since the libav in eac3to can't open it since its a mono?

TDiTP_

20th January 2012, 15:17

how can I find the bit depth of the THD since the libav in eac3to can't open it since its a mono?
If eac3to isn't able to decode lossless-stream, then it can't check his bitdepth, so in this case -analyzebitdepth doesn't work.

An IP BreAKDoWN

20th January 2012, 18:52

If eac3to isn't able to decode lossless-stream, then it can't check his bitdepth, so in this case -analyzebitdepth doesn't work.
Right, do you know a program that can analyze the bit depth in THD format?

Rodeo

20th January 2012, 22:40

Thats very useful thanks!

One question though, how can I find the bit depth of the THD since the libav in eac3to can't open it since its a mono?

Please try to follow the provided instructions. First, b66pak's:

eac3to 00001.m2ts 4:japanese.thd

japanese.thd becomes input.thd in TDiTP_'s instructions:

ffmpeg -i input.thd -acodec pcm_s24le output.wav

So here, the TrueHD file is now 24-bit PCM in WAV…

eac3to input.wav -analyzebitdepth

…thus the old, buggy libavcodec TrueHD decoder is no longer involved.

TDiTP_

21st January 2012, 05:25

Right, do you know a program that can analyze the bit depth in THD format?
i know only eac3to with his -analyzebitdepth. This command can be applied to any lossless-stream or PCM (in WAV/W64).
BTW, BDInfo can check bitdepth of THD incorrectly: look known issues (http://www.cinemasquid.com/blu-ray/tools/bdinfo).
Of course, if libav of eac3to can't decode THD, then -analyzebitdepth does not make sense (because for determine bitdepth you need to decode). In this case you should try fresh ffmpeg.

Floatingshed

25th January 2012, 19:47

Sparktank

25th January 2012, 22:16

I've been using eac3to to join various audio files together. I have been doing it by hand as a commandline and it works well.
However I have a directory containing 37 mp3 files (all same params) that I want to join. A batch file would be nice, perhaps using IN DO, but my brain won't work, please somebody take pity...
Thanks.

>.< multiple file batching can be the morning-coffee-buzzkill.

I really haven't done anything like joining files in eac3to, but I do do some batch files.

For doing multiple files, I just open a text file with the "eac3to" commands, copy and paste them several times.
Going back and modifying only the input/output file names.
And then exit at the very bottom.

Note: I also have windows configured to open CMD in a folder, using the context menu: "Open Comand Promt here"

"path\eac3to.exe" "sourcefile[+sourcefile2]" [trackno:] "[destfile|stdout]" [-options]
"path\eac3to.exe" "sourcefile[+sourcefile2]" [trackno:] "[destfile|stdout]" [-options]
"path\eac3to.exe" "sourcefile[+sourcefile2]" [trackno:] "[destfile|stdout]" [-options]
exit

Then I just copy/paste that into the CMD of the destination folder.
I use "" if the path of my eac3to is in a folder with 2 words including a space "App Bin", and also some of the source files are in folders that contain spaces as well.

I also add in custom logs.
-log=custom_name_codec__etc_with_no_spaces_to_reduce_amount_of_quotes.txt

As for creating a batch script, I'm not that far in batching.
Yet :D

Floatingshed

26th January 2012, 03:06

OK. I've done it. I'm not a whizz at programming so it may not be very elegant but it works. If you want to use it just copy and paste into a batch file (set path to eac3to). Drop into a folder of mp3 files, double click and a single file of them all nicely joined will appear!

FOR %%a IN (*.mp3) DO echo/|set /p =" "%%~NXa"+" + >> temp.txt
set /p inlist= <temp.txt
set newlist=%inlist:~0,-1%
path.to\eac3to.exe" %newlist% "Joined mp3 files.mp3"
del temp.txt

nibus

28th January 2012, 01:06

LeKouz

30th January 2012, 03:38

Hi.

Does anybody know how to set SoX to handle the 64 floating point output from eac3to?
Or in an other question, what kind of 64bit-Stream is eac3to giving out?
I've tried a lot of commands but all I've got is noise and it sounds like playing a CD-ROM in an audio-player.

This is the last command-line i've tried to pipe from eac3to to SoX:
eac3to "test.ac3" stdout.wav -full -simple | sox -V3 --type .f64 --rate 48000 --encoding floating-point --bits 64 --channels 6 --ignore-length - -t wav --encoding floating-point --bits 32 result.wav

"--type raw --rate 48000 --encoding floating-point --bits 64 --channels 6" -> Doesn't works too. Lot of noise.

If i set "--type" to wav, SoX says:
sox WARN wav: wave header missing FmtExt chunk
sox FAIL formats: can't open input `-': Sorry, don't understand .wav size

And if I command eac3to to write the 64bit output directly to a WAV-File, Mediainfo means:
General
Complete name : E:\Tools\eac3to\test.wav
Format : Wave
File size : 668 MiB
Duration : 5mn 4s
Overall bit rate mode : Constant
Overall bit rate : 18.4 Mbps

Audio
ID : 0
Format : PCM
Format profile : Float
Format settings, Endianness : Float
Codec ID : 00001000-0000-0300-8000-00AA00389B71
Codec ID/Hint : IEEE
Duration : 5mn 4s
Bit rate mode : Constant
Bit rate : 18.4 Mbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 64 bits
Stream size : 668 MiB (100%)

nu774

30th January 2012, 05:03

sox WARN wav: wave header missing FmtExt chunk

This is due to SoX wave parser bug. It currently always shows this strange warning to IEEE float wave file with WAVE_FORMAT_EXTENSIBLE header.

sox FAIL formats: can't open input `-': Sorry, don't understand .wav size

This is another thing. Currently SoX cannot read 64bit float wave files. Though one line of modification to the wave parser source code can enable it.
However, since SoX internally treats audio samples in the form of 32bit float, feeding 64bit float PCM to SoX is pointless anyway.

"--type raw --rate 48000 --encoding floating-point --bits 64 --channels 6" -> Doesn't works too. Lot of noise.

You cannot treat wave file as raw PCM. In this case, length of wave header is not divisible by the length of double PCM frame (= 8 * number of channels).
Therefore, access to each PCM samples are not aligned, and produces completely wrong result.
Even if it is aligned, since wave header is treated as audio, you will get some garbage at the beginning.

LeKouz

30th January 2012, 14:42

THX nu774 for the infos :)

geminigod

1st February 2012, 01:51

Nu774, you can go here. https://sourceforge.net/projects/sox/files/

The latest release candidate that just came out addresses some of your concerns. Not sure about handling 64 bit float, but that is a giant waste anyway. You can check the tracker bug status at that website as well.

nu774

1st February 2012, 14:05

AFAIK, this issue (http://sourceforge.net/tracker/?func=detail&aid=3481510&group_id=10706&atid=110706) is not yet addressed. Am I missing something?

Floatingshed

14th February 2012, 13:58

I have a problem getting BluRay info from eac3to. When I run:

eac3to.exe E:\ >text.txt

to generate a text file of disc contents I get a lot of wierd characters (as well as the required data). e.g.

- DTS Master Audio, English, multi-channel, 48kHz

Different text editors display this very differently, so clearly none of them understand it!

Does anyone here? Thanks.

kypec

14th February 2012, 14:56

Different text editors display this very differently, so clearly none of them understand it!

Does anyone here? Thanks.
That's because eac3to outputs special (non-printable) characters into console (to change fore/background colors for example).
Why not use other tool for getting information on Blu-ray disc, BDInfo (http://www.cinemasquid.com/blu-ray/tools/bdinfo#about) comes to mind...

NanoBot

14th February 2012, 14:58

Hi,

I just did a quick test trying to reproduce your problem, but without one special case I can't reproduce it. The textfile produced by

eac3to m: > D:\tmp\test.txt

is nearly perfect readable when I load it into either the standard XP text editor or into Ultraedit.

Nearly means, that the german umlauts and e.g. the french circumflex accent or grave accent are not displayed correctly. But the reason for this behavior is simple: Since I am in germany, my command line uses codepage 850 by default ( should be cp 437 in english language countrys ? ), while Windows uses codepage 1252 by default. So the only thing I have to do to solve the problem is to execute "chcp 1252" on my cli, and all characters are displayed correctly within the Windows editor.

Perhabs your problem has a similar cause ?

skeeve1000

14th February 2012, 21:19

Hi,
since i updated to the latest version of popcorn 1.85.1, of mkvtoolnix 5.3 as well as eac3to i always get an eac3to crash when eac3to starts to work within popcorn.
You can see the errormessage here:
http://www.networkedmediatank.com/showthread.php?tid=20887&page=99
(i asked the popcorn developement first).

May this could be a problem with MS.NET? I´m not sure but it can be possible, that i killed a version of .NET on my computer.

Help would be great.
Thanks in advance.

Skeeve

tebasuna51

15th February 2012, 02:17

Hi, since i updated to the latest version of popcorn 1.85.1,...

Like eac3to version don't change seems a bug in AudioConverter.exe 1.85.1 (maybe capturing the eac3to progress) then we can't help you here.

Try execute the command line out of AudioConverter (in a cmd window) and see if work.

I don't know eac3to dependencies with .NET

skeeve1000

15th February 2012, 10:59

Hi again,

eac3to crashes directly while starting in comand line (e.g. eac3to d:\tmß\xyz.mkv).
It asks me to send a bug report, but i am not sure, if i should do this?
Any further advice?

Skeeve

Floatingshed

15th February 2012, 13:00

That's because eac3to outputs special (non-printable) characters into console (to change fore/background colors for example).
Why not use other tool for getting information on Blu-ray disc, BDInfo (http://www.cinemasquid.com/blu-ray/tools/bdinfo#about) comes to mind...

I am using eac3to to extract files, therefore I wanted to use the info in the text file as a means of identifying and selecting the required tracks. Does anyone know how to get a clean feed of the track info?

Found a workaround, the info is nice and clean in the eac3to log file...

tebasuna51

16th February 2012, 02:30

eac3to crashes directly while starting in comand line (e.g. eac3to d:\tmß\xyz.mkv)

Sorry but I can't reproduce your problem, work fine here. :)

Midzuki

16th February 2012, 07:23

Hi again,

eac3to crashes directly while starting in comand line (e.g. eac3to d:\tmß\xyz.mkv).
It asks me to send a bug report, but i am not sure, if i should do this?
Any further advice?

Skeeve

1) :logfile:

2) Wild guess:
d:\tmß\xyz.mkv

Try renaming that folder to an "ASCII-friendly" :) string.

tebasuna51

16th February 2012, 10:17

...
Try renaming that folder to an "ASCII-friendly" :) string.

Yeah, I think skeeve1000 (from germany) know the problem.

In the command line I say to execute (see http://www.networkedmediatank.com/showthread.php?tid=20887&page=99 ) don't exist non-ASCII chars, for that my :)

Midzuki

16th February 2012, 15:47

Hummm you're right, certainly I need more buckets of coffee per day.

[ Midzuki :uglylol:s himself ]

VincAlastor

16th February 2012, 18:40

hi,
is it possible to use flaccl.dll instead libflac.dll directly in eac3to? i've read flaccl -8 provides better compression than flac -8 - besides it is faster, too.
http://www.cuetools.net/wiki/FLACCL
i know, i could use it with the command line, but channel mapping for each different codec is confusing for me. eac3to gives the right channel map to libflac even the source has a strange 7.1 surround setup, isn't it?
i could not found any limitations for flaccl, so i think it supports 24bit and 8 channels.

mindbomb

23rd February 2012, 05:11

kypec

23rd February 2012, 06:42

is it ever a good idea to update the avcodec.dll that comes with eac3to?

my friend ran into a decoding problem, i thought about suggesting this to him, not really sure if it uses a special one or something.
I'm afraid not. I'd tried that last year and replaced both avcodec.dll and avutil-50.dll with some April-ish 2011 versions -> this prevented eac3to to decode/process even some of the previously working formats :( Unless madshi himself recompiles against more recent ffmpeg/libav libraries (or releases the source code in the wild) then we're out of luck. :mad:

Richard1485

27th February 2012, 18:20

If I run a wav or PCM file through eac3to and output as .lpcm, will the result be both Blu-ray and DVD compliant?

06_taro

29th February 2012, 09:32

VincAlastor
stable released FlacCL#0.3 does not support 24bit, only FlacCL#0.4 in CUETools 2.1.2a does. But it's a bit experimental, and there's at least one known (if rare) bug in it (http://www.hydrogenaudio.org/forums/index.php?showtopic=64628&view=findpost&p=763244).

madshi
eac3to internally encodes flac using max compression level as you mentioned here (http://forum.doom9.org/showthread.php?p=1222771#post1222771). Is the encoded flac file within the flac subset (http://flac.sourceforge.net/format.html#subset) like libflac's -8(without --lax), or non-subset like flacCL's -9~11/flake's -11~12?

edit: re-encoding eac3to generated flac file with flac.exe -8 gives only difference in header, so it seems that eac3to is using exactly the same setting as libflac's -8....

shh

2nd March 2012, 04:53

Where's madshi? Is eac3to dead?

tebasuna51

2nd March 2012, 11:35

madshi is still active here http://forum.doom9.org/showthread.php?t=146228

I can't answer to second question.

szabi

6th March 2012, 09:31

Hi

I tried this:
Extract the DTS core of a DTS-HD track:
eac3to input.dtshd output.dts -core

Why is it not working?
IAW wikibooks it should: click (http://en.wikibooks.org/wiki/Eac3to/How_to_Use)

bye
szabi

madshi

6th March 2012, 09:34

eac3to is not dead, it's just kinda "on hold" for the moment, because I'm busy working on madVR. Once madVR has reached v1.0, I'll get back to eac3to development.

szabi

6th March 2012, 10:12

I tried using gui's, but no luck as well.
Eac3to and More GUI (http://forum.doom9.org/showthread.php?t=135095)
This one got crash all the time.
Another mod of this is Eac3to and More GUI v.1.10.01.
Even the input file is given, it is asking to give again and again, it can not be started.

eac3to µGUI (http://forum.doom9.org/showthread.php?t=156762)
Third one i tried is this.
But the needed button is inactive in the gui.

Is that possible it is just a planned feature which does not work momentarily?

bye
szabi

the_weirdo

6th March 2012, 10:25

@szabi: You may try this GUI: UsEac3to (http://forum.doom9.org/showthread.php?t=145574).

szabi

6th March 2012, 11:17

It worked, many thanks. :thanks:
I would reccommend to take this tool to initial post.
Out of 4 gui this was the only one which could work.

bye
szabi

jpsdr

6th March 2012, 18:53

eac3to is not dead, it's just kinda "on hold" for the moment, because I'm busy working on madVR. Once madVR has reached v1.0, I'll get back to eac3to development.

Thanks for the information !

setarip_old

6th March 2012, 19:00

@szabi

If you can find it, you might also try the MUCH older "EAC3toGUI.exe"...

Richard1485

10th March 2012, 15:49

If I run a wav or PCM file through eac3to and output as .lpcm, will the result be both Blu-ray and DVD compliant?

Any ideas about this?

nevcairiel

10th March 2012, 15:51

Any ideas about this?

Blu-ray and DVD use different types of LPCM encoding, so with some confidence i can say that it won't be compatible with both at the same time.

Richard1485

11th March 2012, 18:22

Blu-ray and DVD use different types of LPCM encoding, so with some confidence i can say that it won't be compatible with both at the same time.

Thanks for the info. In that case, I need to know what commands to input for Blu-ray and for DVD.

tebasuna51

11th March 2012, 20:01

Thanks for the info. In that case, I need to know what commands to input for Blu-ray and for DVD.
That depend of the soft you want use to mux video and audio.
For instance tsMuxeR accept .wav (<4GB) and .w64 (>4GB) to obtain .m2ts files for BD

Richard1485

11th March 2012, 23:32

That depend of the soft you want use to mux video and audio.

Really? I understood that muxers don't actually change files: they just, well, mux them. I don't see how muxing .w64 can be BD compliant, unless tsmuxer changes the file in some way.

mindbomb

12th March 2012, 05:31

so, im not completely clear on the use of flac for things above 6 channels.
WAVEFORMATEXTENSIBLE_CHANNEL_MASK is supposed to be used? and eac3to does this by default?
And decoders should be able to use that for to figure out the channels?

If my ultimate goal is to use this in a matroska file, does the channel mask remain through the muxing process?

tebasuna51

12th March 2012, 13:36

Really? I understood that muxers don't actually change files: they just, well, mux them. I don't see how muxing .w64 can be BD compliant, unless tsmuxer changes the file in some way.

Of course, a muxer can modify the data order (unique diference between lpcm and wav/w64 files)

so, im not completely clear on the use of flac for things above 6 channels.
WAVEFORMATEXTENSIBLE_CHANNEL_MASK is supposed to be used? and eac3to does this by default?
And decoders should be able to use that for to figure out the channels?

If my ultimate goal is to use this in a matroska file, does the channel mask remain through the muxing process?

AFAIK, flac can't store channel mask.
There are a unique channel mask supported for each num_channels type, and undefined for 7 and 8 channels.

me7

12th March 2012, 19:57

eac3to doesn't seem to like 5.0 content. I have a 5.0 DTS Master Audio tracks that I want to encode to 5.0 AC3, but eac3to crashes with an error:

DTS Master Audio, 5.0 channels, 24 bits, 48kHz
(core: DTS, 5.0 channels, 24 bits, 1509kbps, 48kHz)
Decoding with ArcSoft DTS Decoder...
The AC3 encoder received a non-supported data format (pcm, 5, 24, -).

Apparently the included AC3 encoder doesn't like 5 channel content.
If I use eac3to to create a 5.0 wav file and encode it with a standalone AC3 encoder, do I need to worry about channel order?

rapscallion

12th March 2012, 22:49

Audio question about my logic :

If I extract a 385 Mb AC-3, 5.1, 640 Kbps track to 6 wavs, the results are 975 Mb wavs @1152 Kbps each.

Enter the wavs into DTS-MA suite, encode them into DTS-High Res audio track format with a bitrate @ 3840 Kbps.
The result is still lossy ( I know they can't be converted to DTS-HD) but much higher bitrate and, I would think, higher quality than DTS @ 1509 Kbps and the original AC-3 track.

Is my reasoning correct or am I out in left field somewhere ?

Asmodian

12th March 2012, 23:09

... higher quality than DTS @ 1509 Kbps and the original AC-3 track.

You can never get higher quality than the original, a lossy reencode will always be lower quality than the source.

How much lower is dependent on the settings and in your case I would assume not much. Still it is better to just leave it as the AC3 if you can.

Richard1485

12th March 2012, 23:27

Of course, a muxer can modify the data order (unique diference between lpcm and wav/w64 files)

Ah, so the only difference between those types of files is data order, and as long as the muxer accepts the input the result should be compliant. Thanks. That helps.

rapscallion

12th March 2012, 23:36

You can never get higher quality than the original, a lossy reencode will always be lower quality than the source.

How much lower is dependent on the settings and in your case I would assume not much. Still it is better to just leave it as the AC3 if you can.

However, the AC-3 isn't the original, the wavs are. Thus, a re encode of the extracted wavs should result in better quality at the higher bitrate, no?

Edit: This stuff really does make my head hurt.
BTW, I did the encode and the resulting DTS-HR file is ~ 3.2 gb

tebasuna51

13th March 2012, 01:40

...
If I use eac3to to create a 5.0 wav file and encode it with a standalone AC3 encoder, do I need to worry about channel order?

No problem.
You can use also eac3to with 'pipe' if channels aren't 2.0 or 5.1:
eac3to input stdout.wav | Aften -b 640 -readtoeof 1 output.ac3

However, the AC-3 isn't the original, the wavs are.

The decoded ac3 wav's aren't the originals, have the same quality than the ac3.

Asmodian

13th March 2012, 02:07

If I extract a 385 Mb AC-3, 5.1, 640 Kbps track to 6 wavs

However, the AC-3 isn't the original, the wavs are.

If you started with it as your source of the audio the AC3 is your "original". You cannot recover the real original audio; some information was lost when it was compressed to AC3.

As tebasuna51 said the wav files contain the same information as the AC3, just no longer in a compressed form.

rapscallion

13th March 2012, 18:21

If you started with it as your source of the audio the AC3 is your "original". You cannot recover the real original audio; some information was lost when it was compressed to AC3.

As tebasuna51 said the wav files contain the same information as the AC3, just no longer in a compressed form.
Yes, what I should have said is that the wav files are the originals minus the "unnecessary" data that the compression expunges.

So, am I to understand that a 5.1 AC-3 640 Kbps audio is the same quality as it's 1150 Kbps extracted wavs? If that's the case, then shouldn't a 440 Kbps be the same quality as a 640 Kbps if compressed form the same source ?
I just can't get my head around this, so sorry for dwelling on it.

Asmodian

13th March 2012, 18:40

It is lossy compression, an AC3 is not like a zip file where after decompression you get exactly the same data back. Lossy compression drops information it thinks you will not miss (much); the 440 Kbps file dropped more information than the 640 Kbps file. Both dropped some information compared to the original.

It isn't really "unnescessary" data it is just less noticable.

I should also mention that re-encoding your extracted wavs to a 640 Kbps AC3 would lose information again; your new AC3 would be lower quality compaired to the original AC3.

rapscallion

13th March 2012, 18:48

Ok, thanks Asmodian. No how much I googled this, I couldn't find a concise answer.
Especially re the greater the compression, the more information dropped. So the answer to my previous question "5.1 AC-3 640 Kbps audio is the same quality as it's 1150 Kbps extracted wavs? " is YES. Very interesting.

Asmodian

13th March 2012, 20:49

So the answer to my previous question "5.1 AC-3 640 Kbps audio is the same quality as it's 1150 Kbps extracted wavs? " is YES. Very interesting.

You can trust tebasuna51. ;)
We both answered this in the previous two posts.

When you play an AC3 (or any audio file) it is converted to raw digital audio data and sent to your sound card for digital to analogue conversion. "Extracting" to a wav file is like playing the AC3 but saving the digital data to the hard drive instead of sending it to the sound card.

rapscallion

13th March 2012, 21:32

I know you did, I was just confirming that I now get it.

Asmodian

14th March 2012, 00:29

Sorry, I thought you were annoyed I didn't answer your obvious question. I hate ignored questions but left that one out as it was already addressed.

Glad it makes sense now. :)

odin24

18th March 2012, 20:17

Sparktank

19th March 2012, 10:58

Are there any issues converting a 7.1 DTS-HD MA to DD5.1, in regards to the back channels mixing into the surround channels? I use Arcsoft to decode the audio, output to .w64, encode to DD5.1 (libav).

According to tebasuna51, there shouldn't be any problems downmixing 7.1 DTS-HD MA to 5.1 (http://forum.doom9.org/showthread.php?p=1529424#post1529424).
As long as you're using Arcsoft 1.1.0.0 (25/04/2008), it can downmix 7.1 with either proper channel setup or strange setup.
Post 1 (http://forum.doom9.org/showthread.php?p=1519949#post1519949); Post 2 (http://forum.doom9.org/showthread.php?p=1519992#post1519992); Arcsoft version info (http://forum.doom9.org/showthread.php?p=1266679#post1266679)

Not quite sure if I missed anything... :scared:
There is this post (http://forum.doom9.org/showthread.php?p=1520511#post1520511) of some interest.

taiyoyuden

24th March 2012, 23:53

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

Is the ArcSoft DTS Decoder 1.1.0.0+ the recommended/default DTS-HD decoder now? I'm confused because here (http://en.wikibooks.org/wiki/Eac3to/In_Depth_Technical_Explanation#Evaluation_of_available_decoders), the default says Sonic and no mention of the Arcsoft. Thus, I've been using the Sonic 4.3.0.169 decoder for numerous DTS-HD 5.1 tracks.

Is there any difference in sound quality, channel mapping, etc.? Should I have them redone with the ArcSoft? Which version of the ArcSoft DTS Decoder is the best?

tebasuna51

25th March 2012, 02:24

Please read the first post in this thread:

"The Sonic DTS decoder is very good for DTS, DTS-ES, DTS-96/24, DTS-HD Master Audio and DTS-HD High Resolution tracks. The only problem is that it decodes DTS-HD 7.1 tracks only as 5.1.
...
The ArcSoft DTS decoder seems to be perfect for DTS and DTS-HD decoding. It supports every format and channel configuration that exists including 6.1 and 7.1."

I recommend ArcSoft DTS Decoder 1.1.0.0

Richard1485

25th March 2012, 14:59

Tebasuna, taiyoyuden also asked if there is any difference in sound quality between ArcSoft and Sonic when DTS-HD 5.1 tracks are being decoded. I am interested in knowing the answer to this too. The only limitation for Sonic mentioned on the first page is that 7.1 tracks are decoded as 5.1.

tebasuna51

25th March 2012, 15:29

Decoding lossless (DTS-MA) don't exist differences.
Decoding lossy DTS all decoders can have very little differences but is dificult to say what is better.
Even free decoders are good for standard DTS.

Richard1485

26th March 2012, 21:00

Thank you.

Marin85

29th March 2012, 19:43

I am wondering about the precise meaning of the following type of log messages from eac3to: "A remaining delay of +1ms could not be fixed." Does it mean that the audio track still has a residual delay of +1ms and hence I hypothetically need to somehow apply -1ms delay to fix it completely if I want to? Or does it mean that eac3to managed to apply only (X-1)ms delay, where -X is the full audio delay determined by eac3to, and I still need to apply +1 ms (hypothetically) delay to the audio track to fix it completely?

