UsEac3to v1.3.6: a different GUI for eac3to

[tebasuna51]

Seems a problem in your D drive, check it with CrystalDiskInfo or equivalent.
I use C or D drive without problems with eac3to and UsEac3to

[orisvl]

You're right it's an issue with that hdd, it's an external hdd (not health issue). I have problem only with haali. If I demux to raw it works fine.

[rocknard]

E-AC3 really working? I'm trying to convert or decode a EAC3 file and this gives me error. does it decode with libav/ffmpeg (E-)AC3 or i need nero 7 installed?

Thanks.

EDIT: seems that with windows xp i can't use the ffmpeg. And i want to use qaac. I will try belight/besweet

[sneaker_ger]

eac3to can use the libav/ffmpeg decoder but it may not work for all files. If eac3to fails try ffmpeg directly.

[manolito]

Fortunately some nice folks still provide current FFmpeg builds which work under WinXP. This one by Reino (CoRoNe) is my favorite:
http://rwijnsma.home.xs4all.nl/files/ffmpeg/ffmpeg-4.3-dev-327-g83e0b71-win32-static-xpmod-sse.7z

I know that it has no problems decoding and encoding eac3 audio.

To use qaac under XP you need an older version of iTunes or QuickTime. Let me know if I should upload something for you.


Cheers
manolito

[tebasuna51]

The libav dll in use with eac3to is very old and only work with eac3 5.1, with 7.1 crash.
For instance with:

M2TS, 1 video track, 2 audio tracks, 0:01:23, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: E-AC3, English, 5.1 channels, 640kbps, 48kHz
3: E-AC3, Spanish, 7.1 channels, 768kbps, 48kHz, dialnorm: -27dB
(core: AC3, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB)

work with track 2, but crash with track 3, the bug is already reported.

To decode/recode the track 3 you can use the button 'A/V Recode', if you have the appropiate ffmpeg and qaac for you OS.

belight/besweet never can decode eac3.

[rocknard]

eac3to can use the libav/ffmpeg decoder but it may not work for all files. If eac3to fails try ffmpeg directly.

Well, i'm not well versed on ffmepg commands with variables as normalize + downstereo etc (i will check for future reference).

Thank you

Fortunately some nice folks still provide current FFmpeg builds which work under WinXP. This one by Reino (CoRoNe) is my favorite:
http://rwijnsma.home.xs4all.nl/files/ffmpeg/ffmpeg-4.3-dev-327-g83e0b71-win32-static-xpmod-sse.7z

I know that it has no problems decoding and encoding eac3 audio.

To use qaac under XP you need an older version of iTunes or QuickTime. Let me know if I should upload something for you.


Cheers
manolito

Thanks, but i have already one packed by me. I don't know why i forgot to search for a ffmpeg for win xp.


The libav dll in use with eac3to is very old and only work with eac3 5.1, with 7.1 crash.
For instance with:



work with track 2, but crash with track 3, the bug is already reported.

To decode/recode the track 3 you can use the button 'A/V Recode', if you have the appropiate ffmpeg and qaac for you OS.

belight/besweet never can decode eac3.

Thank you for the note.

Well, i will check how to use ffmpeg to reencode a .eac3 7.1/5.1 to 2.0 (+lfe or not) with normalization/drc.

Stereo with 2 channels + subwoofer is equal to 2.1? do i need to put -downstereo or -downdpl (effect surround on stereo = dolby pro logic II or I?) on UsEac3To?

P.S.: behappy works on winxp with one lLSMASH-Works-r784 (for win xp) for .eac3 files (libav or ffmpegsource).

[tebasuna51]

Stereo with 2 channels + subwoofer is equal to 2.1?

Yes, but downmix to 2.1 is not recomended at all.
Downmix only to 2.0

do i need to put -downstereo or -downdpl (effect surround on stereo = dolby pro logic II or I?) on UsEac3To?

-downdpl is only recommended if your audio player is a 5.1 system with DPL decoder. To play in a audio stereo player (TV or standar amplifier) use always -downstereo

[mgutt]

Could you add more parameters to the parameters dropdown or make it editable, please. I miss "down6" for example.

