eac3to - audio conversion tool

[VAMET]

Dear Friends

First of all, hello to everybody.

I would like to know, what such information means during eac3to process:

Case 1:
[a07] Audio overlaps for 25ms at playtime 0:11:08. <WARNING>

Case 2:
[a08] A remaining delay of -9ms could not be fixed.

Case 3:
[a05] Reading RAW/PCM...
[a05] Swapping endian...
[a05] Remapping channels...
[a05] The original audio track has a constant bit depth of 16 bits.

Case 4:
[v02] [1:13:34] The source file seems to be damaged (sync byte missing). <WARNING>
[a03] [1:13:33] The source file seems to be damaged (sync byte missing). <WARNING>

My question is this, beacuse the original disc BD 1:1 has got errors, scratches or something has gone wrong, while copying to HDD via AnyDVD HD?
Which of above cases are the disc faults and should I copy them once more to get rid off?

I would be glad for reply.
Thank you in advance.
Best regards.

Sincerely

[nautilus7]

Case 1:
In seamless braching BDs main movie is split among several .m2ts files. This looks like this:
http://img361.imageshack.us/img361/359/seamlessbranchingaudioo.png

Case 2:
Audio consists of frames of fixed duration (duration depends on audio format). Delay adjustment is done by adding or removing a number of audio frames. If you multiply (audio frame duration) x (number of frames) you don't always result (in fact you never result) with a value that matches the exact delay needed.

Case 3:
Some modifications are done when converting from one format to another. Nothing to worry about.

Case 4:
What is says. It seems like your data are corrupted. Try re-ripping the disc. This is the only thing you have to worry about. If you still get the same message after ripping the disc again, see if you get any errors (corrupted picture, audio/video de-sync) during playback If you can't find a flaw at these points, then everything should be ok.

[VAMET]

Dear nautilus7

Thank you very much for your reply and great explanation.

In Case 3, I have RAW/PCM it's in Casino Royale, you said:
Some modifications are done when converting from one format to another..
I would like to have no conversion. I used .pcm, should I use another format (extension) in eac3to process, to have unconverted RAW/PCM track from BDMV folder structure? Which one should I enter in eac3to process.

Dolby TrueHD = thd+ac3
DTS HD-MA = dtsma
DTS HD High Resolution = dtshr
Dolby Digital = ac3
DTS = dts
RAW/PCM = ??? I used .pcm and conversion is done, I want uncoverted

I thought that for RAW/PCM I should use .pcm.

Best regards.

Sincerely

[tebasuna51]

I thought that for RAW/PCM I should use .pcm.

Nope, .pcm file is audio uncompressed without headers.
Is better use .wav or .w64 (if is >4GB) with the same uncompressed audio data + header.

edit: .wav if you want use after MkvMerge, .w64 if TsMuxer.

[spork985]

http://img63.imageshack.us/img63/3817/71004044.png

Can someone help me with this? I have installed latest versions of eac3to, mkvtoolnix, haali, arcsoft dts decoder, and c++ redistributable. Operating system is windows xp professional sp3 x86. Am I missing any softwares? I have 3 different blurays, all of them do the same thing. I have used eas3to with around 30ish blurays on my laptop (other computer) and never have had a problem. I'm knowledgeable with video and containers and how stuff works, I just can't figure out why eac3to is crashing. Thanks!

It is crashing regardless of the file... Whether it be
eac3to "C:\file.ac3"
eac3to "C:\file.ts"
eac3to "C:\file.m2ts"

[Inspector.Gadget]

You apparently know that nobody can help you with that source, hence obliterating the path name. Yet you ask for help anyway. Please don't ask other users to violate Rule 6.

[spork985]

You apparently know that nobody can help you with that source, hence obliterating the path name. Yet you ask for help anyway. Please don't ask other users to violate Rule 6.