This is just a theoretical question. I am well aware that in general I should not be worrying about 1ms audio delays, nor that I would be able to fix such a small value via eac3to (say, for an AC3 audio track). I would be thankful for a straight and clear answer. I have searched the web and there appear to be some controversial opinions on this matter.

pandv2

29th March 2012, 20:25

AC3 is a packed format. What it means is the samples are grouped and compressed. The packet lenght and duration is fixed and depends on the samplerate (44.1 KHz, 48 KHz...). If you want to do a lossless operation with a AC3 stream you need to do at packet level. Supose (I don't remember the real number) a packet is 16 ms, you want to delay the stream 20 ms, eac3to only can add a complete ac3 silence packet (16 ms) or two (32 ms). In the first case the error is -4ms and in the second +12ms.

Marin85

29th March 2012, 21:52

tebasuna51

29th March 2012, 23:43

Or does it mean that eac3to managed to apply only (X-1)ms delay, where -X is the full audio delay determined by eac3to, and I still need to apply +1 ms (hypothetically) delay to the audio track to fix it completely
That is correct.

Boulder

30th March 2012, 03:22

Regarding audio delay: if DGIndex tells me that the audio track has +144ms delay, do I need to use +144ms or -144ms in eac3to's command line? It's a silly question but there is no obvious answer ;)

Marin85

30th March 2012, 07:26

Regarding audio delay: if DGIndex tells me that the audio track has +144ms delay, do I need to use +144ms or -144ms in eac3to's command line? It's a silly question but there is no obvious answer ;)
I feel like I am being mocked here :) What tebasuna51 (btw, thank you for the clear answer!) explained is - as I suspected - sort of the opposite of what eac3to is actually suggesting by the above message IMHO. It is confusing because eac3to message refers to the delay it tries to apply to the audio track, and not to the (resulting) delay the audio track already has.

Boulder

30th March 2012, 07:33

I feel like I am being mocked here :)Not at all :) It's just very confusing at times. I think I've always used the opposite but since the delays are usually rather small, it's not easy to determine which way is the correct one.

tebasuna51

30th March 2012, 10:53

... if DGIndex tells me that the audio track has +144ms delay, do I need to use +144ms...
Yes, you must always respect the sign.
The info (also MediaInfo) always show the delay than you need apply.

tormento

31st March 2012, 08:56

There is a stream that eac3to really doesn't like:

9: DTS Express, English, 1.0 channels, 24 bits, 96kbps, 48kHz

I can demux it but nothing more. Tried with TotalMedia libraries, LibAV ones but whenever I am going to export to wavs, it tells me some kind of errors. Any idea about how to convert it?

Brazil2

1st April 2012, 09:17

tormento

1st April 2012, 19:20

A sample would be nice for testing purposes :)
If only could be possible to demux it somehow..

Brazil2

2nd April 2012, 14:18

If only could be possible to demux it somehow..
Well, you said you did:
I can demux it but nothing more.

Anyway, post a muxed sample so we can check it ?

Bryce2

9th April 2012, 10:59

Sparktank

10th April 2012, 02:04

Bryce2

10th April 2012, 10:03

Hi to all!
Trying to convert a DTS track to AC3 using "eac3to" leads me to an unusual problem.
DTS (ES):2.77GiB, 4h 23mn = AC3: 1.34GiB, 5h 0mn.
I want to mention that this is only the 2nd time I'm facing this problem after more than 200 conversions. The first time, I found a solution in this forum with adding a parameter to eac3to, but now I'm searching hours & hours with no luck.

Solved:
Using as usual the default command "eac3to.exe input.dts output.ac3" was the reason of this weird effect (for this & only dts stream).
After trial & error and changing the command to "eac3to.exe input.dts output.ac3 -libav" ..did the thing.
Anyway ..:thanks: "Sparktank" for your interest to give a help.

DJ-1

10th April 2012, 11:00

Can I convert maintain DTS-MA track using eac32 ?
Sent from my GT-I9100 using Tapatalk 2

DJ-1

10th April 2012, 11:01

Oops, double post

sneaker_ger

10th April 2012, 14:22

eac3to can do that with the help of external libraries like Arcsoft. Read the start post.

DJ-1

10th April 2012, 14:30

eac3to can do that with the help of external libraries like Arcsoft. Read the start post.
OK, will read it, I was trying out Multi avchd Wichita uses eac32.... tried converting dts-ma to LPCM, (as my media box does play dts, but not .dts-ma..... ) so I was trying g to passtgrough the pre-decoded LPCM stream.. got lots of hissing...nothing more.
I have an Onkyo Tx nr609, (does support dts-ma )
Sent from my GT-I9100 using Tapatalk 2

frumble

11th April 2012, 15:25

Hello,
I have a problem with eac3to that drives me crazy. I want to decode the DTS-HD MA streams of my release of x and encode them to FLAC so that my Linux audio players can play the full sound, not just the DTS core.
I have downloaded eac3to and the ArcSoft DTS Decoder 1.1.0.0 plus the HdBrStreamExtractor GUI.
The output WAVE files (or FLACs) have all 7 channels and the sound is nice but they appear to have the wrong channel mapping: Voices are very low and often come from only one side (I have only stereo speakers) and once in a while come with strange echo. But when I import them into Audacity and play them with the editor the output is correct. This however has no effect on the Audacity export: 6.1 channel export has the same mapping problems as the original file. My players are not the problem, they can play every DTS, AC3 and TrueHD stream correctly.
I read something about "strange setup" but I can't understand if this might be the problem. I have absolutely no knowledge about audio mastering. The issue affects both English and German streams. The Audacity export window offers the option to remap the channels but I have no clue what could be the right choice. So I really hope you can help me: I uploaded a piece of 6.1 export from Audacity in Ogg Vorbis and it would be really nice if someone could tell me the right mapping. Thank you very much in advance!
http://www.2shared.com/audio/boFrWT5B/Export-example-eac3to-wrong-ch.html - I hope not to violate against forum rules with this.

Off this topic I want to say something: It may sound ridiculous but I miss a "tutorial". I lost hours in trying to understand what these freeware tools do and how to set them up with decoder and codecs. The learning curve is very steep but not because I am a idiot but 'cause most of your tools lack a proper documentation. I am a Linux user and I am used to find answers in reading docs.

It's really great that there are freeware tools like eac3to but I can't understand why the authors of such media helpers don't make the source code available. The last version of eac3to is from 2010? I read something about a GPU video encoder the author is writing since then and the promise to get back to work on eac3to when this encoder is final. But since 2010 no progress with eac3to. In this time the program could have been matured and bugs could have been fixed from others but they can't do it because they don't have the source code. It is his right to hold the source for himself but I truly can't understand the reason. It doesn't seams to be his plan to make commercial profit out of it. But no hard feelings, this are just my thoughts.

NanoBot

11th April 2012, 19:27

Hi frumble,

this workaround might help: http://forum.doom9.org/showthread.php?p=1524301#post1524301

sneaker_ger

11th April 2012, 20:40

Unfortunately 6.1 is not defined in FLAC. So players might decode it not as you expect, unless you do a downmix.

In addition to NanoBot's link:
http://forum.doom9.org/showthread.php?p=1538930#post1538930

Sparktank

12th April 2012, 11:25

Unfortunately 6.1 is not defined in FLAC.

This has actually been a bit of a constant problem for FLAC.
One user here goes on about it...
Dear (people), please stop using 6.1 FLAC (http://anonym.to/?http://mod16.org/hurfdurf/?p=184)

I remember reading about 6.1 FLAC in other places.

Personally, I'd stick to the CORE or even a conversion to AC3@640, as 640 is perceived to have transparent quality to the master file.

With some BD rips I do, I'm quite happy using only the core for maximum compatibility with my BD player that can playback MKV (with original h264/VC1/MPEG).

It's not a significant compromise between lossless and core, I'd stick to CORE audio. The human ear wouldn't really be able to pick pu the difference too much.

Midzuki

12th April 2012, 12:06

Another possible approach: drop FLAC :p and "move house" to WavPack :)

frumble

12th April 2012, 14:32

Thank you for your help! So I made WAV files with eac3to and want to convert them to WavPack, but the Encoder says: "can't handle .WAV files larger than 4 GB (non-standard)!" - sadly booth streams are over 4 GB. Also Monkey's audio can't encode them: "Error: 1002".
Export to AC3 with ea3to works fine but I absolutely want a lossless codec (it feels better ;) ). In my desperation I even tried to merge the movie MKV with the WAVE files but MKVmerge GUI gives the error "87".
Both WAVE streams seams to be a little bit strange: Non of my audio players can play more than a few minutes (even they display only a few minutes of playtime) and also Audacity can only show me a few seconds!
The link from @Sparktank is very confusing for me. Maybe I try it later.
Downmixing the stream would be the last option but I would be very pleased if it's possible to preserve the original channels...

Midzuki

12th April 2012, 15:10

^ @ frumble:

1) forget Monkey Audio, it doesn't support multichannel ;

2) try "wavpack --help" in the command prompt ;)

Midzuki

12th April 2012, 15:18

Off this topic...

It's really great that there are freeware tools like eac3to but I can't understand why the authors of such media helpers don't make the source code available...
But since 2010 no progress with eac3to. In this time the program could have been matured and bugs could have been fixed from others but they can't do it because they don't have the source code.

Sadly you're not entirely correct. Open source-code is NO guarantee that there will ever be other capable programmers interested in fixing the bugs of the software. Take a look at MaestroSBT, K-Meleon, mpeg2enc, mkisofs, etc.

frumble

12th April 2012, 16:04

@Midzuki Sorry, the WaPack help page hasn't a option close to "allow input files bigger than 4 GB", or what do you mean?
You are right, the fact that a software is OSS doesn't mean that there are other developers capable with the task. But look at ffmpeg/libav, x264, VLC and mplayer in the media field. It's not soo bad. And consider the chance @madshi, eac3to's author can't work on freeware projects anymore or dies. Then the community would be forced to rewrite the tool from scratch. OSS is freedom and opportunity, not constraint.

Midzuki

12th April 2012, 16:16

@Midzuki Sorry, the WaPack help page hasn't a option close to "allow input files bigger than 4 GB", or what do you mean?
...

-i ignore length in wav header

MuteyM

12th April 2012, 22:35

Hi, I am decoding a DTS-ES 6.1 soundtrack to wav format. If I use the Sonic decoder then everything works correctly. However if I use libav, then I end up with a wav file with a corrupt channel mask. It looks like eac3to is not taking into account libav's lack of back channel decoding:

eac3to j:\backup 1) 4: c:\temp\orig.wav -libav
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 1:38:52, 24p /1.001
1: Chapters, 20 chapters
2: MPEG2, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
4: DTS-ES, English, 6.1 channels, 24 bits, 1509kbps, 48kHz, dialnorm: -4dB
5: Subtitle (PGS), English
6: Subtitle (PGS), French
7: Subtitle (PGS), Spanish
8: Subtitle (PGS), English
a04 The libav DTS decoder doesn't decode the back channels.
a04 Extracting audio track number 4...
a04 Removing DTS dialog normalization...
a04 Removing XCh extension...
a04 Decoding with libav/ffmpeg...
a04 Reducing depth from 64 to 24 bits...
a04 Writing WAV...
a04 Creating file "c:\temp\orig.wav"...

I end up with a wav file containing 5.1 channels of data, but with a channel mask of 0x70f (6.1) instead of the expected 0x60f (5.1). This confuses the heck out of any software that attempts to play it :)

I thought using -down6 might force eac3to to work properly, but this ended up with the same problem. So I think the solution is for madshi to update eac3to to write the correct channel mask when libav decodes 6.1 (and probably 7.1) DTS files.

tebasuna51

13th April 2012, 03:05

I end up with a wav file containing 5.1 channels of data, but with a channel mask of 0x70f (6.1) instead of the expected 0x60f (5.1).
Yes, is a know bug.

frumble

13th April 2012, 18:41

@Midzuki Thank you very much! The conversion worked with all streams and the end MKVs play nice with the WavPack codec! :)

Midzuki

13th April 2012, 18:51

^ Glad to see it worked :)

Now let's spread the word,
WavPack is the future! :sly: :D

Bigmango

20th April 2012, 00:31

Is it possible to convert TrueHD to DTS-HDMA (TrueHD > WAVS > DTS HD Encoder) and keep the same sound for the TrueHD tracks that apply DRC to improve the sound? (at least I think it's the DRC that's doing this ?).

This problem arises with the Transformers 3 TrueHD 7.1 track (and people tell me Iron Man is probably the only other movie doing the same). The converted DTS-HDMA track does not sound the same as the original TrueHD. Some sounds at different moments in the movie have more presence with the original TrueHD track. It makes the sound better, more dynamic.

I think this is caused by the DRC (?) that's applied differently on the different channels at different moments.

The TrueHD track sounds better with PowerDVD 12 compared to the converted DTS-HDMA, but with MPC I don't seem to hear a difference (this leads me to think that MPC doesn't apply the effects and so it doesn't handle the TrueHD metadata correctly (?).

PCM and DTS-HDMA can't modify the sound at playback time in the way TrueHD does. How can I save the wavs with the full TrueHD sound experience?

To summarize, I would like to convert TrueHD to DTS-HDMA and have it sound as good on hardware receivers and PowerDVD (now with eac3to the audio source quality is the same, but many sounds are lacking presence in different parts of the movie).

(This is the first movie I am having this problem with. It seems the Transformers 3 sound engineers wanted to TrueHD track to be reproduced with improvements applied to the Lossless track it contains - and the whole experience is indeed more enjoyable to my ears with these improvements).

PS: sorry mods, I have opened this thread (http://forum.doom9.org/showthread.php?p=1570771#post1570771) before, but I think here is the right place as it concerns eac3to

tebasuna51

20th April 2012, 10:11

Bigmango

21st April 2012, 14:40

Bimango, please continue this discussion in your original thread if you want.
The discussion about this topic in this thread was closed in first month.

eac3to always ignore DRC because is a effect to apply at play time (with 'Night mode' for instance), never when recoding, and don't improve the audio just make it less dynamic.

Ok so if it's not DRC it's another effect that's applied to the sound. (continued in my other thread (http://forum.doom9.org/showthread.php?t=164724)).

Regarding eac3to: Can eac3to be used with another decoder that can save the "complete" TrueHD to wavs? It seems the open source decoders don't fully support TrueHD as they only extract the lossless track without the effects that enhance it by giving more presence to specific sounds at specific times (most movies only play the lossless track as is, so only a handful of movies seem to be concerned by this issue).

Thanks.

tebasuna51

22nd April 2012, 12:13

eac3to save the complete TrueHD to wav's. Don't exist "effects that enhance it by giving more presence to specific sounds".

Joniii

26th April 2012, 11:29

I'm converting 5.1 AC3 from DVD (25.000 -> 23.976), Is it normal that eac3to remaps channels?

eac3to f:\al.ac3 g:\al.ac3 -slowdown
AC3, 5.1 channels, 1:26:43, 384kbps, 48kHz, dialnorm: -27dB
The Nero decoder doesn't seem to work, will use libav instead.
Removing AC3 dialog normalization...
Decoding with libav/ffmpeg...
Remapping channels...
Changing FPS from 25.000 to 23.976...
Encoding AC3 <640kbps> with libAften...
Creating file "g:\al.ac3"...
eac3to processing took 3 minutes, 42 seconds.
Done.

tebasuna51

26th April 2012, 15:49

Yes, internal AC3 channel order is different than standard channel order used to resample and send to encoder.
Don't worry.

ilomambo

5th May 2012, 22:22

Hello, I am having the following problem:

eac3to v3.24
command line: "c:\Program Files\Utils\eac3to\eac3to.exe" "track01.dts" "track01.wav" -libav -simple
------------------------------------------------------------------------------
VOB, 1 audio track, 0:08:57
1: DTS-96/24, 5.1 channels, 24 bits, 1510kbps, 96kHz
Track 1 is used for destination file "track01.wav".
[a01] Extracting audio track number 1...
[a01] Decoding with libav/ffmpeg...
[a01] Reducing depth from 64 to 24 bits...
[a01] Writing WAV...
[a01] Creating file "track01.wav"...
[a01] Clipping detected, a 2nd pass will be necessary. <WARNING>
[a01] Starting 2nd pass...
[a01] Extracting audio track number 1...
[a01] Decoding with libav/ffmpeg...
[a01] Reducing depth from 64 to 24 bits...
[a01] Writing WAV...
[a01] Applying -0.73dB gain...
[a01] Creating file "track01.wav"...
eac3to processing took 1 minute, 27 seconds.
Done.

The wav file seems fine, I can see it plays all 5.1 channels
But look at the file sizes:

>>dir track01*.*
Volume in drive K is Music
Volume Serial Number is 884C-4DCD

Directory of K:\DTS

29/04/2012 14:09 105,261,056 track01.dts
05/05/2012 23:54 987 track01 - Log.txt
05/05/2012 23:54 463,804,460 track01.wav
3 File(s) 569,066,503 bytes
0 Dir(s) 88,401,051,648 bytes free

Is this correct, WAV has a factor of 4.5 times the DTS size?
I don't know, other WAV DTS files I have are much smaller (given the average MB/min of audio, this one is 8:56 min)
Is there any command line switch to keep the WAV size smaller?

tebasuna51

6th May 2012, 02:49

...
Is this correct, WAV has a factor of 4.5 times the DTS size?

eac3to/libav decode your DTS-96/24, 5.1 channels, 24 bits, 1510kbps, 96kHz to a WAV 5.1 channels, 24 bits, 48kHz with a bitrate of:
6 channels x 24 bits x 48KHz = 6912 kb/s

6912/1510 = 4,57

Then, yes, is correct.

Is there any command line switch to keep the WAV size smaller?
You can use -down16 to obtain a WAV 5.1 channels, 16 bits, 48kHz with a bitrate of:
6 channels x 16 bits x 48KHz = 4608 kb/s

4608/1510 = 3,05

ilomambo

7th May 2012, 09:53

You can use -down16 to obtain a WAV 5.1 channels, 16 bits, 48kHz with a bitrate of:
6 channels x 16 bits x 48KHz = 4608 kb/s

4608/1510 = 3,05

Thanks.

I am dumb regarding the technical details. I think the DTS file has the same information as the final WAV, that's why the increase in size seemed too much.
But, I assume from your explanation, that much of the DTS info is replicated to create the 6 CH WAV, and that's why the file is so much bigger, isn't it?

EDIT:
I just looked in another song I have in DTS 5.1 WAV format (in the AC3 filter properties while it was playing) and it showed this:

Decoder:
Stream format: DTS 3/2.1 (5.1) 44100Hz
Bitstream type: 14bit low endian
Frame size: free format
Samples: 1024
Bitrate: unknown
SPDIF stream type: 0xc
Frame interval: 4096
Actual bitrate: 1411kbps
DTS
speakers: 3/2.1 (5.1)
sample rate: 44100Hz
bitrate: 1411kbps
stream: 14bit LE
frame size: 3584 bytes
nsamples: 1024
amode: 9
No CRC

Tebasuna51, If I follow your math 6ch x 14bit x 44KHz = 3696 kbps != 1411 kbps reported by AC3
It is 13:40 min song and the file only takes 141MB
Something is not fitting, according to my understanding

On the other hand when I play the file created by eac3to, AC3 filter shows this:

Input format: PCM24 3/2.1 (5.1) 48000
User format: PCM16 - 0
Output format: PCM16 3/2.1 (5.1) 48000

That's why I think I am using the wrong tool. I just wanted to wrap the DTS file in WAV format, not to convert it. It seems the file I got is pure PCM.

MrVideo

7th May 2012, 11:33

I just wanted to wrap the DTS file in WAV format, not to convert it. It seems the file I got is pure PCM.

WAV is not a wrapper, it is a format, just like MPEG-2 and MPEG-4 are video formats, that get wrapped into various containers. Neither DTS or WAV are containers.

DTS is a compressed audio format, while WAV is not. Hence the reason that the WAV file is larger after you uncompressed the DTS file.

Midzuki

7th May 2012, 13:45

...

DTS is a compressed audio format, while WAV is not.

But yes, .WAV is a RIFF-based container, and may contain compressed audio. Regarding DTS-in-WAV especifically, there are two types, 1) normal, without SPDIF-padding, and with a .dca TwoCC, and 2) hacky, with SPDIF-padding, disguised as stereo PCM @ 32 / 44.1 / 48 kHz.

MrVideo

7th May 2012, 13:53

Interesting. I've never seen anything but WAV uncompressed PCM audio.

Then, that leads me to this question... why put DTS compressed data into a WAV file? I'd just leave it as a DTS file.

tebasuna51

7th May 2012, 15:57

But, I assume from your explanation, that much of the DTS info is replicated to create the 6 CH WAV, and that's why the file is so much bigger, isn't it?
The audio data in DTS is encoded (compressed like a zip) the audio data in WAV file is decoded (uncompressed PCM data)

I just looked in another song I have in DTS 5.1 WAV format
A dts_5.1_wav is not a correct WAV file because the header don't show the content. The fake header show a PCM 2.0 44.1KHz just to lies a burner and burn a CD-Audio.

I just wanted to wrap the DTS file in WAV format, not to convert it.
Then you want burn CD-Audios 5.1
Try spdifer (http://ac3filter.net/wiki/AC3Filter_tools), a AC3Filter tool.

Then, that leads me to this question... why put DTS compressed data into a WAV file?

Like you can read, only to burn a CD Audio.
Some CD players can send the DTS 5.1 and play surround.

ilomambo

7th May 2012, 17:11

Then you want burn CD-Audios 5.1
Try spdifer (http://ac3filter.net/wiki/AC3Filter_tools), a AC3Filter tool.

tebasuna51, you are the man! :goodpost:

:thanks:

spdifer worked like a charm! I got my WAV file 5.1ch, at the same size as the DTS original file !!

... and I learned something about DTS and WAV files.

RazvanuZu

10th May 2012, 16:17

I get an error when I try to use eac3to with SurCode DVD–DTS v1029. Is this a problem with eac3to or SurCode DVD–DTS v1029? I mean, is eac3to incompatible with SurCode DVD–DTS v1029?
When I use SurCode DVD–DTS v1021 I get no error.

The error sounds like: "Pressing the Surcode "Encode" button didn't seem to work", but when I directly use SurCode DVD–DTS v1029 I don't get any errors at all.

What is the latest compatible version of SurCode DVD–DTS?

Thanks.

ramicio

11th May 2012, 17:09

I wish there was a way to have eac3to in Linux without wine. I don't have a GUI installed, so I can't do wine. I run eac3to on my Windows machine to rip discs, but sometimes I forget something and need to fiddle with large files after I've written them, and my disc is across a network. Being able to have it on my Linux would speed things up quite a bit. I would only need basic libavcodec functionality.

Furiousflea

15th May 2012, 00:05

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

Is this still needed, may I ask. Changelog mentions a bug fixed in version 3.21...

fixed: 6.1 DTS decoding with ArcSoft resulted in wrong channel order

Many thanks.

tebasuna51

15th May 2012, 00:25

The decode is OK, only need a remap when downmix 6.1 to 5.1

Furiousflea

17th May 2012, 11:59

The decode is OK, only need a remap when downmix 6.1 to 5.1

Thanks for confirmation :)

Joniii

20th May 2012, 15:55

I have previously converted 25.000 fps audio to 23.976 with -slowdown switch, how do I convert 25.000 to 24.000?

I have import Blu-ray with strange 24p (not 24.000/1.001) h.264 and DTS. I'm trying to add AC3 audio into it from my older PAL DVD.

ramicio

20th May 2012, 17:08

Joniii

21st May 2012, 10:35

Are you even sure the audio is going to line up? What is the fixation with the audio from the DVD?

Anyway, you would first add the "-25.000" switch, to tell it to assume the material is 25 FPS, and then "-changeto24.000".

I've always adjusted the delay after conversion. I'm adding dubbed audio from DVD to import blu-ray for my 3 year old.

I've thought you need to convert audio with -slowdown not -changeto switch?

ramicio

21st May 2012, 13:13

I still don't get it. Slowdown is deprecated, and only for 25 to 23.976 conversion.

Revgen

25th May 2012, 11:01

xkodi

26th May 2012, 09:29

- 1.1.0.0 can decode DTS(-HD) 6.1/6.0 but can't decode non-standard 7.1
- 1.1.0.8 can't decode DTS(-HD) 6.1/6.0 but can decode non-standard 7.1

i don't know why the same wrong information is continue spreading, when there are enough facts proving that's completely wrong statement about 1.1.0.0 vs. 1.1.0.8 regarding non-standard 7.1 (i.e. "strange setup") tracks:

http://forum.doom9.org/showthread.php?p=1471763#post1471763

Boulder

26th May 2012, 10:06

Only 1.1.0.8 seems to work with DTS MA 1.0 files.Is it the only one that can decode such files or what is the issue with other versions?

Revgen

26th May 2012, 10:55

Is it the only one that can decode such files or what is the issue with other versions?

It's the only version that works that I know of.

I have 2 Blu-Ray discs with DTS MA 1.0 tracks. The Big Heat (1953) and The Big Trail (1930). 1.1.0.0, 1.1.0.5, and 1.1.07 give errors everytime I try to decode with eac3to. I also can't play the DTS MA 1.0 audio in TMT either unless 1.1.0.8 is in the codecs folder. I don't have version 1.1.0.1, so I can't comment on whether that version can decode DTS MA 1.0 or not.

pasadena

30th May 2012, 04:37

Hi All,

I'm trying to convert this into a blu-ray structure ISO for playback on my media player.