List of commands:
https://en.wikibooks.org/wiki/Eac3to/How_to_Use#Using_audio_or_video_files_as_input

[tebasuna51]

You can edit UsEac3to.au3 and compile or interpret with free tools https://www.autoitscript.com/site/autoit/downloads/

BTW down6 is already in 'Frequent parameters':

[Natto]

Hi,

I hope someone can answer these questions. I need to encode DTS-HD MA/DTS tracks to AC-3, initially I used the default encoder in eac3to but then I read using the ffmpeg encoder is better. So I used:

DTS as track input and ac3-ffm as track output which creates this command:

5: stdout.w64 | ffmpeg -i - -c:a ac3 -b:a 640k -center_mixlev 0.707 %_5kor.ac3

is this the best way to encode DTS to AC-3?

I noticed after this command runs a yellow bit of text, it says:

"pipe:: corrupt input packet in stream 0"

This isn't present if I use ffmpeg through the A/V Recoder button or eac3to's internal encoder. Are the files damaged or are they fine to use?

Lastly why are some of the newly created AC-3 files shorter in duration compared to the original video? Some are 1 second shorter, are these still fine to use and will they be in sync throughout the whole video?

[SeeMoreDigital]

What playback device do you have? Are you sure it does not support DTS audio playback?

[Natto]

I checked the manual and sadly it doesn't support DST playback, only AC3.

I just demuxed the DST core using tsMuxeR and the length is 5 seconds shorter than the m2ts file. The AC3 file created by UsEac3to using ffmpeg is the same length, the LeeAudBi5 program also states both files are 5 seconds shorter.

Can you please tell me if the files created by UsEac3to using the ffmpeg command above are good for use providing the lengths match? What about the files that are shorter?

[tebasuna51]

The command line is correct and must work fine if source is not corrupt.
But if the output is shorter maybe there are problems, check if there are async at end of movie.

The libdcadec.dll (DTS decoder) included with eac3to is from 2015.
Is better use ffmpeg, with a recent version, through the A/V Recoder.

[Natto]

If I select DTS as track input and ac3-ffm as track output creating this command:

5: stdout.w64 | ffmpeg -i - -c:a ac3 -b:a 640k -center_mixlev 0.707 %_5kor.ac3


is that the same (minus your mix settings) as using the A/V Recode option? I find it quicker to do it as above. I've downloaded the latest version of ffmpeg and I've set the location in the encoders folder.

There are quite a few that have a second shorter duration, a couple show a second shorter duration but then quickly change by adding the one second on. VLC shows 02:18:05 for the m2ts file, 02:18:04 for the AC3 file where as LeeAudBi shows:

Duration ..........: 8284.8 seconds. ( 2 h. 18 m. 4.8 s.)

for the AC3 file.

I will remux a video and check if the output is in sync at the end. Thanks for your help.


Edit:

I've remuxed the film and checked the start, middle and end. Everything, including subtitles, appears to be in sync. The newly created MKV file has a duration of 2:18:04 according to VLC.

I then used BDInfo on the disc folder/files and it states:

Length
------
2:18:04.776


LeAudiBi states the AC3 file is:

2 h. 18 m. 4.8 s

If VLC rounds of to the nearest whole second, the decrease may only be a few hundred milliseconds. Could there be an explanation for this? I think in these instances where an extra second is shown, everything is probably fine, especially as the start, middle and end appear to be sync. The file that is off by 5seconds may be corrupt.

Do I have to worry about the message in yellow after ffmpeg has run? The message appears at the end of each job, just before starting the next job. I added pause to the batch file so I could see what it says.

[tebasuna51]

If you load a 00004.m2ts in UsEac3to and want recode to AC3 the 5 audio track (4 if begining by 0) you can:

1) click in 'A/V Recode' button
2) select map:X ... to 4
3) Recode to E/AC3, ffmpeg: 640
4) Click Run or EnQueue

If you select EnQueue you can see in ...\eac3to\UsEac3to\zzJob_1.cmd the command to execute the job:
"C:\PATH1\ffmpeg.exe" -drc_scale 0 -i "D:\PATH2\00004.m2ts" -map 0:4 -acodec ac3 -center_mixlev 0.707 -ab 640k "D:\PATH2\00004.m2ts_.ac3" 2> "D:\PATH2\00004.m2ts_Job_1.log"


Now ffmpeg is decoder and encoder at same time, but the encoder parameters are identical to the created for eac3to decoder.
The difference is only the decoder, before was the old libdcadec.dll and now is the more updated included in ffmpeg.
And is always better than the decoded data are internal to ffmpeg than send by 'pipe' betwen eac3to and ffmpeg.
Seems than the corrupt packet (yellow message) is now avoided.

