eac3to - audio conversion tool

[Overdrive80]

Hi, I am converting audio between NTSC and FILM and I noticed that the final result, in duration, is distinctive to the expected one.

For example, starting with an audio of the following characteristics:

Duration: 24 min 37 sec 684 ms
Origin: NTSC
Samplerate: 96000 Hz
Bitdepth: 32 float

The conversion factor gives me a samplerate change of 95904.096.

When truncating this amount, in audition change the sampling rate to 95904 and the result is an audio with the following duration: 24 min 39 sec 163 ms

If I use the amount 95905, I get: 24 min 39 sec 147 ms which is approximated to that obtained with eac3to being 146 ms.

I can understand that we are using a multiplo of 5, for reasons that I do not understand, since I have made a conversion table, and this is the only conversion along with PAL to FILM, which disagree with the operations made from another editor. In the latter case, the conversion factor is 92067,932, which must be rounded to 92068 to obtain the expected value. This rounding I can understand but the previous one seems to me that rounding to 95905 is too much.

https://i.postimg.cc/j5CKTRBt/Captura-de-pantalla-2019-04-18-19-09-17.png

[tebasuna51]

Only can confirm your data:

The FILM -> NTSC conversion, with %_.wav -24.000 -changeTo23.976, over 1477.684 seconds wav output 1479.147 seconds when must be 1479.162
There are a error of 15 ms, all other conversions work fine.

You can use -r8brain parameter to obtain a exact output instead use the default ssrc routines.

[rco133]

Hi.

I have just installed a new PC, and am quite puzzled by how eac3to is reacting.

For some reason eac3to is now 2-3 times slower at demuxing than it is on my old system.

Demuxing a bluray on old system takes for example 9 minutes.

Demuxing the same bluray on the new system takes 20-30 minutes.

I am using the same command line on both systems, and have tried to use both HDD and SSD on both systems. I have not been able to find anything that makes the new system demux as fast as the new one though.

I have simply copied the eac3to folder from the old system to the new one. so there shouldn't be any changes in the installation.

I have also tried to do the demux through MeGUI, but I think it also uses eac3to, and is just as slow.

Old system is an 8 core i7 6900k and new one is 16 core i9 9960x.

Any ideas as to what might cause this slowdown?

Thanks in advance.

rco133

[Overdrive80]

Ok, thanks Tebasuna51 for the workaround.

[Overdrive80]

Hi again, I am trying use this line command in batch file: "-24.000 -changeTo23.976 -down32 -normalize -resampleTo48000 -r8brain", but process stop (blocked or paused) when *.pass1 file reach 1 GB.

Why this limitation?

Source file:

Origin: NTSC
Samplerate: 96000 Hz
Bitdepth: 32 float
Size: 3GB


Batch file:

@echo off
Title NTSC to FILM
COLOR A8

set "eac3to=C:\Portables\MeGUI\tools\eac3to\eac3to.exe" rem Put you directory for eac3to

md %~dp0Completed

for %%@ in (*.wav) do (

"%eac3to%" "%%@" "%%~dpn@_FILM.wav" -24.000 -changeTo23.976 -down32 -normalize -resampleTo48000 -r8brain -log=NUL
move "%%@" "%~dp0Completed" /Y
)

pause&exit

[Sparktank]

Aside from diagnostics working with blurays,etc.
Is there any advantage to using eac3to over FFMPEG for transcoding BluRay audio (THD/DTSHDMA/etc) to something else?

I notice that FFMPEG won't do stats about gaps like eac3to does.
But, FFMPEG is more friendly for updates.

[tebasuna51]

Hi again, I am trying use this line command in batch file: "-24.000 -changeTo23.976 -down32 -normalize -resampleTo48000 -r8brain", but process stop (blocked or paused) when *.pass1 file reach 1 GB.

I don't know what is the problem.

The process work without -normalize?

Maybe is not necesary at all or can be do in a second separate pass, now you force to do the resample for -changeTo23.976 and the -resampleTo48000 two times, in pass1 and in pass2.

[Bandits]

Hi.

I have just installed a new PC, and am quite puzzled by how eac3to is reacting.

For some reason eac3to is now 2-3 times slower at demuxing than it is on my old system.

Demuxing a bluray on old system takes for example 9 minutes.