I have no problems extracting all the files using either evodemux or HD-DVD steam extractor but cannot remux this particluar TrueHD track.

I've spent the past week trying to work out issues that exist with remuxing TrueHD tracks.

I tried to convert the TrueHD track to DTS but it comes up saying that it does not support the sampling rate.

Here's an ouput log:

eac3to v3.24
command line: "D:\Converter Programs\HdBrStreamExtractor_0.8\eac3to.exe" "D:\Black and White Night" 1) 4:"D:\bwnight\1_4_audio.dts" -progressnumbers
------------------------------------------------------------------------------
EVO, 1 video track, 3 audio tracks, 1 subtitle track, 1:04:34
1: Chapters, 18 chapters
2: h264/AVC, 1080i60 /1.001 (16:9)
3: E-AC3, English, 5.1 channels, 3024kbps, 48kHz, -66ms
4: TrueHD, English, 5.1 channels, 96kHz, -66ms
5: E-AC3, English, 2.0 channels, 1023kbps, 48kHz, -66ms
6: Subtitle (DVD), Japanese
[a04] Extracting audio track number 4...
[a04] Decoding with libav/ffmpeg...
[a04] Applying RAW/PCM delay...
[a04] Writing WAVs...
[a04] Creating file "D:\bwnight\1_4_audio.L.wav"...
[a04] Creating file "D:\bwnight\1_4_audio.R.wav"...
[a04] Creating file "D:\bwnight\1_4_audio.LFE.wav"...
[a04] Creating file "D:\bwnight\1_4_audio.C.wav"...
[a04] Creating file "D:\bwnight\1_4_audio.SR.wav"...
[a04] Creating file "D:\bwnight\1_4_audio.SL.wav"...
[a04] The original audio track has a constant bit depth of 24 bits.
Encoding DTS <1536kbps> with Surcode...
Surcode DTS Encoder doesn't support this samplingrate. <ERROR>

I use evodemux to get the THD file but I can't do anything else with it. Tsmuxer bombs out saying it can't detect stream type so it can't read and import it

Eac3to cannot convert it to thd+ac3.

eac3to v3.24
command line: eac3to demuxed.thd output.thd+ac3
------------------------------------------------------------------------------
TrueHD, 5.1 channels, 96kHz
Decoding with libav/ffmpeg...
Encoding AC3 <640kbps> with libAften...
Initialization of the AC3 encoder failed. <ERROR>
Aborted at file position 262144. <ERROR>

I've also used a program called pcm2tsmu which converted the TrueHD track to an LPCM stream sampled at 96KHz which tsmuxer was able to use and process. When I created a BD structure through Tsmuxer and played it through my media player, the audio cames out as white noise.

Is converting a TrueHD track to an LPCM stream going to be inferior in any way?

If anyone has this disc, can you please let me know if you had any luck successfully converting this to blu-ray with the TrueHD track intact.

Any help would be much appreciated.

Cheers

tebasuna51

30th May 2012, 15:40

@pasadena
- With THD track: I don't know if this can work because I can't test with a THD 96 KHz, but try this:

"D:\Converter Programs\HdBrStreamExtractor_0.8\eac3to.exe" "D:\Black and White Night" 1) 4:"D:\bwnight\1_4_audio.thd+ac3" -resampleTo48000 -progressnumbers

- With DTS-MA: you can't use Surcode because don't support 96 KHz and DTS-MA, only standard DTS is not recommended.
If you have DTS-HD Master Audio you can extract the monowavs and after encode to DTS-MA

"D:\Converter Programs\HdBrStreamExtractor_0.8\eac3to.exe" "D:\Black and White Night" 1) 4:"D:\bwnight\1_4_audio.wavs" -progressnumbers

- With LPCM, same quality than THD but biger size.
You have two options:

1) Decode to w64 (tsMuxeR must accept PCM in w64 container):

"D:\Converter Programs\HdBrStreamExtractor_0.8\eac3to.exe" "D:\Black and White Night" 1) 4:"D:\bwnight\1_4_audio.w64" -progressnumbers

2) Decode to special PCM format for tsMuxeR with pcm2tsmu:

"D:\Converter Programs\HdBrStreamExtractor_0.8\eac3to.exe" "D:\Black and White Night" 1) 4: stdout.pcm -progressnumbers | "YOUR-PATH-TO\Pcm2Tsmu.exe" - "D:\bwnight\1_4_audio.pcm" -s 96000

pasadena

30th May 2012, 17:32

@tebasuna51

Thanks for getting back to me.

I'll give these a try in the morning and let you know the outcome.

Much appreciated.

Cheers.

Vasch the stampede

3rd June 2012, 12:54

There will be any chance that eac3to become multithreaded?

ramicio

4th June 2012, 16:03

Another possible approach: drop FLAC :p and "move house" to WavPack :)

Except problems arise when you have files bigger than 4 GiB. Some filters handle it, some don't. For Windows you can use madFLAC for correct channel assignments, or WavPack's splitter filter and decoder handles files > 4 GiB. Outside of Windows, you're screwed on both fronts. Both codecs still need work, and it's pathetic that they've been basically "completed" years ago despite some problems. All FLAC would have to do is switch to using WAVEFORMATEXTENSIBLE channel masks.

Joniii

5th June 2012, 20:21

Stupid question, I have a m2ts file with DTS (delay -9ms). eac3to applied dts delay when demuxing but said a remaining delay of +1ms could not be fixed. I know 1ms is nothing but being a perfectionist, do I have to add -1ms or +1ms with mkvtoolnix when remuxing?

Little confused because the file has -9ms delay but after fixing it, it ends up having +1ms delay.

hello_hello

6th June 2012, 02:45

Maybe the extra 1ms delay is caused by some sort of padding used at the beginning of the audio file and that's as close as eac3to could get it? Just a guess....
To answer you question though I'd assume it's a positive delay of 1ms which needs to be added manually.

tebasuna51

6th June 2012, 08:25

Correct the delay without recode is only possible adding/cutting complete DTS frames (512 samples at 48KHz = 10.667 ms each frame).
Here 1 frame is cutted then 10.67ms - 9ms = 1.667ms.
Like audio have 1.667 ms less duration the delay in container must be +1, or +2 if you want, but is unnecesary.

Video frames have a duration of 40 ms (25 fps) then delays of +/- 20 ms are absolutely undetectable.

Joniii

6th June 2012, 20:58

Thx for the explanation, I have always wondered why it always leaves +1ms.

ramicio

8th June 2012, 17:06

I really wish logging could be disabled. There should be support for some sort of .ini file in the eac3to directory for a few settings, like which method for progress display, and if it's just the hyphen bar, then you should be able to specify how many characters wide your terminal is. Who uses the default width of 80? I don't get why progress numbers need to start a new line. It's difficult for me to judge progress because my terminals are 170 wide, and will likely grow as resolutions keep going up.

sl1pkn07

10th June 2012, 16:16

little question. why not use libmatroska/libebml instead haalisplitter/mkvtoolnix?

mastrandrea

11th June 2012, 02:23

Is this the right tool to convert a 3.0 AC3 track (Front: L R, Side: C) to a 2.0 AC3 track? If not, which software do you suggest me to use?

tebasuna51

11th June 2012, 09:18

eac3to only accept 5.1 to convert to 2.0.

You can try:
- BeHappy, with NicAc3Source(Down2) like Source and a DSP Normalize.
- BeLight, adjusting azid output like Stereo and Normalize.

ramicio

11th June 2012, 14:04

FFDShow, GraphEdit, and the WavDest filter. Set FFDShow to mix to 2.0 and normalize the levels.

mastrandrea

11th June 2012, 19:04

Thank you both!

mp3dom

15th June 2012, 13:13

I have a question that probably was already answered but I can't find a clear post (most of them refers to dts-hd 6.1 to 5.1 remapping channels). I have a 6.1 interleaved wave with the following mappings:

Left
Right
Center
LFE
Left Surround
Right Surround
Back-Center Surround

I want to downmix to 5.1 with eac3to. Is this channelmap already ok for what eac3to expect, or do I need to switch some channel (like put Back Center Surround between LFE and Left Surround?).
Thanks!

ramicio

15th June 2012, 13:47

If they are already in a WAV then they are likely in the correct order. WAV has a standardized channel order. Are the surrounds "back" or "side?"

mp3dom

15th June 2012, 14:07

The surrounds are all back channels. The interleaved wave was made from 7 mono files (I think using SoundForge or Nuendo)

ramicio

15th June 2012, 14:08

Well, if you put the files in the correct order to interleave the file, and it has the correct channel mask, then there will be zero problems.

Carpo

15th June 2012, 15:32

instead of reading 583 pages and trying to use the search here that never finds what i'm after, I will ask a question that has already been asked, but is there any chance of eac3to being updated to use LAV filters instead of/ as well as haali?

ramicio

15th June 2012, 15:36

Isn't Haali just a splitter? I see a file in the eac3to directory called avcodec.dll. Is this not libavcodec, which that LAV Filters thing is based off of?

Carpo

15th June 2012, 16:22

it would still need to be made aware of lav and what it can do

ramicio

15th June 2012, 16:23

What?

Carpo

15th June 2012, 16:26

you cant just drop a dll in the folder and expect it to know what LAV is and what lav can do, avcodec-lav-54.dll which is the lastest version that comes with LAV filters may have changed from the version that comes with eac3to and cause errors

ramicio

15th June 2012, 16:27

But who cares? It does all it needs to do. Just because libavcodec has been updated, that doesn't mean eac3to will ever support writing to those extra formats. I don't really get what you're after here.

Carpo

15th June 2012, 16:33

in order for it to see mkv's or in this case mux them it uses haali

For video muxing you need:
(1) Haali Matroska Muxer

I am asking that LAV Splitter be added there, so we can use that instead of having to install and use haali

ramicio

15th June 2012, 16:35

What's wrong with using Haali? It's much better than the crap in libavcodec.

Carpo

15th June 2012, 16:36

nothing is wrong with it, and libavocdec is far from crap, have you ever used LAV filters?

ramicio

15th June 2012, 16:40

There's no need for me to. It's the same thing as FFDShow. MPC-HC's internal MKV filter is not as good as Haali.

I also have zero hope in eac3to ever being updated.

sneaker_ger

15th June 2012, 18:46

Even LAV uses Haali's code.

Carpo

15th June 2012, 23:12

ffdshow uses it mplayer, lav - quite a few things do

v___v

16th June 2012, 21:01

Hihihi.

If what I'm about to ask seems redundant or as if I didn't look through the proper channels (internet/documentation) thoroughly enough, I apologize. I really did try to look for the answer beforehand.

I'm trying to decode the lossless portion of DTS-HD MA audio. But I'm not sure what I am doing is working.

This is the file (http://susepaste.org/view/raw/30197475) I am working with. This is the output (http://susepaste.org/view/raw/93951891) of my eac3to setup with the -test parameter.

Alright, so when I try eac3to file.dts file.flac or eac3to file.dts file.wav, I get the following output for instance of FLAC (http://susepaste.org/view/raw/52117386) or wav (http://susepaste.org/view/raw/21417270) (also put mediainfo output for the output file there).

It says it's "decoding with ArcSoft DTS Decoder" but the following line mentions the core's information...

So here's what I'm wondering:

- Is it really decoding the lossless -> flac/wav? This seems like it's decoding the lossy core then transcoding that to flac/wav. So lossy -> lossless ?

- Is it possible that I am doing it wrong? (incorrect parameters, wrong thing to use, etc)

- Is it possible that this has something to do with the versions of ArcSoft/eac3to I have?

Any clarification/assistance would be helpful.

tebasuna51

16th June 2012, 23:38

...
- Is it really decoding the lossless -> flac/wav?
Yes
This seems like it's decoding the lossy core then transcoding that to flac/wav. So lossy -> lossless ?
Nope, the line:
"(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48kHz)"
is only a info. ArcSoft decode the DTS-MA lossless.
All is OK.

Leinad4Mind

29th June 2012, 00:26

I've an DTS-HD Master Audio 7.1 and I want to encode to AAC-LC 7.1.

If I extract the core:
eac3to input.dtshd output.dts -core

I get an 5.1 .dts. After that I can encode to aac... but I will have an AAC-LC 5.1. I want 7.1 :(

If I try to extract on pcm:
eac3to input.dtshd output.wavs

The program begin to extract 8 tracks... but it goes wrong... after some minutos one error comes up: "The ArcSoft DTS Decoder reported an error while decoding. Aborted at file position 11567589376"

I've tried to .flac too, but withou sucess, same error.

eac3to v3.24
command line: eac3to "INPUT_ING.dtshd" "OUTPUT_ING.flac"
------------------------------------------------------------------------------
DTS Master Audio, 7.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48kHz)
Decoding with ArcSoft DTS Decoder...
Encoding FLAC with libFlac...
Creating file "OUTPUT_ING.flac"...
The ArcSoft DTS Decoder reported an error while decoding. <ERROR>
Aborted at file position 1167589376. <ERROR>

What I'm doing wrong?

Cheers!

Adub

29th June 2012, 00:32

(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48kHz)

The core IS 5.1, that's why it's called the core. You won't get 7.1 by extracting the core.

Asmodian

29th June 2012, 00:34

That looks like a corrupted DTS-HD file to me, given the error is in ArcSoft not eac3to I don't think a setting will help.

Maybe there was an error while ripping (I assume this file was from a bluray?).

Leinad4Mind

29th June 2012, 00:39

I know if I extract the core I get an 5.1 xD I was just explaining.

And exactly, Asmodian, its from an bluray.

eac3to v3.24
command line: eac3to "Adormecida_ING.dtshd" "Adormecida_ING_dts.wavs" -16 -48000 -8 -little
------------------------------------------------------------------------------
DTS Master Audio, 7.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48kHz)
Decoding with ArcSoft DTS Decoder...
Writing WAVs...
Creating file "Adormecida_ING_dts.L.wav"...
Creating file "Adormecida_ING_dts.C.wav"...
Creating file "Adormecida_ING_dts.BL.wav"...
Creating file "Adormecida_ING_dts.SL.wav"...
Creating file "Adormecida_ING_dts.R.wav"...
Creating file "Adormecida_ING_dts.LFE.wav"...
Creating file "Adormecida_ING_dts.BR.wav"...
Creating file "Adormecida_ING_dts.SR.wav"...
The ArcSoft DTS Decoder reported an error while decoding. <ERROR>
Aborted at file position 1167589376. <ERROR>

So... the dtshd must be corrupt. I will extract again from the bluray. Do you recomend TSMuxer or HdBrStreamExtractor to extract the audio?

Cheers!

sl1pkn07

29th June 2012, 01:25

Snowknight26

29th June 2012, 17:17

Does it really matter? It's 1 millisecond.

Asmodian

29th June 2012, 18:22

It isn't even a matter of noticing the 1 ms. You would still seak to the same frame given that a single frame is displayed for >16ms even with 60 fps material so nothing changes.

Maybe eac3to is setting chapter points to the nearest frame start time while the other programs are less aware of (or do not care about) the exact frame start/stop times. Or maybe they are rounding all the times to the nearest 10ms? I often see only two digits for ms time stamps in editing programs (HH:MM:SS.ss not HH:MM:SS.sss).

It would be nice to know why of course but I would trust eac3to over the others if I had to pick one.

frank

2nd July 2012, 07:43

AFAIK this 1 ms difference is a rounding error. Very annoying!

mkv is very picky.
When you jump to a chapter it will never get the key frame on that chapter point if the time is 1 ms shorter!
mkv jumps to the next frame starting WITHIN the time -40 ms...-41 ms (1 / fps). That is the key frame or the last decoded frame before the time stamp.

As workaround I add 2 ms to the chapters on every rip.

Leinad4Mind

3rd July 2012, 14:02

try eac3to directly

go to root of BD rip folder (example: c:\Adormecida) and tip "eac3to.exe 1) adormecida.aac". this extract directly from m2ts if BDMV is rip/unlocked by anydvdHD/dvdfabHD/makemkv backup option

I've tried that, but I got just 1h of the movie :(

eac3to v3.24
command line: eac3to.exe 1) adormecida.aac
------------------------------------------------------------------------------
M2TS, 2 video tracks, 7 audio tracks, 21 subtitle tracks, 1:15:10, 24p /1.001
1: Chapters, 30 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: DTS Master Audio, English, 7.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48kHz)
5: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz
6: AC3, English, 2.0 channels, 192kbps, 48kHz
7: AC3, Portuguese, 5.1 channels, 640kbps, 48kHz
8: AC3, Polish, 5.1 channels, 640kbps, 48kHz
9: AC3, Russian, 5.1 channels, 640kbps, 48kHz
10: AC3, Arabic, 5.1 channels, 640kbps, 48kHz
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), English
14: Subtitle (PGS), English
15: Subtitle (PGS), English
16: Subtitle (PGS), Portuguese
17: Subtitle (PGS), Polish
18: Subtitle (PGS), Russian
19: Subtitle (PGS), Arabic
20: Subtitle (PGS), Portuguese
21: Subtitle (PGS), Polish
22: Subtitle (PGS), Russian
23: Subtitle (PGS), Arabic
24: Subtitle (PGS), Portuguese
25: Subtitle (PGS), Polish
26: Subtitle (PGS), Russian
27: Subtitle (PGS), Arabic
28: Subtitle (PGS), Portuguese
29: Subtitle (PGS), Polish
30: Subtitle (PGS), Russian
31: Subtitle (PGS), Arabic
Track 4 is used for destination file "adormecida.aac".
[a04] Extracting audio track number 4...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Encoding AAC <0.50> with NeroAacEnc...
[a04] [1:00:41] The source file seems to be damaged (sync byte missing). <WARNING>
[a04] [1:00:41] The source file seems to be damaged (sync byte missing). <WARNING>
[a04] [1:00:41] The source file seems to be damaged (discontinuity). <WARNING>
[a04] The ArcSoft DTS Decoder reported an error while decoding. <ERROR>
Aborted at file position 15091105792. <ERROR>

This never happend to me before... Maybe the Bluray has some protections?! I will try to use DVD PassKey, and see if I pass to the hard drive without corrupting the audio.

Sparktank

3rd July 2012, 21:24

I just came across something interesting.
I bought Aliens on Blu-Ray and ripped it to MKV using MakeMKV (free while in Beta) and ran it through eac3to.
I found the audio for the director's cut had a range of bit-depth.

Here's the log:

eac3to v3.24
command line: "C:\Apps\eac3to\eac3to.exe" "G:\1080p\Aliens\Aliens.mkv" 1: Aliens.h264 2: Aliens.ac3 -448 3:Aliens_2.ac3 4: Aliens_3.ac3
------------------------------------------------------------------------------
MKV, 1 video track, 3 audio tracks, 2:34:27, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: DTS Master Audio, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
"Lossless"
3: AC3 Surround, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
"2/0"
4: AC3, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
"2/0"
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a03] Extracting audio track number 3...
[a04] Extracting audio track number 4...
[a03] Removing AC3 dialog normalization...
[a04] Removing AC3 dialog normalization...
[a02] Decoding with ArcSoft DTS Decoder...
[a02] Encoding AC3 <448kbps> with libAften...
[v01] Creating file "Aliens.h264"...
[a02] Creating file "Aliens.ac3"...
[a03] Creating file "Aliens_2.ac3"...
[a04] Creating file "Aliens_3.ac3"...
[a02] Original audio track: max 24 bits, average 19 bits, most common 16 bits.
Video track 1 contains 222185 frames.
eac3to processing took 1 hour, 14 minutes.
Done.

I haven't tried from the disc directly. Going to try that later and see if it's relatively the same.
Need to update DVDfab Passkey first.

Also haven't tried the theatrical edition either.

But from all the Blu-Rays I have, I've not seen that before.

Is that some kind of error or is there such a thing as VBR for Bit-Depth?
I know there's usually constant of 20 inside a 24 container.
so confused right now.
*sips coffee*

sl1pkn07

4th July 2012, 00:48

if try "eac3to 1) -check" ?

-check checks if the source EVO/(M2)TS file is clean.

setarip_old

4th July 2012, 01:56

@Sparktank

Hi!

What, if any, problems does this apparent anomaly cause for you?

Sparktank

4th July 2012, 05:12

@Sparktank

Hi!

What, if any, problems does this apparent anomaly cause for you?

Well, no problems at all.

Just curious as to what this is all about; never have seen this information before in eac3to. :p

Audio plays back fine.

ramicio

5th July 2012, 15:08

It just shows you the true bit depth of the audio. Some of that audio is 24 bits, and some is 16 bits. The total content of each average together to be 19 bits. Since it's only 19 bits average, most of it is 16 bits. It doesn't matter, anyway, since you converted to ac3, which doesn't have a bit depth, but it's decoded to 16-bit accuracy. Sometimes I see a most common 20 bits. I don't understand why there is no 20 bit, it has to be put into a 24 bit container.

xkodi

5th July 2012, 22:42

hi All, i missed that ETSI actually released DTS-HD specifications:

http://www.etsi.org/deliver/etsi_ts/102100_102199/102114/01.03.01_60/ts_102114v010301p.pdf

and i'm not sure if that was covered here.

so, i remember back in the old days when all DTS-HD headers were just partially guessed by 'madshi', i.e. mainly the channel configuration information from the header and eac3to was the only application that was actually able to display with '-logdts' undocumented switch and use that information from the DTS-HD headers.

so, that same limitation, i.e. lack of knowledge about DTS-HD headers was the reason that eac3to wasn't and it still is not able to decode all DTS-HD stream properly - for example "7.1 (strange setup)", because that same issue for DTS was solved by idea i suggested several times:

http://forum.doom9.org/showthread.php?p=1180214#post1180214

and at the end 'madshi' implemented, i.e. patching the DTS headers to channel configuration that is properly decoded and then remap the channels, as mentioned here:

http://forum.doom9.org/showthread.php?p=1180224#post1180224

so, the same idea was hard to achieve if not even impossible with DTS-HD due to lack of knowledge about the DTS-HD headers, but it seems with the ETSI PDF document with the DTS-HD specification that is no longer true.

so, eac3to is slowly getting outdated and it seems new open source initiative of new alternative to eac3to is a good idea - i'm sure i'm not the only one that is interested "7.1 (strange setup)" and other not properly decoded DTS-HD tracks by eac3to to be fixed. however, maybe netirely new tool is too much work and as a first step and proof of concept small tool that patches the DTS-HD headers, then use eac3to to decode and then another small tool to remap the channels is more feasible option, i.e. set of small tools to patch what eac3to is lacking as functionality. anyone interested?

73ChargerFan

7th July 2012, 19:51

I think you're underestimating the work involved duplicating eac3to. Just about every free tool out there uses it as a component.

BugiBugBug

10th July 2012, 08:20

I'm consistently experiencing sync problems when recoding the DTS track in my previously ripped MKVs to multichannel AAC.
The method I use is to load a given 720p MKV containing a DTS track, using the HdBrStreamExtractor with ArcSoft and NeroAAC, demuxing the H264 to a *.264 file, transcoding the DTS directly to AAC using -quality=0.45, and demuxing any contained SRT.

This goes without any problems (although it does almost every single time warn about some non-standard video bitstream stuff):

MKV, 1 video track, 1 audio track, 1:45:55, 24p /1.001
1: h264/AVC, English, 1280x536 23.975p
2: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz
[v01] The video bitstream is encoded in a non-standard framerate. <WARNING>
[v01] The video bitstream framerate field doesn't match the container framerate. <WARNING>
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Decoding with ArcSoft DTS Decoder...
[a02] Encoding AAC <0.45> with NeroAacEnc...
[v01] Creating file "D:\1_1_video.h264"...
Video track 1 contains 152373 frames.
eac3to processing took 20 minutes, 35 seconds.
Done.

When remuxing the stuff back to MKV with MKVToolnix, the audio is consistently a fraction of a second before the video; I've tested this with three video's now. The sync problem worsens over the course of the video; at the end the sync is worse than in the beginning.
Clearly my first idea is that this could have something to do with audio being sped up because of a 23,976 to 24 fps 'conversion' in the transcoding process. But I did not tell eac3to to do that. Maybe I'm missing something?

I've also noted that the resulting MKVs all have a 9ms delay applied to the audio; I do not understand where this comes from (it isn't present in mkvtoolnix stream settings), since no stream has this value incorporated, at least so MediaInfo tells me.

Does anyone have an idea what causes this?

Edit: I've tested setting mkv to 24p for muxing, but that doesn't work in the slightest bit; audio and video are now completely out of sync. This is driving me nuts - I had to free up disc space and deleted the source files, so now I'm stuck with 80GB of single streams that produce out-of-sync movies. *insert random power-curse*

kypec

10th July 2012, 08:46

When remuxing the stuff back to MKV with MKVToolnix, the audio is consistently a fraction of a second before the video; I've tested this with three video's now. The sync problem worsens over the course of the video; at the end the sync is worse than in the beginning.
Clearly my first idea is that this could have something to do with audio being sped up because of a 23,976 to 24 fps 'conversion' in the transcoding process. But I did not tell eac3to to do that. Maybe I'm missing something?

I've also noted that the resulting MKVs all have a 9ms delay applied to the audio; I do not understand where this comes from (it isn't present in mkvtoolnix stream settings), since no stream has this value incorporated, at least so MediaInfo tells me.

Does anyone have an idea what causes this?

