eac3to - audio conversion tool

[73ChargerFan]

Try BDInfo, it will decode all MPLS files and give you the track lengths. I'm guessing you'll find seven 20 minute playlists. BDInfo does make mistakes sometimes, though.

[Killroy™]

I just bought the Clone Wars Season one box set.

After ripping with AnyDVD HD, I use eac3to to select one of the episodes and rip out the video, audio & chapter data and process further from there.

However, eac3to is only showing 1 "episode" (for lack of a better term) which I don't believe is actually an episode, and there's only video - duration 15:02. No audio, no chapters.

I know there are 7 separate episodes on this disc but, for some reason, eac3to is not finding them.

Anyone else struck this yet and, if so, found a way around it?

Good luck with this. I found the same thing until I found sort of a solution:

I had to hunt down the .mpls that showed the actual playlist and found that the "Play All" is listed on 00020.mpls but I found another surprise:

c:\eac3to317>eac3to f:\swcwdisc1\bdmv\playlist\00200.mpls
1) 00200.mpls, 2:41:03
[95+96+97+96+98+96+99+96+101+96+102+96+103+96].m2ts
- Chapters, 35 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48khz
- AC3, French, multi-channel, 48khz
- AC3, German, multi-channel, 48khz
- AC3, Spanish, multi-channel, 48khz
- AC3, Spanish, multi-channel, 48khz
- AC3, Japanese, multi-channel, 48khzDo you see that repeating 00096.m2ts file? Well, that little sucker is a 5 minute copyright placecard...yes, 5 minutes repeated after each episode.

Even if you decide to leave it in place and just skip it with the "Next Chapter" button (if you include chapters) you will find that those 5 minute files really screw up the synch for each episode after it. The first episode is fine but later on it gets worse and worse.

The only way to do it is to remux each individual episode as its own file. At least you know the correct order by looking at the mpls file.

[73ChargerFan]

The following should work.

eac3to f:\swcwdisc1\bdmv\stream\00095.m2ts+f:\swcwdisc1\bdmv\stream\00097.m2ts+f:\swcwdisc1\bdmv\stream\00098.m2ts+f:\swcwdisc1\bdmv\stream\00099.m2ts+f:\swcwdisc1\bdmv\stream\00101.m2ts+f:\swcwdisc1\bdmv\stream\00102.m2ts+f:\swcwdisc1\bdmv\stream\00103.m2ts -demux

To get a chapter list, create one manually by adding the length of each successive episode to the prior chapter time.

[Roscoe62]

The following should work.

eac3to f:\swcwdisc1\bdmv\stream\00095.m2ts+f:\swcwdisc1\bdmv\stream\00097.m2ts+f:\swcwdisc1\bdmv\stream\00098.m2ts+f:\swcwdisc1\bdmv\stream\00099.m2ts+f:\swcwdisc1\bdmv\stream\00101.m2ts+f:\swcwdisc1\bdmv\stream\00102.m2ts+f:\swcwdisc1\bdmv\stream\00103.m2ts -demux

To get a chapter list, create one manually by adding the length of each successive episode to the prior chapter time.

Thanks for this. I'm just going to do this individually for each episode, but so far this procedure is working fine. Thanks again! :)

[umaximus]

Is it possible that TrueHD track doesn't have audio gaps/overlaps on seamless branching disc while AC3 has?

