UsEac3to v1.3.6: a different GUI for eac3to

[Axeldook]

Congratulations for your speakers and ears if you can distinguise between a eac3 640 Kb/s and other 2300 Kb/s.

I make some test with ffmpeg and ignore the -aq (quality audio encode to obtain VBR) parameter and output always the default 448 Kb/s.
"I make some test with ffmpeg and ignore the -aq (quality audio encode to obtain VBR) parameter "

how can i do this please?

[SeeMoreDigital]

Agreed... 640Kbps is more than good enough for 5.1 channel EAC3 (Dolby Digital+) audio.

[tebasuna51]

how can i do this please?

Using UsEac3to:
1) Load a file, select a audio track, Output format 'ac3-ffm' and 'Add'

2) Edit the generated CLP, for instance:
2: stdout.w64 | ffmpeg -i - -c:a ac3 -b:a 640k -center_mixlev 0.707 %_2eng.ac3
until obtain:
2: stdout.w64 | ffmpeg -i - -c:a eac3 -aq Q %_2engQ.eac3

Put any value for Q but the output is always 448 Kb/s
With other codecs, for instance AAC, Q can be 0.1,...,1.0,...,2.0,...
2: stdout.w64 | ffmpeg -i - -c:a aac -aq 1.5 %_2eng1.5.aac
with VBR output

[Axeldook]

Using UsEac3to:
1) Load a file, select a audio track, Output format 'ac3-ffm' and 'Add'

2) Edit the generated CLP, for instance:
2: stdout.w64 | ffmpeg -i - -c:a ac3 -b:a 640k -center_mixlev 0.707 %_2eng.ac3
until obtain:
2: stdout.w64 | ffmpeg -i - -c:a eac3 -aq Q %_2engQ.eac3

Put any value for Q but the output is always 448 Kb/s
With other codecs, for instance AAC, Q can be 0.1,...,1.0,...,2.0,...
2: stdout.w64 | ffmpeg -i - -c:a aac -aq 1.5 %_2eng1.5.aac
with VBR output
great, thank you so much for the info. will try now and report back !

take care

[Axeldook]

Using UsEac3to:
1) Load a file, select a audio track, Output format 'ac3-ffm' and 'Add'

2) Edit the generated CLP, for instance:
2: stdout.w64 | ffmpeg -i - -c:a ac3 -b:a 640k -center_mixlev 0.707 %_2eng.ac3
until obtain:
2: stdout.w64 | ffmpeg -i - -c:a eac3 -aq Q %_2engQ.eac3

Put any value for Q but the output is always 448 Kb/s
With other codecs, for instance AAC, Q can be 0.1,...,1.0,...,2.0,...
2: stdout.w64 | ffmpeg -i - -c:a aac -aq 1.5 %_2eng1.5.aac
with VBR output
Just tried this... using this:

2: stdout.w64 | ffmpeg -i - -c:a eac3 -aq 1 %_2engQ.eac3

Media info still shows constant...

Audio
Format : E-AC-3
Format/Info : Enhanced AC-3
Commercial name : Dolby Digital Plus
Duration : 9 min 46 s
Bit rate mode : Constant
Bit rate : 448 kb/s
Channel(s) : 6 channels
Channel layout : L R C LFE Ls Rs
Sampling rate : 48.0 kHz
Frame rate : 31.250 FPS (1536 SPF)
Compression mode : Lossy
Stream size : 31.3 MiB (100%)
Service kind : Complete Main

[tebasuna51]

Like I say you the -aq parameter is ignored and apply the default constant bitrate.

[gonca]

Is there any way to encode to eac3 with a core for blu ray compatibility?
High priced solutions are a no go, this is only a hobby

[tebasuna51]

See the SeeMoreDigital answer: https://forum.doom9.org/showthread.php?p=1949474#post1949474

[Axeldook]

Hi,

if i want to try QAAC, where do i put the exe so the app sees it ?

