UsEac3to v1.3.6: a different GUI for eac3to

[tebasuna51]

To make a mkv directly from BluRay source I recommend you use MakeMkv (https://www.makemkv.com/) instead.

[szabi]

ffmpeg simply doesn't support encoding >6 channels eac3.

https://trac.ffmpeg.org/ticket/3595

Hi
The quoted answer to my question is almost 2 years old.
Are there any development since regarding 7.1 ch E-AC3 (dolby digital plus) encoding?
regards
szabi

[tebasuna51]

Seems developers aren't interested in 7.1 channels (like me).

[szabi]

Hi
I am interested too.
More and more movies are out with DD+ 7.1ch.
When I use search so many result have the same interest.
I really do not understand why developers are not move forward to.
Anyway thanks.
I ask it again after 2 years passed. :D
Regards
szabi

[SeeMoreDigital]

Hi
I am interested too.
More and more movies are out with DD+ 7.1ch.
When I use search so many result have the same interest.
I really do not understand why developers are not move forward to.
Anyway thanks.
I ask it again after 2 years passed. :D
Regards
szabiHmmm....

If more and more movies are already encoded with 7.1 DD+, why is it so important for you to generate 7.1 DD+ encodes yourself?

[szabi]

If 7.1 DD+ is a trend and more widely used, why a developer does not have interest?

[SeeMoreDigital]

If 7.1 DD+ is a trend and more widely used, why a developer does not have interest?If 7.1 DD+ is a trend, then extract and use the original DD+ stream from the Blu-ray disc!

[tebasuna51]

7.1 is a commercial placebo to force people to buy new systems, 5.1 is more than enough for 2D surround.

We have only 2 ears, with 4 speakers we can obtain a full 2D surround.
We can add the subwoffer speaker for low frecuencies, but don't need a specific channel (LFE) all amplifiers/receivers can extract low frecuencies from all channels and send them to subwoffer.
Even the Center speaker is unnecesary, the same signal in FL-FR make a phantom Center.

A 4.0 is enough for a 2D surround.

[SeeMoreDigital]

7.1 is a commercial placebo to force people to buy new systems, 5.1 is more than enough for 2D surround.

We have only 2 ears, with 4 speakers we can obtain a full 2D surround.

Personally, I quite like 7.1 audio in movies...

But I did grow-up listening to quadraphonic music and preferred it to many 5.1 music mixes.

[chros]

But I did grow-up listening to quadraphonic music and preferred it to many 5.1 music mixes.
Me too!

[tormento]

Even the Center speaker is unnecesary, the same signal in FL-FR make a phantom Center.
I don't agree on this.

Dialogues are usually on Center and some movies are really difficult to hear about words with lot of movie background noise.

The possibility to increase Center volume is priceless.

[tebasuna51]

Dialogues are usually on Center and some movies are really difficult to hear about words with lot of movie background noise.

This is other problem related with the volume mix.
With a proper mix the problem don't exist.

Of course have the main dialogs on Center channel make easy correct bad volume balance on some movies.

BTW I can accept 3 speakers to play front audio with full volume, my objection is use 4 speakers for surround audio, most the time the volume of SL,SR,BL,BR are very low and are unnecesary, is enough 2 speakers for the volume needed (and for 2D surround).

[szabi]

If 7.1 DD+ is a trend, then extract and use the original DD+ stream from the Blu-ray disc!

Yeah, but it is not an answer to my question about developer being disinterested.

[nevcairiel]

Interest in relatively old commercial codecs (face it, EAC3 isn't exactly new anymore) comes from need - wanting to actually use it. Especially when we're talking about encoders.
Perhaps wanting to actually encode DD+ 7.1 isn't really something that comes up often outside of commercial use-cases. You yourself have not exactly elaborated why wanting to encode it correlates with more movies using it.

Or of course someone could offer to pay a developer for this feature.