Pineapple Express

eac3to v3.17
command line: eac3to 1) 2: video.h264 3: audio.wav 5: audio.ac3
------------------------------------------------------------------------------
M2TS, 1 video track, 7 audio tracks, 15 subtitle tracks, 1:57:26, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB)
4: TrueHD/AC3, French, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB)
5: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Extracting TrueHD stream...
[a03] Removing TrueHD dialog normalization...
[a05] Extracting audio track number 5...
[a03] Decoding with libav/ffmpeg...
[a05] Removing AC3 dialog normalization...
[a03] Writing WAV...
[a03] Creating file "audio.wav"...
[v02] Creating file "video.h264"...
[a05] Creating file "audio.ac3"...
[a03] [libav] End of stream indicated <WARNING>
[a03] The original audio track has a constant bit depth of 16 bits.
[a03] Caution: The WAV file is bigger than 4GB. <WARNING>
[a03] Some WAV readers might not be able to handle this file correctly. <WARNING>
[a05] Audio overlaps for 7ms at playtime 0:10:27. <WARNING>
[a05] Audio overlaps for 8ms at playtime 0:27:14. <WARNING>
[a05] Audio overlaps for 10ms at playtime 0:32:23. <WARNING>
[a05] Audio overlaps for 18ms at playtime 0:44:12. <WARNING>
[a05] Audio overlaps for 31ms at playtime 0:44:54. <WARNING>
[a05] Audio overlaps for 28ms at playtime 0:56:01. <WARNING>
[a05] Audio overlaps for 9ms at playtime 1:02:00. <WARNING>
[a05] Audio overlaps for 15ms at playtime 1:10:16. <WARNING>
[a05] Audio overlaps for 14ms at playtime 1:12:12. <WARNING>
[a05] Audio overlaps for 32ms at playtime 1:43:47. <WARNING>
[a05] Audio overlaps for 25ms at playtime 1:52:10. <WARNING>
[a03] Superfluous zero bytes detected, will be stripped in 2nd pass.
[a03] Starting 2nd pass...
[a03] Reading WAV...
[a03] Stripping zero bytes...
[a03] Writing WAV...
[a03] Creating file "audio.wav"...
[a03] Caution: The WAV file is bigger than 2GB. <WARNING>
[a03] Some WAV readers might not be able to handle this file correctly. <WARNING>
[a05] Starting 2nd pass...
[a05] Realizing (E-)AC3 gaps...
[a05] Creating file "audio.ac3"...
Video track 2 contains 168949 frames.
eac3to processing took 47 minutes, 28 seconds.
Done.

[nautilus7]

Yes, that's very usual. TrueHD frames are ~3ms, while AC3 ones are 32ms long. IIRC, eac3to won't report any overlaps if they are too short.

[wolfbane5]

I was checking out We Were Soldiers and noticed that it fronts a DTS-ES audio track. When I used the -down6 option, it created the .wavs but said that Surcode wasn't installed. It is installed, however, I've encountered this problem before and have simply ignored it since I've been unable to find a solution to it. My question is: does the -down6 option mix/rewrite the channels so that the BC .wav track is somehow included in the other 6 channels or does it simply ignore the BC .wav track created when you convert the .dts into .wavs? If it's the latter, then I can simply take the long route and create the .wavs from the .dts, ignoring the BC .wav track and then use Surcode to create a 5.1 dts audio track.

Thanks for the help in advance.

[nautilus7]

From 1st post:

-down6 --> downmix 7 or 8 channels to 6 channels

[wolfbane5]

Alright, so if that's the case, then I need to use -down6. But this leads to my problem of eac3to thinking Surcode isn't installed. Is there a way to use the Arcsoft DTS Decoder manually?

[TinTime]

If you already have a DTS track why decode it and encode to DTS again? Why not just keep it?

[SomeJoe]

1. Why the need to convert DTS-ES into something else? The DTS-ES stream is fully compatible with any DTS receiver, even if that receiver is 5.1 or doesn't recognize the ES extensions.

2. DTS-ES can come in two forms:

A) DTS-ES 5.1 Matrix, which derives the back center channel from matrix encoding on the Left Surround/Right Surround channels. This is the same way that Dolby Digital EX works. There is no difference between a DTS-ES 5.1 Matrix stream and a standard DTS 5.1 stream -- both contain 5.1 discrete channels of audio.

B) DTS-ES 6.1 Discrete, which actually has a discrete back center channel in the DTS stream. This is not nearly as common as the DTS-ES 5.1 Matrix described above. But this stream will still play back properly on 5.1 equipment, just without the back center channel.

If your current stream is DTS-ES 5.1 Matrix (likely), then there is no need for -down6, since the stream is already 5.1 channels.