About the duration: an AC3 stream is a sucession of frames of 32 ms, with 258899 frames the duration is 02:18:04.768, if video legth is 2:18:04.776 the encoder create a new frame until 258900 and 02:18:04.800 with 24 ms silence at end.
Some soft round to 02:18:04 or 02:18:05 don't worry.

[Natto]

Sorry for the delay in responding and thank you for taking the time to explain and for helping.

With the videos I have already remuxed with the eac3to ffmpeg command, do I need to create the audio again as you have described and then remux the videos again with the new audio?
I ask because I'm not sure if the pipe message means there is a problem. If it's OK, I will leave them but make use of the A/V Recode button going forward.

I noticed something with 4K films, I encoded audio and ffmpeg is selecting the wrong track, I think it's being caused by the chapters track and/or Dolby Vision. Should I be
starting my count at zero from the last video track?

For instance, if I want the:

5: DTS Master Audio, Korean, 5.1 channels, 24 bits, 48kHz

track from:

M2TS, 2 video tracks, 5 audio tracks, 6 subtitle tracks, 1:59:47, 12p
1: Chapters, 20 chapters
2: h265/HEVC, 2160p24 (16:9), 10 bits
3: h265/HEVC, 1080p24 (16:9), 10 bits - Dolby Vision Enhancement Layer
4: DTS Master Audio, German, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
5: DTS Master Audio, Korean, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
6: AC3, Korean, 2.0 channels, 192kbps, 48kHz
7: AC3, Korean, 2.0 channels, 192kbps, 48kHz
8: AC3, Korean, 2.0 channels, 192kbps, 48kHz
9: Subtitle (PGS), German
10: Subtitle (PGS), German
11: Subtitle (PGS), English
12: Subtitle (PGS), German
13: Subtitle (PGS), German
14: Subtitle (PGS), German

Would track 3 = 0, track 4 = 1 and track 5 = 2?

Thank you again for explaining, I'm happy knowing the audio showing a second difference isn't a problem. I did think it would be strange if so many discs would have been the problem.

I have found a tiny bug in your program, the job queue follows this pattern:

1 to 9
a - z
[
and then \

Slashes aren't allowed in Windows filenames/folders, so the job isn't created.

[tebasuna51]

... do I need to create the audio again as you have described and then remux the videos again with the new audio?
I ask because I'm not sure if the pipe message means there is a problem.
I don't know, check the end of movies for async.
Maybe the 'pipe' system crash in your PC, I never see that message before.

I noticed something with 4K films, I encoded audio and ffmpeg is selecting the wrong track, I think it's being caused by the chapters track and/or Dolby Vision. Should I be starting my count at zero from the last video track?

To know the track order than ffmpeg uses load the m2ts and click in 'MkvExtract/INF', that show ffmpeg info.


I have found a tiny bug in your program, the job queue follows this pattern:

1 to 9
a - z
[
and then \

Take this bug like a number of jobs limit (36).

[Natto]

I will check to make sure everything is in sync.

Thank you for telling me how to find the track with MkvExtract/INF, it's both quicker and easier that way.

I understand why you intend to leave it as the limit, I don't think many users will require more jobs and I probably won't need to go above 36 many times.

I received a warning on another file, is this a problem?:

[tebasuna51]

No problem, is only a warning about a parameter (drc_scale) not used in this case.

[mgutt]

I have a problem with some DTS audio tracks of an MKV file. If I convert them through PopCorn MKV AudioConverter or eac3to the audio is out of sync. If I use XMedia Recode it produces a correct result. But XMedia Recode is not build to convert multiple files. This means I need more recent ffmpeg libav dlls like as explained in "\legal stuff\ffmpeg\compiling\readme.txt" of eac3to:
Here's how the ffmpeg/libav dlls shipping with eac3to were compiled:

(1)
install TDM-GCC -> http://tdm-gcc.tdragon.net/
install MSYS -> http://www.mingw.org/wiki/MSYS

(2)
create empty folder "c:\msys\home\ffmpeg"

(3)
git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg

(4)
apply patch "ac3dec.patch"

(5)
start msys
cd ../ffmpeg
configure --enable-shared --disable-static --enable-memalign-hack

(6)
make install

(7)
c:\msys\local\bin\avcodec-54.dll
c:\msys\local\bin\avutil-52.dll


I tried to follow these steps, but I really don't know what I'm doing so it failed. Is anyone of you able to build the DLLs with a more recent version of ffmpeg? I would donate something if it works and of course everyone should be able to use them.