[tebasuna51]

You must have the qaac.exe (and the subfolder QTfiles64) in the same folder than eac3to (my C:\Portable\eac3to) or the same than UsEac3to (my C:\Portable\eac3to\UsEac3to) or in 'Encoders Folder' (Settings) like me in C:\Portable\0:

Directorio de C:\Portable\0
...
21/04/2019 16:15 654.336 opusenc.exe
23/04/2021 16:32 2.017.280 qaac.exe
29/04/2021 11:06 <DIR> QTfiles64
15/11/2018 21:28 282.816 soundstretch.exe
...
Directorio de C:\Portable\0\QTfiles64
15/01/2019 01:27 87.864 ASL.dll
15/01/2019 01:28 9.112.888 CoreAudioToolbox.dll
15/01/2019 01:27 2.252.088 CoreFoundation.dll
17/09/2015 20:09 3.072 icudt55.dll
15/01/2019 01:27 97.080 libdispatch.dll
15/01/2019 01:28 1.987.896 libicuin.dll
15/01/2019 01:28 1.358.136 libicuuc.dll
11/06/2011 01:15 608.080 msvcp100.dll
11/06/2011 01:15 829.264 msvcr100.dll
15/01/2019 01:27 216.376 objc.dll

Of course if your OS is only 32 bits the subfolder (and all the files 32 bit) must be only C:\Portable\0\QTfiles

[Krautmaster]

i'm still confused how to get the best audio at low bitrate from my LG OLED to my Q950A soundbar.

The 7.1True HD tracks consume a ton of space. I read that Plex is able to convert 7.1 true hd to 7.1 DD+, but i wonder how. FFMPEG or the thread here ends up having 5.1.

The LG can also convert internally AAC 7.1 to DD for ARC but I cant say of it does the same in 7.1 or 5.1 as this dumb soundbar does not tell ya how many channels it receives.


Why the heck cant the ffmpeg internal eac3 encoder handle 7.1 DD+ reencode?? Its only 2 channels more.

[tebasuna51]

If you want surround 3D you can't recode the tracks TrueHD/EAC3 Atmos or DTS:X you must preserve the original tracks.

If you are talking about surround 2D you don't need at all 7.1 channels, is more than enough 5.1 channels to obtain a full 2D surround.

[tebasuna51]

There are a new v1.3.2 (download link in first post)
Only solve a bug.

Changelog
=========
- v1.3.2 2021-10-31 Solved bug regresion in Merge Wav's function.

[szabi]

Hi

I asked it 5 years ago (https://forum.doom9.org/showthread.php?p=1753342#post1753342) then 3 years ago (https://forum.doom9.org/showthread.php?p=1846996#post1846996) and last year (https://forum.doom9.org/showthread.php?p=1914526#post1914526) as well.
It is time to ask again.
Are there any development regarding 7.1 channel dolby digital plus encoding?

Regards
szabi

[tebasuna51]

The answer is not far:
Looking for a eac3 7.1 encoder (https://forum.doom9.org/showthread.php?t=177346)

[karbre]

I'm currently trying to re-encode a Dolby TrueHD 7.1 + Atmos track to DTS-HD MA 7.1 in order to remove the Atmos extensions (it is revisionist in this case and I don't want them in my file). As the DTS-HD MA encoder needs separate channel wav files, I used UsEac3To v1.3.1 to convert the thd to w64 and now want to use ffmpeg to extract the 8 mono tracks.

However I'm confused about the channel layout (for use in the ffmpeg command), as MediaInfo reports different channel layouts on the source .thd and the resulting .w64: .thd is reported as "L R C LFE Ls Rs Lb Rb", while the result w64 is reported as "L R C LFE Lb Rb Ls Rs".

Is it correct that UsEac3To changed the channel layout? Can I trust the MediaInfo report for the further steps of my workflow?

[tebasuna51]

You have two options with UsEac3to:

1) Extract the THD without Atmos metadata (remain 7.1 lossless with less size than the DTS-MA):

a) put in COMMAND LINE PARAMETERS the ffmpeg filter:
-bsf:a truehd_core -c:a copy
b) Clik in 'A/V Recode', 'Add to DEC' in 'Use Command Line Parameters', thd in 'Decode, lossless, WavDts'
EDIT: must be 'Replace ENC' instead 'Add to DEC'

Run or Enqueue

2) If you still want recode to DTS-MA you can obtain the monowavs directly with eac3to:

%_.wavs

And eac3to never change the channel layout, MediaInfo can show the internal order of channels and can be different in a encoded file than in wav/w64.
Don't worry, the decoder output the correct channel order.