As a contrast, the ability to decode Blu-ray style DD+ 7.1 was actually added to FFmpeg only fairly recently. But encoding would be quite a different cup of tea (also, Blu-ray style DD+ 7.1 is quite different to eg. streaming)

[tebasuna51]

@szabi
Here there are a Test_Chan_71 (http://www.mediafire.com/file/ebmub78swa6agbp/Test_Chan_71.7z/file) file with 8 audio tracks.
Say me how do you listen each track in your 7.1 audio system.

[tebasuna51]

There are a new v1.3.0 (download link in first post)

Changelog:

- v1.3.0 2020-08-03 Incluided mlp.exe (Extract TrueHD from .mlps, see 1.2).
Improved include of FFMPEG functions (see 2.1 y 2.2).
ffmpeg can now encode all audio formats.

Please read the Readme to know more about the changes directed to use UsEac3To like a GUI for ffmpeg audio encoding.

[SeeMoreDigital]

Has anyone been able to download this new version?

Windows Security reports that it has detected a virus. And deletes the software download!

[tebasuna51]

Is easy to compile or work like interpreted soft (open .au3 with AutoIt3_x64.exe), the GUI don't need be fast.

[LUCHOO]

There are a new v1.3.0 (download link in first post)

Changelog:

- v1.3.0 2020-08-03 Incluided mlp.exe (Extract TrueHD from .mlps, see 1.2).
Improved include of FFMPEG functions (see 2.1 y 2.2).
ffmpeg can now encode all audio formats.

Please read the Readme to know more about the changes directed to use UsEac3To like a GUI for ffmpeg audio encoding.


thanks tebasuna ..

Has anyone been able to download this new version?

Windows Security reports that it has detected a virus. And deletes the software download!

I disable security for a few minutes ..

https://www.virustotal.com/gui/file/a6dddaf2a841828645b2857c184d7106ff40b7f71abbdca4fd5eae87c13d14ed/detection

[tebasuna51]

My new Samsung TV don't play anymore .avi's with video MPEG4-ASP (XviD or DivX), then I need recode the video to MPEG-AVC (h264) or HEVC (h265)

0) When I load my Old.avi in UsEac3to I get a message:
"The format of the source file could not be detected. <ERROR>"

And 'Track Input' is empty and 'Output Format' show 'A.Tools'

1) Now I can write in CLP (Command Line Parameters):
-c libx264 -x264-params crf=19 -map 0:1 -acodec copy

2) And press 'MkvExtract/INF' button

3) In the window 'EACTO INFO' is show a ffmpeg info, and a new 'ffmpeg extract' window is open. See the attached image.

4) The track '0-v-avi' is selected (track 0 - type video - extension to extract avi). Let it selected to use 'map 0:0' like prefix to parameters.

5) Check 'Use CLP instead -xcodec copy' (here -vcodec like is video track)
to replace -vcodec copy by the CLP content.

6) Change output extension to 'mkv'. Of course you can use 'mp4'.

7) 'Run' or 'EnQueue'

If you select 'EnQueue' you can see the full command in C:\Portable\eac3to\UsEac3to\zzJob_1.cmd:

"C:\Portable\0\ffmpeg.exe" -i "C:\tmp\Old.avi" -map 0:0 -c libx264 -x264-params crf=19 -map 0:1 -acodec copy "C:\tmp\Old.avi_0.mkv" 1> "C:\tmp\Old.avi_Job_1.log"
rem (edited)-Encoders folder -----0)------- --4)--- ---------------------1) + 5)------------------------ Same as input----6)- *

1bis) Of course you can have different presets to encode to x264 or x265 or recode the audio to AAC or any preference stored in the ...\UsEac3to\Vid_enc.par and recover it with the 'Add' and 'F-FFMPEG' in 'More Parameters'

(*) ffmpeg output the log by stderr then I need change '1>' with '2>', corrected in next version.

[tebasuna51]

thanks tebasuna ...

You are welcome.

How are you, all ok?

[LUCHOO]

Yes tebasuna .. everything is ok. thanks.