[tebasuna51]

Is anyone of you able to build the DLLs with a more recent version of ffmpeg?

I only know madshi to do that.

The problem is also when you convert the DTS's with standalone ffmpeg using 'A/V Recode' from UsEac3to?

[nevcairiel]

I tried to follow these steps, but I really don't know what I'm doing so it failed. Is anyone of you able to build the DLLs with a more recent version of ffmpeg?

You cannot use a much newer version of ffmpeg because the API changed, as indicated by the version in the DLL name advancing. Current versions are avcodec-58 and avutil-56, hence no longer compatible with software written against 54 and 52 respectively.

[mgutt]

You cannot use a much newer version of ffmpeg because the API changed

And maybe something between those old and the newest version? I don't know when this bug has been solved in ffmpeg. :confused:

@tebasuna51
I didn't tested it, but as XMedia Recode uses ffmpeg I thought this should be the reason.

Should I test it with different versions of ffmpeg. Which would be the last version that should be compatible with eac3to?

[tebasuna51]

... Which would be the last version that should be compatible with eac3to?

When use 'A/V Recode' function from UsEac3to eac3to it's not used at all, then you can use any ffmpeg version. Of course the last one is the recommended.

[SeeMoreDigital]

I have a question with regard to my 'quadraphonic' (4-channel) disc back-ups!

The problem I have is that my OPPO plays the original quadraphonic disc's okay but it doesn't like playing the 4-channel pcm.wav back-ups or even 4-channel FLAC encodes :scared:

So my question is... How easy is it to add a silent or empty 'front centre' channel to the original 4-channel pcm.wav file using UsEac3to?


Cheers

[tebasuna51]

eac3to can't add a new channel but you can use UsEac3to like a ffmpeg GUI like show the image.

About the filter:
-af "pan=5.0|FL < FL|FR < FR |FC < .0FR |BL < BL |BR < BR"

The Surround channels can be named like Side (SL-SR) or Back (BL-BR).
ffmpeg accept both like input but write the channelmask like you put (here BL-BR) maybe your player need the other channelmask, check it:

-af "pan=5.0(side)|FL < FL|FR < FR |FC < .0FR |SL < BL |SR < BR"

If you just EnQueue the job you can see in:
...\UsEac3to\zzJob_1.cmd
the command line to use ffmpeg directly.

EDIT: if your source is not a wav file maybe you need force the ffmpeg decoder (to 1) before 4) Add the filter
The empty Front Center channel is added with FC < .0FR (any channel amplified by 0)

[SeeMoreDigital]

Hi and thanks...

I think I may be doing something wrong: -

C:\UsEac3to>"C:\UsEac3to\ffmpeg.exe" -drc_scale 0 -i "D:\Andy Jackson\[2014] Signal To Noise (pcm)\Signal To Noise.wav" -vn -acodec pcm_s24le -af "pan=5.0|FL < FL|FR < FR |FC < .0FR |SL < BL |SR < BR" "D:\Andy Jackson\[2014] Signal To Noise (pcm)\Signal To Noise.wav_.wav"
ffmpeg version git-2020-03-23-ba698a2 Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 9.2.1 (GCC) 20200122
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
libavutil 56. 42.101 / 56. 42.101
libavcodec 58. 76.100 / 58. 76.100
libavformat 58. 42.100 / 58. 42.100
libavdevice 58. 9.103 / 58. 9.103
libavfilter 7. 77.100 / 7. 77.100
libswscale 5. 6.101 / 5. 6.101
libswresample 3. 6.100 / 3. 6.100
libpostproc 55. 6.100 / 55. 6.100
Guessed Channel Layout for Input Stream #0.0 : 4.0
Input #0, wav, from 'D:\Andy Jackson\[2014] Signal To Noise (pcm)\Signal To Noise.wav':
Duration: 00:40:52.79, bitrate: 9216 kb/s
Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 96000 Hz, 4.0, s32 (24 bit), 9216 kb/s
Codec AVOption drc_scale (percentage of dynamic range compression to apply) specified for input file #0 (D:\Andy Jackson\[2014] Signal To Noise (pcm)\Signal To Noise.wav) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some decoder which was not actually used for any stream.
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s24le (native) -> pcm_s24le (native))
Press [q] to stop, [?] for help
[Parsed_pan_0 @ 0000020416609500] Channel "SL < BL " does not exist in the chosen layout
[AVFilterGraph @ 00000204165c1800] Error initializing filter 'pan' with args '5.0|FL < FL|FR < FR |FC < .0FR |SL < BL |SR < BR'
Error reinitializing filters!
Failed to inject frame into filter network: Invalid argument
Error while processing the decoded data for stream #0:0
Conversion failed!
End job.
Press any key to continue . . .