[karbre]

Thanks, that was super helpful! I used the ffmpeg filter which I did not know before.

Option 1a) however somehow led to a 5.1 file instead of 7.1, and also with bigger file size than the 7.1 produced by the other version. Also the log produced the warning "Only mono and stereo are supported at the moment.", which does not really make sense considering the output was 5.1.

But well, one way that works is enough for me ;)

[tebasuna51]

Option 1a) however somehow led to a 5.1 file instead of 7.1, and also with bigger file size than the 7.1 produced by the other version. Also the log produced the warning "Only mono and stereo are supported at the moment.", which does not really make sense considering the output was 5.1

Seems ffmpeg recode the file to thd instead only extract the core deleting the Atmos metadata.

My fault: to copy instead recode we need 'Replace ENC' with CLP

In the image the complete steps to do the procedure.
The output is
command line:eac3to 00001.m2ts_2eng.thd_.thd
------------------------------------------------------------------------------
TrueHD, 7.1 channels, 48kHz

with sizes:
60.654.740 00001.m2ts_2eng.thd
45.019.306 00001.m2ts_2eng.thd_.thd

[szabi]

The answer is not far:
Looking for a eac3 7.1 encoder (https://forum.doom9.org/showthread.php?t=177346)

Seems still no solution.
Maybe next year.

[DanDare1983]

I have a quick question. Can you queue up movies using the demux option as I only seem to be able to demux one folder at a time. When you have a lot of movies to demux, clicking on each file you want to demux takes time so just by clicking demux and adding to queue would be alot easier.

[tebasuna51]

If you put in COMMAND LINE PARAMETERS:

%_.*

You can EnQueue the demux job.

Load other file, without clear the CLP, and EnQueue...

[DanDare1983]

Thankyou for your help, works perfect. I have one more question, basically its in regards to save destination. For some reason the demuxed files go to the actual program folder? I've tried same as input and also user defined but no good. Also when i demux a movie with two films on one disc or one of the extras along with the main movie they both go in the same folder and tend to overwrite each other, is there a way to keep seperate?

[tebasuna51]

For some reason the demuxed files go to the actual program folder? I've tried same as input and also user defined but no good.

Work fine here.
Remember define the 'User defined' in Settings -> Output folder
When it is not defined use the program folder.

Also when i demux a movie with two films on one disc or one of the extras along with the main movie they both go in the same folder and tend to overwrite each other, is there a way to keep seperate?

You always can modify the sufix:

%_Main.*
and
%_Extra1.*
...

[DanDare1983]

Work fine here.
Remember define the 'User defined' in Settings -> Output folder
When it is not defined use the program folder.



You always can modify the sufix:

%_Main.*
and
%_Extra1.*
...
Thankyou as always for getting back to me with an answer. I've just realised that all the files with % next to them go in the correct folder but when I simply type
-demux the demuxed files go in the program folder. Do I have to put something like %-demux to get the files to the same folder as the source? Also I didn't know that the demux files can be automatically remuxed with mkvmerge, I only found out whilst trying some of the options. This saves alot of time although theirs a few things I would like to ask? 1) when I picked all tracks including chapters and run with runmkv I noticed its just the video and main audio that are remuxed without commentary, subs, chapters etc 2) is their any settings I can use as I'm not used to mkvmerge command, only the gui 3) I have a few movies with extra black video/padding, coloured bars and extra video which has nothing to do with the movie, do you know how I would trim the video etc to make it right. I've been directed to a program called dgdemux which from what I've seen is a very capable program dealing with video with these problems, although I find it quite limited as theirs no audio conversion and you can't queue up files which isn't good when you have lots of files to demux

[tebasuna51]

I've just realised that all the files with % next to them go in the correct folder but when I simply type
-demux the demuxed files go in the program folder. Do I have to put something like %-demux to get the files to the same folder as the source?

The character '%' is invalid for filenames and the GUI parse the CLP to replace that character by the folder (input folder or user defined) and the name of input file.