Always using your GUI .. for many things .. with the implementation of ffmpeg filters it will be much better ..

we will do tests ..

[SeeMoreDigital]

My new Samsung TV don't play anymore .avi's with video MPEG4-ASP (XviD or DivX), then I need recode the video to MPEG-AVC (h264) or HEVC (h265)
Out of interest...

What model of Samsung TV do you have? And did you check to see if it's able to play MPEG-4 ASP video streams when they are placed within the .MP4 or .MKV containers?

[tebasuna51]

Samsung UE40NU7125 (not very new, from 2018).

In .avi or .mkv can't play MPEG4-ASP.

[roisiano]

Thanks again for this excellent gui, tebasuna51.

[tebasuna51]

See the attached image to explain the method.

1) Load the file or container in UsEac3To (or drag&drop in EAC3TO INFO window)

2) With F-FFMPEG in 'More parameters' click in 'Add' to open the 'FFMPEG functions' window.

3) Select the desired function '71-51c' for this sample.

4) You can edit the function here and when finish click in 'Done'

5) The function is traslated to 'COMMAND LINE PARAMETERS' window. Also here can be edit for instance to change the compand parameters (https://forum.doom9.org/showthread.php?p=1926777#post1926777) to points=-90/-84|-8/-2|-6/-1|-0/-0.1

6) Now you must click in 'A/V Recode' to open the 'Audio Recode' window.
If you want the output in a 'User defined' folder, instead the same folder than input, please select this option before click in 'A/V Recode'

7) First select the track to recode (if the input is a container).
Remember: for eac3to the first track is the '1', but for ffmpeg the first one is the '0'.
Here we select the track '1' to open the track number '2: TrueHD' for eac3to. Also force ffmpeg decoder.

8) Select 'Replace DEC' to use the -filter complex... in CML instead the default decoder parameters.

9) Select the output format and quality. Here EAC3 at 640 Kb/s

10) Run or EnQueue the job.

[2-perf]

Hi, a friend supplied me with this DTS-HD Master Audio 5.1 file and I would need to convert it to an RF64 file so to bring it Adobe Premiere, Can I do that with UsEac3to GUI 1.3.0? Thanks!!!

[tebasuna51]

Is not a standard conversion but eac3to can do it.

Select 'Output format' w64 and 'Add'.

In the CLP edit the:
%_.w64
to obtain:
%_.rf64

And 'RUN CL'

[2-perf]

Thank you tebasuna51!

Is not a standard conversion but eac3to can do it.

Select 'Output format' w64 and 'Add'.

In the CLP edit the:
%_.w64
to obtain:
%_.rf64

And 'RUN CL'

[2-perf]

Is there a way to extract chapters from an MKV file with eac3to? Thanks!

[tebasuna51]

eac3to can't extract chapters, but UsEac3to is also a GUI for MkvExtract (Settings -> Path to MkvMerge):

Click in 'MkvExtract/INF' -> Extract Chapters OGM (or XML), Add and Run or Enqueue

[SeeMoreDigital]

gMKVExtractGUI v2.5.2.0 works great at extracting .mkv contained streams for me...

[2-perf]

Thank you so much.

I think I got it now. :)

eac3to can't extract chapters, but UsEac3to is also a GUI for MkvExtract (Settings -> Path to MkvMerge):

Click in 'MkvExtract/INF' -> Extract Chapters OGM (or XML), Add and Run or Enqueue

[2-perf]

gMKVExtractGUI v2.5.2.0 works great at extracting .mkv contained streams for me...

So does eac3to. I'd rather use only one tool in my workflow.

[tebasuna51]

So does eac3to. I'd rather use only one tool in my workflow.

Like I say eac3to don't extract chapters.

Is UsEac3to than work like a GUI for MkvExtract.exe (in the same folder than MkvMerge.exe because is distributed in the same MkvToolNix package).

You don't need download gMKVExtractGUI (not distributed in MkvToolNix package) if you have already UsEac3to.