Edit: I've tested setting mkv to 24p for muxing, but that doesn't work in the slightest bit; audio and video are now completely out of sync. This is driving me nuts - I had to free up disc space and deleted the source files, so now I'm stuck with 80GB of single streams that produce out-of-sync movies. *insert random power-curse*
I use eac3to only for audio re-encoding tasks, DTS->AAC like you do. Then I just use MKVToolnix to remux original MKV with M4A audio and I've never experienced any sync problems. That 9ms delay is probably a default AAC decoder delay which is inherently present in every AAC audio file, don't worry about it, mkvmerge just does its job reliably.

BugiBugBug

10th July 2012, 08:54

I use eac3to only for audio re-encoding tasks, DTS->AAC like you do. Then I just use MKVToolnix to remux original MKV with M4A audio and I've never experienced any sync problems. That 9ms delay is probably a default AAC decoder delay which is inherently present in every AAC audio file, don't worry about it, mkvmerge just does its job reliably.

Then this is -very- weird, because every file I remux is out of sync. One worse than the other (two were completely unwatchable).
I did have problems with my hard drives - it seemed the AHCI modus used on my AMD SB700 rig caused some writing/reading errors, maybe some bits have been 'shifted around' without me knowing. It never told me though, untill I started getting errors of programs being corrupt and such. Now I reinstalled my rig in RAID mode (single discs), which allows the use of RAIDXpert, which monitors stuff like that. Who needs S.M.A.R.T. if it doesn't work...
Luckily I know now I did not do anything wrong out of stupidity...

sneaker_ger

10th July 2012, 14:21

Mkvmerge automatically reads and applies the audio delay if present in apple style in the mp4 container. This is the case for files created by NeroAacEnc (used by eac3to). That explains the 9ms delay - not the progressive desync, though. The video is probably not strictly constant 24/1.001 fps.

BugiBugBug

10th July 2012, 15:26

Yes, I've been able to solve the problem for some of the movies by manually setting the frame duration to 24/1001 - this somehow proved to work best.
The one that proved progressively out of sync I removed already, so no way to test that. My guess is it would have improved as well.
The very badly out of sync ones were probably damaged; it was quite blocky here and there, artifacts showed up indicating a broken stream.

I've now reverted to only demuxing/transcoding the audio track, and using the original MKV stream to remux using the new audio (and leaving out the DTS). This has been showing the best results, consistently.

So I guess it's best to change as little as possible, and only mux/remux what you need to mux/remux. It also saves me a lot of time, since demuxing triple number amounts of gigabytes is not something you do very lightly.

PS. On a side note; WHY for gods sake does everyone keep using the horribly inefficient DTS? Please just use AAC already! Who needs 1500kbit/s for a simple audio track?! It's nuts!

ramicio

10th July 2012, 15:27

BugiBugBug

10th July 2012, 15:50

The video shows as 23.975 FPS. What are you setting it to for muxing?

Eac3to is fine for tasks other than just audio encoding. If you're getting corrupt data, it's likely a hardware issue, not eac3to.
Good god, I did not even notice that five, I'm so used to seeing a particular number that I actually start seeing it when it isn't there...
Obviously the framerate that should have been used is different from the one of the stream; the log tells 24/1001, and the stream is 23.975. I'm guessing this very minor fraction caused the problems, and why setting it manually to 24/1001 solved these problems: an encode will in theory not be anything but one of the official frame durations. 23,975 is not the right number, and must be incorrect. 0,001 frame/s less does sum up to a larger delay in the end than in the beginning, because the video runs slower than the audio. This is really the culprit.

So the moral of the story is - keep track of your eac3to log files, and use another number if problems arise. And look at the numbers -closely- :)

marcusj0015

12th July 2012, 17:31

Hey Madshi, I have a question about Eac3to. Can I add flac specific options to the command line for Eac3to? Because Flac doesn't support writing files over 2GB, but eac3to can because it doesn't use Flac's built in IO stuff, and basically, I want to compress the flac file as heavily as possible. I'm only encoding once, might as well compress it as much as possible.

The commands from Flac I want to use are " -8 -e -p" -8 means best compression as I'm sure you know, but you may not know about the other ones. -e is "Exhaustive model search (expensive!). Normally the encoder estimates the best model to use and encodes once based on the estimate. With an exhaustive model search, the encoder will generate subframes for every order and use the smallest."

and -p is "Do exhaustive LP coefficient quantization optimization."

Basically, it'll slightly improve the file size, and over two hour file, I think it'd be worth it. anyway, how can I do this, and if I can't, can you add it?

Descriptions from Flacs documentation.

ramicio

12th July 2012, 20:55

-8 already implies the -e flag, and -p is not guaranteed to actually decrease file size.

Asmodian

13th July 2012, 01:36

Based on this post (http://forum.doom9.org/showthread.php?p=1222771&highlight=flac+compression+level#post1222771) from Madshi I think eac3to defaults to using compression level 8 for Flac.

marcusj0015

15th July 2012, 14:51

Thanks guys, I appreciate it. :)

rapscallion

15th July 2012, 17:24

Remember that theres a bigger Problem with the NeroAudio Decoder in eac3to! The DRC isnt removed properly so the Decoder is not recommended

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

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

AC hasn't been around for some time now.

Does anyone know if the above applies to both DD and True-HD ?

`Orum

21st July 2012, 16:42

Had an interesting problem today when trying to go from AC3 -> FLAC with a channel remap (in preparation for oggenc2 encoding). Here's the log:

AC3, 2.0 channels, 1:35:39, 224kbps, 48kHz
The Nero decoder doesn't seem to work, will use libav instead.
Decoding with libav/ffmpeg...
Remapping channels...
Reducing depth from 64 to 24 bits...
Encoding FLAC with libFlac...
Creating file "audio_bonus-oggorder.flac"...
Clipping detected, a 2nd pass will be necessary.
Starting 2nd pass...
Decoding with libav/ffmpeg...
Remapping channels...
Reducing depth from 64 to 24 bits...
Encoding FLAC with libFlac...
Applying -0.22dB gain...
The libav decoder crashed.
Creating file "audio_bonus-oggorder.flac"...
Aborted at file position 262144.

It handles the first pass without issue, and then, almost immediately upon attempting the second pass, it crashes? Strange...

Any idea how to work around it? Only thing I can think of is to possibly drop in a new libav decoder, but not sure what is compatible.

tebasuna51

22nd July 2012, 00:54

I never see that.

Try with only one pass.
Ignoring the "Clipping detected..." with:

eac3to input.ac3 stdout.wav -no2ndpass | OggEnc2 -q X -o output.ogg -

Or applying manually the needed -0.22dB gain...

eac3to input.ac3 stdout.wav -0.22dB | OggEnc2 -q X -o output.ogg -

`Orum

22nd July 2012, 19:14

Or applying manually the needed -0.22dB gain...

eac3to input.ac3 stdout.wav -0.22dB | OggEnc2 -q X -o output.ogg -
I tried this but it throws an error: Command line parameter "0.22dB" is unknown. I think it expects whole numbers for this?

It does work with -no2ndpass, but then it has some, albeit minor, clipping.

tebasuna51

23rd July 2012, 09:48

I think it expects whole numbers for this?
You are right seems only work with integer values.

It does work with -no2ndpass, but then it has some, albeit minor, clipping.
Don't worry with the clipping, they are always encoder/decoder imperfections, the original source can't have peaks over 0dB.

Recode from lossy formats never can be exact.

ramicio

23rd July 2012, 15:31

Eac3to is due for a refresh...........................................

Overdrive80

23rd July 2012, 15:33

Had an interesting problem today when trying to go from AC3 -> FLAC with a channel remap (in preparation for oggenc2 encoding). Here's the log:

[CODE]AC3, 2.0 channels, 1:35:39, 224kbps, 48kHz
The Nero decoder doesn't seem to work, will use libav instead...

Do you have tried to use nero ac3 decoder??? Apparently, it isnt installed.

dream88

24th July 2012, 11:26

Eac3to is due for a refresh...........................................

Madashi has stated that once MadVR reaches v1.0 work on eac3to will continue.

`Orum

24th July 2012, 23:22

Do you have tried to use nero ac3 decoder??? Apparently, it isnt installed.

No. IIRC you needed a fairly old version of Nero for this, and I have a newer version installed which I needed for some new features (I forget what exactly).

rapscallion

24th July 2012, 23:38

Overdrive80

25th July 2012, 20:19

If eac3to doesnt remove DRC properly and I dont want use libavcodec. What could it be other option??? Maybe Audition??

ramicio

26th July 2012, 18:26

How doesn't it remove DRC properly? It's just a gain metadata. Why don't you want to use libavcodec? Why would one want to go from AC3 to FLAC?

rapscallion

26th July 2012, 18:44

@ramico, see my post above re the same, #11708.

ramicio

26th July 2012, 18:45

What?

rapscallion

26th July 2012, 18:46

What do you mean "what"? Did you read the post and the links.

ramicio

26th July 2012, 18:49

First off, my first post wasn't even replying to you, it was to Overdrive80. Secondly, I read it, and I don't see what it has to do with anything I said. The problem is not with eac3to, it's with a codec. I was asking Overdrive80 what was wrong with libavcodec, because it will function perfectly.

rapscallion

26th July 2012, 20:15

Whoa, slow down ther ramico.

I know you weren't replying to me, but Overdive's question was in response to my post, right above his.
You commented re DRC and metadata so my post has everything to do with what you said. I was hoping you could clarify things for him and me.

My post(s), I asked this back on the 15th as well (w/no responses): http://forum.doom9.org/showpost.php?p=1582897&postcount=11699 referenced the Nero codec, which ACrowley stated removal of drc was a problem.

We are here to help each other, no ?

Overdrive80

27th July 2012, 00:52

Ramicio, I havent any reason for not use libavcodec. My question is mainly based on the curiosity to know if there are alternatives.

my question is a consequence of assertion of rapscallion. Is there technical data for it??? I actually use nero for disable drc on my projects, and it worry me.

Thanks

ramicio

27th July 2012, 16:37

Why not just use libavcodec, which comes with eac3to, instead of a third party codec that doesn't even work?

rapscallion

27th July 2012, 17:10

Why not just use libavcodec, which comes with eac3to, instead of a third party codec that doesn't even work?
From the first post in this thread:
The libav/ffmpeg (E-)AC3 and DTS decoders work really well, but the output is not 100% identical to the Dolby/DTS reference decoders.

So, it has always seemed safer to me ( and Overdreive) to use Nero. Until, of course, the drc question. Do you know if the "problem" applies to both DD and TrueHD ?

ramicio

27th July 2012, 17:14

It shouldn't apply to TrueHD. It's really MLP in disguise, and is lossless. Specs weren't hidden for it. EVERY different lossy decoder will give a different decoded result. This is the nature of all of this psychoacoustic lossy pseudoscientific pseudomathematic BS. Want consistency? Go lossless.

rapscallion

27th July 2012, 17:20

It shouldn't apply to TrueHD. Go lossless.
Thanks for the info...that is what I was hoping !

Always go lossless. However, and just my personal prefrence, I use eac3to to create the wavs from a BD TrueHD track and then use DTS MA Suite to create a DTS-HD track to mux back with the original video.

Thus, my reason for asking.

ramicio

27th July 2012, 17:21

Why the love for DTS-HD?

rapscallion

27th July 2012, 17:23

Well, and this has certainly been a matter of discussion on the boards, I'm not a fan of DRC.

ramicio

27th July 2012, 17:27

I really don't get what you're talking about again. Just ripping the track will remove the DRC metadata. The audio inside of TrueHD will remain lossless. TrueHD > DTS-HD Master IMO because it allows you to run just the lossless portion without a core. But I don't like either codec in the end. I use FLAC.

rapscallion

27th July 2012, 17:46

Ok, I left out another important reason.

When playing a BD, with a TrueHD track, in the bedroom non-surround system, the volume is always too low, even when turned up to the tv's max.

ramicio

27th July 2012, 17:49

But this is before you do anything to it. Before you were obviously referring to copying your discs onto your computer. The DRC would be removed.

Playing back a blu-ray the TV would be decoding an AC-3 track, anyway, not a TrueHD track, or even a DTS-HD track.

I am still not following you.

rapscallion

27th July 2012, 18:20

BD to pc, pc to disc, to be played on a SA player.

Correct, however the core volume between DD and DTS is dramatically different.
I've found that eac3to doesn't extract TrueHD correctly (Mediainfo doesn't recognixe it and VLC won't play it), but it does fine with wavs as the output.

Tsmuxer demuxes TrueHD with the DRC intact.

Maybe the trouble following is that we have 2 completely differen goals. Thank you for the info, I do learn something new everyday even though I'm retired from 35 years in the audio business.

ramicio

27th July 2012, 18:26

BD to pc, pc to disc, to be played on a SA player.

I just have to ask. Why? I am assuming "pc to disc" means you are ripping it, doing something to it, and burning it to another disc. Eac3to does just fine demuxing TrueHD tracks and stripping the core. Trusting VLC for anything is part of your problem. Playing this back on a TV I wouldn't even bother using any audio track except the mandatory 2.0 DD.

rapscallion

27th July 2012, 18:37

Correct, ripping it. Doing something is eliminating all the fluff extras, warnings, duplicate studio trailers, menus, etc.

I mentioned the bedroom TV, however the sytem in our den is a complete surround sytem so just the 2.0 track is a no go.

VLC does play the Tsmuxer demuxed HD track fine but not the same by eac3to. Then there's Mediainfo.
You've got me curious though...gonna run a few tests.

ramicio

27th July 2012, 18:39

I just don't get the point of ripping something just to burn it again to remove the fluff. People remove the fluff when ripping to hard drives. I would just get into the program of using networked media players, store them on a NAS, and just use something like FLAC for the audio.

rapscallion

27th July 2012, 19:42

Well, to each his own. I didn't want to take this thread too far OT.

I prefer using an SA player. Additionally, I only do this for discs that have TrueHD, extremely long load times ( a problem w/my Pio BDP-51) or discs that require a new firmware update (my Sammy 1200 doesn't get fw updates anymore).

Just tested a True HD track w/eac3to and it will extract the TrueHD or the core , but not demux both the way Tsmuxer does.

I was mistaken re VLC, I should have stated Media Player Classic won't play the eac3to TrueHD.

ramicio

27th July 2012, 19:45

Have you tried something like "eac3to <input> track: output.thd+ac3"? I don't get what effect the codec would have on load times.

rapscallion

27th July 2012, 19:49

I don't use the command line, I use the eac3to and More GUI.

Codec doesn't affect load times, of course. I'm talking about the rip removing all the Java that does affect it.

ramicio

27th July 2012, 19:52

And that GUI won't show you the command? I don't get the point of what you're doing. It just seems like a giant waste of time. You're better off just getting a TV with Netflix or something, if you like wasting money.

rapscallion

27th July 2012, 19:59

And that GUI won't show you the command?
Yes, however you can't edit/add to it.

ramicio

27th July 2012, 20:00

Seems like a piece of crap to me. Eac3to is pretty straight forward to use. Typing commands is trivial compared to the time it takes to rip.

Sparktank

28th July 2012, 00:01

I've been curious about the undocumented swtiches and options that eac3to can use.

I've stumbled upon some here and there through the thread and kept an extra text file handy for such occasions. Also found some pointed out in the chagelog.

Is there a command or switch to display a complete list of operations?

-fullhelp

Or something like that?
I tried to google using different search functions and found pretty much the ones I ran into.

-normalize
-analyzebitdepth
-check16bit
-no2ndpass
-neroaacenc="differentpathifnotdefaulteac3tofolder"
-mono
-progressnumbers

robpdotcom

28th July 2012, 02:17

I don't use the command line, I use the eac3to and More GUI.

You could try UsEac3To. It has an option to extract as thd+ac3.

Abradoks

28th July 2012, 04:34

Is there a command or switch to display a complete list of operations?
There was an attempt to document them on wikibooks. Feel free to add anything missing there (http://en.wikibooks.org/wiki/Eac3to/How_to_Use#Command_Line_Syntax).

Sparktank

28th July 2012, 08:04

There was an attempt to document them on wikibooks. Feel free to add anything missing there (http://en.wikibooks.org/wiki/Eac3to/How_to_Use#Command_Line_Syntax).

:goodpost: Wow, I didn't even see that page before!
Thanks!

tebasuna51

28th July 2012, 13:09

There was an attempt to document them on wikibooks. Feel free to add anything missing there (http://en.wikibooks.org/wiki/Eac3to/How_to_Use#Command_Line_Syntax).

I add some parameters and info to this page.

rapscallion

28th July 2012, 18:07

You could try UsEac3To. It has an option to extract as thd+ac3.
Thank you, I wasn't aware of that GUI....works great.

Lincoln Burrows

1st August 2012, 16:36

Is there a way for me to use this kind of command without changing the bitrate for 128/44?

eac3to originalfile.mp3 stdout.wav -slowdown | lame - new-file-created.mp3

And how do I change the fps from a file (also without changing the bitrate/Hz) from 23.976 to 24? Or 24 > 23.976?

P.S. It seems the command is:

eac3to original-file new-file -23.976 -changeTo 24.000-23.976 = define fps from the source + new fps.

However, if I am dealing with AC3, the bitrate is also changed (this time, to 448 Kbps). Why?

sneaker_ger

1st August 2012, 17:31

Changing the speed means eac3to has to re-encode the audio and 448 is the default bitrate.

tebasuna51

2nd August 2012, 01:18

With Lame you can specify the bitrate (CBR 128 Kb/s is the default) with:

CBR: lame -b 160 - new-file-created.mp3
VBR: lame -V 2 - new-file-created.mp3
With V (0 best .. 5 medium .. 9 low quality)

With internal AC3 encoder (448 default for stereo, 640 for 5.1)
you can use -192, -224, -256, -320, -384, -448, -512, -576 or -640 to specify your desired bitrate

Skwelcha

9th August 2012, 21:58

I get a "Patching bitdepth to 24 bits..." message when I try to extract the DTS core of a 16bit DTS-MA track and the DTS core is 24bit after that. Shouldnt it bet 16bit?

Snowknight26

9th August 2012, 23:23

Ignore that. Lossy formats don't have a bit-depth.

ramicio

10th August 2012, 14:52

Because you are extracting the lossy core. DTS is usually labeled as 24 bit despite people claiming that there is no such thing as bit depth in lossy formats.

Skwelcha

10th August 2012, 16:52

alright weird though that mediainfo reports 16bit sometimes and other times 24bit DTS tracks, but I will ignore it from now on then.
Thanks for the help.

NanoBot

10th August 2012, 21:26

The header of the DTS core contains a field "Source PCM resolution", which should indicate the bit depth of the original pcm files, before they where compressed with the DTS codec. Nevertheless, this value can be altered manually during / after the encoding, so it can be a fake value. And even if the source PCM really was 24bit PCM, this bit depth is likely to be almost lost, since the DTS core, as already said, is a lossy compression.

Therefore, if I would like to decode a DTS core to a lossless format like PCM or flac using eac3to, I would always use the "-down16" switch, since the additional size of a 24bit target file IMHO is wasted space on my HDD.

C.U. Nanobot

MGSteve

14th August 2012, 09:08

Has anyone had issues extracting the Right eye channel from Despicable Me 3D? Every time I do it, I get an out of memory error.

Only the right eye, everything else extracts fine. It even does it if I try and extract it straight from the disc.

ramicio

19th August 2012, 18:53

tebasuna51

20th August 2012, 11:09

So, for 5.1 eac3to writes a 0x0000060F dwChannelMask, but MadFLAC ignores that and does 0x0000003F. I know madFLAC doesn't change all channel masks, because I have some that are odd and they still show through (0x00000107). Seeing that both things are by the same author, why 0x0000060F? 5.1 uses rear speakers, not side. The "S" when labeling "SR" or "SL" doesn't mean "side," it means "surround." Obviously the audio device needs the correct channel mask for mixing or whatever, so I think when eac3to deals with 5.1 it should do 0x0000003F.
This was discussed already in this thread.

Until XP SP2, if I remember well, only the channel mask 0x0000003F was defined by M$ like standard 5.1 (Surround channels -> Back channels BL-BR), but after XP SP2 the channel mask 0x0000060F was the preferred to 5.1 (Surround channels -> Side channels SL-SR), maintaining the old mask like valid 5.1

Then new soft, M$ compliant, must accept both mask like equivalent for 5.1, and old soft maybe can reject 0x0000060F.
I recommended to madshi use always 0x0000003F, but ...

Also, if I strip the header off of a WAV file and specify the input parameters, does it eac3to read the standard WAV channel ordering or does it assume some other funky (lpcm) ordering?
If you use, for a 6 chan, 16 bits, 48 KHz:

-override -little -6 -16 -48000

the channel order is assumed like WAV

-override -big -6 -16 -48000

the channel order is assumed like PCM

ramicio

20th August 2012, 17:27

This was discussed already in this thread.

Until XP SP2, if I remember well, only the channel mask 0x0000003F was defined by M$ like standard 5.1 (Surround channels -> Back channels BL-BR), but after XP SP2 the channel mask 0x0000060F was the preferred to 5.1 (Surround channels -> Side channels SL-SR), maintaining the old mask like valid 5.1

Then new soft, M$ compliant, must accept both mask like equivalent for 5.1, and old soft maybe can reject 0x0000060F.
I recommended to madshi use always 0x0000003F, but ...

It's weird. Eac3to writes 0x0000060F, but madFLAC ignores that and sends 0x0000003F to the audio renderer. As long as madFLAC gives me 0x0000003F, I'm happy.

If you use, for a 6 chan, 16 bits, 48 KHz:

-override -little -6 -16 -48000

the channel order is assumed like WAV

-override -big -6 -16 -48000

the channel order is assumed like PCM

Thank you :)

Furiousflea

24th August 2012, 07:38

Hi there, getting a problem with two RAW\PCM 2.0 24bit 96khz tracks on "Beatles - Yellow Submarine" Bluray...

[07:31:59] Retrieving features
[07:31:59] Arguments: "C:\Users\Rob\Desktop\YELLOW SUBMARINE"
[07:31:59] Feature Retrieval Completed
[07:32:00] Retrieving streams
[07:32:00] Arguments: "C:\Users\Rob\Desktop\YELLOW SUBMARINE" 1) -progressnumbers
[07:32:01] Stream Retrieval Completed
[07:33:08] Extracting streams
[07:33:08] Arguments: "C:\Users\Rob\Desktop\YELLOW SUBMARINE" 1) (HAVE CHOPPED THIS OFF TO STOP MESSING UP FORUM FORMATTING - JUST COMPRESSING AUDIO TO FLAC AND EXTRACTING SUBTITLES\VIDEO)
[07:33:09] Creating file "D:\source\yellow submarine\1_1_chapter.txt"...
[07:33:09] s09 Extracting subtitle track number 9...
[07:33:09] a04 Extracting audio track number 4...
[07:33:09] v02 Extracting video track number 2...
[07:33:09] v02 Muxing video to Matroska...
[07:33:09] a05 Extracting audio track number 5...
[07:33:09] a05 Reading RAW/PCM...
[07:33:09] a05 Swapping endian...
[07:33:09] a05 Encoding FLAC with libFlac...
[07:33:09] a08 Extracting audio track number 8...
[07:33:09] a03 Extracting audio track number 3...
[07:33:09] a03 Decoding with ArcSoft DTS Decoder...
[07:33:09] a03 Encoding FLAC with libFlac...
[07:33:09] s10 Extracting subtitle track number 10...
[07:33:09] a04 Reading RAW/PCM...
[07:33:09] a04 Swapping endian...
[07:33:09] a04 Encoding FLAC with libFlac...
[07:33:12] a04 The ts pcm reader received a PCM block in an unexpected format.
[07:33:12] a05 The ts pcm reader received a PCM block in an unexpected format.
[07:33:13] a03 Creating file "D:\source\yellow submarine\1_3_audio.flac"...
[07:33:14] a08 Creating file "D:\source\yellow submarine\1_8_audio.ac3"...
[07:34:36] s10 Creating file "D:\source\yellow submarine\1_10_subtitle.sup"...
[07:35:52] s09 Creating file "D:\source\yellow submarine\1_9_subtitle.sup"...

Any ideas, thanks.

EDIT - Did a bit of investigating, looks like there were 3 m2ts in the mpls, the first one had didn't have PCM streams in it and the last one was just black for a few seconds.

Just loaded the m2ts directly and seems to have resolved the problem except no chapters hope I can get those out another way but I guess this is RESOLVED.

ramicio

24th August 2012, 15:08

That Blu-ray is an absolute mess to work with. The tracks are 24/48, even though they report as 24/96.

jpsdr

25th August 2012, 11:05

I've encouter cases where chapters are not on the m2ts file but only on the .mpls file. You have to get them from it.

mastrandrea

31st August 2012, 00:21

ramicio

31st August 2012, 14:37

eac3to <input> <output> -down6 -640 (ac3 bit rate, can be any standard bit rate number)

the_weirdo

31st August 2012, 15:27

I need to convert a 7-channels dts to ac3 5.1.
The channels are mapped as this:
Front: L C R, Side: L R, Back: C, LFE / Front: L C R, Side: L R, LFE (from MediaInfo).

I do not own any commercial DTS encoder, are there any other tools I can use with eac3to to process this file?

Thanks
eac3to <input> <output> -down6 -640 (ac3 bit rate, can be any standard bit rate number)

I like to add that according this post (http://forum.doom9.org/showthread.php?p=1524791&post1524791) and this post (http://forum.doom9.org/showthread.php?p=1524301#post1524301) you'll need to remap channels when downmix 6.1 to 5.1 with eac3to.