[raquete]

The DTS-ES stream is fully compatible with any DTS receiver, even if that receiver is 5.1 or doesn't recognize the ES extensions. seems that have space for my fast question.
i have DTS receiver 6.1 and DTS-ES encoder but how to author & burn in medias(what 'format' to chose) ?
(with or without video, doesn't matter, the target is audio)

Happy New Year ! :)

[wolfbane5]

SomeJoe,

I was converting the DTS-ES into DTS since I don't have a receiver and was planning on watching the movie on my computer, which has 5.1 audio.

As for Matrix or Discrete, MediaInfo is telling me this:

General
Format : DTS
Format/Info : Digital Theater Systems
File size : 1.46 GiB
Duration : 2h 18mn
Overall bit rate : 1 510 Kbps

Audio
Format : DTS
Format/Info : Digital Theater Systems
Format profile : ES
Duration : 2h 18mn
Bit rate mode : Constant
Bit rate : 1 510 Kbps
Channel(s) : 7 channels
Channel positions : Front: L C R, Rear: L C R, LFE
Sampling rate : 48.0 KHz
Resolution : 24 bits
Stream size : 1.46 GiB (100%)

Am I correct in assuming that's DTS 6.1 Discrete?

[tebasuna51]

You only need eac3to to know this. Check the DTS with eac3to:

DTS-ES, 5.1 channels, 0:00:22, 16 bits, 1510kbps, 48khz (Matrix)
DTS-ES, 6.1 channels, 0:00:22, 16 bits, 1510kbps, 48khz (Discrete)

[tebasuna51]

i have DTS receiver 6.1 and DTS-ES encoder but how to author & burn in medias(what 'format' to chose) ?
The same than standard DTS.

To burn CD or play with some players (WD TV Live for instance) you need dtswav.

To mix with video or play on PC or some players (Xtreamer) you can use the .cpt (compact format) with the extension renamed to .dts.

Maybe some authoring program need the .dts (padded)

[ACrowley]

one question :

I never thought about but can ec3to outout 32bit float .wav and .wavs from any Source ?
Maybe -down32 works as a override ?

EDIT: no...doesnt work.

[TinTime]

You can try using -full.

However I don't think eac3to will ever output at a higher bit depth (i.e. pad with zeroes) than is necessary.

So with the Nero decoder...
eac3to input.ac3 output.wav -full
...will result in 24bit output but...
eac3to input.ac3 output.wav -25.000 -changeTo24.000 -full
...will result in 64bit float output.

[raquete]

The same than standard DTS.

To burn CD or play with some players (WD TV Live for instance) you need dtswav.

To mix with video or play on PC or some players (Xtreamer) you can use the .cpt (compact format) with the extension renamed to .dts.

Maybe some authoring program need the .dts (padded) very clever with full details.

thank you so much! :)

[Thunderbolt8]

what does it mean again when I have a 7.1 track with a "strange setup" note? anything I have to look out for? and what was it again with a DTS-HD MA track with -2db dialnorm & eac3to saying "decoding this track with arcsoft resulsts in low volume" (its the same track in both cases)

guess I have a problem now, because arcsoft results in low volume and sonic can only do 5.1?

[xkodi]

Arcsoft can't do bit-perfect decode of DTS-HD MA 7.1 "strange setup" files, while Sonic can but only 5.1 channels, look at this posts and all links in it to understand more:

http://forum.doom9.org/showthread.php?p=1179218#post1179218

[Thunderbolt8]

hm so whats the standing regarding this problem, is madshi willing to do this patch so that arcsoft can properly recognize the channel order?

and does the low volume decoding problem result from this strange channel order or from those -2db dialnorm value?

additionally, the acrsoft decoder says that track has constant 24-bit, while sonic says the 5.1 track is only 16-bit?

[SpaceAgeHero]

Hey folks,

I am having some troubles getting the Arcsoft DTS Decoder to work with eac3to.
Recently I installed Arcsoft Total Media Theatre and updated to 3.0.1.160.
I don't know what version of the dts decoder is included in this package.
Eac3to simply won't recognize the decoder. I'm currently running Windows 7 x64 Ultimate.
I also tried adding "c:\Program Files (x86)\ArcSoft\TotalMedia Theatre 3\" and/or "c:\Program Files (x86)\ArcSoft\TotalMedia Theatre 3\Codec\" to my environment path. c:\Program Files\Common Files\ArcSoft is not there!