[tyee]

Sorry, I haven't read all this thread but just found this program. Using Win7 with dpi scaling I get improper gui buttons/text. Did anyone mention this previously and is there a fix?

https://i.imgur.com/gUT4X1A.png

[tebasuna51]

@tyee
Seems you have dpi scale to 150%

UsEac3to work fine for dpi scale 100%

You have the code in the first post, search each

GUICreate(...)

and add after a:

GUISetFont(5.7)
to override the default 8.5 size to be show at 150%

Maybe also other controls must be modified.

BTW if you scale the text to 150% maybe is because you can't read it at 100%, and if I make the font small the text become unreadable for you also.

If you have a screen with resolution 1920x1080 and need a dpi scale of 150 % to read text maybe is better preserve the dpi scale at 100 % and low the resolution to 1280x720

[TK423]

Hi all. Noob here, so please take it easy on me.

I've been using USEac3to v.1.3.0 to convert DTS HD-MA tracks to FLAC.

I've been dragging and dropping my MKV file into USEac3to, and selecting DTS under Track Input Format and flac under Output Format. I click Add, and then I click RUN CL.

https://i.imgur.com/Se53bjM.png

This then brings up a CL window, and the FLAC file is then created in my directory.

I recently discovered there is another way of converting a track, this way by clicking A/V Recode, then clicking Decode,Lossless,WavDts and selecting flac and then clicking Run.

https://i.imgur.com/QCnw5Zw.png

This then brings up a CL window as well, but shows different stuff, but also eventually spits out a FLAC file of the track, though this method seems to take a little longer.

Is one of these methods superior to the other? Is something different happening between these two conversion methods, is one creating a higher quality FLAC file than the other?

Really appreciate your patience and any assistance anyone might be able to provide. Thanks!

[guest]

Hi all. Noob here, so please take it easy on me.

I've been using USEac3to v.1.3.0 to convert DTS HD-MA tracks to FLAC.

I've been dragging and dropping my MKV file into USEac3to, and selecting DTS under Track Input Format and flac under Output Format. I click Add, and then I click RUN CL.

https://i.imgur.com/Se53bjM.png

This then brings up a CL window, and the FLAC file is then created in my directory.

I recently discovered there is another way of converting a track, this way by clicking A/V Recode, then clicking Decode,Lossless,WavDts and selecting flac and then clicking Run.

https://i.imgur.com/QCnw5Zw.png

This then brings up a CL window as well, but shows different stuff, but also eventually spits out a FLAC file of the track, though this method seems to take a little longer.

Is one of these methods superior to the other? Is something different happening between these two conversion methods, is one creating a higher quality FLAC file than the other?

Really appreciate your patience and any assistance anyone might be able to provide. Thanks!

Hi, I would think that they would turn out to be the same, however, what you could do, is do a conversion each way, then check them using Mediainfo, and compare the spec's...eg:- bitrate (kb/s), bit (16 or 24, etc), channels....

[tebasuna51]

Is one of these methods superior to the other? Is something different happening between these two conversion methods, is one creating a higher quality FLAC file than the other?

If you enqueue the jobs you can see (simplified) the differences:

1) eac3to input.dts output.flac
eac3to decode with libdcadec.dll and encode with libFLAC.dll (32 bits last version 1.3.3 2019-06-12) and --compression-level-8
I obtain a compression of 28,56 % and a speed of 21x with a sample in my PC.

2) ffmpeg -i input.dts -compression_level 12 output.flac
ffmpeg 64 bits decode with included libdcadec (most recent version) and libFLAC (I don't know the version but I supose is updated) but with best compression level (same quality because is lossless but less size)
I obtain a compression of 28,54 % and a speed of 14x with the same sample.

We obtain a very little gain in size but is slow than eac3to.

BTW I read than this compression level is less compatible with some devices and in the next UsEac3to version I will let the default level (5) like the more compatible.
With 5 I obtain a compression of 28,91 % and a speed of 66x with the same sample.