Also, if you mean you don't have any commercial DTS decoder (like ArcSoft or Sonic) then eac3to will using libav to decode and it can only output 5.1 anyway. Not sure in this case you'll need to remap or not.

tebasuna51

31st August 2012, 19:54

The ramicio solution must work always.

- The source is 7.1 then not needed remap (only 6.1 need)
- If a decoder like ArcSoft is not installed (Sonic don't decode 7.1, I think) the -down6 is ignored.

BTW, there are demo versions of ArcSoft decoder than work with eac3to. Use the search.

the_weirdo

31st August 2012, 22:04

...a 7-channels dts to ac3 5.1.
The channels are mapped as this:
Front: L C R, Side: L R, Back: C, LFE / Front: L C R, Side: L R, LFE (from MediaInfo)...

Well, from what he states in his post and information from MediaInfo, I guess his DTS is 6.1, not 7.1.

tebasuna51

31st August 2012, 22:46

Well, from what he states in his post and information from MediaInfo, I guess his DTS is 6.1, not 7.1.

Sorry, you are rigth. My mistake.

Then, if you have ArcSoft/Sonic decoder the remap is necesary:

eac3to <input> <output> -down6 -0,1,2,3,5,6,4

mastrandrea

1st September 2012, 10:24

Thank you guys, I've tried with the Arcsoft Decoder (1.1.0.7) and your options, but it always seems that the ouput is very compressed and with no low frequencies...

tebasuna51

1st September 2012, 11:30

... but it always seems that the ouput is very compressed and with no low frequencies...

Then there are two options:

- Or you have a very expensive high quality audio system and very good ears (Please say your player, amplifier/receiver and speakers).

- Or your audio system have a different calibration for DTS and AC3.

BTW, the low frequencies are very easy to code, and have no sense to ear differences, maybe at high frequencies...

Azal

3rd September 2012, 12:36

I just came across to this phenomenon that a LPCM track extracted off of a .M2TS with eac3to was about 50% smaller than what I got when I demuxed the same audio track with tsMuxeR. You can find a screenshot in a link below.

http://img405.imageshack.us/img405/8383/extractedlpcmtrackshrun.png
(audio.wav is extracted with eac3to and 00000.track_4352.wav with tsMuxeR from same source - on the side you can see both WAV's opened in MPC with property pages opened)

I started suspecting something was wrong when I noticed that the extracted WAV had changed in attributes from 24bit to 16bit and was noticeably small in size. Also the bitrate had dropped from stated original 2304 to 1536kbps. I tried tsMuxeR and got a WAV corresponging the original track. This worries me alot, because this is the first time I noticed this and I've been extracting LPCM tracks with eac3to a long time, because MKVtoolnix cannot read them directly from the M2TS files. It seems like only the "core" track of the LPCM was extracted with eac3to with a loss of quality. It makes me wonder how many LPCM tracks I've downgraded and remux back into MKV unnoticed? I should say I've never really familiarized myself with the advanced settings of eac3to and I might just be doing something wrong command-wise. Can you please shed light into this - why am I getting smaller WAV tracks with eac3to and is it at an expense of quality, not by some redundant overhead data?

sneaker_ger

3rd September 2012, 13:25

Azal

3rd September 2012, 13:49

No quality is lost. As the log says:
http://www.abload.de/img/extractedlpcmtrackshr3cppx.png

Of each 24 bits, 8 bits are simply zeros, which can be safely removed. This saves a lot of space and results in correct 16 bit depth instead of the misleading "24 bits" it is supposed to have.

This sounds like great news. Also a lot like misleading advertising as this is from a retail Michael Bublé concert Blu-Ray (notice the audio details in a link below).
http://www.blu-ray.com/movies/Michael-Buble-Caught-in-the-Act-Blu-ray/4643/
Apologizes if I make you repeat yourself but I've never really got about to get into and read what is the difference of 16 or 24bit compared to simple kbps and is it that the kbps drop is also result of the bits "downgrade"? Nothing is really lost in that regards either?

sneaker_ger

3rd September 2012, 14:01

48 kHz * 24 Bit * 2 (Channels) = 2304 kbps
48 kHz * 16 Bit * 2 (Channels) = 1536 kbps

Hope that helps.

Azal

3rd September 2012, 14:38

48 kHz * 24 Bit * 2 (Channels) = 2304 kbps
48 kHz * 16 Bit * 2 (Channels) = 1536 kbps

Hope that helps.

Thank you sir!

Keiyakusha

3rd September 2012, 15:43

By the way about this stripping of zeroes. It works with any input or only LPCM and THD (maybe others)? I mean if I converted audio to flac using different software, if I will run it through eac3to it will still be able to detect zero bytes?
Edit: Never mind. Tried myself. Yes it will.

Joniii

4th September 2012, 14:28

Anyone know why eac3to gives this error about Surcode 1.0.29 on Win7/8 64-bit. I remember it working fine on 32-bit, but haven't been able to use it on 64-bit OS.

G:\eac3to>eac3to D:\Test 1) 3: e:\test.dts
M2TS, 1 video track, 6 audio tracks, 12 subtitle tracks, 2:30:40, 24p /1.001
1: Chapters, 28 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 5.1 channels, 16 bits, 48kHz
4: AC3, English, 5.1 channels, 640kbps, 48kHz
5: DTS, French, 5.1 channels, 24 bits, 1509kbps, 48kHz
6: AC3, French, 5.1 channels, 640kbps, 48kHz
7: DTS, Spanish, 5.1 channels, 24 bits, 1509kbps, 48kHz
8: AC3, Spanish, 5.1 channels, 640kbps, 48kHz
9: Subtitle (PGS), English
10: Subtitle (PGS), English
11: Subtitle (PGS), French
12: Subtitle (PGS), Spanish
13: Subtitle (PGS), Norwegian
14: Subtitle (PGS), Danish
15: Subtitle (PGS), Swedish
16: Subtitle (PGS), Finnish
17: Subtitle (PGS), Icelandic
18: Subtitle (PGS), Dutch
19: Subtitle (PGS), French
20: Subtitle (PGS), Spanish
a03 Extracting audio track number 3...
a03 Reading RAW/PCM...
a03 Swapping endian...
a03 Remapping channels...
a03 Writing WAVs...
a03 Creating file "e:\test.R.wav"...
a03 Creating file "e:\test.SL.wav"...
a03 Creating file "e:\test.C.wav"...
a03 Creating file "e:\test.LFE.wav"...
a03 Creating file "e:\test.L.wav"...
a03 Creating file "e:\test.SR.wav"...
a03 The original audio track has a constant bit depth of 16 bits.
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.29.0.
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

G:\eac3to>

ramicio

4th September 2012, 14:49

Why not just stop messing around with old archaic garbage and just compress that PCM track to FLAC and see what the resulting bit rate is? 16-bit 5.1 tracks tend to be around the same or much much lower than 1,536 kbps, and it's lossless instead of lossy.

Joniii

4th September 2012, 14:59

ramicio

4th September 2012, 15:29

There's no excuse to have a surround system and still not have HDMI.

Shevek

4th September 2012, 15:52

There's no excuse to have a surround system and still not have HDMI.

Sorry, but I have a c.2001 Pioneer 5.1 surround system which doesn't have HDMI and only has 2 optical audio inputs.

It still behaves flawlessly and I have no intention of upgrading until it stops working.

ramicio

4th September 2012, 15:53

Cheap... Why even bother with an HD picture then if an old CRT still functions? Why pursue anything better, ever?

Shevek

4th September 2012, 16:12

Cheap... Why even bother with an HD picture then if an old CRT still functions? Why pursue anything better, ever?

Not cheap at all!!! Your comparison doesn't hold water.

CRT vs HDTV is a HUGE difference.

All my devices connect video to my 2008 HDTV via HDMI

Audio where required is optical to the Amp for 5.1

The only thing I don't in my setup is 7.1 audio.

5.1 vs 7.1 is not such a big difference to me.

ramicio

4th September 2012, 16:14

The same argument could be had for the worthless old DVD lossy formats and the new lossless formats brought on by Blu-ray...

Shevek

4th September 2012, 16:31

The same argument could be had for the worthless old DVD lossy formats and the new lossless formats brought on by Blu-ray...

My point is, that updating my CRT to HDTV was worth doing regardless of the state of my old CRT.

Upgrading my 5.1 non HDMI amp purely to allow me to have 7.1 lossless audio is only going to be worth it once the amp dies.

There's nothing "cheap" about that.

ramicio

4th September 2012, 16:33

Shevek

4th September 2012, 16:45

Sure there is. It's also not an amp, it's a receiver. They do more than just amplify or decode a signal, and newer receivers do more with other devices than just take HDMI audio from a computer or STB. It's cheapness. It's '90s technology.

Well, if NASA had said similar then the Shuttle would have stopped flying many years before :)

YOU may not be happy with 5.1 audio

I however am, no point in arguing any further.

Potaytoes, potahtoes, tomaytoes, tomahtoes :)

ramicio

4th September 2012, 16:50

What are you talking about? Cheapness is why the Soviet space program never went anywhere. The antithesis of cheapness is why the US went to the moon. Space programs that are visible to the public are just pure entertainment. Hidden technology is centuries ahead of what the public has knowledge of. The shuttle was 1930s technology.

When did I say anything about 5.1 not being enough? My comments were about lossy garbage, not 5.1 versus 7.1. If you are still a cheapass and need 5.1 to go over TOSLINK there are things that encode on the fly.

Brazil2

4th September 2012, 17:14

Cheapness is why the Soviet space program never went anywhere.
That's why the only way to actually send people and supply to the I.S.S. and to bring people back to Earth from the I.S.S. is the cheap but solid as a rock good old Soviet Soyuz :p

ramicio

4th September 2012, 17:15

A space station isn't the moon...

e-t172

4th September 2012, 21:07

The same argument could be had for the worthless old DVD lossy formats and the new lossless formats brought on by Blu-ray...

No it can't. The difference between SD and HD is obvious to anyone. Hearing the difference between AC3/DTS and lossless is much, much more difficult.

tebasuna51

4th September 2012, 22:01

Please guys stop this off topic discussion.

Joniii

5th September 2012, 05:19

Cheap... Why even bother with an HD picture then if an old CRT still functions? Why pursue anything better, ever?

Cheap? My TV was 5500 € when I bought it, however I don't really see why I should upgrade my receiver yet even though it would only be around 600 €. As long as it works I'm happy with DTS.

Joniii

5th September 2012, 09:56

ramicio

5th September 2012, 15:25

Yeah, cheap.

LigH

5th September 2012, 18:18

Which ArcSoft software is suitable to get a version of the DTS decoder which works with eac3to? Are trial versions acceptable?

Sometimes the ArcSoft DTS decoder doesn't work even after you've installed the full retail software. This can usually be fixed by manually adding the ArcSoft "Bin" folder (e.g. "C:\Program Files\Common Files\ArcSoft\Bin") to your environment path. If you don't know how to do this, google "environment path".

I installed the trial of TotalMedia Theatre 5. But there is no path "C:\Program Files\Common Files\ArcSoft\Bin". Instead, there is a "C:\Program Files\Common Files\ArcSoft\esinter\Bin" (which doesn't seem to contain DLLs eac3to may need). Files which seem to be interesting (e.g. DtsDec.dll or dtsdecoderdll.dll) are in "C:\Program Files\ArcSoft\TotalMedia Theatre 5\Codec"; but adding that to the PATH doesn't help. So which DLL or AX is eac3to looking for?

sneaker_ger

5th September 2012, 18:35

Should be dtsdecoderdll.dll. Try copying to SysWOW64 or to the eac3to folder. Did you restart the system after adding the folder to PATH?

LigH

5th September 2012, 18:43

Yes, I restarted.

I copied dtsdecoderdll.dll to the eac3to folder, but that didn't work either.

Groucho2004

5th September 2012, 18:47

Yes, I restarted.

I copied dtsdecoderdll.dll to the eac3to folder, but that didn't work either.

I copied these files to my eac3to directory:
ASAudioHD.ax (don't forget regsvr32)
DtsDec.dll
dtsdecoderdll.dll
checkactivate.dll
MagCore.dll
MagPCMac.dll
MagUIEngine.dll
MagUIImage.dll
MagUIInter.dll

tebasuna51

5th September 2012, 19:00

Try with:

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

Maybe checkactivate.dll is not present in last's TMT versions, but eac3to need this file and work with the same file from first's versions.

ASAudioHD.ax must be registered in your DirectShow system.

LigH

5th September 2012, 19:08

Maybe checkactivate.dll is not present in last's TMT versions, but eac3to need this file and work with the same file from first's versions.

So it seems that the most recent version 5.3.1.146 is not compatible anymore. I will have to try to find an older version of TMT5.

sneaker_ger

5th September 2012, 19:22

My last post was indeed wrong, I just re-checked and I needed all files listed by tebasuna51 (except dtsdec.dll, it seems) and have the .ax registered.

LigH

5th September 2012, 19:29

Even an older installer I found (5.0.1.x) does not contain a "checkactivation.dll"; I hope you will be able to find a way to support newer versions of the software... or maybe another ArcSoft software (besides TMT5) is usable?

sneaker_ger

5th September 2012, 19:31

Since LAV works using only dtsdecoderdll.dll it probably means that madshi tried to avoid legal issues by checking for that file, too. We'll have to wait for him to start working on eac3to again.

tebasuna51

5th September 2012, 19:52

So it seems that the most recent version 5.3.1.146 is not compatible anymore. I will have to try to find an older version of TMT5.

Try add the needed file from here (http://forum.doom9.org/showthread.php?p=1392750#post1392750)

LigH

5th September 2012, 20:39

Interesting collection. The added stub of "checkactivation.dll" was certainly useful...

phate89

5th September 2012, 22:16

hi, i have nero 6 installed with nero ac3 decoder, but i discover that with some audio track do something strange and make me loose sync of the track, is it possible to set a different encoder by default (without have to set everytime manually with "-libav) for ac3?

tebasuna51

6th September 2012, 02:01

Sorry, but I think is not possible override the default decoder.

mastrandrea

6th September 2012, 19:08

Is there a way to convert 6 mono wavs into 5.1 ac3?
Thanks in advance

LigH

6th September 2012, 19:20

Yes. Even if (probably?) not with eac3to.

BeSweet supports it via a *.mux file (you can use BeLight as GUI to create that and feed it into BeSweet). Unfortunately, the quality of the AC3Enc plugin is quite bad, and many versions have been bugged regarding a too low volume.

Also, Aften supports reading from six mono channel WAVs. WAV to AC3 Encoder (http://code.google.com/p/wavtoac3encoder/) is a useful GUI here. The Aften AC3 encoder is not yet close to commercial encoders, but still possibly the best current freeware AC3 encoder. WAV2AC3 uses Aften as stand-alone encoder; eac3to uses Aften as a library.

the_weirdo

6th September 2012, 19:21

Or this way:
http://forum.doom9.org/showthread.php?p=1582629#post1582629

LigH

6th September 2012, 19:32

Yes, ffmpeg's AC3 encoder developed a bit since that AC3Enc plugin for BeSweet.

Interleaving the six channels to a 6-channel file may pass the 4 GB limit, therefore the W64 target is recommendable. The usual WAV header has only 32 bit size fields in their chunks.

Joniii

8th September 2012, 14:41

Question about RAW/PCM channel remapping. If I demux RAW/PCM from Blu-ray to RAW, eac3to swaps endian and remaps channels. If I later mux that RAW to Blu-ray M2TS with tsMuxeR, wouldn't I then have wrong channel order?

Oh, and second question if anyone here has some ideas. What would be the best way to join The Lord of The Rings EE main movies as those are on two discs? I would like to get those to mkv.

Shevek

8th September 2012, 15:05

Oh, and second question if anyone here has some ideas. What would be the best way to join The Lord of The Rings EE main movies as those are on two discs? I would like to get those to mkv.

MKVMerge (and it's GUI) has an append function. Add part 1 video and audio then click the append button in the GUI to add part 2 video and audio

Joniii

8th September 2012, 17:08

MKVMerge (and it's GUI) has an append function. Add part 1 video and audio then click the append button in the GUI to add part 2 video and audio

Thx, I'll try that. :)

Joniii

8th September 2012, 18:08

Didn't work, playback got screwed where the files were joined also the playtime of the file was over 5 hours.

manolito

10th September 2012, 00:15

I have a long 6-ch AAC file which I want to convert to a 6-ch AC3 file. The source file has been created by Nero AAC encoder, here is the MediaInfo analysis:
Audio
ID : 1
Format : AAC
Format/Info : Advanced Audio Codec
Format-Profil : LC
Codec-ID : 40
Dauer : 2h 19min
Bitraten-Modus : variabel
Bitrate : 321 Kbps
maximale Bitrate : 380 Kbps
Kanäle : 6 Kanäle
Kanal-Positionen : Front: L C R, Side: L R, LFE
Samplingrate : 48,0 KHz
Stream-Größe : 320 MiB (100%)
verwendete Encoder-Bibliothek : Nero AAC codec 1.5.4.0
Kodierungseinstellungen : -q 0.40
Kodierungs-Datum : UTC 2012-09-08 16:18:01
Tagging-Datum : UTC 2012-09-08 17:37:15

The file plays in all my software players. But eac3to somehow cannot be convinced to convert this file:
eac3to v3.24
command line: eac3to.exe i:\test.m4a i:\test.ac3
------------------------------------------------------------------------------
The format of the source file could not be detected. <ERROR>

Giving the file a different extension like .mp4 or .aac does not help either. What's wrong here?

Cheers
manolito

tebasuna51

10th September 2012, 00:49

To decode AAC with eac3to you need Nero 7 decoder installed (like say first post in this thread).

You can use Foobar2000 or BeHappy-BassAudio to recode AAC -> AC3 directly.
Or you can decode AAC to WAV (with Faad) and after use eac3to.

You can't change the extension of the file. A .aac file is AAC audio with ADTS headers. A .m4a file (mp4 with only audio) is AAC audio in mp4 container.

manolito

10th September 2012, 05:21

Thanks tebasuna for your quick reply.
To decode AAC with eac3to you need Nero 7 decoder installed (like say first post in this thread).
Sorry, I will certainly not install bloated software like Nero7 on my machine...:(
I have NeroAacDec in my eac3to folder though, and NeroAacDec can certainly decode my source file. Why in the world can't eac3to use NeroAacDec for decoding AAC files?

You can use Foobar2000 or BeHappy-BassAudio to recode AAC -> AC3 directly.
Or you can decode AAC to WAV (with Faad) and after use eac3to.
Yes, this will probably work, but frankly I am too lazy to learn using these programs (I read that BeHappy is a nightmare to set up) for just the simple task to transcode AAC to AC3.

Cheers
manolito

sneaker_ger

10th September 2012, 05:27

ffmpeg is fairly easy to use for this task:
ffmpeg -i input.m4a -ab 448k output.ac3

Sadly, eac3to cannot utilize neroaacdec.exe, nor can it handle AAC audio in mp4 container. (But ironically, it cannot create AAC audio without mp4 container...)

manolito

10th September 2012, 20:37

Thanks sneaker_ger,
I know that ffmpeg can make these conversions. Frankly I am a little worried about the quality. Do current ffmpeg versions use Aften for AC3 encoding? I could not detect "libaften" anywhere in a couple of recent binaries.:confused:

In the meantime I made my own little tool for AAC to AC3 conversions. It uses faad and Aften. Here is the readme:
AAC2AC3 - Simple audio conversion utility
-------------------------------------------

Converts AAC audio files to AC3 using FAAD and Aften.
Large multichannel files are supported.

Downmix of multichannel sources to stereo is supported,
also the encoding bitrate can be specified (CBR only).
Other effects like normalizing, DRC profiles or Dialog
Normalization are not supported.

Usage:
Use 'Drag and Drop' or 'Copy and Paste' to feed a source file
to the executable. A target file of the same name and the
extension .ac3 will be created in the source folder.

Known issues:
The executable is a compiled batch file. If your virus scanner
has a problem with it, file a complaint with the manufacturer...

For anyone interested, you can download it here:
http://www.sendspace.com/file/oxrf7s

Cheers
manolito

sneaker_ger

10th September 2012, 21:02

Aften is based on the ffmpeg encoder. I think that actually the ffmpeg one has had a few minor improvements while aften was already stalled.

Boulder

11th September 2012, 15:48

Has anyone tried to compile new avcodec.dll and avutil-50.dll to use with eac3to? I'm not sure if there have been any big changes but the ones used by eac3to are over 2 years old now.

kypec

11th September 2012, 20:44

Has anyone tried to compile new avcodec.dll and avutil-50.dll to use with eac3to? I'm not sure if there have been any big changes but the ones used by eac3to are over 2 years old now.
Few months ago I tried to replace those DLLs with newer versions - it didn't work. :(
madshi should have released this as open-source long time ago...

Boulder

11th September 2012, 20:47

The changes made by madshi are there in the "legal stuff" folder, so it should be possible to build new libraries. I was just wondering whether anyone's tried that out yet.

sexconker

11th September 2012, 23:54

Then this is -very- weird, because every file I remux is out of sync. One worse than the other (two were completely unwatchable).
I did have problems with my hard drives - it seemed the AHCI modus used on my AMD SB700 rig caused some writing/reading errors, maybe some bits have been 'shifted around' without me knowing. It never told me though, untill I started getting errors of programs being corrupt and such. Now I reinstalled my rig in RAID mode (single discs), which allows the use of RAIDXpert, which monitors stuff like that. Who needs S.M.A.R.T. if it doesn't work...
Luckily I know now I did not do anything wrong out of stupidity...

I had a hell of a time dealing with a recent video.
I was using eac3to to convert the DTS-HD MA track to a 7.1 AAC (Arcsoft to Nero), and the audio would become progressively out of sync regardless of how I muxed it.

I tried all manner of fiddling / manually specifying the video framerate, but nothing worked. In the end (2 days of slamming my head against a wall), I ended up going with 5.1 instead of 7.1 and everything worked fine.

No idea what's up. Next job I do I'll post about it here if I have the same issue.

nibus

12th September 2012, 02:32

Didn't work, playback got screwed where the files were joined also the playtime of the file was over 5 hours.

There's a number of topics about LOTR EE and merging the discs. Do a forum or google search on it.

Basically you have to cut off the last few seconds of the end of part 1, and about 1 second from the beginning of part 2. You can do this in mkvmerge.

This is because the audio stream ends before the video stream.

the_weirdo

12th September 2012, 11:40

The changes made by madshi are there in the "legal stuff" folder, so it should be possible to build new libraries. I was just wondering whether anyone's tried that out yet.

Those patches are too old, some are even redundant (for example, libavcodec can now output AC-3 and DTS as floating point via API call), so I don't think they can be easily applied to latest libav* codebase. Also, there are a lot API/ABI changes in libav* during past 2 years, so even someone manage to apply those patches and build new libraries, I doubt they can be used with current eac3to.

Now only hope that Madshi will soon release madVR 1.0 then he can return to develop eac3to.

ramicio

13th September 2012, 23:04

Sorry, but 99% of the people seem to be happy with the progress bar.

You took a poll on this? This isn't the days of 640x480 CRTs. People are running higher resolutions than full HD now, so I doubt too many people user tiny 80-column terminals.

Foofaraw

15th September 2012, 03:11

If one has PAL DVD audio of a US TV show (ie, originally NTSC) the speech sounds sped up - can eac3to re-pitch the sound so it sounds like the original, but where the total duration hasn't changed?

And if yes, what are the switches ?

Using -slowdown makes the resulting file longer in duration.

sneaker_ger

15th September 2012, 09:39

Eac3to does not offer pitch change without duration change.
You could either speedup both audio and video or use a different program to do the change, like e.g. SoX.

Foofaraw

15th September 2012, 14:10

Eac3to does not offer pitch change without duration change.
You could either speedup both audio and video or use a different program to do the change, like e.g. SoX.

Yeah I don't like to mess with the video. So I'll try looking at Sox - thanks for the suggestion sneaker_ger :)

Joniii

18th September 2012, 09:22

There's a number of topics about LOTR EE and merging the discs. Do a forum or google search on it.

Basically you have to cut off the last few seconds of the end of part 1, and about 1 second from the beginning of part 2. You can do this in mkvmerge.

This is because the audio stream ends before the video stream.

Thanks, found them. There was someone who figured the exact time to cut at the end of each disc one but it doesn't seem to be that accurate as there is still 107ms gap on audio. Is there any free program that can open m2ts and displays a frame numbers and timestamp so I can find the exact frame where to cut?

Dunno, maybe it would be a good option on eac3to if it could cut the video if audio ends before video?

ramicio

19th September 2012, 19:18

Why can't you just open this up so people can fix all of these stupid little issues and maybe make it cross-platform?