It happens with ANY FILE, that is just the example I used. I don't like the path name. I own AnyDVD and I own the Bluray disc and I copied it to my hard drive as the one legal backup allowed by US law. It happens wit ANY FILE not just this particular one, I need help on the error not the content.

Perhaps these will help.

http://thumbnails16.imagebam.com/7337/04ccf373362574.gif (http://www.imagebam.com/image/04ccf373362574) http://thumbnails3.imagebam.com/7337/2574ab73362579.gif (http://www.imagebam.com/image/2574ab73362579)

[madshi]

@madshi could you add downmix to stereo (beside the current downmix to dpl2) to your tool...dpl2 is not at all suited for headphones...
_
Is there a standard for mixing 5.1 to stereo? If so, what is the downmixing matrix?

I'm trying to remux some blurays using clown bd and it works most of the time however I have problem with 2 movies, kill bill 2 and sin city.
There are no errors in clown bd however there are 2 warnings in the logs.
Sin City: [a03] This track begins with a non-major frame. <WARNING>
Kill Bill 2: [a03] Caution: The WAV file is bigger than 4GB. <WARNING> [a03] Some WAV readers might not be able to handle this file correctly. <WARNING>
I'm using MPC HC with internal ffmpeg filter to play blurays and both movies work just fine when they are in full bluray structure on my hd but when I try to play them after I've used clown bd there is no audio or video and the screen is just black and the time stays at "00:00:00".
So does anyone have a clue where the remux goes wrong?
I don't know what clown bd does. Does it put all audio tracks into the MKV? In which format? Maybe you should ask the clown bd author for help? My first guess would be that either clown bd does something wrong, or that your DirectShow filter chain needs some tweaking.

Hi,
I've been using this just fine for about a year or so now for converting dts files in an mkv container (from mkvmerge) to ac3 so I can play them back on my WD media player.

I'm having this weird problem now where the file converts from dts to ac3 and right after it's finished, my computer crashes and reboots with a "windows has recovered from a serious error" message. So the conversion is working but it's crashing my comp every time.
That's weird. Try deleting the eac3to folder and redownloading it. Try deleting the two WAV files from the eac3to folder. Does that help? If not, you may have to follow setarip_old's suggestion or even reinstall the OS. I don't think this is an eac3to issue.

I would like to have no conversion. I used .pcm, should I use another format (extension) in eac3to process, to have unconverted RAW/PCM track from BDMV folder structure?
What is your final aim? You can't play .pcm files.

It happens with ANY FILE [...].

Perhaps these will help.
Looks like the FLAC decoders crashes. Don't be confused. When eac3to tries to detect the source's file format, it calls the FLAC decoder to check whether it's a FLAC file. Even if it's not. Well, anyway. My first suggestion would be to delete the eac3to folder and redownload it. My best guess is that the libFlac.dll file is corrupted on your PC.

[b66pak]

Is there a standard for mixing 5.1 to stereo? If so, what is the downmixing matrix?

avisynth script for 5.1 to Stereo NO LFE:
function Dmix5.1Stereo(clip a) # 5.1 Channels L,R,C,NO-LFE,SL,SR -> Stereo
{
flr = GetChannel(a, 1, 2)
fcc = GetChannel(a, 3, 3)
lrc = MixAudio(flr, fcc, 0.3694, 0.2612)
blr = GetChannel(a, 4, 5)
return MixAudio(lrc, blr, 1.0, 0.3694)
}

sox remix line for 5.1 to Stereo NO LFE:
remix -m 1v0.3694,3v0.2612,5v0.3694 2v0.3694,3v0.2612,6v0.3694

avisynth script for 5.1 to Stereo with LFE:
function Dmix5.1Stereo(clip a) # 5.1 Channels L,R,C,LFE,SL,SR -> Stereo
{
flr = GetChannel(a, 1, 2)
fcc = GetChannel(a, 3)
lfe = GetChannel(a, 4)
lfc = MixAudio(fcc, lfe, 0.2071, 0.2071)
mix = MergeChannels(lfc, lfc)
lrc = MixAudio(flr, mix, 0.2929, 1.0)
blr = GetChannel(a, 5, 6)
return MixAudio(lrc, blr, 1.0, 0.2929)
}

sox remix line for 5.1 to Stereo with LFE:
remix -m 1v0.2929,3v0.2071,4v0.2071,5v0.2929 2v0.2929,3v0.2071,4v0.2071,6v0.2929
_