Much more fast than eac3to, more compatible with a little more size but much better than the DTS-MA with a compression of 45,18%

[EDIT]
Until next UsEac3to version remember the method (explained in the readme.txt) to override the UsEac3to default: -compression_level 12

Put in the COMMAND LINE PARAMETERS:

-acodec flac -compression_level 5

or any other value, and select in 'Use Command Line Parameters' : 'Replace ENC'

In the next version 1.3.1 to override the default (5) we need only in CLP:
-compression_level 8
and 'Add to ENC'

With ffmpeg -compression_level 8 (maybe like eac3to) I obtain a compression of 28,84 and a speed of 53x.
Fast but with a little more size than eac3to (I don't know for what, maybe different flac versions).

[tebasuna51]

There are a new v1.3.1 (download link in first post)

Changelog
=========
- v1.3.1 2021-05-26 Default flac compression level (5) instead 12.
Some ffmpeg improvements. AvsCutter function.

I added a new function, inspired in this thread (https://forum.doom9.org/showthread.php?p=1943598#post1943598), to help users than need something like the MeGUI Avs Cutter.

We know the range of frames of a video and need extract/recode the audio.
If the video trim is, for example:

Trim(30,120) (both frames included)

We can do (see the attached image):
1) Load the audio or container, any of the ffmpeg supported like input, in UsEac3to.
2) Select destination folder
3) Open the FFMPEG functions window
4) Select the video fps
5) Select Start frame (30)
6) Select End frame (120)
7) Click in AvsCutter button
8) The needed ffmpeg -atrim function appears in the window
(the End frame is replaced with the audio frame equivalent for Start frame, but the end time in atrim correspond to the end of the 120 video frame)
9) Click in Done and the function is transfered to the CLP and close the FFMPEG functions window
10) Now click in A/V Recode to open the Audio Recode window
11) Select track, 1 for first audio, 2 for second, etc.
The track order for ffmpeg is different from eac3to, the first track is 0 instead 1.
12) Add the audio filter needed from the CLP
13) Select the output audio format
14) Run or EnQueue

[LUCHOO]

Hi

Excellent tebasuna .. Thanks!

[tebasuna51]

You are welcome LUCHOO, nice to read you here.

[wolflop]

Hmm, tried to recode a wave file to an *.thd file. But it says always cannot find ffmpeg

[tebasuna51]

@wolflop: You must read the Readme.txt

Settings -> Encoder folder

- Folder with de/encoders: when the soft need a de/encoder search at UsEac3to
folder, after in the folder indicated here and, at last, in the ec3to folder.

You must provide all encoders/decoders you want use.
Download ffmpeg from https://www.gyan.dev/ffmpeg/builds/

[Axeldook]

Hello,

great tool and guide, thank you.

im encoding all my atmos/true hd to eac3 at 2300 mbs (my tv doesnt like these formats)

is there any way to encode eac3 with variable bit rate (rather than constant)

thanks

[tebasuna51]

im encoding all my atmos/true hd to eac3 at 2300 mbs

EAC3 support until 6.144 Mb/s but in UsEac3to I only offer until 640 Kb/s because it is transparent (without audible difference with lossless) at this bitrate.

And ffmpeg AFAIK only offer constant bitrate for EAC3 encoder, I never see -aq options for eac3.

[Axeldook]

thanks for coming back to me.

yea, i noticed upto 6.1 mbs but 2305 EAC3 saves me some file size and also (appears) to give much better sound (im only re-encoding from truehd source) via my atmos capable amp (using samsung tv (with plex) which doesnt like atmos or dts but can passthrough eac3 regardless of bit rate...)

[tebasuna51]

but 2305 EAC3 saves me some file size and also (appears) to give much better sound...

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.

[Axeldook]

"I make some test with ffmpeg and ignore the -aq (quality audio encode to obtain VBR) parameter "

how can i do this please?