If you use ' % -demux' and have 'User defined' folder selected work fine, but if you have the input folder selected eac3to show:
Input and output file names overlap. <ERROR>
and the output go to program folder.

For that I say you: ' %_.*', the '*' is equivalent to demux, but with the '_' like sufix to override the overlap of names (you can use anything like sufix).

Also I didn't know that the demux files can be automatically remuxed with mkvmerge, I only found out whilst trying some of the options. This saves alot of time although theirs a few things I would like to ask? 1) when I picked all tracks including chapters and run with runmkv I noticed its just the video and main audio that are remuxed without commentary, subs, chapters etc

Please read '7) Run and MkvMux' in the Readme_UsEac3To.txt file.
Only work for some process:
1) First run eac3to for one/zero video and one/two audios.
Any other track is ignored after by mkvmerge.
2) After run mkvmerge with the video extracted (without audio, subs or chapters) or the input file (preserving video, subs and chapters) and the 1/2 audios extracted by eac3to.

2) is their any settings I can use as I'm not used to mkvmerge command, only the gui

Nope, the UsEac3toGUI try to help you for frequent process but can't cover all options.
The generic use is demux/convert all the tracks of input files with eac3to, do some auxiliary process (modify video, audio, subs,...) and after use MkvToolNix to obtain the final output.

3) I have a few movies with extra black video/padding, coloured bars and extra video which has nothing to do with the movie, do you know how I would trim the video etc to make it right. I've been directed to a program called dgdemux which from what I've seen is a very capable program dealing with video with these problems, although I find it quite limited as theirs no audio conversion and you can't queue up files which isn't good when you have lots of files to demux

To cut video without recode it you can try AviDemux or LosslessCut, but not always is possible a exact cut, maybe you need recode the video with other tools.

[tebasuna51]

Right now encoding 7.1 audio tract to AAC creates flawed files.
REAR_SURROUND channels are mapped to the SURROUND channels.
Content of SURROUND channels is completely lost and replaced with empty (silent) tract.

The 4 encoders included in UsEac3toGUI-Audio Recode (qaac, Nero, Fghaac and ffmpeg) work fine with 7.1 AAC.

I am not 100% confident if the issue is with decoding ...

Seems is a problem with your decoder, use ffmpeg (libav) to decode.

[KoVaR]

You are right.
AAC files indeed have a proper 7.1 audio.
It is me AV receiver that can't handle it.
Bummer ☹

Thank you for your involvement.

[DanDare1983]

Hi,

I'm having trouble adding single m2ts files. I would like something like 0007m2ts+0008m2ts+00012 as the first m2ts in the playlist is just 2 secs of padding causing audio to out of sync. I try to add 1 m2ts and then another but it keeps opening as a new file and doesn't add to the existing m2ts. Any help would be appreciated. Thankyou

[tebasuna51]

Let in a folder only the m2ts you want.
Load the first and click '+ Sources'

[DanDare1983]

Let in a folder only the m2ts you want.
Load the first and click '+ Sources'

OK thankyou so much for your help and also for such a fantastic program. Oh and one more thing, just using m2ts doesn't get the chapters. Would I have to get them separately from the playlist and would I have to delay the chapters?

[tebasuna51]

Would I have to get them separately from the playlist and would I have to delay the chapters?

Exactly, the 'SRT/../TRIM' button can help you to the delay.

[DanDare1983]

Exactly, the 'SRT/../TRIM' button can help you to the delay.

Thankyou so much the padded m2ts is showing at 1sec 1ms so if I was to delay the chapters how would I put it.....101ms?

[tebasuna51]

1sec 1ms = 1001 ms

[DanDare1983]

Hi, i would like to convert my audio commentaries to aac but i'm not sure where to start as i've never used the format before. I understand that i'd need to download the qaac application but would i do from there as i believe its not as straight forward as that. Any help would be much appreciated.

[tebasuna51]

Audio FAQ 3. How can I encode AAC? (https://forum.doom9.org/showthread.php?p=424070#post424070)

[DanDare1983]

Hi,

Thankyou so much for posting the link. It looks complex to me and i'm finding it hard. I did read from the readme notes from your program that neroaac could be used. However this program is very old and no longer in development so i'm not sure if theirs anything else that can just be dropped into the UsEac3to folder which is any better than neroaac? Any help would be appreciated as i'm reading that aac is now preferred to ac3.