[spork985]

Looks like the FLAC decoders crashes. Don't be confused. When eac3to tries to detect the source's file format, it calls the FLAC decoder to check whether it's a FLAC file. Even if it's not. Well, anyway. My first suggestion would be to delete the eac3to folder and redownload it. My best guess is that the libFlac.dll file is corrupted on your PC.
I redownloaded and tried it and it does the same thing. I hashed all the files with md5 and they match the hashes of the fiels on the computer that does work.

[madshi]

avisynth script for 5.1 to Stereo NO LFE:
sox remix line for 5.1 to Stereo NO LFE:
avisynth script for 5.1 to Stereo with LFE:
sox remix line for 5.1 to Stereo with LFE:
avisynth and sox seem to use the same matrixes? But is this a standard matrix? Or did just somebody say "this sounds ok" and everyone else is copying the parameters?

I redownloaded and tried it and it does the same thing. I hashed all the files with md5 and they match the hashes of the fiels on the computer that does work.
Hmmmm... Maybe your PC doesn't like the libFlac.dll version for whatever reason. You could try replacing the libFlac.dll with an older version. eac3to ships with libFlac 1.2.1. You can get older versions here in the "flac-win" section:

http://sourceforge.net/projects/flac/files/

You need to download the "flac-x.x.x-devel-win.zip" file. That's the only one where the libFlac.dll file is in.

[tebasuna51]

avisynth and sox seem to use the same matrixes? But is this a standard matrix? Or did just somebody say "this sounds ok" and everyone else is copying the parameters?

Is the standard matrix:

fl' = FL + sqr(2)/2 x FC + BL
fr' = FR + 0.7071 x FC + BR

now we have the same acustic power for FC when add the part in FL and FR.

If all channels have a peak at same time we can have overflow reaching values until 2.7071, then we need divide the coefficients by 2.7071:
1/2.7071 = 0.3694
0.7071/2.7071 = 0.2612

fl' = 0.3694 x FL + 0.2612 x FC + 0.3694 x BL
fr' = 0.3694 x FR + 0.2612 x FC + 0.3694 x BR

Now the max value is:
0.3694 + 0.2612 + 0.3694 = 1
After the downmix is recommended a Normalize to avoid low volume.

[madshi]

Is the standard matrix:

fl' = FL + sqr(2)/2 x FC + BL
fr' = FR + 0.7071 x FC + BR

now we have the same acustic power for FC when add the part in FL and FR.
Ok, thanks, that makes a lot of sense.

Now the big question is: How should I make this available in eac3to? Right now "-down2" invokes Dolby ProLogic II. How should I name the options for Dolby ProLogic II and for this other downmixing matrix? Maybe I should use "-down2" for the normal matrix and "-downDpl2" for Dolby ProLogic II? Any other suggestions?

If all channels have a peak at same time we can have overflow reaching values until 2.7071, then we need divide the coefficients by 2.7071:
1/2.7071 = 0.3694
0.7071/2.7071 = 0.2612

fl' = 0.3694 x FL + 0.2612 x FC + 0.3694 x BL
fr' = 0.3694 x FR + 0.2612 x FC + 0.3694 x BR

Now the max value is:
0.3694 + 0.2612 + 0.3694 = 1
After the downmix is recommended a Normalize to avoid low volume.
I'll simply skip the 2.7071 division. If there's overflow/clipping, eac3to will detect that automatically and fix it in a 2nd run, resulting in normalized audio.