Demuxing the same bluray on the new system takes 20-30 minutes.

I am using the same command line on both systems, and have tried to use both HDD and SSD on both systems. I have not been able to find anything that makes the new system demux as fast as the new one though.

I have simply copied the eac3to folder from the old system to the new one. so there shouldn't be any changes in the installation.

I have also tried to do the demux through MeGUI, but I think it also uses eac3to, and is just as slow.

Old system is an 8 core i7 6900k and new one is 16 core i9 9960x.

Any ideas as to what might cause this slowdown?

Thanks in advance.

rco133

Are you demuxing straight from the Blu-Ray? If you are, are you using the same Blu-Ray drive?

Riplock is a drive function that can slow your reading of video files. Ensure your drive does not use Riplock.

[tebasuna51]

@Sparktank
- Decode audio from THD/DTSHDMA/etc can be do with ffmpeg, more actualized than eac3to decoder dll's.

- Also ffmpeg have the best free AC3 encoder or the output can be piped to other free encoders like qaac.

- But I don't know how ffmpeg can manage audio from multiple m2ts sources like can be present in BD's.
Like you say eac3to can manage gaps/overlaps in the best know way.
By the moment I extract the audio tracks from BD's with eac3to (gui's UsEac3to or MeGUI) and recode after to desired format with ffmpeg or qaac (gui's UsEac3to, MeGUI or BeHappy).

[Overdrive80]

I don't know what is the problem.

The process work without -normalize?

Maybe is not necesary at all or can be do in a second separate pass, now you force to do the resample for -changeTo23.976 and the -resampleTo48000 two times, in pass1 and in pass2.

Tested separating process and same result. New script:

@echo off
Title NTSC to FILM
COLOR A8

IF NOT EXIST "%~dp0Completados" (md "%~dp0Completados")

set "eac3to=C:\Users\Isra-PC\Desktop\eac3to\eac3to.exe" rem Put you directory for eac3to

for %%@ in (*.wav) do (
::First pass, conversion
"%eac3to%" "%%@" "%temp%\temp.wav" -24.000 -changeTo23.976 -resampleTo48000 -down32 -r8brain -log=NUL

::Two pass, normalization
"%eac3to%" "%temp%\temp.wav" "%%~dpn@_FILM.wav" -normalize -down32 -log=NUL

::Remove temps files
del "%temp%\temp.wav"

::Move originals to completed
move "%%@" "%~dp0Completados"
)

pause&exit

Will try ffmpeg:

"%ffmpeg%" -hide_banner -nostdin -y -channel_layout stereo -i "%input%" -filter:a "atempo=0.9990009" -c:a pcm_f32le -ar 48000 -vn -sn "%temp%\temp.wav"

[an3k]

If anyone wants to get SurCode DTS Encoder working on Windows 10 this is how it goes:

1) Get the "SurCode DVD-DTS Professional" Setup, you can buy it at https://www.minnetonkaaudioshop.com/epages/MASIShop.sf/seccaf11703b2/?ObjectPath=/Shops/MASIShop/Products/3001-00106-000
2) Get "UniExtract" from https://www.legroom.net/software/uniextract (you want the "UniExtract Binary Archive" which is the "portable" version)
3) Start UniExtract.exe and select the Data1.cab from the "SurCode DVD-DTS Professional" zip. - It will extract the contents into a directory called "data1".
4) Create a directory called "Minnetonka Audio Software" inside "C:\Program Files (x86)"
5) Copy or Move directory "data1" into "C:\Program Files (x86)\Minnetonka Audio Software"
6) Rename directory "C:\Program Files (x86)\Minnetonka Audio Software\data1" to "C:\Program Files (x86)\Minnetonka Audio Software\SurCode DVD-DTS"
7) Create a new text document on your Desktop or somewhere else and rename the file "install.reg". Make sure it is not called "install.reg.txt" (show file-extensions). It should have a green-ish icon in front of the filename.
8) Copy&Paste the text from the quote-block below into that "install.reg" file.
9) Double-click the "install.reg" file and let Windows import the Registry settings.
10) run eac3to.exe -test and you should have SurCode support.
11) Activate your copy of "SurCode DVD-DTS Professional".