LigH

20th September 2012, 06:55

Cross-platform? How? It mainly uses software which is only available for Windows, if it doesn't use l(ib)av(codec|format) as fallbacks.

ramicio

20th September 2012, 13:15

libav is cross-platform. It mainly uses libav, and Windows stuff secondary. For most of the Windows-based stuff, there are better tools out there, anyway. Once DTS-HD gets built into libav, there is no reason eac3to couldn't be cross-platform. AC-3 decoding is handled just fine with libav. No need for Nero there. Surely there are other AAC encoders, no need for Nero again. You can't even have anything >7 installed, so yeah, really good there. I don't know why anyone would waste their time encoding lossy DTS in this day and age, so no need for SurCode. Go with 640 kbps AC-3. And for MKV muxing, well mkvtoolnix is cross-platform already, so no need for this stupid Haali muxer. This tool needs a major update, or for the source to be opened. It should not be relying on commercial software just to perform basic functions. The functionalities can be free, but someone needs to stop making tiny incremental updates of MadVR and just be done with it already. It's going to be 5-10 years before I see 1.0 being released, and by then there will already be another format for movies out there.

the_weirdo

20th September 2012, 14:57

Personally, I don't care if eac3to will be cross-platform or not (even I'm using multi-OS now, my main OS is still Windows anyway). But I agree with ramicio, this tool needs a major update. IMHO, if madshi is not going to maintain eac3to anytime soon, he should open its source so that others may contribute, fix and update it. He doesn't intend to commercialize it, after all. He said that he would resume its development once madVR 1.0 is released. But now when he's added micro version number to madVR with every increment took months to be done, then who know when will he reach his goal?

However, this is just my opinion. ea3to is a madshi's software, so he have full rights to do anything with it. I'll respect his decision on this problem regardless.

mbcd

20th September 2012, 15:39

Why dont people start to program an opensource variant themselves?

Madshi did eac3to in some years, but he had very hard work to get all information. Today its mostly all known.
Mostly there are programs out there that did a partial job that eac3to did, so maybee there is sourcecode available.

If a small group would take such a project, it should not take more than some months to get a better version of it.
And even if it takes more time ... there is something opensource ...

Overdrive80

22nd September 2012, 08:23

In my opinion if he would want to do his software as commercial, should not share his code. If this not the case, maybe would create a group of developers. Secure that on this forum everyone join it.

xkodi

23rd September 2012, 22:25

Today its mostly all known.
Mostly there are programs out there that did a partial job that eac3to did, so maybee there is sourcecode available.

If a small group would take such a project, it should not take more than some months to get a better version of it.
And even if it takes more time ... there is something opensource ...

i agree with you at least as far as DTS-HD decoding is concerned, which i will give for example, you can read my post here:

http://forum.doom9.org/showthread.php?p=1581365#post1581365

that post in short: before nothing about the DTS-HD headers was known, now there is PDF with the full specification. also, as it's clear from that post "eac3to" can't decode properly some DTS-HD streams, which is now very easy to be fixed with the idea i shared in that same post. in fact i coded simple test program of that idea as "proof-of-concept". last, but not least how to access directly Arcsoft decoder is now available as open-source thanks to "LAV Audio decoder" project. so, those 3 things: full specification of DTS-HD headers, the idea to patch the DTS-HD headers to "trick" Arcsoft decoder to decode properly even the streams it can't and direct access to Arcsoft decoder as how's done in "LAV Audio decoder" gives much better "eac3to" at least for DTS-HD streams and that can be coded in hours actually - as i've just mentioned i made very simple "proof-of-concept" for first two ideas and then after i patched the stream use "eac3to" to decode it. however, it seems there are no just people with enough free time and enthusiastic about such project to work on it.

73ChargerFan

24th September 2012, 02:29

eac3to is much more than a dts-hd to something else transcoder.

Anakunda

29th September 2012, 12:16

Hello,

would eac3to be able to use as input an specific audio track of MKV container directly?
Such that I have Matroska file with one video and 2 audio tracks, and I want to convert the 2nd audio track to <something else> without necessity of extracting it via mkvextract first...

73ChargerFan

29th September 2012, 15:28

Yep, just tried it, converted ac3 to aac with nero encoder. Worked great. But it won't replace the track within mkv, that will still require MkvToolNix.

Anakunda

29th September 2012, 15:34

Yep, just tried it, converted ac3 to aac with nero encoder. Worked great. But it won't replace the track within mkv, that will still require MkvToolNix.

That's good news. But how exactly do I tell eac3to to take the 2nd (non-default) audio track...I assume eac3to always picks the 1rst track..

73ChargerFan

29th September 2012, 16:31

Use the following command to list the tracks:
> eac3to mystuff.mkv

Then I'll wager track 1 is video, track 2 is audio 1, track 3 is audio 2
so use the following command to convert to aac whatever:
>eac3to mystuff.mkv 3: mystuff.aac

or to convert to ac3:
>eac3to mystuff.mkv 3: mystuff.ac3

Anakunda

29th September 2012, 16:39

>eac3to mystuff.mkv 3: mystuff.aac

Ah so I thougt the numbering only can be used for direct demuxing.
:thanks:

73ChargerFan

29th September 2012, 16:47

Hey, I didn't know about direct demuxing of mkv until you asked and I tried it.

Incast

29th September 2012, 17:40

Firstly, thanks to all involved with this great tool.

Embarrassing question but I'm going to ask it anyway. When eac3to reports an uncorrectable audio delay after demuxing a TS, should I be inputting that value into MKVmerge as the delay for the audio track or the inverse?

E.g.
"A remaining delay of +14ms could not be fixed"

What should my entry into mkvmerge for the delay entry on the audio track be: +14ms or -14ms?

Boulder

29th September 2012, 18:00

You need to put +14ms there.

Incast

29th September 2012, 18:11

Perfect - thanks for clarifying.

Thunderbolt8

16th October 2012, 19:56

is resampling (with eac3to) from a lower to a higher kHz frequency lossless? is it somewhat comparable when increasing the bitdepth from 16 to 24-bit that there will be then 8 zero bits and the actual audio quality remains the same?

Asmodian

16th October 2012, 21:52

No it isn't, it is like resizing a video.

Xorp

27th October 2012, 17:34

madshi, a few times I've had eac3to introduce video errors into the video.mkv. The only way I've caught them is by noticing them during playback. I don't think there's any way to scan for them right? In the below example I see the artifacts with multiple decoders, including lav, coreavc, mpc-hc internal and VLC.

playback from original m2ts: http://imageshack.us/a/img24/6346/eac3toerror1.jpg
playback from eac3to remuxed mkv: http://imageshack.us/a/img109/8760/eac3toerror2.jpg

Did not receive any error messages from eac3to.

The artifacts are gone after re-entering the same command and checking the re-remux, but I don't know if the 2nd one is 100% clean either.

Should this never happened or is the issue is caused by my computer?

madshi

27th October 2012, 17:42

That shouldn't be happening, Xorp. Can you reproduce such an error reliably? If the same problem occurs at exactly the same frame, when remuxing the same movie twice, then that sounds like a likely eac3to bug. In that case could you try to create a sample which shows this problem?

Marhaba

30th October 2012, 14:31

Hi...

Since Last Few Days I'm Struggling To Downgrade With My eac3to - Below Is The Error.

eac3to v3.24
command line: eac3to324\eac3to.exe D:\input.dts D:\output.dts -768
------------------------------------------------------------------------------
DTS Master Audio, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
Decoding with ArcSoft DTS Decoder...
Writing WAVs...
Creating file "D:\output.SL.wav"...
Creating file "D:\output.L.wav"...
Creating file "D:\output.R.wav"...
Creating file "D:\output.LFE.wav"...
Creating file "D:\output.SR.wav"...
Creating file "D:\output.C.wav"...
The original audio track has a constant bit depth of 24 bits.
Encoding DTS <768kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.23.0.
Surcode says/asks: "Parallel port lock required to run this program.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

I Have Tried All Available Solutions , Plz Guide Me.

LigH

30th October 2012, 15:45

You got a SurCode DTS Encoder which only works when you put a copy-protection dongle into your parallel (old-style printer) port.

Are you sure you legally bought your copy?

tebasuna51

30th October 2012, 16:03

BTW, I can't understand this recode.

If you want DTS extract the 'core' 1509 Kb/s.
If you want less bitrate, recode to AC3 640 Kb/s from DTS-MA, much better than DTS 768 Kb/s.

Snowknight26

31st October 2012, 02:54

Any chance of ditching GDS mux in favor of ffmpeg?

Gew

2nd November 2012, 20:07

Hi guys!

Say (for instance) I have a an old DVD film that has mono audio, yet they have duped the audio to 2 channel Dolby (AC3) for it's DVD release.

And oh, I know, sometimes they have used artificial tools to "post-produce" true stereo so that i.e. sounds coming from the right goes on right channel, etc.

Anyways, say I just want my computer backup to be of mono audio. I used to use LAME for audio, and then I just pulled the "-m m" syntax to downmix stereo to mono. This worked a charm.

Now I've started using AVC+AAC. Weird though, I use the eac3to utility, and can't seem to find a way to downmix stereo. If AC3 has e.g. 5.1 sound I can use "eac3to in.ac3 out.m4a -down2" to easily downmix the output to stereo.

Hence, I was really hoping for a "-down1" option, yet this won't work. Any easy approach that I have missed out on, that makes this downmix possible with one simple eac3to line. Or will I simply need to combine several tools?

Cheers!

tebasuna51

2nd November 2012, 21:24

There are a eac3to parameter, -mono, to extract the Center channel from a 5.1 or the Left channel from 2.0, but don't exist a downmix 2.0 -> mono.

You can use BeHappy to decode AC3, downmix to mono, maybe normalize, and convert to AAC.

Snowknight26

3rd November 2012, 19:57

madshi,

This TrueHD sample (http://stfcc.org/misc/eac3to_corrupt_mlp.thd) does not decode properly with eac3to but ffmpeg can decode it fine. Might be time to release a new version? :p

eac3to:
eac3to.exe eac3to_corrupt_mlp.thd eac3to_corrupt_mlp.flac
TrueHD, 7.1 channels, 48kHz
Decoding with libav/ffmpeg...
Remapping channels...
Encoding FLAC with libFlac...
[mlp @ 008f5d90] Restart header sync incorrect (got 0x0598)
The libav decoder reported error -1 while decoding.
Creating file "eac3to_corrupt_mlp.flac"...
Aborted at file position 262144.

ffmpeg:
ffmpeg.exe -i eac3to_corrupt_mlp.thd eac3to_corrupt_mlp.flac
ffmpeg version N-46146-g11d695d Copyright (c) 2000-2012 the FFmpeg developers
built on Oct 29 2012 18:10:27 with gcc 4.7.2 (GCC)
configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger
--enable-libspeex --enable-libtheora --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs
--enable-libxvid --enable-zlib
libavutil 52. 1.100 / 52. 1.100
libavcodec 54. 69.100 / 54. 69.100
libavformat 54. 35.100 / 54. 35.100
libavdevice 54. 3.100 / 54. 3.100
libavfilter 3. 20.109 / 3. 20.109
libswscale 2. 1.101 / 2. 1.101
libswresample 0. 16.100 / 0. 16.100
libpostproc 52. 1.100 / 52. 1.100
[truehd @ 0000000003822f80] max_analyze_duration 5000000 reached at 5000000
[truehd @ 0000000003822f80] Estimating duration from bitrate, this may be inaccurate
Input #0, truehd, from 'eac3to_corrupt_mlp.thd':
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0: Audio: truehd, 48000 Hz, 7.1, s32
Output #0, flac, to 'eac3to_corrupt_mlp.flac':
Metadata:
encoder : Lavf54.35.100
Stream #0:0: Audio: flac, 48000 Hz, 7.1, s16, 128 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (truehd -> flac)
Press [q] to stop, [?] for help
size= 8095kB time=00:00:44.64 bitrate=1485.5kbits/s
video:0kB audio:8087kB subtitle:0 global headers:0kB muxing overhead 0.099678%

bugmen0t

4th November 2012, 23:01

Hi

I found 3 blurays that i cant demux with eac3to 3.24

Brave 3D, Brave 2D and Finding Nemo 3D.

All those are Disney blurays, i allways get this error :

a04 libav Restart header sync incorrect (got 0x0598)
a04 The libav decoder reported error -1 while decoding.

I was able to demux those with tsmuxer.

Could you please have a look at it ?

Thanks

LOG :

F:\BDREMUXES\Brave (2012) 3D>eac3to.exe E:\BDMV\PLAYLIST\00802.mpls -demux
M2TS, 2 video tracks, 6 audio tracks, 4 subtitle tracks, 1:34:12, 108.889p
1: Chapters, 37 chapters
2: h264/AVC (left eye), 1080p24 /1.001 (16:9)
3: h264/AVC (right eye), 1080p24 /1.001 (16:9)
4: TrueHD/AC3, English, 7.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
5: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
6: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
7: AC3, French, 5.1 channels, 512kbps, 48kHz
8: AC3, French, 5.1 channels, 640kbps, 48kHz
9: AC3, Spanish, 5.1 channels, 640kbps, 48kHz
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), French
13: Subtitle (PGS), Spanish
Creating file "00914 - Chapters.txt"...
a04 AC3 encoding doesn't support back channels. Will mix them into the surround.

v02 Extracting video track number 2...
a04 Extracting audio track number 4...
a05 Extracting audio track number 5...
a09 Extracting audio track number 9...
s11 Extracting subtitle track number 11...
a06 Extracting audio track number 6...
v03 Extracting video track number 3...
s10 Extracting subtitle track number 10...
s13 Extracting subtitle track number 13...
s12 Extracting subtitle track number 12...
a04 Extracting audio track number 4...
a08 Extracting audio track number 8...
a07 Extracting audio track number 7...
v03 Creating file "00914 - 3 - h264 (right eye), 1080p24.h264"...
a04 Extracting TrueHD stream...
a04 Extracting TrueHD stream...
a04 Decoding with libav/ffmpeg...
a04 Remapping channels...
a04 Mixing surround channels...
a04 Encoding AC3 <640kbps> with libAften...
a04 libav Restart header sync incorrect (got 0x0598)
a04 The libav decoder reported error -1 while decoding.

jj666

4th November 2012, 23:52

Those blu-ray's you mention aren't even released yet :-P

Have you tried latest FFMPEG standalone, like the chap posting above you did?

Cheers,

-jj-

Overdrive80

5th November 2012, 04:08

Joniii

5th November 2012, 06:34

When I encode 2.0 AC3 file to DTS, it output 5.0 DTS, anyone know why? I use Surcode 1.0.23.

Sparktank

5th November 2012, 08:28

When I encode 2.0 AC3 file to DTS, it output 5.0 DTS, anyone know why? I use Surcode 1.0.23.

Why not just manage it yourself in SurCode than using eac3to?

You can set it up to create a 2.0 in SurCode.

rapscallion

8th November 2012, 21:36

Atak_Snajpera

8th November 2012, 22:34

libaften is an ac3 encoder not truehd decoder.

tebasuna51

8th November 2012, 22:42

I notice in the eac3to dir there is only libaften.dll, dtd 2008.
No ffmpeg files at all.

avcodec.dll and avutil-50.dll are the special madshi dll's for libavcodec support. Only madshi can update that.

rapscallion

8th November 2012, 23:18

libaften is an ac3 encoder not truehd decoder.
Ok, then I'm a bit confused by this in Madshi's first post in this thread:
The libav/ffmpeg TrueHD decoder works perfectly fine including full 7.1 decoding.

What libav, ffmpeg or other file(s), do I need to decode TruHD 7.1 to wavs?

BTW, I use the gui eac3to and More.

@tebasuna51, avcodec.dll and avutil-50.dll are in my dir and are dated 2010. Are they relevant to TruHD 7.1 decoding?

tebasuna51

9th November 2012, 12:02

... avcodec.dll and avutil-50.dll are in my dir and are dated 2010. Are they relevant to TruHD 7.1 decoding?

Yes, inside there are the TruHD libav/ffmpeg decoder.

rapscallion

9th November 2012, 16:23

Thank you...it can be a bit confusing because of the names.

Any idea why they're causing this error (from previous post) on a retail THD 7.1 track? :

Writing WAVs...
[libav] Substream 0 parity check failed <WARNING>
[libav] Substream 0 checksum failed <WARNING>
[libav] Substream 0 length mismatch. <WARNING>
The libav decoder reported error -1 while decoding. <ERROR>
Aborted at file position 262144. <ERROR>

Snowknight26

9th November 2012, 16:56

No one knows if the discs are bad or if they are using a new format that the old versions of ffmpeg that eac3to uses don't support, but a solution was given so why not go with that?

rapscallion

9th November 2012, 17:02

No one knows if the discs are bad or if they are using a new format that the old versions of ffmpeg that eac3to uses don't support, but a solution was given so why not go with that?

If you're talking about ffmpeg, I wouldn't have a clue on how to apply that with the Eac3to and More Gui.

I'll try to demux with Tsmuxer

Snowknight26

9th November 2012, 17:33

Extract the TrueHD track as you would normally then use ffmpeg to convert it to whatever (intermediate) format you need. I showed how to do the latter part in my first post.

rapscallion

9th November 2012, 17:40

Extract the TrueHD track as you would normally then use ffmpeg to convert it to whatever (intermediate) format you need. I showed how to do the latter part in my first post.
Yes, I saw that, however, that's way beyond my understanding/ability.

Snowknight26

9th November 2012, 17:53

http://ffmpeg.zeranoe.com/builds/

Grab the latest static release. Extract somewhere.

Go to the bin folder and start a command prompt window there.

ffmpeg -i file.thd file.flac

Voila.

rapscallion

9th November 2012, 18:03

Thanks for you help and response.

Yes, just did that from here : http://www.videohelp.com/tools/ffmpeg

Which linked to the site you posted.

Then downloaded the WinFF front end. However, no output option for wavs.
Doing the cmd line your way has output of flac. Will I be able to convert that to 8 wavs?

I'll give it a try.

Edit: must be something funky with this track as I received this :

"Invalid data found when processing input"

sneaker_ger

9th November 2012, 18:04

Try:
eac3to input.flac output.wavs

Snowknight26

9th November 2012, 18:58

"Invalid data found when processing input"

Rename the input to .thd.

rapscallion

9th November 2012, 19:23

Rename the input to .thd.
Done, but still getting errors. This track is definitely skunked.

[truehd @ 024c0900] Lossless check failed - expected 4f, calculated 53.
[truehd @ 024c0900] mlpparse: Parity check failed.
[truehd @ 024c0900] Lossless check failed - expected 33, calculated 83.
[truehd @ 024c0900] mlpparse: Parity check failed.
[truehd @ 024c0900] Lossless check failed - expected 9d, calculated 72.
[truehd @ 024c0900] mlpparse: Parity check failed.
[truehd @ 024c0900] Lossless check failed - expected b2, calculated 41.
[truehd @ 024c0900] Invalid nonrestart_substr.
Error while decoding stream #0:0: Invalid data found when processing input
[truehd @ 024c0900] Stream parameters not seen; skipping frame.
Last message repeated 32 times
[truehd @ 024c0900] mlpparse: Parity check failed.
[truehd @ 024c0900] Lossless check failed - expected da, calculated 77.
[truehd @ 024c0900] mlpparse: Parity check failed.
[truehd @ 024c0900] Lossless check failed - expected f0, calculated 3f.
[truehd @ 024c0900] mlpparse: Parity check failed.

Snowknight26

9th November 2012, 20:47

Most likely because the source uses seamless branching. Only 6 bytes of probably over 3GB is corrupt. You won't even hear it.

rapscallion

9th November 2012, 21:56

Most likely because the source uses seamless branching. Only 6 bytes of probably over 3GB is corrupt. You won't even hear it.
Thanks, that is probably the issue, because the source does have seamless branching.
Though I've never had an issue with seamless branching audio tracks before.

When I demuxed the track, Tsmuxer stated "overlaps" at 2 different time points and "fixed" it.

Problem is that I need to convert to wavs, not use the THD track in it's current format.

Furiousflea

11th November 2012, 02:32

No one knows if the discs are bad or if they are using a new format that the old versions of ffmpeg that eac3to uses don't support, but a solution was given so why not go with that?

Solution doesn't work for "Brave".

Countless lossless check failed/parity check/numerous other errors when using FFMPEG command line.

There's about 5 gaps in audio ranging from 2ms to 30ms so...

Kinda stuck until something gets updated.

Furiousflea

11th November 2012, 02:32

Thanks, that is probably the issue, because the source does have seamless branching.
Though I've never had an issue with seamless branching audio tracks before.

When I demuxed the track, Tsmuxer stated "overlaps" at 2 different time points and "fixed" it.

Problem is that I need to convert to wavs, not use the THD track in it's current format.

That's not the issue, there's something different about these tracks.

Snowknight26

11th November 2012, 05:10

Solution doesn't work for "Brave".

Countless lossless check failed/parity check/numerous other errors when using FFMPEG command line.

It does work. There are 6 errors total, all which are only one byte.

r0lZ

11th November 2012, 12:34

@madshi
As you can see here (http://forum.doom9.org/showthread.php?p=1600104#post1600104), I'm trying to implement TrueHD support in my BD3D2MK3D GUI. Currently, the libav decoder crashes with this message: "Too many audio samples in frame" and when that problem occurs, eac3to hangs.
Is it a solution for that problem? Can you add a check in the next eac3to version to return an error instead of hanging?
(Sorry if that bug has already been reported. I'm too lazy to read the whole thread!)
Thanks!

Furiousflea

11th November 2012, 13:56

It does work. There are 6 errors total, all which are only one byte.

Maybe with your track, which film is it exactly? I'm talking about "Brave" the 2D version I'm trying to remux the main movie but with FLAC audio instead of the TrueHD 7.1 track.

eac3to crashes when using my nero audio encoder (I realize it can't do 7.1 but thought I'd try just to see).

libav give the message you talk about so no-go there.

But when using FFMPEG after demuxing the .THD track, I get about 4 different types error continuously throughout the whole stream...

1. Too many audio sample in frame
2. audio sync mismatch

Well basically lots of red error messages for first 14 mins or so.

The real problem though is that even if this output is close to the original, there's still the audio gaps that can't be removed.

Joniii

11th November 2012, 14:43

I demuxed Pixars new Finding Nemo BD and got some warnings.

First this when demuxing, I wonder if the audio is 14 frames off or did eac3to fix the video overlap?

eac3to v3.24
command line: eac3to F:\FINDING_NEMO 1) 2: d:\nemo.h264 3: d:\nemo.thd
M2TS, 1 video track, 8 audio tracks, 13 subtitle tracks, 1:40:54, 24p /1.001
1: Chapters, 32 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 7.1 channels, 48kHz
(embedded: AC3 EX, 5.1 channels, 640kbps, 48kHz)
...
[a03] Extracting audio track number 3...
[a03] Extracting TrueHD stream...
[v02] Extracting video track number 2...
[v02] Creating file "d:\nemo.h264"...
[a03] Creating file "d:\nemo.thd"...
[v02] Video overlaps for 14 frames at playtime 0:00:49. <WARNING>
Video track 2 contains 145150 frames.
eac3to processing took 5 minutes, 31 seconds.
Done.

Then when trying to extract ac3 core from the TrueHD I just demuxed, I get this.

eac3to d:\nemo.thd h:\nemo.ac3 -core
TrueHD, 7.1 channels, 48kHz
AC3 encoding doesn't support back channels. Will mix them into the surround.
Decoding with libav/ffmpeg...
Remapping channels...
Mixing surround channels...
Encoding AC3 <640kbps> with libAften...
[mlp @ 009e2d10] Substream 0 parity check failed
[mlp @ 009e2d10] Substream 0 checksum failed
[mlp @ 009e2d10] Substream 0 length mismatch.
The libav decoder reported error -1 while decoding.

Extracting ac3 directly from BD:

eac3to F:\FINDING_NEMO 1) 3: h:\nemo.ac3 -core
M2TS, 1 video track, 8 audio tracks, 13 subtitle tracks, 1:40:54, 24p /1.001
1: Chapters, 32 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 7.1 channels, 48kHz
(embedded: AC3 EX, 5.1 channels, 640kbps, 48kHz)
...
a03 Extracting audio track number 3...
a03 Extracting AC3 stream...
a03 Creating file "h:\nemo.ac3"...
a03 Audio overlaps for 7ms at playtime 0:04:46.
a03 Audio overlaps for 28ms at playtime 0:05:38.
a03 Audio overlaps for 20ms at playtime 1:33:24.
a03 Starting 2nd pass...
a03 Realizing (E-)AC3 gaps...
a03 Creating file "h:\nemo.ac3"...
Video track 2 contains 145150 frames.
eac3to processing took 4 minutes, 52 seconds.
Done.

Joniii

11th November 2012, 14:51

Also with the new Total Recall I get this:

H:\eac3to>eac3to d:\total.thd e:\total.ac3 -core
TrueHD, 5.1 channels, 48kHz
Decoding with libav/ffmpeg...
Encoding AC3 <640kbps> with libAften...
Creating file "e:\total.ac3"...
[mlp @ 00712d70] Lossless check failed - expected 0, calculated 93
-[mlp @ 00712d70] Lossless check failed - expected 0, calculated ac
-[mlp @ 00712d70] End of stream indicated
[mlp @ 00712d70] Lossless check failed - expected 0, calculated b3
[mlp @ 00712d70] Lossless check failed - expected 0, calculated 1c
-[mlp @ 00712d70] End of stream indicated
[mlp @ 00712d70] Lossless check failed - expected 0, calculated 26
-[mlp @ 00712d70] Lossless check failed - expected 0, calculated 91
.......

Snowknight26

11th November 2012, 20:47

The real problem though is that even if this output is close to the original, there's still the audio gaps that can't be removed.

So use eac3to to demux the TrueHD track then use ffmpeg to convert it to whatever format you want.

Joniii

13th November 2012, 07:44

Joniii

13th November 2012, 07:50

Anyone know why EAC3TO adds +5ms delay when doing FLAC -> AC3?