[jpsdr]

Hello.
Question about use. If i want to extract the raw video track from an .mts/.m2ts file, to get an untouched raw video data stream, to be able to use it later in authoring SW like Scenarist.
Is the following enough : eac3to file.mts -demux
or, is there others options to add ? (Maybe to extract only video if possible).
If i want the results file(s) to be put on another place than the source file, what the command line should be ?
Thanks.

[tebasuna51]

Now the big question is: How should I make this available in eac3to? Right now "-down2" invokes Dolby ProLogic II. How should I name the options for Dolby ProLogic II and for this other downmixing matrix? Maybe I should use "-down2" for the normal matrix and "-downDpl2" for Dolby ProLogic II? Any other suggestions?
To be backward compatible I suggest use a new parameter "-downstereo" for the new mix and let "-down2" with the Dolby ProLogic II behaviour.

I'll simply skip the 2.7071 division. If there's overflow/clipping, eac3to will detect that automatically and fix it in a 2nd run, resulting in normalized audio.
You are right, maybe you can use the same procedure for dpl2 with:

fl' = FL + 0.7071 x FC + 0.866 x BL + 0.5 x BR
fr' = FR + 0.7071 x FC - 0.5 x BL - 0.866 x BR

[Midzuki]

Let's say, I create an MKV which contains DTS audio, encoded from 16-bit .WAVs. When I demux the MKV with eac3to, this always "patches" the demuxed DTS to "24-bits". What is the point of this behavior? :confused: :confused: :confused:

[SomeJoe]

Let's say, I create an MKV which contains DTS audio, encoded from 16-bit .WAVs. When I demux the MKV with eac3to, this always "patches" the demuxed DTS to "24-bits". What is the point of this behavior? :confused: :confused: :confused:

DTS and Dolby Digital store audio information in the frequency domain. As such, they do not have a "bit depth". When PCM audio (time-domain) is recreated from them during decoding, the audio may in fact decode to values that have data into the 8 additional bits. Those bits may not be accurate with respect to the source, but are accurate with respect to the frequency-domain representation in the compressed audio.

Remember that a 16-bit PCM source is itself an approximation of an analog waveform, and as such is inaccurate because of the 16-bit quantization.

[tebasuna51]

Let's say, I create an MKV which contains DTS audio, encoded from 16-bit .WAVs. When I demux the MKV with eac3to, this always "patches" the demuxed DTS to "24-bits". What is the point of this behavior? :confused: :confused: :confused:

This is because standard DTS isn't lossless. The internal precission of the samples, in frequency domain, is equivalent (not equal) to 24 bits in time domain (no matter what is the source bitdepth).

Then decode to 24 bits is always better than decode to 16 bits, BTW if you want only 16 bits you can add -down16.

edit: I don't see the SomeJoe post with, more or less, the same thing.

[b66pak]

@madshi in my opinion we should have:


-down2stereo
-down2dpl
-down2dpl2
-lfe


the present "-down2" will be equal to "-down2dpl2" for backward compatibility...

also for everybody to be happy i think you should also give full access to the mixer with something like:


-down2custom 1v0.3694,3v0.2612,5v0.3694 2v0.3694,3v0.2612,6v0.3694


this will also solve the present eac3to inability to downmix anything but 5.1...

for example a 3ch (FL,FR,FC) to stereo will luke like:


-down2custom 1v0.5858,3v0.4142 2v0.5858,3v0.4142

_

[tebasuna51]

...
also for everybody to be happy i think you should also give full access to the mixer with something like:


-down2custom 1v0.3694,3v0.2612,5v0.3694 2v0.3694,3v0.2612,6v0.3694

...

That's could be interesting.