[tebasuna51]

It looks complex to me and i'm finding it hard.

Uncompress that (https://www.mediafire.com/file/fhya6zujyk7fa2a/qaac64.7z/file) in UsEac3to folder

[DanDare1983]

Uncompress that (https://www.mediafire.com/file/fhya6zujyk7fa2a/qaac64.7z/file) in UsEac3to folder

Thankyou for the link. I've dropped it in and all seems to be working without problem. This is the setting.......stdout.wav | qaac -V 99 --ignorelength --adts --no-delay -o %_4eng.aac do i need to change anything or is this perfectly fine?

[tebasuna51]

Maybe audio commentaries don't need high quality (-V 99) you can modify to 91 or even 82.

[DanDare1983]

Hi, everything seems to be working well. I would like some advice if you could help...qaac is the recommended aac encoder but what is now the recommended ac3 encoder? Obviously aften has been aroun years and thats what i originally used but i'm sure something else has now come and taken its place? Also i'm trying to convert several audio tracks with qaac but it only allows one and i cant add anything else after i've added an audio track to convert. Do you know why that is? Thankyou

[tebasuna51]

... but what is now the recommended ac3 encoder?

Now the free ac3 encoder recommended is ffmpeg.

Also i'm trying to convert several audio tracks with qaac but it only allows one and i cant add anything else after i've added an audio track to convert. Do you know why that is?

When use external encoders (qaac, ffmpeg, ... with the 'pipe' method) only one track at time is allowed. You can EnQueue more than one job.

[von Suppé]

My apologies if this has been addressed before.
When I import a h264 elementary stream into UsEac3To, I get this error: "The format of the source file could not be detected. <ERROR>"

This happens with at least the streams demuxed with ffmpeg (clever ffmpeg GUI) and mkvextract (gMKVExtract GUI). When I let eac3to do the demuxing, the resulting videostream is accepted with no problems. Is this by design?

[tebasuna51]

You are right, eac3to can't detect the .h264 extracted with mkvextract or ffmpeg, but yes the extracted (bigger than the others) by eac3to.

BTW eac3to is not designed to work with video streams and can't do anything with them. What do you want make with eac3to and a h264 stream?

[von Suppé]

When changing h264 framerate I often demux with eac3to and then import the elementary stream. For years I've done this way to avoid container issues.
Since it's now topical in the MKVToolnix thread, it reminded me and just wondered why demuxed streams from other tools wouldn't work.

[tebasuna51]

New UsEac3to 1.3.3 version.
By demand I add a simple GUI for thdmerge (https://forum.doom9.org/showthread.php?p=1965067#post1965067) in the old 'MkvExtract/INF' button. Now:

Auxiliary Tools
===============
1) MkvEx/Thds/INF
-----------------
...
1.3) If the input file is a .thd, exist a .ac3 with the same name and you have
thdmerge.exe in one of the UsEac3To folders, make a .thd+ac3. New in 1.3.3.

Changelog
=========
- v1.3.3 2022-03-08 GUI for thdmerge in button MkvEx/Thds/INF (1.3).

[tebasuna51]

Reload the v1.3.3 to support thdmerge v1.1 (https://forum.doom9.org/showthread.php?p=1965366#post1965366) with the -t parameter

[von Suppé]

Thanks for the thdmerge support.

[von Suppé]

Hi tebasuna51,

About the "XLL output not lossless" message, I read this last post of VH thread: https://forum.videohelp.com/threads/400707-UsEac3to-1-3-0-DTS-HD-MA-to-FLAC#post2651323

Doesn't eac3to need both the 32 and 64 bit files, and maybe you want to update the eac3to package?

[tebasuna51]

Like eac3to.exe is a 32 bit exe the dll's can be only 32 bits.

Only madshi must update the eac3to package.

About that problem I edited, long time ago, the first post in eac3to thread:

Links
...
dcadec: https://github.com/foo86/dcadec [Recommended replace the libdcadec.dll (https://forum.doom9.org/showthread.php?p=1953397#post1953397) from the eac3to package with the updated v2.0 in the link]
...