5.1 AC3 to FLAC with EAC3TO, I then edit the file with Audacity and save as FLAC. Then again use EAC3TO to encode it back to AC3. After it has encoded it, it is off by +5ms when comparing to the original AC3 or FLAC saved by Audacity.

sneaker_ger

13th November 2012, 07:55

Encoder delay caused by MDCT (https://en.wikipedia.org/wiki/Modified_discrete_cosine_transform), not the fault of eac3to. Many lossy formats suffer from it.

Joniii

13th November 2012, 08:09

Encoder delay caused by MDCT (https://en.wikipedia.org/wiki/Modified_discrete_cosine_transform), not the fault of eac3to. Many lossy formats suffer from it.

Thanks :)

tebasuna51

13th November 2012, 09:44

Anyone know why EAC3TO adds +5ms delay when doing FLAC -> AC3?

You can avoid this delay with external Aften encoder using the -pad parameter:

eac3to input stdout.wav | Aften -b 640 -pad 0 -readtoeof 1 - undelayed.ac3

Without problems if sound begin with silence, else the first 5.33 ms can be distorted.

Joniii

13th November 2012, 16:29

You can avoid this delay with external Aften encoder using the -pad parameter:

eac3to input stdout.wav | Aften -b 640 -pad 0 -readtoeof 1 - %_.ac3

Without problems if sound begin with silence, else the first 5.33 ms can be distorted.

Thanks :)

sneaker_ger

13th November 2012, 18:13

Yes, for movies this is usually fine. No one cares about a few milliseconds of a 2 hour movie that are probably just silence anyways.

Joniii

14th November 2012, 07:44

Yes, for movies this is usually fine. No one cares about a few milliseconds of a 2 hour movie that are probably just silence anyways.

Yeah, noone will notice it and it doesn't bother me either but it's a plus if you can get rid of that +5.33ms in the progress. Also I wanted to understand why and where it is added.

G_M_C

14th November 2012, 10:17

Maybe the problems with truehd are a manifestation of the dreaded cinavia watermarking?

Furiousflea

14th November 2012, 17:02

So use eac3to to demux the TrueHD track then use ffmpeg to convert it to whatever format you want.

That doesn't remove gaps since no decoding\encoding is taking place.

This is still a problem. Brave is currently unwatchable with demux with eac3to -> FFMPEG encoding to FLAC because of the gaps. Audio is a good 200-300ms out.

Maybe eac3to can resolve gaps by simply demuxing doing a 2nd pass with some formats - but not THD - which is what we're dealing with here.

Snowknight26

14th November 2012, 18:13

That doesn't remove gaps since no decoding\encoding is taking place.

Sure it can, just use the .gaps file in conjunction with the FLAC file that ffmpeg creates.

henryperu77

14th November 2012, 19:59

Several new BD's with TrueHD have problems, for example it's impossible to directly encode TrueHD to DTS or AC3 with EAC3TO, it gives some red warnings and hangs. Also demuxing THD works but demuxing THD with AC3 fails.

[mlp @ 00b42d10] Substream 0 parity check failed
[mlp @ 00b42d10] Substream 0 checksum failed
[mlp @ 00b42d10] Substream 0 length mismatch.
a03 The libav decoder reported error -1 while decoding.

Hi i have a .thd audio file that i encoded with Dolby Media Encoder, i tried to make a thd+ac3 in order to mux on a .ts file but i get that same error

TrueHD, 5.1 channels, 48kHz
Decoding with libav/ffmpeg...
Encoding AC3 <640kbps> with libAften...
libav Substream 0 parity check failed
libav Substream 0 checksum failed
libav Substream 0 length mismatch.
The libav decoder reported error -1 while decoding.

Any help?

madshi

14th November 2012, 20:25

@tebasuna51, would you be willing to make a list of the most important things to fix in / add to eac3to? Not too big things, please, just bugfixes and minor changes/improvements. Thanks!

And for those of you who have problems with specific TrueHD tracks, please upload small samples. E.g. 2MB per sample should be enough. But before you upload, please make sure that the problem also occurs with the sample and not just with the full file, otherwise the less won't help. Thx.

henryperu77

14th November 2012, 20:39

How i can make a 2mb sample of a truehd file?? what program could i use?

Atak_Snajpera

14th November 2012, 22:49

dgsplit?

henryperu77

15th November 2012, 00:15

Ok i used dgsplit and this is the sample

http://www.mediafire.com/download.php?oopeorxdvqnk77v

thanks in advance

Snowknight26

15th November 2012, 00:40

@tebasuna51, would you be willing to make a list of the most important things to fix in / add to eac3to? Not too big things, please, just bugfixes and minor changes/improvements. Thanks!

You just made several people very happy. :)

And for those of you who have problems with specific TrueHD tracks, please upload small samples. E.g. 2MB per sample should be enough. But before you upload, please make sure that the problem also occurs with the sample and not just with the full file, otherwise the less won't help. Thx.

http://stfcc.org/misc/eac3to_corrupt_mlp.thd

Joniii

15th November 2012, 07:48

Here's a small THD sample from Finding Nemo. It demuxed fine from BD but the demuxed file doesn't seem to have AC3 core, also when trying to encode to AC3 it fails. I also have this exact same problem with Total Recall.

H:\eac3to>eac3to e:\nemo.thd d:\nemo.ac3
TrueHD, 7.1 channels, 48kHz
AC3 encoding doesn't support back channels. Will mix them into the surround.
Decoding with libav/ffmpeg...
Remapping channels...
Mixing surround channels...
Encoding AC3 <640kbps> with libAften...
[mlp @ 00b42d10] Substream 0 parity check failed
[mlp @ 00b42d10] Substream 0 checksum failed
[mlp @ 00b42d10] Substream 0 length mismatch.
The libav decoder reported error -1 while decoding.

Sample (http://sdrv.ms/TKwiu4)

jj666

15th November 2012, 12:28

Hi Madshi,

There are two outstanding fixes mentioned here (http://forum.doom9.org/showthread.php?p=1428982#post1428982) (and the post above) regarding the stripping of DTS-HD Master Audio Suite headers needing adjustment, and issues with audio delays when two passes are needed.

Cheers,

-jj-

tebasuna51

15th November 2012, 12:53

@tebasuna51, would you be willing to make a list of the most important things to fix in / add to eac3to? Not too big things, please, just bugfixes and minor changes/improvements. Thanks!

Maybe the best option is open the source code like was sugessted many times in this thread.

BTW, I open a new thread eac3to v3.24 Bugs & Improvements (http://forum.doom9.org/showthread.php?p=1600693#post1600693)

madshi

15th November 2012, 15:49

Thanks, I'll have a look at that new thread. It might take a week or two or three before I get to this, though.

sporic

16th November 2012, 21:30

rapscallion

16th November 2012, 21:48

What's new in these TrueHD tracks that's causing these problems?
Someone previously suggested that it has to do with seamless branching. Although I've had no problem in the past with that.

tebasuna51

16th November 2012, 22:34

Someone previously suggested that it has to do with seamless branching.

Nope, I extracted the .thd's directly from .m2ts's (with eac3to and tsMuxeR) of BD Brave, and when try decode with eac3to all crash with this error.

ffmpeg decode the thd's without problem.

Maybe -libav need a update.

rapscallion

16th November 2012, 22:41

Nope, I extracted the .thd's directly from .m2ts's (with eac3to and tsMuxeR) of BD Brave, and when try decode with eac3to all crash with this error.

ffmpeg decode the thd's without problem.

Maybe -libav need a update.
Funny, ffmpeg didn't work for me :(

[truehd @ 023cc440] max_analyze_duration 5000000 reached at 5000000
[truehd @ 023cc440] Estimating duration from bitrate, this may be inaccurate
Input #0, truehd, from 'track_4352.thd':
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0: Audio: truehd, 48000 Hz, 7.1, s32
Output #0, flac, to 'track_4352.flac':
Metadata:
encoder : Lavf54.36.100
Stream #0:0: Audio: flac, 48000 Hz, 7.1, s32, 128 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (truehd -> flac)
Press [q] to stop, [?] for help
Lossless check failed - expected 00, calculated f4.s/s
Lossless check failed - expected 00, calculated 65.s/s
Lossless check failed - expected 00, calculated dc.s/s
End of stream indicated.40:25.24 bitrate=2975.9kbits/s
[truehd @ 024c0900] restart header sync incorrect (got 0x0685)
Error while decoding stream #0:0: Invalid data found when processing input
Last message repeated 3 times
[truehd @ 024c0900] restart header sync incorrect (got 0x0d60)
Error while decoding stream #0:0: Invalid data found when processing input
Last message repeated 3 times
[truehd @ 024c0900] too many audio samples in frame
Error while decoding stream #0:0: Invalid data found when processing input
[truehd @ 024c0900] too many audio samples in frame
Error while decoding stream #0:0: Invalid data found when processing input
[truehd @ 024c0900] too many audio samples in frame
Error while decoding stream #0:0: Invalid data found when processing input
Last message repeated 1 times
[truehd @ 024c0900] too many audio samples in frame
Error while decoding stream #0:0: Invalid data found when processing input
Last message repeated 1 times
[truehd @ 024c0900] restart header sync incorrect (got 0x1556)
Error while decoding stream #0:0: Invalid data found when processing input
[truehd @ 024c0900] Lossless check failed - expected 07, calculated e0.
size= 2195470kB time=01:40:53.96 bitrate=2970.8kbits/s
video:0kB audio:2195462kB subtitle:0 global headers:0kB muxing overhead 0.000367
%

C:\Dvd_rw\ffmpeg\bin>

Plus, I could extract with eac3to, but not create "wavs".

tebasuna51

17th November 2012, 03:19

With the thd extracted from 00954.m2ts (Brave BD):

ffmpeg version N-44601-gcb3591e Copyright (c) 2000-2012 the FFmpeg developers
built on Sep 19 2012 16:28:01 with gcc 4.7.1 (GCC)
...
[truehd @ 023ecf60] max_analyze_duration 5000000 reached at 5000000
[truehd @ 023ecf60] Estimating duration from bitrate, this may be inaccurate
Input #0, truehd, from 'D:\Temp\t003.thd':
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0: Audio: truehd, 48000 Hz, 7.1, s32
Output #0, wav, to 'audio.wav':
Metadata:
encoder : Lavf54.27.101
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 7.1, s16, 6144 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (truehd -> pcm_s16le)
Press [q] to stop, [?] for help
End of stream indicated.02:56.87 bitrate=6143.2kbits/s
size= 136856kB time=00:03:02.47 bitrate=6144.0kbits/s
video:0kB audio:136856kB subtitle:0 global headers:0kB muxing overhead 0.000049%

ron spencer

17th November 2012, 03:58

I have same issue...eac3to and tsmuxer both crash...is there a way to demux from the m2ts in the playlist...I like to convert to ac-3 448...why is this truehd stream so weird...it has worked b4

soneca

17th November 2012, 05:03

With the thd extracted from 00954.m2ts (Brave BD):

tebasuna51,

This same blu-ray(Brave) shows me the main audio track as Dolby Digital Plus(E-AC-3) with 7.1 channels.

Joniii

17th November 2012, 08:26

Thanks, I'll have a look at that new thread. It might take a week or two or three before I get to this, though.

Put madVR on hold. Ripping Blu-rays should be priority.

lmaolmao

17th November 2012, 10:24

eac3to does not seem to be able to extract the .thd+ac3 on the brave/nemo discs it tries to start making an ac3 core using aften.

set it to demux just .thd and it spits out a file, set it to .ac3 and it spits out a file (neither one encoding), but when you want a .thd+ac3 file it fails.

M2TS, 1 video track, 8 audio tracks, 13 subtitle tracks, 1:40:54, 24p /1.001
1: Chapters, 32 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 7.1 channels, 48kHz
(embedded: AC3 EX, 5.1 channels, 640kbps, 48kHz)
4: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
5: AC3 EX, Czech, 5.1 channels, 640kbps, 48kHz
6: AC3 EX, Slovak, 5.1 channels, 640kbps, 48kHz
7: AC3 EX, Arabic, 5.1 channels, 640kbps, 48kHz
8: AC3 EX, Russian, 5.1 channels, 640kbps, 48kHz
9: AC3 EX, Ukrainian, 5.1 channels, 640kbps, 48kHz
10: AC3 EX, Croatian, 5.1 channels, 640kbps, 48kHz
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), Czech
14: Subtitle (PGS), Slovak
15: Subtitle (PGS), Arabic
16: Subtitle (PGS), Russian
17: Subtitle (PGS), Ukrainian
18: Subtitle (PGS), Croatian
19: Subtitle (PGS), Slovak
20: Subtitle (PGS), Arabic
21: Subtitle (PGS), Ukrainian
22: Subtitle (PGS), Croatian
23: Subtitle (PGS), Czech
a03 AC3 encoding doesn't support back channels. Will mix them into the surround.

a03 Extracting audio track number 3...
a03 Extracting TrueHD stream...
a03 Extracting audio track number 3...
a03 Extracting TrueHD stream...
a03 Decoding with libav/ffmpeg...
a03 Remapping channels...
a03 Mixing surround channels...
a03 Encoding AC3 <640kbps> with libAften...
a03 libav Substream 0 parity check failed
a03 libav Substream 0 checksum failed
a03 libav Substream 0 length mismatch.
a03 The libav decoder reported error -1 while decoding.

I hadn't noticed this mentioned in the thread. Is this why some are having trouble remuxing?

hoju3508

17th November 2012, 23:52

@lmaolmao, this "a03 The libav decoder reported error -1 while decoding." is a known problem for eac3to.

BigPines

18th November 2012, 03:45

@lmaolmao, this "a03 The libav decoder reported error -1 while decoding." is a known problem for eac3to.

I've been using eac3to for years and never got this error until Brave. Unfortunately, this is a pretty bad bug. It makes it impossible to demux Brave or convert the audio using eac3to.

Bummer...

Mike

ron spencer

18th November 2012, 04:28

you can demux the thd only...that is easy.

then use ffmpeg to convert to ac3 448...I just did that ...seems perfect. you just need and extra step.

hopefully eac3to can be fixed for this.

lmaolmao

18th November 2012, 09:25

@lmaolmao, this "a03 The libav decoder reported error -1 while decoding." is a known problem for eac3to.

maybe i am not making myself clear. This is happening before the -1 error.

with all other truehd discs you can extract a .thd file, the .ac3 core or a .tdh+ac3 file. And it just demuxes it (no encoding).

With BRave and Nemo, you can extract a .thd or the .ac3 core fine. But when extracting the .thd+ac3, eac3to initialises aften to make the ac3 track instead of just demuxing. Which is odd.....

Boulder

18th November 2012, 10:08

But when extracting the .thd+ac3, eac3to initialises aften to make the ac3 track instead of just demuxing. Which is odd.....Post the commandline so we can see if there's anything wrong with that.

lmaolmao

18th November 2012, 10:27

Post the commandline so we can see if there's anything wrong with that.

Heres the full command line/output.

P:\22>"D:\VID TOOLS\eac3to\eac3to.exe" "Y:\BDMV\PLAYLIST\00800.mpls" 1) 3: "P:\22\Audio.thd+ac3"
M2TS, 1 video track, 8 audio tracks, 13 subtitle tracks, 1:40:54, 24p /1.001
1: Chapters, 32 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 7.1 channels, 48kHz
(embedded: AC3 EX, 5.1 channels, 640kbps, 48kHz)
4: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
5: AC3 EX, Czech, 5.1 channels, 640kbps, 48kHz
6: AC3 EX, Slovak, 5.1 channels, 640kbps, 48kHz
7: AC3 EX, Arabic, 5.1 channels, 640kbps, 48kHz
8: AC3 EX, Russian, 5.1 channels, 640kbps, 48kHz
9: AC3 EX, Ukrainian, 5.1 channels, 640kbps, 48kHz
10: AC3 EX, Croatian, 5.1 channels, 640kbps, 48kHz
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), Czech
14: Subtitle (PGS), Slovak
15: Subtitle (PGS), Arabic
16: Subtitle (PGS), Russian
17: Subtitle (PGS), Ukrainian
18: Subtitle (PGS), Croatian
19: Subtitle (PGS), Slovak
20: Subtitle (PGS), Arabic
21: Subtitle (PGS), Ukrainian
22: Subtitle (PGS), Croatian
23: Subtitle (PGS), Czech
a03 AC3 encoding doesn't support back channels. Will mix them into the surround.

a03 Extracting audio track number 3...
a03 Extracting TrueHD stream...
a03 Extracting audio track number 3...
a03 Extracting TrueHD stream...
a03 Decoding with libav/ffmpeg...
a03 Remapping channels...
a03 Mixing surround channels...
a03 Encoding AC3 <640kbps> with libAften...
a03 libav Substream 0 parity check failed
a03 libav Substream 0 checksum failed
a03 libav Substream 0 length mismatch.
a03 The libav decoder reported error -1 while decoding.

tebasuna51

18th November 2012, 10:30

I can confirm the issue, I have the same log than lmaolmao:
a03 AC3 encoding doesn't support back channels. Will mix them into the surround.
a03 Extracting audio track number 3...
a03 Extracting audio track number 3...
a03 Extracting TrueHD stream...
a03 Extracting TrueHD stream...
a03 Decoding with libav/ffmpeg...
a03 Remapping channels...
a03 Mixing surround channels...
a03 Encoding AC3 <640kbps> with libAften...
a03 libav Substream 1 parity check failed
a03 libav Substream 1 checksum failed
a03 libav Substream 1 length mismatch.
a03 The libav decoder reported error -1 while decoding.
with this command line:
eac3to.exe "D:\Temp\BRAVE\" 1) 3: "D:\Temp\BRAVE\0_eng3.thd+ac3"

madshi

18th November 2012, 10:43

A sample of the Brave's movie m2ts would be helpful.

lmaolmao

18th November 2012, 10:43

I'll grab a bit of the first m2ts. And make sure it does show this error.

Edit: Well, if the first m2ts is done directly then it extracts fine......do you still want the sample?

It only does it when the playlist is selected...

r0lZ

18th November 2012, 11:20

sporic

18th November 2012, 14:12

i have also noticed a very strange thing with the brave 3d blu-ray. (i don't have the 2d bd to check if it has the same problem.)

z:\bdmv\playlist>eac3to 00800.mpls 1)
m2ts, 2 video tracks, 6 audio tracks, 4 subtitle tracks, 1:33:37, 108.607p
1: Chapters, 37 chapters
2: H264/avc (left eye), 1080p24 /1.001 (16:9)
3: H264/avc (right eye), 1080p24 /1.001 (16:9)
4: Truehd/ac3, english, 7.1 channels, 48khz
(embedded: Ac3, 5.1 channels, 640kbps, 48khz)
5: Ac3 surround, english, 2.0 channels, 320kbps, 48khz
6: Ac3 surround, english, 2.0 channels, 320kbps, 48khz
7: Ac3, french, 5.1 channels, 512kbps, 48khz
8: Ac3, french, 5.1 channels, 640kbps, 48khz
9: Ac3, spanish, 5.1 channels, 640kbps, 48khz
10: Subtitle (pgs), english
11: Subtitle (pgs), english
12: Subtitle (pgs), french
13: Subtitle (pgs), spanish

as you can see, stream 7 is strange. It has an unusual bit rate, and, as far as i know, that stream does not exist in the bd. There is only one french audio track (listed by eac3to as stream #8). Tsmuxer does not show that strange french stream.

When eac3to tries to demux that track, it issues the warning "a07 this track is not clean.", but it continues anyway. The resulting ac3 file plays two times faster than it should in all audio players, and the sound is strange.

cinavia!!! :d

r0lZ

18th November 2012, 14:31

Sorry. That might be a noob question, but I don't understand why cinavia should need an additional audio track. The real French audio is in track #8, and if it is protected by cinavia, it is possible to demux it anyway. So, what's the benefit in adding a fake French track?
Also, it seems strange that only the French audio has that additional track, but not the main English or the Spanish audio tracks.
Honestly, I know almost nothing about cinavia, but afaik the protection is simply a watermark, that should not prevent eac3to to demux the track. (Playing it on cinavia protected hardware is another story.)
If I have missed something, can you lead me to a thread with additional info?

BigPines

18th November 2012, 16:27

Honestly, I know almost nothing about cinavia, but afaik the protection is simply a watermark, that should not prevent eac3to to demux the track. (Playing it on cinavia protected hardware is another story.)

This is exactly right. I don't think what we are seeing is Cinavia. I think it is some new variant of TrueHD that we have not encountered before.

I will see if I can get a sample together for Madshi.

Mike

Chumbo

18th November 2012, 17:00

[EDIT] I moved request to the new bugs/feature requests thread.

BigPines

19th November 2012, 02:38

I am posting the bugs and sample files in the new thread: http://forum.doom9.org/showthread.php?t=166487&page=2

Mike

BigPines

19th November 2012, 04:43

Sparktank

19th November 2012, 05:53

For anyone interested, I found a bit of a work-around for the Brave/Finding Nemo True-HD seamless branching problem.

I used MakeMkv to transcode the TrueHD 7.1 into FLAC 7.1 and then used eac3to to transcode the FLAC into WAVs. I will probably use this method until there is an official fix for eac3to.

I hope this helps someone else.

Mike

That's the same solution posted earlier here for using ffmpeg. Just an extra step/program.
MakeMKV has an option to use ffmpeg for audio transcoding.

You forgot to mention that you need to go into options and enable "Advanced options" and tell MMKV where it can find the binary for ffmpeg.
FFmpeg binarys can be found at zeranoe's site.
His site is mentioned on the official ffmpeg website, if most don't know where to find him already.

BigPines

19th November 2012, 08:55

That's the same solution posted earlier here for using ffmpeg. Just an extra step/program.
MakeMKV has an option to use ffmpeg for audio transcoding.

You forgot to mention that you need to go into options and enable "Advanced options" and tell MMKV where it can find the binary for ffmpeg.
FFmpeg binarys can be found at zeranoe's site.
His site is mentioned on the official ffmpeg website, if most don't know where to find him already.

You may not have noticed that the solution posted earlier was reported not to work by a follow-up post.

True, MakeMkv uses ffmpeg for the audio transcoding. However, my understanding is MakeMkv comes with a patched version that will not down-convert everything to 16 bits. Admittedly, I don't really know much about the full abilities of ffmpeg. Can it handle seamless branching like MakeMkv can? At any rate, MakeMkv is a valuable program notwithstanding it's reliance on ffmpeg for audio transcoding. It is a worthwhile tool to have in one's arsenal and I choose to use it.

You don't need to tell MakeMkv where the ffmpeg binaries are anymore. That was an older version but you are correct in that advanced options must be enabled to use the FLAC profile.

Just trying to help.

Mike

Joniii

19th November 2012, 09:33

maybe i am not making myself clear. This is happening before the -1 error.

with all other truehd discs you can extract a .thd file, the .ac3 core or a .tdh+ac3 file. And it just demuxes it (no encoding).

With BRave and Nemo, you can extract a .thd or the .ac3 core fine. But when extracting the .thd+ac3, eac3to initialises aften to make the ac3 track instead of just demuxing. Which is odd.....

I can confirm I have the same thing with Nemo.

H:\eac3to>eac3to D:\FINDING_NEMO 1) 3: Y:\nemo.thd+ac3
M2TS, 1 video track, 1 audio track, 1 subtitle track, 1:40:54, 24p /1.001
1: Chapters, 32 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 7.1 channels, 48kHz
(embedded: AC3 EX, 5.1 channels, 640kbps, 48kHz)
4: Subtitle (PGS), English
a03 AC3 encoding doesn't support back channels. Will mix them into the surround.
a03 Extracting audio track number 3...
a03 Extracting TrueHD stream...
a03 Decoding with libav/ffmpeg...
a03 Remapping channels...
a03 Extracting audio track number 3...
a03 Mixing surround channels...
a03 Extracting TrueHD stream...
a03 Encoding AC3 <640kbps> with libAften...
[mlp @ 009c2d10] Substream 0 parity check failed
[mlp @ 009c2d10] Substream 0 checksum failed
[mlp @ 009c2d10] Substream 0 length mismatch.
a03 The libav decoder reported error -1 while decoding.

lostclusters

20th November 2012, 02:19

I did a -demux on the Brave 7.1 soundtrack and it seemed to work. But I get intermittent clicks, pops, and sound drop outs in the resulting .mkv. I'll bet some sort of copy protection.

Edit: Might be the source, checking.

Thunderbolt8

20th November 2012, 23:23

isnt there any way or hasnt anyone be able to come with some idea how to process THD files with gaps in the meantime? or maybe some idea to get round this problem with the final file still being an in sync THD file? there should be some way, somehow, shouldnt there?

BigPines

21st November 2012, 02:35

See my post above. It works fine and everything is in sync.

Mike

rapscallion

24th November 2012, 23:12

However, my understanding is MakeMkv comes with a patched version that will not down-convert everything to 16 bits.
You don't need to tell MakeMkv where the ffmpeg binaries are anymore. That was an older version but you are correct in that advanced options must be enabled to use the FLAC profile.

Just trying to help.

Mike
V 1.7.9, which is the latest, there is the option to put in the path to ffmpeg.

There is no "patched" version in the MakeMkv folder that i can see, unless it's mmffmpeg.exe.

Edit: tried the path to mmffmpeg and it worked. I was now able to create the Mkv, with flac audio, and then save the flac to 8 wavs, via eac3to. Finding Nemo Btw, which failed using the straight/unpatched standalone ffmpeg.