A 'Quadro' audio (2/2.0 : FL,FR,BL,BR) can be downmixed to Dpl2 with:

-down2custom 1v1.0000,3v0.8660,4v0.5000 2v1.000,3v-0.5000,4v-0.8660

(the sample is for remember accept the '-' sign)

[spork985]

Hmmmm... Maybe your PC doesn't like the libFlac.dll version for whatever reason. You could try replacing the libFlac.dll with an older version. eac3to ships with libFlac 1.2.1. You can get older versions here in the "flac-win" section:

http://sourceforge.net/projects/flac/files/

You need to download the "flac-x.x.x-devel-win.zip" file. That's the only one where the libFlac.dll file is in.

FLAC 1.1.4 worked, flac 1.2.0 would not. Thank you very much, I have been going crazy about this for a few days now!

[Midzuki]

tebasuna51 wrote:

BTW if you want only 16 bits you can add -down16.

Doesn't work. MediaInfo says the demuxed DTS still "is" 24-bits. Fortunately, there are workarounds — namely, TSMuxer and AVI-Mux GUI.

SomeJoe wrote:

Remember that a 16-bit PCM source is itself an approximation of an analog waveform, and as such is inaccurate because of the 16-bit quantization.

Even a 64-bit @ 768kHz PCM file will be "just an approximation". Remember, there is no such thing as a "perfect" Analog-To-Digital conversion. :)

Anyway, thanks for the accurate answers.
Actually, I was expecting to receive an answer from madshi himself, but it seems he really is "much busier" than I thought. ;)

[Snowknight26]

It doesn't matter what MediaInfo says about lossy tracks regarding bitness. It's not 16-bit and it's not 24-bit.

[raquete]

It doesn't matter what MediaInfo says about lossy tracks regarding bitness. It's not 16-bit and it's not 24-bit.

all right, can be but...
i was sending pm to Midzuki telling that i have few dvds where olaying DTS from video_ts folder my receiver/decoder inform in the display: 24b..this info stay there until the dvd end!
others dvds don't inform and i don't know how to encode in 24b.
i need lessons and explanations about this informations. :confused: how and what program can be used to do that?!?
Porcupine Tree is one of my DVDs that give 24b info, i have others!

@ Midzuki
i could not resisit to post here too but i'm waiting yours infos too. :)

[Midzuki]

It doesn't matter what MediaInfo says about lossy tracks regarding bitness. It's not 16-bit and it's not 24-bit.

I already knew that, since DTS "Digital Surround" is a lossy format. BUT, a binary file-comparison between a 16-bit DTS and a "patched-to-24-bits" DTS shows they are different beasts (on a "frame-by-frame" level, I think).

[Midzuki]

4 raquete: lossy "24-bit DTS" requires 24-bit sources
@ 96kHz or higher.

Also, take a look at your PM inbox. :rolleyes:

[raquete]

4 raquete: lossy "24-bit DTS" requires 24-bit sources
@ 96kHz or higher.

Also, take a look at your PM inbox. :rolleyes:
great, i got the lesson Mid.

thanks so much! :)

[tebasuna51]

One more time, AC3/DTS standard (lossy) don't have bitdepth. The parameter quality for this files is the bitrate, forget the bitdepth.

Only DTS have a value in the header than say the bitdepth of the source wav files used to create the dts. And Surcode put always 24 bits also when source wavs is 16 bits, then is a useless info. I say to madshi don't put this info (sometimes wrong) with dts files because confuse the users, like you can see.

AC3/DTS files must be decoded to, at least, 24 bit to obtain the best aproach to the source, no matter the original bitdepth.

[raquete]

ok tebasuna,
i was curious based on some informations from my decoder like i posted here:
http://forum.doom9.org/showthread.php?p=1386384#post1386384

to tell the true i was not only confused, i was really lost with that infos. :)

thanks so much!

[Midzuki]

Only DTS have a value in the header than say the bitdepth of the source wav files used to create the dts. And Surcode put always 24 bits also when source wavs is 16 bits, then is a useless info.

AC3/DTS files must be decoded to, at least, 24 bit to obtain the best aproach to the source, no matter the original bitdepth.