Perhaps someone can help me out via E-Mail or private messages.
I do not want to bloat this thread with this issue.
Thank you!

[Leiw]

great program :) works fine for all my conversion needs

[Capsbackup]

@SpaceAgeHero;
Try this thread out, should help you.
http://forum.doom9.org/showthread.php?t=148324&page=2

[PowerGamer]

I have "Pixar Short Films Collection" Bluray. "Mike's New Car" movie on this bluray has the following tracks:

eac3to.exe "PIX1EGD1\BDMV\PLAYLIST\00008.mpls"
1) 00008.mpls, 00006.m2ts, 0:03:49
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, multi-channel, 48khz
...

I demuxed video and audio tracks with eac3to, tsmuxer and xport. Audio track in .pcm format created by eac3to and xport are bitwise identical. But the same audio track in .wav format produced by eac3to and tsmuxer are different.

Command lines used:

eac3to vs. xport:
eac3to.exe "PIX1EGD1\BDMV\PLAYLIST\00008.mpls" -demux
xport.exe -h "PIX1EGD1\BDMV\STREAM\00006.m2ts" 1 1 1

eac3to vs. tsmuxer:
eac3to.exe "PIX1EGD1\BDMV\PLAYLIST\00008.mpls" 2: audio_eac3to.wav
tsMuxeR.exe 1.meta .\

1.meta file content:
MUXOPT --demux
A_LPCM, "PIX1EGD1\BDMV\STREAM\00006.m2ts", track=4352, lang=eng, mplsFile=00008

Upon further analysis of .wav files created by eac3to and tsmuxer I discovered that these two .wav files contain different value in the dwChannelMask field in the .wav file header: tsmuxer .wav file has 0x0000003F and eac3to .wav file has 0x0000060F. Otherwise headers of both .wav files contain bitwise identical information. In the audio data contained within these two .wav files there were about 880 miscompared bytes (see wav_diff.txt attached to this post).

According to http://www.microsoft.com/whdc/device/audio/multichaud.mspx:

dwChannelMask=0x0000003F (tsmuxer):
FRONT_LEFT, FRONT_RIGHT, FRONT_CENTER, LOW_FREQUENCY, BACK_LEFT, BACK_RIGHT

dwChannelMask=0x0000060F (eac3to):
FRONT_LEFT, FRONT_RIGHT, FRONT_CENTER, LOW_FREQUENCY, SIDE_LEFT, SIDE_RIGHT.

My questions are:
1. Is it correct to say that eac3to or tsmuxer performed some kind of transformation of audio signal according to different dwChannelMask value which caused miscompared bytes in .wav files?
2. Which .wav file (produced by eac3to or tsmuxer) is "better" or "more identical to the source" and why?

[Abradoks]

neroAacEnc expects floating-point number as "-q" parameter while eac3to accepts only two decimal places. It is important because bitrate may vary significantly over 0.01, e.g. q=0.37 -> 240kbps, q=0.38 -> 370kbps.

[tebasuna51]

...
My questions are:
1. Is it correct to say that eac3to or tsmuxer performed some kind of transformation of audio signal according to different dwChannelMask value which caused miscompared bytes in .wav files?
2. Which .wav file (produced by eac3to or tsmuxer) is "better" or "more identical to the source" and why?
I can't see your wav_diff.txt (Attachements Pending Approval), use http://pastebin.com/ to upload txt files, but:

a) The Channel Mask 0x0000060F is the recommended by M$ for wav 5.1, after XP SP2, but the old 0x0000003F must be also compatible for 5.1. Then:
0x0000003F old, but compatible with old and new soft.
0x0000060F new, but can be a problem with old soft.

b) The audio data bit difference between eac3to and TsMuxer is a know, and reported, bug in TsMuxer extraction. Don't use TsMuxer to extract only to mux.