Cheers

[SeeMoreDigital]

Ahaaaa...

I have just used the following command line: -af "pan=5.0|FL < FL|FR < FR |FC < .0FR |BL < BL |BR < BR"

And the software is working :)

UPDATE 1: Okay... I've just transferred the new 5-channel pcm.wav' file onto a USB flash drive and plugged it into my OPPO. But I'm only able to hear audio from the front left and right speakers!

General
Complete name : D:\Andy Jackson\[2014] Signal To Noise (pcm)\5-channel encode\Signal To Noise.wav
Format : Wave
File size : 3.29 GiB
Duration : 40 min 52 s
Overall bit rate mode : Constant
Overall bit rate : 11.5 Mb/s
Writing application : Lavf58.42.100

Audio
Format : PCM
Format settings : Little / Signed
Codec ID : 00000001-0000-0010-8000-00AA00389B71
Duration : 40 min 52 s
Bit rate mode : Constant
Bit rate : 11.5 Mb/s
Channel(s) : 5 channels
Channel layout : L R C Lb Rb
Sampling rate : 96.0 kHz
Bit depth : 24 bits
Stream size : 3.29 GiB (100%)

UPDATE 2: I've just tried converting the '5-channel pcm.wav' file to a '5-channel.flac' file. And the FLAC file has come out at only 830MB. Which would suggest that 3 out of the 5 channels are silent...

General
Complete name : D:\Andy Jackson\[2014] Signal To Noise (pcm)\5-channel encode\Signal To Noise.flac
Format : FLAC
Format/Info : Free Lossless Audio Codec
File size : 830 MiB
Duration : 40 min 52 s
Overall bit rate mode : Variable
Overall bit rate : 2 839 kb/s

Audio
Format : FLAC
Format/Info : Free Lossless Audio Codec
Duration : 40 min 52 s
Bit rate mode : Variable
Bit rate : 2 839 kb/s
Channel(s) : 5 channels
Channel layout : L R C Lb Rb
Sampling rate : 96.0 kHz
Bit depth : 24 bits
Compression mode : Lossless
Stream size : 830 MiB (100%)
Writing library : libFLAC 1.3.2 (UTC 2017-01-01)

[tebasuna51]

Sorry, seems a recent change in ffmpeg, if select like output SL-SR the filter must be

-af "pan=5.0(side)|FL < FL|FR < FR |FC < .0FR |SL < BL |SR < BR"

(previous post edited)

[tebasuna51]

Try now with SL-SR, or try to flac if your OPPO suport it.

Or maybe to flac 5.1 (the empty channels are compressed without size):

-af "pan=5.1(side)|FL < FL|FR < FR |FC < .0FR |LFE < .0FR |SL < BL |SR < BR"

[SeeMoreDigital]

So... I decided to start again using a smaller (470MB) '4-channel pcm.wav' audio file as a source', and: -

With this command line: -af "pan=5.0(side)|FL < FL|FR < FR |FC < .0FR |SL < BL |SR < BR", I hear no audio from the side channels on my OPPO.

However with command line: -af "pan=5.1(side)|FL < FL|FR < FR |FC < .0FR |LFE < .0FR |SL < BL |SR < BR", I hear audio from all the channels perfectly on my OPPO. So that's great.

But wait... There's more...

I re-encoded the original '4-channel pcm.wav' audio file to a '4-channel. flac' audio file, and the OPPO played all the channels perfectly too!

So I'm both happy and confused.

[SeeMoreDigital]

Using the same '4-channel pcm.wav' file' as a source, I've just performed another test using this much shorter command line: -af "pan=5.1(side)". And it generated exactly the same result as this command line: -af "pan=5.1(side)|FL < FL|FR < FR |FC < .0FR |LFE < .0FR |SL < BL |SR < BR".

It's all like living in "The Matrix" stuff to me....

[tebasuna51]

ffmpeg is always a surprise.

About flac: yes, the default channel layout for 4 channels is: front left, front right, back left, back right, then is correct with your source.

[SeeMoreDigital]

About flac: yes, the default channel layout for 4 channels is: front left, front right, back left, back right, then is correct with your source.It's certainly a weird one.