Windows Registry Editor Version 5.00

[HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion\App Paths\surcodedvd.exe]
"Path"="C:\\Program Files (x86)\\Minnetonka Audio Software\\SurCode DVD-DTS"
@="C:\\Program Files (x86)\\Minnetonka Audio Software\\SurCode DVD-DTS\\surcodedvd.exe"

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Minnetonka Audio Software]

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Minnetonka Audio Software\SurCode DVD-DTS]
"Home"="C:\\Program Files (x86)\\Minnetonka Audio Software\\SurCode DVD-DTS"

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Minnetonka Audio Software\SurCode DVD-DTS\1.0.10]


Q: Why?
A: Because the Installer used by Minnetonka for the SurCode DVD-DTS Encoder is so old it doesn't run on Windows 10 (even though it just copies files from A to B and writes some Registry values, nothing fancy).

Q: Is this piracy?
A: No. We just eliminate that crappy Installer. Activation of the product is done by the surcodedvd.exe itself.

Q: But I can use the encoder already without activating it. So this is piracy?
A: Congratulations, you have found the trial period :)

[Overdrive80]

Thanks for you tip, for complete it:

@echo off
Title Install SurCode
Color E0

goto check_Permissions

:check_Permissions
echo Administrative permissions required. Detecting permissions...

net session >nul 2>&1
if %errorLevel% == 0 (
goto main
) else (
ECHO.
ECHO. Failure: Current permissions inadequate. & ECHO.
ECHO. Press key for trying again.
pause>nul&exit
)


:main
ECHO The process has begin...

::Set paths to your files.
set "uniExtract=%~dp0uniextract161_noinst\UniExtract.exe"
set "dataCab=%~dp0Surcode DVD Pro DTS Encoder v1.0.29\Data1.cab"

::Extract the content
IF EXIST "%~dp0SurCode DVD-DTS" (
DEL "%~dp0SurCode DVD-DTS" /Q /F
)

"%UniExtract%" "%dataCab%" "%~dp0SurCode DVD-DTS"

::Create folder system
IF NOT EXIST "%ProgramFiles(x86)%\Minnetonka Audio Software" md "%ProgramFiles(x86)%\Minnetonka Audio Software"

::Move folder extracted to system
robocopy "%~dp0SurCode DVD-DTS" "%ProgramFiles(x86)%\Minnetonka Audio Software\SurCode DVD-DTS" /E /njh /njs /ndl /nc /ns >nul

::Install key registry
reg add "HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion\App Paths\surcodedvd.exe" /ve /t REG_SZ /d "C:\Program Files (x86)\Minnetonka Audio Software\SurCode DVD-DTS\surcodedvd.exe" /f >nul
reg add "HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion\App Paths\surcodedvd.exe" /v "Path" /t REG_SZ /d "C:\Program Files (x86)\Minnetonka Audio Software\SurCode DVD-DTS" /f >nul
reg add "HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Minnetonka Audio Software\SurCode DVD-DTS" /v "Home" /t REG_SZ /d "C:\\Program Files (x86)\\Minnetonka Audio Software\\SurCode DVD-DTS" /f >nul
reg add "HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Minnetonka Audio Software\SurCode DVD-DTS\1.0.10" /ve /t REG_SZ /f >nul

cls&ECHO The process has ended.
pause>nul&exit

[rco133]

Are you demuxing straight from the Blu-Ray? If you are, are you using the same Blu-Ray drive?

Riplock is a drive function that can slow your reading of video files. Ensure your drive does not use Riplock.

No, I am demuxing from SSD drive.

What I have found is that if I use megui and only select the tracks that I want, instead of using -demux directly with eac3to it goes much faster. Pretty much 4-5 times faster than if I extract all tracks.

If I use tsmuxer to extract all tracks, it takes half the time that it takes with eac3to.

I have no explanation, other then it must be something about how eac3to writes stuff to disk, which gives a big negative impact on Windows 10 compared to Windows 7. Other wise I have no explanation for what I am seing.