So, let's just hope there don't exist too many
" 'dumb' DTS decoders ". :)

[madshi]

To be backward compatible I suggest use a new parameter "-downstereo" for the new mix and let "-down2" with the Dolby ProLogic II behaviour.
in my opinion we should have:


-down2stereo
-down2dpl
-down2dpl2
-lfe


the present "-down2" will be equal to "-down2dpl2" for backward compatibility...
Hmmmmm... I think I like b66pak's idea of appending a qualifier to "-down2", however, I don't really like the term "stereo" here, especially because "-down2stereo" is kind of double information. Do you guys have a good idea how to term the "straight" downmixing matrix? Maybe "-down2straight"? Or "-down2simple"? Or something else?

also for everybody to be happy i think you should also give full access to the mixer
That's could be interesting.
I'll think about that. Don't know right now how difficult it would be to implement...

You are right, maybe you can use the same procedure for dpl2 with:

fl' = FL + 0.7071 x FC + 0.866 x BL + 0.5 x BR
fr' = FR + 0.7071 x FC - 0.5 x BL - 0.866 x BR
Yeah, good thinking, will do that!

If i want to extract the raw video track from an .mts/.m2ts file, to get an untouched raw video data stream, to be able to use it later in authoring SW like Scenarist.
Is the following enough : eac3to file.mts -demux
or, is there others options to add ? (Maybe to extract only video if possible).
If i want the results file(s) to be put on another place than the source file, what the command line should be ?
"-demux" is ok, but it will demux all video, audio and subtitle tracks. You can do "eac3to file.mts c:\whatever\video.h264", if the video track is h264 (use .mpeg2 or .vc1 for other video types). That will demux only the video track, and this way you can also choose the target file path.

Let's say, I create an MKV which contains DTS audio, encoded from 16-bit .WAVs. When I demux the MKV with eac3to, this always "patches" the demuxed DTS to "24-bits". What is the point of this behavior? :confused: :confused: :confused:
Two facts:

(1) Each DTS frame contains a header field which informs us about whether the original source was 16bit or 24bit. In theory this header field is only for our information and should not be used during decoding (except for DTS-HD MA where due to being lossless the final output should have the same bitdepth as the original source). Because this header field is only for our information, and should not really be used during decoding, it should not *harm* to modify it (again: the exception is DTS-HD MA).

(2) There are some decoders which always limit their output to the bitdepth of the original audio source - even for lossy DTS. That means, if the header field says that the original audio data was only 16bit, then the decoder only outputs 16bit, too. This is *BAD* because lossy formats decode always to higher than 16bit (doesn't matter which bitdepth the original source had), and if the decoder downconverts that to 16bit, we lose quality.

Now combine the two facts above and you will see that what eac3to does, shouldn't ever harm, but it will sometimes (with stupid decoders) help to get higher quality playback.

Got it?

FLAC 1.1.4 worked, flac 1.2.0 would not. Thank you very much, I have been going crazy about this for a few days now!
<shameless self promotion> I wish all software had such exact crash reporting as eac3to has. I think with most other software you'd be stuck and never know why it didn't work on your PC. </shameless self promotion>

[Midzuki]

Originally posted by madshi
Got it?

Yes, sir:

1) there is a certain DTS encoder that is not, let's say,
"as smart as it should be" :D ;

2) In the "dictionary of advertising and marketing" :) ,
"24-bit DTS" is a misnomer for *96kHz* :devil:

:thanks:

[madshi]

there is a certain DTS-HD encoder that is not, let's say, "as smart as it should be" :D ;
Not sure what you're saying here...

In the "dictionary of advertising and marketing" :) ,
"24-bit DTS" is a misnomer for *96kHz* :devil:
From a technical point of view 24bit and 96kHz have *nothing* to do with each other. Totally separate things. Don't let yourself be confused by the DTS marketing department...

[tebasuna51]

Hmmmmm... I think I like b66pak's idea of appending a qualifier to "-down2", however, I don't really like the term "stereo" here, especially because "-down2stereo" is kind of double information. Do you guys have a good idea how to term the "straight" downmixing matrix? Maybe "-down2straight"? Or "-down2simple"? Or something else?