The 4-channel (quadraphonic) DVD-Audio disc plays fine in the OPPO. But the extracted '4-channel mlp/pcm.mlp' or '4-channel pcm.wav' files do not. But '4-channel flac.flac' files play fine.

Also, I've just discovered that '4-channel dts.mka' contained files play okay in the OPPO too...

Nothing is every straight forward is it :p

[jlw_4049]

Anyway to scale the GUI larger/easier to read fonts?

[SeeMoreDigital]

Anyway to scale the GUI larger/easier to read fonts?Out of interest... What resolution display do you have?

[jlw_4049]

Out of interest... What resolution display do you have?

I have a 1080p monitor. I can read the font easy but after long hours (my eye sight isn't perfect) it makes it rather difficult to read.

The program doesn't enlarge with the default windows DPI scalier. So I was hoping there was a way to adjust it in the programs .ini or something.

[tebasuna51]

You have here (https://forum.doom9.org/showthread.php?p=1905871#post1905871) a image sample in my 1080 display, I can't test diferent resolution or text size in screen configuration. BTW you have the source .au3 and autoit (https://www.autoitscript.com/site/) is free, modify the soft at your taste.

[jlw_4049]

You have here (https://forum.doom9.org/showthread.php?p=1905871#post1905871) a image sample in my 1080 display, I can't test diferent resolution or text size in screen configuration. BTW you have the source .au3 and autoit (https://www.autoitscript.com/site/) is free, modify the soft at your taste.

I don't see where to enlarge the screen size. This is way above my head :P

[tormento]

Do you mind use different folders for the needed programs?

I use MeGUI as standard repository for my encoders and I really don't like to put everything in eac3to folder or single folder whatsoever.

Thanks ;)

[tebasuna51]

Maybe MeGUI can change their behavior to fit with foobar2000 and UsEac3to, and use a unique folder for audio encoders.

Remenber a path for any encoder is complex.

[Rumbah]

Do you mind use different folders for the needed programs?

I use MeGUI as standard repository for my encoders and I really don't like to put everything in eac3to folder or single folder whatsoever.

Thanks ;)

You could use symbolic links of your file system to set a link for the different files from the eac3to folder to your repository.
That way you could have your one tool source without any changes for any other programs you might want to use.

[tymoxa]

@tebasuna51
Is possible to fix an error (Source file not found) that UsEac3to throws when PATH to input file(s) contains cyrillic symbols?

[tebasuna51]

Is possible to fix an error (Source file not found) that UsEac3to throws when PATH to input file(s) contains cyrillic symbols?

Sorry, many options in UsEac3to work by .cmd (or .bat) files with a special set of characters.

I have, inside UsEac3to, a translation between standard european character page to this special set, but I don't know how to make other translations.

In the download there are the autoit source, try to make this yourself.

[kdantas]

Hi, sorry my noob question... how can I get the .log of the eac3to demux using UsEac3to Gui?

[tebasuna51]

You can see the log in the main window, you can edit it, and you can save it with the 'Save Log' button in upper right corner ( in Input folder or Used Defined folder).

If you EnQueue the job the Save log is automatic.

[kdantas]

You can see the log in the main window, you can edit it, and you can save it with the 'Save Log' button in upper right corner ( in Input folder or Used Defined folder).

If you EnQueue the job the Save log is automatic.

Thanks @ tebasuna51, your information was perfect. One more question, for now, is it possible in the "Run and MkvMux" process to include the subtitles and chapters file?

[tebasuna51]

If the source is a mkv the "Run and MkvMux" preserve video, subs and chapters, only audio (1 or 2 tracks) are modified with the changed by eac3to.

If you need modify subs you need run MkvMuxer separately.

"Run and MkvMux" is ussefull for some audio recode (DTS to AC3 or AAC) for play in devices without DTS decode support.

[kdantas]

If the source is a mkv the "Run and MkvMux" preserve video, subs and chapters, only audio (1 or 2 tracks) are modified with the changed by eac3to.

If you need modify subs you need run MkvMuxer separately.

"Run and MkvMux" is ussefull for some audio recode (DTS to AC3 or AAC) for play in devices without DTS decode support.

But, using "Run and MkvMux" after the files are demuxed by eac3to, doesn't mkvmerge use these files to create the .mkv file? In my mind, the source would be the files that eac3to extracts from blu-ray, in my case. Or am I wrong in that thought?