eac3to - audio conversion tool

[the_weirdo]

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]

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)

[Brazil2]

<Left blank, please delete>

[sneaker_ger]

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]

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]

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]

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]

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]

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]

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]

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]

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]

@ 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]

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]

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]

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]

EDIT: Solved with nero micro/lite

[fano]

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]

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.

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*

[fano]

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_]

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]

.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]

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]

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]

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]

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]

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]

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]

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]

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]

Mux to a different drive or use mkvmerge.

[doom-nine]

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.

[Snowknight26]

Not with this program. Try WaveWizard, Audacity, etc.

[ramicio]

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.

[doom-nine]

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]

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]

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]

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]

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]

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]

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_]

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.

You can't change DRC without reencoding, but you can change DN to to any value: http://forum.doom9.org/showthread.php?p=1493146#post1493146 .

[Abradoks]

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]

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, for one, could use such a utility right now though. :(

[tebasuna51]

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]

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]

And wath is the delay than show eac3to?

[rapscallion]

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]

"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]

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 !!