To answer your questions:
1) The channelmask are both valids and don't modify the audio data, the problem is the TsMuxer's bug.
2) Of course the eac3to wav is the "exact" audio source.

[AnryV]

eac3to v3.17
command line: 00000_1_02.dtshd dtshd.wavs
------------------------------------------------------------------------------
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 "dtshd.C.wav"...
Creating file "dtshd.R.wav"...
Creating file "dtshd.SR.wav"...
Creating file "dtshd.LFE.wav"...
Creating file "dtshd.L.wav"...
Creating file "dtshd.SL.wav"...
Creating file "dtshd.BR.wav"...
Creating file "dtshd.BL.wav"...


00000_1_02.dtshd has the following scheme of channells (according to Scenaris and BDReauthor):
C,L,R,LFE,Lsr,Rsr,Lss,Rss

To what channels in this scheme do correspond eac3to's channels SL,SR,BL, BR?

[tebasuna51]

00000_1_02.dtshd has the following scheme of channells (according to Scenaris and BDReauthor):
C,L,R,LFE,Lsr,Rsr,Lss,Rss

To what channels in this scheme do correspond eac3to's channels SL,SR,BL, BR?
SL (Side Left) is Lss (Left Surround Side?), SR is Rss.
BL (Back Left) is Lsr (Left Surround Rear?), BR is Rsr.

[PowerGamer]

I can't see your wav_diff.txt (Attachements Pending Approval), use http://pastebin.com/ to upload txt files, but:

a) The Channel Mask 0x0000060F is the recommended by M$ for wav 5.1, after XP SP2, but the old 0x0000003F must be also compatible for 5.1. Then:
0x0000003F old, but compatible with old and new soft.
0x0000060F new, but can be a problem with old soft.

b) The audio data bit difference between eac3to and TsMuxer is a know, and reported, bug in TsMuxer extraction. Don't use TsMuxer to extract only to mux.

To answer your questions:
1) The channelmask are both valids and don't modify the audio data, the problem is the TsMuxer's bug.
2) Of course the eac3to wav is the "exact" audio source.

Tebasuna51, thanks for your explanations. Meanwhile I tryed to demux that audio track using Blu-Ray Demuxer Pro (http://dvd-logic.com/bddemuxerpro.php) and much to my dismay found out that it produces exactly the same .pcm file as eac3to/xport but completely different .wav file that practically in every byte (apart from header which is the same as tsmuxer's .wav file) differs from both eac3to and tsmuxer versions. Still all .wav files have exactly the same size and play fine in media player.

Once again it is unclear which .wav output is better: eac3to or Blu-Ray Demuxer Pro? And what is the difference between .pcm and .wav files and what kind of transformation is involved when converting from .pcm to .wav?

[tebasuna51]

Once again it is unclear which .wav output is better: eac3to or Blu-Ray Demuxer Pro?
I never support commercial soft.
If the wav audio data output is different (the header may be different), then the Blu-Ray Demuxer Pro is wrong because the eac3to output is exact (not better, is exact).

And what is the difference between .pcm and .wav files and what kind of transformation is involved when converting from .pcm to .wav?
The audio data is stored with different byte order (Big endian, Litle Endian) and the audio channels are stored also with different order.
The wav file have a header, the pcm is only audio data without header.

[PowerGamer]

The audio data is stored with different byte order (Big endian, Litle Endian) and the audio channels are stored also with different order.
The wav file have a header, the pcm is only audio data without header.