Anyway, I will just use megui and select the few tracks that I need, which will make the process much shorter, eventhough it also uses eac3to.

rco133

[an3k]

Anyway, I will just use megui and select the few tracks that I need, which will make the process much shorter, eventhough it also uses eac3to.

rco133

When MeGUI is using eac3to to demux the audio tracks you can open the task manager, go to "Processes" (the first tab), right-click onto the column header (Name, Status, CPU, etc.) and enable "command line". Go to the eac3to process launched by MeGUI and check the actual command line eac3to is invoked with. Maybe this gives a hint.

[Wildfire]

Perhaps it's just me, but I seem to be unable to find a way to automate converting DTS-HD audio tracks from MKV files (or better said, replacing the DTS-HD tracks with the core DTS tracks).

Currently, I'm using gMKVExtractGUI to extract the DTS-HD audio track from each file, then I extract the core DTS track using eac3to and finally I use MKVToolnix to replace the DTS-HD track with the core DTS track.

For dozens of files, that's a lot of work. Surely there's a way to automate the process, but for the moment I'm at a loss how to automate the process. I'm sure using eac3to and MKVToolnix there must be some way to script the entire process...

[sneaker_ger]

mkvmerge has a --reduce-to-core option.
https://mkvtoolnix.download/doc/mkvmerge.html

or ffmpeg dca_core bitstream filter
https://ffmpeg.org/ffmpeg-bitstream-filters.html#dca_005fcore

[Wildfire]

mkvmerge has a --reduce-to-core option.
https://mkvtoolnix.download/doc/mkvmerge.html

Thanks for the tip, now I'll have to figure out how to automate the process. Unfortunately, it's not as easy as "mkvmerge --reduce-to-core *.mkv *-DTS.mkv" or something like that :P

[Vicio]

Hi Guys, I'm having a question regarding audio conversion 7.1 (DTS MA-HD / TrueHD) to 5.1 AC3.
I was reading this tutorial: https://bodoicuho.ucoz.ru/ and I noticed that the conversion process for AC3 audios is somewhat laborious, needing to convert it to .w64 with eac3to and then using Sound Forge to get the RMS value from the central band and finally convert it for Dolby Digital.

However this method seems to me a bit old and searching here I came across this other method (which seems to me simpler and more current) of @tebasuna51: https://forum.doom9.org/showthread.php?t=175044

https://forum.doom9.org/showpost.php?p=1850153&postcount=14689

Where it recommends using the Megui for this purpose, the doubt is even with the Megui still need to convert the audio to .w64 and do the Sound Forge step to get the RMS value?

Below I'll be leaving two samples of two audios 7.1.
https://send.firefox.com/download/bff23123430354f6/#311orhBTI8w70DP2CMUTCA

I hope I have managed to express myself and apologize for my English, I am using a translator.

[tebasuna51]

Where it recommends using the Megui for this purpose, the doubt is even with the Megui still need to convert the audio to .w64 and do the Sound Forge step to get the RMS value?

MeGUI, or other free software, can't encode to Dolby Digital.
The best AC3 free encoder (ffmpeg) used with MeGUI or BeHappy is not a Dolby Digital encoder.

The recommended Dialog Normalization and Dynamic Range Compression with this encoder is the default (-31 dB and None), then is not necesary know the RMS value of center channel in Sound Forge (or with other method).

Take in mind than you are recoding a previous track, not encoding material recorded by you. Many encoder options in commercial encoders (filters, phase shift, etc.) are now unnecesary.

[Vicio]

MeGUI, or other free software, can't encode to Dolby Digital.
The best AC3 free encoder (ffmpeg) used with MeGUI or BeHappy is not a Dolby Digital encoder.
Yes, I'm aware of this, Megui is just a GUI, I just wanted to ask the question straight to the point without dwelling on technical terms. :)

The recommended Dialog Normalization and Dynamic Range Compression with this encoder is the default (-31 dB and None), then is not necesary know the RMS value of center channel in Sound Forge (or with other method).
Okay, that's what I wanted to hear. :goodpost:

Take in mind than you are recoding a previous track, not encoding material recorded by you. Many encoder options in commercial encoders (filters, phase shift, etc.) are now unnecesary.
Understood, noted here.
And thank you very much for the help, I read more half of the pages of this topic and I observed that you are a helpful person.
Greetings from Brazil.