No problem, my vote for "-down2simple", "straight" is more dificult to spell, for no english people, and can cause typos.

Related with this kind of downmix uses, maybe you can add support for Lame.exe encoder (like NeroAacEnc or with dll).
Many people want mp3 output for compatibility issues (device or container avi related).

We can use parameters like -192 to CBR encode (Lame -b 192). Valid values:
[32,40,48,56,64,80,]96,112,128,160,192,224,256,320
Maybe we can forget low values, not recommended at all.

For VBR we can use the same parameter than for NeroAacEnc, with this conversion:
-quality=1.0 -> -b 320 (high possible with Lame
-quality=0.9.. -> -V 0
-quality=0.8.. -> -V 1
-quality=0.7.. -> -V 2
-quality=0.6.. -> -V 3
-quality=0.5.. -> -V 4 (default VBR)
-quality=0.4.. -> -V 5
-quality=0.3.. -> -V 6
-quality=0.2.. -> -V 7
-quality=0.1.. -> -V 8
-quality=0.0.. -> -V 9
To avoid the strange behaviour of Lame parameter (high V less quality)

I know this can be do with stdout.wav | Lame ... (like my GUI UsEac3to do) but other GUI's (Clown_BD, HdBrStreamExtractor-MeGUI, ...) can't convert to mp3.

[Midzuki]

Originally Posted by Midzuki
there is a certain DTS-HD encoder that is not, let's say, "as smart as it should be" :D
Not sure what you're saying here...

tebasuna51 had written:

And Surcode put always 24 bits also when source wavs is 16 bits,

Conclusion: I was not talking about Surcode.

[yesgrey]

Hmmmmm... I think I like b66pak's idea of appending a qualifier to "-down2", however, I don't really like the term "stereo" here, especially because "-down2stereo" is kind of double information. Do you guys have a good idea how to term the "straight" downmixing matrix? Maybe "-down2straight"? Or "-down2simple"? Or something else?

Here is my suggestion...
Changing the "2" to "To", like this:

-downTo6 (previously down6)
-downTo2
-downTodpl
-downTodpl2
-downTo "some specification to decide" (custom mix)

This would have the benefit of not using the same command for downmixing channels as for reducing the bit depth (down16...23).
Another idea is using "mixTo" instead of "downTo".

[yesgrey]

also for everybody to be happy i think you should also give full access to the mixer with something like:

-down2custom 1v0.3694,3v0.2612,5v0.3694 2v0.3694,3v0.2612,6v0.3694


I also like the full access to the mixer idea, and I would like if you would also add another option in case you decide to go for it.
Using the custom mix only for scanning the file and then output the peak value in dB.
Why I need this?
I always use the hardware mixer of my soundcard to mix from 5.1 to 4.0, and with that info I could set the right values to avoid any clipping without lowering the volume too much. Currently I have it set to safe values, which give me low volume in a lot of movies...

[Midzuki]

Example:

--- create a stereo DTS file @ 48kHz @ 384kbps;

--- run "eac3to input.dts output.wav";

--- play "output.wav" in your favorite player;

*giggles*

OTOH,

"ffmpeg -i input.dts output.wav" works properly.

[nurbs]

Try "eac3to input.dts output.wav -simple"

[b66pak]

i am ok with "-down2simple" too...
_

[Midzuki]

Try "eac3to input.dts output.wav -simple"

Doesn't work.

[VAMET]

Dear Friends

eac3to is a very good application to demux original Blu-Ray Disc.
I used it very often and then my movies I watch on Dune HD Base 3.0.

Is there any possibility to demux with eac3to, but with such flag, option, to have subtitles ready for remux with tsMuxeR with final result: specified subtitles always on during movie playback?

I have audio English and Polish subtitles, but when I watch such movie after eac3to and tsMuxeR process, I need to choose manualy my Polish subtitles.