Taking that into account (thanks again for explaining) I found out the following (EAC - channel data in .wav file produced by eac3to, BDR - same in Blu-ray Demuxer .wav, PCM - corresponding data in .pcm file):

PCM EAC BDR
0100 0001 0001
0200 0002 0002
0000 0000 0000
ffff 0000 ffff
0000 ffff 0000
0000 0000 0000

ffff ffff ffff
ffff ffff ffff
0000 0000 0000
0100 0001 0001
0000 0001 0000
0100 0000 0001

0000 0000 0000
0000 0000 0000
0000 0000 0000
0000 fffe 0000
0000 0000 0000
feff 0000 fffe

0100 0001 0001
0000 0000 0000
0100 0001 0001
ffff 0003 ffff
0000 ffff 0000
0300 0000 0003

ffff ffff ffff
0100 0001 0001
feff fffe fffe
0000 fffc 0000
0000 0000 0000
fcff 0000 fffc

So apart from Big-Endian to Little-Endian convertion (done by each program) eac3to takes 6th channel from .pcm writes it at position of 3rd channel and pushes one position below 4th and 5th channels. Can someone please explain what is the meaning of this channel rearrangement? (Headers of eac3to and Blu-ray Demuxer .wav files differ only by dwChannelMask: 0x0000060F (eac3to), 0x0000003F (Blu-ray Demuxer)).

[tebasuna51]

So apart from Big-Endian to Little-Endian convertion (done by each program) eac3to takes 6th channel from .pcm writes it at position of 3rd channel and pushes one position below 4th and 5th channels. Can someone please explain what is the meaning of this channel rearrangement? (Headers of eac3to and Blu-ray Demuxer .wav files differ only by dwChannelMask: 0x0000060F (eac3to), 0x0000003F (Blu-ray Demuxer)).
Like I say before the audio data in .pcm have different channel order than in wav container:

pcm: FL, FR, FC, BL, BR, LF
wav: FL, FR, FC, LF, BL, BR
then the LF channel in PCM (6th) must go to 4th position, BL (4th) to 5th and BR (5th) to 6th.

Seems BDR don't make the channel arrangement.
Is easy to verify the problem, if you play the BDR.wav with a multichannel capable player must listen the Low Frequency channel in the Back Right speaker.

[Stephen R. Savage]

Is there any way to implement audio cutting (by timecodes) in eac3to? A sample-accurate way to cut and join audio clips without intermediary files or pipes would be great. Perhaps something like "eac3to in.m2ts 2:out.flac -remove 00:00.000-01:30.000,20:30.000,22:00.000"

[TinTime]

@PowerGamer

Specifically the channels must be interleaved in this order (http://www.microsoft.com/whdc/device/audio/multichaud.mspx#EKLAC) for wav pcm.

[tebasuna51]

Is there any way to implement audio cutting (by timecodes) in eac3to? A sample-accurate way to cut and join audio clips without intermediary files or pipes would be great. Perhaps something like "eac3to in.m2ts 2:out.flac -remove 00:00.000-01:30.000,20:30.000,22:00.000"

Something like that:

v2.84
...
* new option for removing or looping audio data, e.g. "-edit=0:20:47,-100ms"

[umaximus]

I got loads of lossless check failed errors on Animatrix's TrueHD track while extracting/transcoding. Are this safe to overlook? How much of ms is failed byte?