SquallMX

25th November 2012, 00:48

I have also noticed a very strange thing with the Brave 3D blu-ray. (I don't have the 2D BD to check if it has the same problem.)

Z:\BDMV\PLAYLIST>eac3to 00800.mpls 1)
M2TS, 2 video tracks, 6 audio tracks, 4 subtitle tracks, 1:33:37, 108.607p
1: Chapters, 37 chapters
2: h264/AVC (left eye), 1080p24 /1.001 (16:9)
3: h264/AVC (right eye), 1080p24 /1.001 (16:9)
4: TrueHD/AC3, English, 7.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
5: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
6: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
7: AC3, French, 5.1 channels, 512kbps, 48kHz
8: AC3, French, 5.1 channels, 640kbps, 48kHz
9: AC3, Spanish, 5.1 channels, 640kbps, 48kHz
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), French
13: Subtitle (PGS), Spanish

As you can see, stream 7 is strange. It has an unusual bit rate, and, as far as I know, that stream does not exist in the BD. There is only one French audio track (listed by eac3to as stream #8). tsMuxeR does NOT show that strange French stream.

When eac3to tries to demux that track, it issues the warning "a07 This track is not clean.", but it continues anyway. The resulting ac3 file plays two times faster than it should in all audio players, and the sound is strange.

There`re 2 French tracks on the Blu-ray, Dolby Digital Plus 7.1 at 896 (as reported by Total Media Theater) and a standard Dolby Digital 5.1 at 640 Kbps (as reported by Total Media Theater). Maybe eac3to is detecting only the core of the 7.1 DD Plus track?

r0lZ

25th November 2012, 00:54

Thanks for the info. I suppose you're right, as it detects it as 5.1 @512Kbps. Obviously, it's something that needs to be fixed.

phate89

26th November 2012, 03:40

Hi.. I found a bug and i don't know if it's eac3to's fault or (probably) a nero ac3 decoder bug.
I use a lot eac3to to edit audio tracks losslessly and sync with other sources. I also have nero 6 iinstalled and working and i use it to decode ac3 usually. but i recently discover that if i edit an ac3 track (2.0 or 5.1 doesn't matter) the editing is ok, but if i try after to convert the ac3 synced file to wav with nero decoder it looses all the silence i insert in various points in the track... How is that possible? If i set "-libav" all works well, indeed if i downmix the edited 5.1 track to 2.0 it keeps the sync and it works well because it uses by default libav (is this a bug or ment to be libav with downmix?).
It is eac3to or nero related?

aneurysm

27th November 2012, 14:28

Just wanted to thank some people here with their fixes / work around's for transcoding THD files with Finding Nemo.
I tried that solution and worked fine.
Extracted THD
ffmpeg THD to flac 7.1
eac3to flac to 5.1 wav files
surcode 5.1 wav files to DTS file
ran through eac3to again to remove zero padding.
Result is perfect.

tnx !! Will try this with Brave too once I get it.

One thing I did (maybe) different from others who can't get it to work, I first used txmuxer to create 1 m2ts file from the playlist.
then used eac3to to extract the thd from that file.

rapscallion

27th November 2012, 17:28

I did a very similiar process with Nemo, but am wondering about the resulting file sizes.

Original THD, w/core : 5.1 gb

Extracted thd (no core) : 4.6 gb
ffmpeg THD to flac (w/expected 4 errors) : 2.2 gb
eac3to flac to 8 wavs : 851 mb each
DTS-HD MA suite to DTSenc.dtshd (w/core) 7.1 : 2.6 gb

I have done many THD to DTS-HD conversions in the past and have never seen this big a descrepancy in file sizes.

The DTS result syncs up with the movie just fine, however there is a difference in the bitrates to the original THD track. I imagine it's still lossless, but who knows?

aneurysm

27th November 2012, 17:34

rapscallion

27th November 2012, 17:38

No, not yet.

Here's my errors:

Input #0, truehd, from 'nemo.thd':
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0: Audio: truehd, 48000 Hz, 7.1, s32
Output #0, flac, to 'nemo.flac':
Metadata:
encoder : Lavf54.36.100
Stream #0:0: Audio: flac, 48000 Hz, 7.1, s32, 128 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (truehd -> flac)
Press [q] to stop, [?] for help
Lossless check failed - expected 00, calculated f4.s/s
Lossless check failed - expected 00, calculated 65.s/s
End of stream indicated.33:23.61 bitrate=2977.4kbits/s
[truehd @ 024c0900] Lossless check failed - expected 00, calculated f6.
End of stream indicated.40:24.57 bitrate=2976.1kbits/s
[truehd @ 024c0900] Lossless check failed - expected 00, calculated 88.
size= 2195486kB time=01:40:54.04 bitrate=2970.8kbits/s
video:0kB audio:2195477kB subtitle:0 global headers:0kB muxing overhead 0.000367

Edit: Btw, the difference in bitrates that I mentioned above is substantial.

The original THD average BR is 6333 kbps, while the Flac and DTS-HD are 2970 kbps.
I beleive those 4 errors screwed the pooch on this track.

aneurysm

27th November 2012, 19:36

rapscallion

27th November 2012, 20:23

Weird, I only have these 2 :
Lossless check failed - expected 00, calculated f4.s/s
Lossless check failed - expected 00, calculated 65.s/s

and that is indeed a big difference in BR.
for me it's ok since I just needed DTS
Yes, weird. I ran it twice with the same results.

SquallMX

28th November 2012, 04:17

tebasuna51

28th November 2012, 10:33

The original THD is 24 bits, the FLAC/DTS files could be downsampled to 16 bit, usually 16 bits DTS-HD MA files have an average bitrate of 2.5 Mbps. Differences between 24 and 16 bits files are pretty much impossible to hear.

Hello, SqualMX!

But this isn't the case, the wav's files have the correct size for a mono, 48 Khz, 24 bit, 1h 43m

Here the strange size is the 4.6 GB for the TrueHD file.
I always see a size for thd between flac and dtshd sizes.
The size and the errors let me suspect for a incorrect extraction.

And, maybe, the wav's have garbage at end because the length of the movie is only 1:40:54, but in sync with the video.
The size of monowav's for 1:40:54 must be 831 MB instead 851.

aneurysm

28th November 2012, 17:18

Anyone found a solution/fix for Brave yet ?
indeed lots of errors there, also with ffmpeg thd to flac

EDIT: I continued like yesterday with Nemo, and yes more errors it seems. but the end result is still nice in sync.
kewl :)

Jan Marijniszoon

29th November 2012, 15:12

Hello,

I am curious about something...

I decoded an ac3-file to pcm wav with eac3to which used the libav/ffmpeg decoder.
Then i did the same thing in Linux, but there I did it straight with ffmpeg (version 0.10.3).

I compared the outputs in a wave editor and discovered that eac3to produces files with 'larger / louder' waves than ffmpeg.

How can this be while eac3to is also using the ffmpeg decoding system?

Thanks in advance for your insights.

EDIT: this is not due to the removal of dialog normalisation; I already ruled that out.

kypec

29th November 2012, 16:29

Perhaps DRC (Dynamic Range Compression) is applied in one and not in the other? Just a wild guess...

rapscallion

29th November 2012, 17:20

Jan Marijniszoon

29th November 2012, 19:31

Perhaps DRC (Dynamic Range Compression) is applied in one and not in the other? Just a wild guess...

Do you perhaps have any insight how to disable it in ffmpeg? I tried the option "-drc_scale 0" but that didn't do much.

Probably because they're not the same decoders. Eac3to uses a "patched" version that only madshi can supply.

And let me guess, madshi's version is much better tuned right?

Jan Marijniszoon

30th November 2012, 17:45

I fixed it; I had put the option in the wrong order in the command line.

Using this:

ffmpeg -drc_scale 0.000000 -i sound.ac3 -f wav sound.wav

gave me the exact same wav-file as was produced by eac3to.

Thank you for getting me in the right direction.

kws53

2nd December 2012, 18:01

I have also noticed a very strange thing with the Brave 3D blu-ray. (I don't have the 2D BD to check if it has the same problem.)

Z:\BDMV\PLAYLIST>eac3to 00800.mpls 1)
M2TS, 2 video tracks, 6 audio tracks, 4 subtitle tracks, 1:33:37, 108.607p
1: Chapters, 37 chapters
2: h264/AVC (left eye), 1080p24 /1.001 (16:9)
3: h264/AVC (right eye), 1080p24 /1.001 (16:9)
4: TrueHD/AC3, English, 7.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
5: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
6: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
7: AC3, French, 5.1 channels, 512kbps, 48kHz
8: AC3, French, 5.1 channels, 640kbps, 48kHz
9: AC3, Spanish, 5.1 channels, 640kbps, 48kHz
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), French
13: Subtitle (PGS), Spanish

As you can see, stream 7 is strange. It has an unusual bit rate, and, as far as I know, that stream does not exist in the BD. There is only one French audio track (listed by eac3to as stream #8). tsMuxeR does NOT show that strange French stream.

When eac3to tries to demux that track, it issues the warning "a07 This track is not clean.", but it continues anyway. The resulting ac3 file plays two times faster than it should in all audio players, and the sound is strange.

Same problem here - did anyone notice the 108.607p in the opening line? My log for the 2D version is similar but produces another bizarre number - 95.907p.

M2TS, 1 video track, 7 audio tracks, 8 subtitle tracks, 1:33:37, 95.907p
1: Chapters, 37 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 7.1 channels, 48kHz (embedded: AC3, 5.1 channels, 640kbps, 48kHz)
4: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
5: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
6: AC3, French, 5.1 channels, 512kbps, 48kHz
7: AC3, French, 5.1 channels, 640kbps, 48kHz
8: AC3, Spanish, 5.1 channels, 640kbps, 48kHz
9: AC3, English, 2.0 channels, 320kbps, 48kHz
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), French
13: Subtitle (PGS), Spanish
14: Subtitle (PGS), English
15: Subtitle (PGS), French
16: Subtitle (PGS), Spanish
17: Subtitle (PGS), English

Cheers,
Kurt

damorsoft

4th December 2012, 04:29

I have been trying for days to make the code below work.
If I set createnowindow=true and
disable standard output eac3to runs fine in the command box but just locks up when run with the command box hidden.

This same routine works for mkvmerge tsmuxer and eac3to to read the mpls files.\
I have tried read, readline and readtoend all without success.

Any thoughts thanks

Public Function StartFile(ByVal MyApp As String, ByVal MyTail As String) As Integer
Dim myProcess As New Process()
'Dim test As String = Chr(8)
myProcess.StartInfo.WorkingDirectory = Application.StartupPath
myProcess.StartInfo.FileName = MyApp
myProcess.StartInfo.Arguments = MyTail
myProcess.StartInfo.UseShellExecute = False
'myProcess.StartInfo.CreateNoWindow = false
myProcess.StartInfo.CreateNoWindow = True
myProcess.StartInfo.RedirectStandardOutput = True
'myProcess.StartInfo.RedirectStandardError = True
Try
myProcess.Start()
myProcess.PriorityClass = ProcessPriorityClass.Idle
While (myProcess.HasExited = False)
Application.DoEvents()
Dim sLine As String = myProcess.StandardOutput.ReadToEnd
If (Not String.IsNullOrEmpty(sLine)) Then
TextBox14.Text = sLine
End If

If UserCancel Then
myProcess.Kill()
ProcEC = 9
Return ProcEC
End If
End While
ProcEC = myProcess.ExitCode
Catch ex As Exception
MsgBox(" Error(s) with startfile " & MyApp & " " & ex.Message)
ProcEC = 99
End Try
myProcess.Close()
Return ProcEC
End Function

Soul Taker

14th December 2012, 18:52

Hi there. I'm a bit confused about something, here it is...

1: Chapters, 9 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, Japanese, 2.0 channels, 24 bits, 48kHz
4: DTS Master Audio, Japanese, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
5: TrueHD/AC3, Japanese, 5.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 448kbps, 48kHz)
6: AC3, Japanese, 2.0 channels, 384kbps, 48kHz
7: Subtitle (PGS), Japanese
a04 Extracting audio track number 4...
a04 Decoding with ArcSoft DTS Decoder...
a04 Encoding FLAC with libFlac...
a04 Creating file "C:\Users\ST\Desktop\1_4_audio.flac"...
a04 The original audio track has a constant bit depth of 16 bits.
a04 Superfluous zero bytes detected, will be stripped in 2nd pass.
a04 Starting 2nd pass...
a04 Decoding FLAC...
a04 Reducing depth from 24 to 16 bits...
a04 Encoding FLAC with libFlac...
a04 Creating file "C:\Users\ST\Desktop\1_4_audio.flac"...
a04 The processed audio track has a constant bit depth of 16 bits.
Video track 2 contains 133434 frames.
eac3to processing took 10 minutes, 12 seconds.
Done.

-Is reducing the audio depth from 24 to 16 bits... normal? and i have to say, this the first time that happened to me.

rapscallion

14th December 2012, 19:33

At first glance, far too many Japanese audio/sub tracks. :D

Seriously though, completely normal when there is padding present. Happens all the time. By eac3to removing it doesn't change the quality at all. Still lossless.

BTW, no one can accuse you of being a post whore, 1 post in 4 years!!

Soul Taker

14th December 2012, 21:13

^^^

That's Berserk movie for ya... too much tracks and you only need 2 (facepalm).
LMFAO!! I'm not a forum guy... and i don't know why can't i find it interesting to "camp" 24/7 :p
BTW, i don't even post that much in our own forum "non-american,EU guy"
Anyway, thanks for your reply dude

سلام :D <~ :p

bilditup1

18th December 2012, 02:12

Hey all,
I tried about five times now to encode the DTS-MA audio from 'Fellowship of the Ring' using Arcsoft, eac3to, and Nero. Every single time, instead of a 1h:45min multichannel file, I get a 2:00:48 file that's all distorted and useless. I have no idea why this is happening. It's a 6.1 DTS, which there was some info about earlier in the thread about remappings and such from about a year ago. The information does not seem applicable though, since it's about a year old, and since I don't receive any error messages about channel mappings (eac3to automatically detects that it's a 6.1 file and doubles the back channel in order to facilitate compatibility with NeroAacEnc). In any case, I tried what was suggested - manual channel mapping plus the '-double7' switch - and it changed nothing. I also tried to encode just the DTS-ES core - that didn't help at all. Thanks to the Arcsoft filters, the DTS file does play in MPCHC, but a) with terrible seeking b) a reported running time of 05:24:44. I am using Arcsoft 1.1.0.8, eac3to 3.24, and Nero 1.5.4.0.
Anybody know how to encode this thing?
-e

bilditup1

18th December 2012, 05:28

Hmm, after tracking down several different versions, this seems to be an issue with any version of Arcsoft above 1.1.0.1 with 6.1 tracks. Getting the old DLL is the key.

kypec

18th December 2012, 22:08

Hmm, after tracking down several different versions, this seems to be an issue with any version of Arcsoft above 1.1.0.1 with 6.1 tracks. Getting the old DLL is the key.
Yes, exactly. I'm using version 1.1.0.0 and this works just fine, see the log below:eac3to v3.24
command line: "C:\Program Files (x86)\AVTools\eac3to\eac3to.exe" "I:\TLOTR\FOTR Part 1\BDMV\" 1) 3: D:\Source\TLOTR_EE\fotr1_eng.w64 -down16 -down6 -progressnumbers -log="fotr1_eng.log"
------------------------------------------------------------------------------
M2TS, 1 video track, 7 audio tracks, 24 subtitle tracks, 1:45:43, 24p /1.001
1: Chapters, 27 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 EX, Polish, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
5: AC3 EX, Thai, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
7: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
8: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
9: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
10: Subtitle (PGS), English
11: Subtitle (PGS), Chinese
12: Subtitle (PGS), Korean
13: Subtitle (PGS), Chinese
14: Subtitle (PGS), Czech
15: Subtitle (PGS), Hebrew
16: Subtitle (PGS), Polish
17: Subtitle (PGS), Romanian
18: Subtitle (PGS), Thai
19: Subtitle (PGS), English
20: Subtitle (PGS), Chinese
21: Subtitle (PGS), Korean
22: Subtitle (PGS), English
23: Subtitle (PGS), Chinese
24: Subtitle (PGS), Korean
25: Subtitle (PGS), English
26: Subtitle (PGS), Chinese
27: Subtitle (PGS), Korean
28: Subtitle (PGS), English
29: Subtitle (PGS), Chinese
30: Subtitle (PGS), Korean
31: Subtitle (PGS), English
32: Subtitle (PGS), Polish
33: Subtitle (PGS), Thai
[a03] Extracting audio track number 3...
[a03] Remapping channels...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Mixing surround channels...
[a03] Reducing depth from 24 to 16 bits...
[a03] Writing W64...
[a03] Creating file "D:\Source\TLOTR_EE\fotr1_eng.w64"...
[a03] The original audio track has a constant bit depth of 24 bits.
[a03] The processed audio track has a constant bit depth of 16 bits.
Video track 2 contains 152090 frames.
eac3to processing took 8 minutes, 17 seconds.
Done.

Brom

19th December 2012, 01:20

What's the proper way to transcode DTS-ES 5.1 (matrixed 6.1) to FLAC?

Is there any decoder which outputs DTS-ES 5.1 to 6.1?
Is the matrix known?

sneaker_ger

19th December 2012, 01:33

6.1 channel order is not defined in FLAC, so there is no "proper way" to do it.
http://flac.sourceforge.net/format.html#frame_header

Brom

19th December 2012, 02:16

More interested in getting the correct 7 channel wavs from a DTS-ES 5.1 Matrix.

But the reference FLAC library does the following since FLAC 1.1.3 (27-Nov-2006):
Now properly supports AIFF and WAVEFORMATEXTENSIBLE multichannel input, performing necessary channel reordering both for encoding and decoding. WAVEFORMATEXTENSIBLE channel mask is also saved to a tag on encoding and restored on decoding for situations when there is no natural mapping to FLAC channel assignments.http://flac.sourceforge.net/changelog.html

DarkSpace

21st December 2012, 16:11

What's the proper way to transcode DTS-ES 5.1 (matrixed 6.1) to FLAC?

Is there any decoder which outputs DTS-ES 5.1 to 6.1?
Is the matrix known?
I'm not sure if there's any decoder that natively outputs 6.1 audio for DTS-EX 5.1 (DTS-ES means 6.1 discrete channels, whereas DTS-EX means the back channel is matrix-encoded into the 5.1 channels present), but if you're using ArcSoft DTS Decoder in eac3to, you could try the -7 switch which (undocumented) asks ArcSoft to output 6.1 (7 channels). I am however not sure if that will restore the 7th channel, though, it might just as well output a mix of the present channels or add a silent back channel.

EDIT
More interested in getting the correct 7 channel wavs from a DTS-ES 5.1 Matrix.
if it's just the mono wavs for each channel you want, then you can also try decoding into those directly, using a commandline like
eac3to "input" 1:"output.wavs" -7
under the assumtion that the track number is 1.

quartex

26th December 2012, 12:29

noob question:
convert an AC3 448 Kbps track to an AC3 640Kbps track improve quality?

LigH

26th December 2012, 12:53

Of course not. Dolby Digital (AC3) is a lossy format which stores frequencies of the audible spectrum with a limited precision and suppresses frequencies it calculates as subjectively inaudible. The source (with 448 kbps) already lost some frequency parts of the original sound which was recorded on the movie set. Raising the bitrate afterwards doesn't bring these lost frequencies back, they will still be lost. You cannot "invent" more quality than the source contained. And each conversion to a lossy format — even with a higher bitrate than the source — reduces the quality even more.

sporic

27th December 2012, 02:53

Any fix for Brave and Total Recall yet?

bilditup1

29th December 2012, 04:18

Hmm...so I have a DTS track with 5 channels. It isn't a theater DTS track...just DTS from a commercial Blu-Ray that has no LFE (Adventures of Tintin series from the early 90s). I want to encode this to AAC with a fake LFE so that the usual suspects won't choke on it (iOS hardware decoding). Using the -6 option with eac3to yields a '6-channel' track consisting of the right and left channels and 4 channels of silence, so that's not it. Is there some easy switch that'll enable me to do this with eac3to, or is a more manual approach required?

ED: Looking into it more, seems to be that the commercial DVD itself only had LPCM at 2.0, and the release group botched the audio encode somehow.

tebasuna51

29th December 2012, 11:58

I have a DTS track with 5 channels ...from a commercial Blu-Ray that has no LFE... Using the -6 option with eac3to yields a '6-channel' track consisting of the right and left channels and 4 channels of silence...
Please put the log.

BTW, seems your DTS 5.0 have 3 channels of silence.
Maybe you can try something like:

eac3to input.dts stdout.wav | sox -t wav - -t wav - remix -m 1 2 | NeroAacEnc -q 0.5 -ignorelength -if - -of output.m4a

jpsdr

30th December 2012, 13:39

Does someone know how i can extract the 6 channels from a .m2ts file when audio is PCM ? I want to have all 6 channels in separate files, raw format is fine.

Thanks.

bilditup1

30th December 2012, 21:07

Please put the log.

BTW, seems your DTS 5.0 have 3 channels of silence.
Maybe you can try something like:

eac3to input.dts stdout.wav | sox -t wav - -t wav - remix -m 1 2 | NeroAacEnc -q 0.5 -ignorelength -if - -of output.m4a

Right, I figured out that it was a bad encode. I don't think the resulting 2ch files (which register as 6ch, with 4ch of silence) are any bigger than normal 2ch files. They're just recognized as 6ch files (as they should be) Will the above code take only the front two channels? Will using -down2 instead of that in this case muck up the audio or something?

tebasuna51

31st December 2012, 03:40

Will the above code take only the front two channels?
Yes.

Will using -down2 instead of that in this case muck up the audio or something?
Nope, because -down2 can't be applied to a 5.0 source, must be 5.1.

tebasuna51

31st December 2012, 03:44

Does someone know how i can extract the 6 channels from a .m2ts file when audio is PCM ? I want to have all 6 channels in separate files, raw format is fine.

eac3to input.m2ts PCM_TRACK: output.wavs

madshi

2nd January 2013, 21:24

eac3to v3.25 released

http://madshi.net/eac3to.zip

* fixed: option "-down6" didn't handle 6.1 tracks properly
* fixed: option "-down6" for 7.1 content lowered side channels in volume
* fixed: 6.1 DTS/E-AC3 libav decoding resulted in incorrect 70f channel mask
* fixed: a couple of funny AC3 channel configs resulted in swapped channels
* fixed: option "-check" failed for SRT subtitles
* fixed: MKV tracks without language information didn't default to "English"
* fixed: 16bit MLP decoding didn't work at all
* fixed: DTS 2.1 decoding had empty sub channel when using ArcSoft decoder
* fixed: muxing h264 video streams without VUI eventually crashed
* fixed: SMPTE 302M 20bit PCM tracks didn't extract properly
* improved AC3 seamless branching handling by detection of identical frames
* improved DTS seamless branching handling by detection of identical frames
* updated to latest ffmpeg/libav revision
* added support for libav DTS 6.1 and 7.1 decoding (only in core, no DTS-HD)
* added support for encoding all AC3 channel configurations
* added support for demuxing Blu-Ray primary E-AC3 tracks (AC3 core)
* libav is now the default decoder for AC3 and E-AC3
* option "-down2" is now deprecated, but still supported
* option "-downDpl" mixes down to Dolby Pro Logic II
* option "-downStereo" mixes down to standard stereo
* added support for mixing any channel configuration to stereo/DPL
* DPL downmixing doesn't lower volume, anymore (except when clipping occurs)
* editing audio tracks by adding silence is now less restrictive
* trying to mux right eye stream to MKV errors out now, instead of crashing
* doing "eac3to 1) movie.mkv" for 3D BDs now properly skips right eye stream
* FLAC encoding is now done with optimized block sizes for "LossyWAV" files
* encoder bitdepth for lossy DTS tracks is no longer displayed
* option "-test" no longer checks mkvtoolnix
Thanks to tebasuna51 for moderating the bug/improvement list, and for investigating the "duplicate frame" stuff for seamless branching Blu-Rays. I didn't know that overlapping AC3 and DTS frames were actually bit-wise identical! With this information eac3to can now discard such frames immediately, which means that it's no longer necessary to do a 2nd pass for seamless branching Blu-Rays to fix audio overlaps. This new logic should also avoid any audio glitches at the seamless branching edit points - because simply no edits are necessary, anymore, at all. Please note that this only applies to AC3 and DTS (and DTS-HD/MA) tracks. It does not apply to LPCM and TrueHD tracks. However, overlaps with LPCM and TrueHD tracks are rather small, so overlap editing is often not necessary for such tracks, anyway. The new logic also only works if the overlapping AC3 and DTS frames are truely identical. They have been with the 2 Blu-Rays I've tested with. But I don't know for sure if that's the case for *all* seamless branching Blu-Rays. So there's a chance the old overlap logic might still be used for some Blu-Rays.

It seems that mono TrueHD decoding is currently broken when using libav/ffmpeg decoding, but I'm not 100% sure because I only have one such sample and it's pretty short. Maybe you guys can double check (and provide me with a longer sample)? This looks like a libav/ffmpeg bug to me, though, if it's really a bug at all. At least Brave TrueHD decoding seems to be working fine now.

Please note that although eac3to now supports parsing, demuxing and core-extraction for Blu-Ray primary E-AC3 tracks, currently no decoder is available which can decode the full audio information with all 6.1/7.1 channels. Currently only the AC3 core can be decoded which results in 5.1 output only.