[zeropc]

Guys, can eac3to no longer slowdown videos in h265?
I get the following error message

C:\Users\zeropc>"\\Mac\Home\Desktop\eac3to more gui\eac3to\eac3to.exe" "X:\evil2mkv"
1: "\\Mac\Home\Desktop\dead2\hdr.h264" 2: "\\Mac\Home\Desktop\dead2\dv.h264" -slowdown
MKV, 2 video tracks, 2 audio tracks, 1 subtitle track, 1:24:32, 24p
1: h265/HEVC, 2160p24 (16:9), 10 bits
2: h265/HEVC, 1080p24 (16:9), 10 bits - Dolby Vision Enhancement Layer
3: DTS Master Audio, English, 1.0 channels, 16 bits, 48kHz
(core: DTS, 1.0 channels, 768kbps, 48kHz)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
5: Subtitle (PGS), English
v01 The video framerate is correct, but rather unusual.
v02 The video framerate is correct, but rather unusual.
This video conversion is not supported.

[tebasuna51]

eac3to can't edit h265 (slowdown), but you want also recode to h264, for that the message is:
"This video conversion is not supported."

[zeropc]

eac3to can't edit h265 (slowdown), but you want also recode to h264, for that the message is:
"This video conversion is not supported."

AHHH!!!

Would be sweet if madshi can add this option for h265 :)

[SeeMoreDigital]

eac3to can't edit h265 (slowdown), but you want also recode to h264, for that the message is:
"This video conversion is not supported."Out of interest... Has anybody created a tool that can change some of the attributes of h265 elementary streams?

[nevcairiel]

Out of interest... Has anybody created a tool that can change some of the attributes of h265 elementary streams?

Depending on what partciular properties you are talking about, the h265_metadata bitstream filter in ffmpeg can change a bunch of them.

[sneaker_ger]

* hevc_metadata (https://ffmpeg.org/ffmpeg-bitstream-filters.html#hevc_005fmetadata)

[SeeMoreDigital]

Marvellous...

Thank-you Mr sneaker or is it Mr ger ;)

[nevcairiel]

How misleading, the file was called h265_metadata_bsf.c :D

[an3k]

I noticed a bug with eac3to v3.34 and Surcode DTS Encoder. When you convert a multichannel AC3 or WAV (for FLAC, etc.) to DTS it works fine. If you however want to convert a singlechannel (1.0 aka. mono) to DTS eac3to - or better Surcode throws an error. The very same file eac3to doesn't handle correctly is perfectly working if used directly inside the Surcode DTS Encoder software.

The error is:
C:\Program Files (x86)\MeGUI\tools\eac3to>eac3to.exe "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.wav" "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.dts"
WAV, 1.0 channels, 2:32:13, 24 bits, 1152kbps, 48kHz
Reading WAV...
Writing WAVs...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.C.wav"...
The original audio track has a constant bit depth of 24 bits.
Encoding DTS <768kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.29.0.
Surcode says/asks: "Invalid Wave File D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.dts.".
Surcode says/asks: "At least one valid source file must be specified to encode.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

Somehow eac3to pushes the destination filename to Surcode as the source filename.

Another bug is that if your source file is named eg. "foobar.C.wav" and you want to convert it to AC3 or DTS eac3to wants to convert that file to "foobar.C.wav" to make it compatible with the format AC3 or DTS expects. However, it throws an error because it just overwrote the source audio file with the temporary one :)

[tebasuna51]

@an3k
Encode a monowav to standard DTS 768 Kb/s with Surcode is usseless, you obtain a file with the same size and less quality than the PCM 16 bits (WAV).

eac3to don't need manage that conversion.

I can't reproduce the name problem encoding to AC3.

[an3k]

@an3k
Encode a monowav to standard DTS 768 Kb/s with Surcode is usseless, you obtain a file with the same size and less quality than the PCM 16 bits (WAV).

eac3to don't need manage that conversion.

I can't reproduce the name problem encoding to AC3.

The sense may be non-existend. I agree with you on that point but non-the-less it is a bug in eac3to.

Using the same file directly in Surcode works (even though the resulting DTS file is 5.1 channels). Because of that I went and created a DTS-HD MA using the DTS Master Audio Suite which created a mono file.

Yeah, if the target is AC3 it works without issues.