eac3to v3.17
command line: eac3to 1) 2: video.vc1 4: audio.wavs
------------------------------------------------------------------------------
M2TS, 1 video track, 8 audio tracks, 18 subtitle tracks, 1:40:50, 24p /1.001
1: Chapters, 10 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
4: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
5: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: AC3, Italian, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
7: AC3, Japanese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
8: TrueHD/AC3, Japanese, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
9: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
10: AC3, Portuguese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
11: Subtitle (PGS), English
12: Subtitle (PGS), French
13: Subtitle (PGS), Italian
14: Subtitle (PGS), Italian
15: Subtitle (PGS), Dutch
16: Subtitle (PGS), Spanish
17: Subtitle (PGS), Portuguese
18: Subtitle (PGS), Japanese
19: Subtitle (PGS), English
20: Subtitle (PGS), Japanese
21: Subtitle (PGS), French
22: Subtitle (PGS), Italian
23: Subtitle (PGS), Italian
24: Subtitle (PGS), Dutch
25: Subtitle (PGS), Spanish
26: Subtitle (PGS), Portuguese
27: Subtitle (PGS), Japanese
28: Subtitle (PGS), English
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Writing WAVs...
[v02] Creating file "video.vc1"...
[a04] Creating file "audio.R.wav"...
[a04] Creating file "audio.L.wav"...
[a04] Creating file "audio.LFE.wav"...
[a04] Creating file "audio.SR.wav"...
[a04] Creating file "audio.C.wav"...
[a04] Creating file "audio.SL.wav"...
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated fe <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 8c <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 68 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated f5 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 73 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated e4 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 2e <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 26 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated b7 <WARNING>
[v02] Video overlaps for 7 frames at playtime 0:53:49. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:06:51. <WARNING>
[v02] Video overlaps for 7 frames at playtime 1:33:01. <WARNING>
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] Audio overlaps for 43ms at playtime 0:18:49. <WARNING>
[a04] Audio overlaps for 43ms at playtime 0:28:12. <WARNING>
[a04] Audio overlaps for 43ms at playtime 0:45:06. <WARNING>
[a04] Audio overlaps for 42ms at playtime 0:53:50. <WARNING>
[a04] Audio overlaps for 42ms at playtime 1:06:54. <WARNING>
[a04] Audio overlaps for 43ms at playtime 1:33:01. <WARNING>
[a04] Superfluous zero bytes detected, will be stripped in 2nd pass.
[a04] Starting 2nd pass...
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Reducing depth from 24 to 16 bits...
[a04] Writing WAVs...
[a04] Realizing RAW/PCM gaps...
[a04] Creating file "audio.R.wav"...
[a04] Creating file "audio.L.wav"...
[a04] Creating file "audio.SR.wav"...
[a04] Creating file "audio.SL.wav"...
[a04] Creating file "audio.C.wav"...
[a04] Creating file "audio.LFE.wav"...
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated fe <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 8c <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 68 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated f5 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 73 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated e4 <WARNING>
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 2e <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 26 <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated b7 <WARNING>
[a04] The processed audio track has a constant bit depth of 16 bits.
Video track 2 contains 145067 frames.
eac3to processing took 3 hours, 23 minutes.
Done.

[AnryV]

I think that eac3to (or ArcSoft decoder ??) uncorrectly decode 7.1 - L,R,C,LFE,Ls,Rs,Lsr,Rsr scheme.


eac3to v3.17
command line: eac3to Ls_for_eac3to.dtshd Ls.wavs
------------------------------------------------------------------------------
DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48khz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
CAUTION: Decoding this track with ArcSoft results in low volume. <WARNING>
Decoding with ArcSoft DTS Decoder...
Writing WAVs...
The original audio track has a constant bit depth of 24 bits.
Creating file "Ls.L.wav"...
Creating file "Ls.R.wav"...
Creating file "Ls.C.wav"...
Creating file "Ls.LFE.wav"...
Creating file "Ls.BL.wav"...
Creating file "Ls.BR.wav"...
Creating file "Ls.SL.wav"...
Creating file "Ls.SR.wav"...
eac3to processing took 4 seconds.
Done.


1. Ls.dtshd - is the file encoded by DTS-HD Encoder Suite from wavs from <Original>. Scheme - 7.1 - L,R,C,LFE,Ls,Rs,Lsr,Rsr
2. Ls_for_eac3to.dtshd - the same file but without header which don't recognized by eac3to
3. <Original> dir - contains original 8 wavs (1 second duration)
4. <eac3to> dir - contains 8 wavs decoded from Ls_for_eac3to.dtshd by eac3to
5. <StreamPlayer> dir - contains 8 wavs decoded from Ls.dtshd by DTS-HD StreamPlayer

As result - <StreamPlayer> fully identical <Original> but BL and BR files from <eac3to> contains not Ls and Rs from <Original> but mixed Ls+Lsr and Rs+Rsr

Sample
http://multi-up.com/201196

[Thunderbolt8]

look at the link of post #9676

[AnryV]

look at the link of post #9676
Problem, as I do understand, was not solved?

[umaximus]

I have a little question. In the past I have demux the PCM tracks from blu-rays and save them. As I can see from the logs, eac3to have remapped the channels already, but now when I want to transcode PCM to WAVS (for use with DTS-HD mas suite) eac3to is again remapping them which I think is not right. Is there a switch to turn off remapping of the channels?