There is no option to automaticaly choose specified subtitles on Dune HD Base 3.0. So I would like to create BDMV folder structure, but with subtitles, which are always on from the beginning of the movie.

Is it possible?

I will be glad for any help.
Thank you.
Best regards.

Sincerely

[setarip_old]

@VAMET

Hi!

Try "multiAVCHD"...

[Adub]

Okay, I seem to have encountered a bug:

Source: Inglorious Bastards Blu-ray, US.
Problem:
Eac3to appears to hang, or "deadlock", with no CPU usage after finding clipping in the DTS-MA audio track that is being converted.

However, it only does this when the "resampleto48000" commandline flag is present. In addition, eac3to was complaining about the resampling and told me to use r8brain, if that is any help.

Example:
Microsoft Windows [Version 6.1.7600]
Copyright (c) 2009 Microsoft Corporation. All rights reserved.

C:\Users\Adub>"C:\multiAVCHD\tools\eac3to\eac3to.exe" "D:\Multimedia\BluTemp\ING
_BASTERDS\BDMV\STREAM\00010.m2ts" 3: "c:\00010.m2ts.t2.ac3" -resampleTo48000 -3
84 -r8brain -progressnumbers
analyze: 1%
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M2TS, 1 video track, 3 audio tracks, 5 subtitle tracks, 2:33:00, 24p /1.001
1: Chapters, 34 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: DTS, Spanish, 5.1 channels, 24 bits, 768kbps, 48khz
5: DTS, French, 5.1 channels, 24 bits, 768kbps, 48khz
6: Subtitle (PGS), English
7: Subtitle (PGS), Spanish
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), French
a03 The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
a03 The libav DTS decoder doesn't decode the full DTS-HD information.
a03 Extracting audio track number 3...
a03 Extracting DTS core...
a03 Decoding with libav/ffmpeg...
a03 Remapping channels...
a03 Resampling to 48khz...
a03 Encoding AC3 <384kbps> with libAften...
a03 Creating file "c:\00010.m2ts.t2.ac3"...
process: 1%
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a03 Clipping detected, a 2nd pass will be necessary.


If there is anything that you need from me, please, let me know.

[Jynx980]

I can't get eac3to to recognize the Monsters Inc. Bonus disc on Blu Ray. I get "HD DVD / Blu-Ray disc structure not found." from either a rip or actual disc in drive. It's the regular commercial disc, plays fine, just cant get any info on it with eac3to. This is the first time I have had a problem with this. Suggestions?

[NanoBot]

Are you sure that the bonus disc is a Bluray ? Amazon.com tells me "DVD Of Film Plus Original DVD Bonus", so the bonus disk might be a DVD, which of course can't be prcessed with eac3to.

[JnZ]

I'm wondering if would be posible support user defined FPS conversion.
For example I have audio from 23.976fps version, and video 25.050fps, so I need use something like this commandline:
eac3to source.mp3 dest.wav -23.976 -changeTo25.050

It will be nice to support reading these user defined float values.
:thanks:

Anyway thanks for superrior software.

[Jynx980]

It's Blu-Ray. File Size is about 21GB. The one on Amazon has 4 discs, two blu-ray and two dvd. It has the same structure as other blu-rays; BMDV, CERTIFICATE, PLAYLIST, STREAM... Not sure how it's different than any other one.

[tebasuna51]

@madshi, feature request if is easy:

When a gain is applied by request (+3dB) eac3to send the message:
Applying +3dB gain...

Is possible send a equivalent message when a 2nd pass for 'Clipping detected', or -normalize parameter, apply a gain?
The user then know the amount of gain needed.
The value can be in dB, % or coefficient, the more easy way for you.

Thanks.

[liquidator87]

When using a command line like this:

eac3to.exe source dest -slowdown -295ms

is the delay applied before or after the framerate conversion?

Looking at the program output it seems before, just to be sure :)