[an3k]

I noticed another weird behavior and don't know if eac3to is doing something wrong or me not understanding it.

eac3to v3.34
command line: eac3to.exe "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.ac3" "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.wavs"
------------------------------------------------------------------------------
AC3, 5.1 channels, 2:32:13, 448kbps, 48kHz
Decoding with libav/ffmpeg...
Reducing depth from 64 to 24 bits...
Writing WAVs...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.L.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.SL.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.C.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.SR.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.LFE.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.R.wav"...
Clipping detected, a 2nd pass will be necessary. <WARNING>
Starting 2nd pass...
Decoding with libav/ffmpeg...
Reducing depth from 64 to 24 bits...
Writing WAVs...
Applying -0.55dB gain...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.L.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.LFE.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.R.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.C.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.SL.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.SR.wav"...
eac3to processing took 5 minutes, 17 seconds.
Done.

eac3to says Reducing depth from 64 to 24 bits... but MediaInfo says the ac3 is 16 bits. The second audio stream is TrueHD (MediaInfo says it's 16 bits too) and if I transcode that into individual wavs eac3to correctly use 16 bits.

[sneaker_ger]

AC3 and other lossy codecs don't have any inherent bitdepth despite what MediaInfo might suggest. The decoder can decode at 64 bits but it would be to wasteful to store 64 bit PCM/WAV so eac3to reduces the depth.

[an3k]

Oh, I see. Thank you for the information.

What about Dolby TrueHD? MediaInfo says 24 bits but eac3to says 16 bits and is reducing from 24 to 16 bits.

I want to keep as much quality as possible and the size of the intermediate wav files isn't a thing I have to worry about since I'm going to encode to a different format.

[sneaker_ger]

Lossless codecs do have "inherent" bitdepths. In your case it was encoded as 24 bits. But eac3to detects that throughout the complete file all of the respective 8 minor bits are 0s. So it decides to encode to 16 bit instead. This has absolutely no influence on quality.

[MeteorRain]

Is there any way to stop eac3to from adding DELAY to output file name?

I'm writing script to batch extract eac3 from files and convert them to wav for further processing.

eac3to x.mkv x.wav tells me This audio conversion is not supported..

eac3to x.mkv x.eac3 then eac3to x.eac3 x.wav works.
However eac3to will add DELAY XYZ ms to the file name so it's kinda hard to script that into a makefile.
Since this delay is usually less than 1 frame, I really don't care if it's applied correctly.

[sneaker_ger]

I would recommend using ffmpeg. It seems better at reading from mkv and also has a more feature-complete eac3 decoder IIRC.
ffmpeg -i "x.mkv" -map 0:a:0 "x.wav"

It can correct delays, too. Hope this is correct:
ffmpeg -i "x.mkv" -map 0:a:0 -af aresample=first_pts=0 "x.wav"

[MeteorRain]

Thanks. I'm aware of ffmpeg but still prefer eac3to if possible.

[thecoreyburton]

I often use eac3to to demux BD streams. In the past I've noted a common delay of just above 21ms (there's a remaining delay of less than 1ms) when dealing with DTSHD track transcoding (eg. DTSHD > FLAC). At the time, I read up on this a fair bit and did some of my own testing, eventually concluding that a simple "-21ms" parameter would be ideal when dealing with DTSHD in this way.

There's plenty of discussion about this, but a lot of it seems related to specific issues and I can't find anything conclusive about whether or not this is the right way to approach this. I'm not sure if there is a "correct" way to do this (for instance a trim based on samples / frames to make it align perfectly or some internal function that does this), or if I should be touch this value at all. I believe it's an encoder delay (and not a decoder delay, but I could be wrong) and so it stands to reason that if this is true, perhaps some BDs have been authored with compensation for this delay in mind.

I was wondering if the way I'm handling it is the correct procedure, or is there a better way?

[tebasuna51]

It is know than encoders add initial silence samples to initialize some encoders buffers, for instance if you recode to AC3 you obtain a delay of 5 ms, until 38 ms with some AAC encoders.

But with some encoders that delay is stored in metadata and any decoder must restore the audio without delay. For instance AAC in mp4 container (.m4a) must play without delay.
I can't reproduce the problem recoding DTSHD to FLAC, for me the obtained .flac is decoded without delay.