[Thunderbolt8]

Problem, as I do understand, was not solved?no, apparently there havent been enough discs yet with this problem to justify the work

[bassgoonist]

I have some DVD-A disc in 5.0. eac3to seems to have no idea what to do with these. It won't downmix, it won't do DTS or AC3. If I open it up in audacity and insert an empty track where the LFE would be it works just fine. Is there anyway eac3to could have an option to do this?

[Tiziano]

Sorry, perhaps I'm very dum but I need a help. Sorry if I'm disturbing you continuously

I tried to convert a 7.1 ( 8 channels ) interleaved WAV file into an e-AC3 file ... I mean: it should be Dolby Digital Plus standard ... but I was not able to do it.

I'm wondered if anybody can help me giving the correct settings or command line in order to get this .WAV correctly converted ...

I thank you very much

[utenteanonimo64]

I have some DVD-A disc in 5.0. eac3to seems to have no idea what to do with these. It won't downmix, it won't do DTS or AC3. If I open it up in audacity and insert an empty track where the LFE would be it works just fine. Is there anyway eac3to could have an option to do this?

Why don't you use DVD-A Explorer? You can extract the multichannel data or downmix to 2.0. Once you have the WAV files then eac3to will do whatever you want with them...

[tebasuna51]

...
I tried to convert a 7.1 ( 8 channels ) interleaved WAV file into an e-AC3 file ... I mean: it should be Dolby Digital Plus standard ... but I was not able to do it.
...

Aften, the ac3 encoder included with eac3to, isn't a Dolby Digital Plus encoder, you can obtain only 5.1 standard ac3.

[tebasuna51]

I have some DVD-A disc in 5.0. eac3to seems to have no idea what to do with these. It won't downmix, it won't do DTS or AC3. If I open it up in audacity and insert an empty track where the LFE would be it works just fine. Is there anyway eac3to could have an option to do this?

To convert a wav file 5.0 you can use WavToAc3Enc or eac3to with external Aften.exe encoder:

eac3to source50 stdout.wav | Aften -pad 0 -readtoeof 1 -exps 32 -s 1 -b 640 - output.ac3

where:
-exps 32 -s 1 -b 640
is the best quality (and slow encode) than Aften can offer.

[deathlord]

Hi

Can I have eac3to use the source filename for the destination?
I mean in the command line, like eac3to source.wav dest.wav

[twazerty]

@madshi:

Awhile back I saw clipping audio when transcoding MP2->AC3. You mentioned it was due to internal bit depth restrictions in libavcodec, which could probably be overcome with a more recent build of libavcodec. Did you ever integrate this?

Noticed that the clipping problem isn't fixed yet. Can we expect a fix?

[blackcell]

For some unknown reason, Eac3to is unable to extract the BD disc "The Express". It gets nearly finished then aborts.

Log file:
eac3to v3.17
command line: "C:\Tools\RipBot\Tools\eac3to\eac3to.exe" "D:\BDMV\STREAM" 1) 1: "C:\tmp\The Express - Chapters.txt" 2: "C:\tmp\The Express 1080p VC-1.mkv" 3: "C:\tmp\The Express - 3 DTS Master.flac" 6: "C:\tmp\The Express - 6 English Subtitle.sup" -log="C:\tmp\The Express - Log.txt"
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 5 subtitle tracks, 2:09:42, 24p /1.001
1: Chapters, 20 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: DTS, Spanish, 5.1 channels, 24 bits, 768kbps, 48khz
5: AC3 Surround, French, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: Subtitle (PGS), English
7: Subtitle (PGS), Spanish
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), French
Creating file "C:\tmp\The Express - Chapters.txt"...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[s06] Extracting subtitle track number 6...
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "C:\tmp\The Express - 3 DTS Master.flac"...
[s06] Creating file "C:\tmp\The Express - 6 English Subtitle.sup"...
Reading the source file failed. <ERROR>
Aborted at file position 24368906240. <ERROR>

Anyone else seen this issue with other BD's? Any workarounds?