[zeropc]

Are we going to see a new version that supports HEVC? Last version is almost a year old

[sneaker_ger]

eac3to v3.32 "added support for h265/HEVC bitstream parsing". So your feature request seems too unspecific. Either way eac3to development seems pretty much stalled. Not much you can do except make a proper feature request on madshi's tracker and keep fingers crossed he finds time/interest to work on it again ...

[ryu34]

Hey I'm trying to use a sony x700 uhd bd player to play back my 4k uhd remuxes from a usb flash drive. It doesn't support true-hd playback from mkv containers, it can only do true-hd from m2ts, however dts-hd ma works fine from either. I thought i could over come this by just using the m2ts straight from the rip but when i playback that file i lose the ability to seek or skip chapters, anyone have any suggestions barring getting a new playback device? Thanks

[SeeMoreDigital]

Hey I'm trying to use a sony x700 uhd bd player to play back my 4k uhd remuxes from a usb flash drive. It doesn't support true-hd playback from mkv containers, it can only do true-hd from m2ts, however dts-hd ma works fine from either. I thought i could over come this by just using the m2ts straight from the rip but when i playback that file i lose the ability to seek or skip chapters, anyone have any suggestions barring getting a new playback device? ThanksSadly... With regard to av file/container support, all hardware playback devices have their limitations. And just-so-you-know, the Panasonic UHD players are just as limited as the Sony's in this regard!

[ryu34]

OK new question what kind of tools do i need to use with eac3to to demux 4k UHD dics rips and remux them into my own custom m2ts , edit i tried ts muxer and the audio desyncs from the video when using a true-hd track demuxed from the disc rip using eac3to.

[tebasuna51]

I don't know any free tool to manage 4k UHD m2ts.

tsMuxeR can't do it, read this (https://forum.doom9.org/showthread.php?p=1880216#post1880216), now the source code is released and maybe it can work in the future.

[stax76]

Can somebody with a trustworthy reputation provide updated libraries?

https://github.com/staxrip/staxrip/issues/32

[Sparktank]

Can somebody with a trustworthy reputation provide updated libraries?

https://github.com/staxrip/staxrip/issues/32

you can use Rarewares for libflac update (remember to use only the x86 build for eac3to).
www.rarewares.org/lossless.php

They use latest library.
According to hydrogenaud.io forums, the only real differences for FLAC 1.3.2 and 1.3.3 is mainly for behind the curtain updates.
Nothing useful for the end-user and not worth re-compressing your existing library that has used 1.3.2 in the past.

[stax76]

you can use Rarewares for libflac update (remember to use only the x86 build for eac3to).
www.rarewares.org/lossless.php

They use latest library.
According to hydrogenaud.io forums, the only real differences for FLAC 1.3.2 and 1.3.3 is mainly for behind the curtain updates.
Nothing useful for the end-user and not worth re-compressing your existing library that has used 1.3.2 in the past.

Thanks, I found everything I needed.

[samf88]

Hi all... not sure if this doubles as a help thread but hoping someone can spot where I'm going wrong with eac3to. I searched and found others have had this error, but help threads have fizzled out before anything was resolved.

All I want to do is demux *all* video/audio/subs/chapters tracks from a BD playlist on my hard drive to a separate directory (obviously I don't want to demux within the BD structure). I don't want to convert or encode anything, but it seems to think that's what I want to do. Where am I going wrong?

C:\>"D:\eac3to.exe" "F:\BD\MOVIE" 2) -demux "F:\eac3toDemux"
M2TS, 1 video track, 1 audio track, 6 subtitle tracks, 0:21:02, 24p /1.001
1: MPEG2, 1080p24 /1.001 (16:9)
2: AC3, English, 2.0 channels, 192kbps, 48kHz
3: Subtitle (PGS), English
4: Subtitle (PGS), Chinese
5: Subtitle (PGS), Dutch
6: Subtitle (PGS), French
7: Subtitle (PGS), Italian
8: Subtitle (PGS), Spanish
Track 2 is used for destination file "".
This audio conversion is not supported.

I understand I can manually select each track and give it a filename, but I was hoping to do that stuff further on in my workflow as this isn't a one off job.

All of the eac3to help pages seem to imply the -demux flag is for demuxing *everything* so I'm not sure where I'm going wrong. Thanks in advance!