UsEac3to v1.3.6: a different GUI for eac3to [Archive]

Ryu77

23rd October 2009, 08:42

One more time:
http://www.ebu.ch/CMSimages/en/tec_doc_t3324-2007_tcm6-53801.pdf
http://www.hemagazine.com/node/Dolby_TrueHD_DTS-MA_versus_Uncompressed_PCM

So what are you saying? Those articles don't compare Dolby 640 against DTS 1536. One did mention that they are both considered "excellent" in this bitrate range. Hardly a concise scale of performance. Back to the previous post...

BTW, I don't know for what you need a encoder less efficient and compatible than AC3. If you need encode is better AC3 448 than DTS 768, and AC3 640 better than DTS 1536

This statement says to me that you are implying that DTS is less efficient than Dolby Digital, and that Dolby at 640 Kbit/s is better than DTS at 1536 Kbit/s.

For my previous post, a simple yes or no answer would have suffice. I am extremely sorry to challenge you on this on your own thread as it seems like you put together a nice GUI...

However, there is no way that Dolby at 640 Kbit/s sounds better than DTS at 1536 Kbit/s. Unless you are using equipment that can't reproduce the differences. I can easily hear the difference every, single time.

Different codecs are obviously designed differently and some have sweet spots at different bitrates. If you were to say that Dolby at 640 Kbit/s performs better than DTS at 768 Kbit/s, this might indeed be possible. I just find the level of detail and clarity in DTS 1536, almost as good as it's lossless counterpart.

tebasuna51

23rd October 2009, 11:38

This statement says to me that you are implying that DTS is less efficient than Dolby Digital, and that Dolby at 640 Kbit/s is better than DTS at 1536 Kbit/s.
Yes.
For my previous post, a simple yes or no answer would have suffice. I am extremely sorry to challenge you on this on your own thread as it seems like you put together a nice GUI...
Don't worry, is only a copy-paste. :)
However, there is no way that Dolby at 640 Kbit/s sounds better than DTS at 1536 Kbit/s. Unless you are using equipment that can't reproduce the differences. I can easily hear the difference every, single time.
I never say "sounds better".

The test is: listen the original uncompressed, the DTS encoded and the AC3 encoded.
Maybe for you the DTS sounds better than original uncompressed, but is not the question.

shon3i

23rd October 2009, 13:10

there is no way that Dolby at 640 Kbit/s sounds better than DTS at 1536 Kbit/sHow hell no? like comparing simple algo (DTS) with some hevy efficient algo (AC3). Like comparing MPEG2 and MPEG4 and say that MPEG2 still better. AC3 @ 640kbps clearly produce higher sound quality than DTS @ 1536kbps, even AC3@448kbps can be comparable to DTS 1536kbps.
I proved to myself several times on different equipment, even now when I had access to the DTS-HD MA Encoding Suite, which mean i encode my samples with up to date DTS encoding tool.

honai

23rd October 2009, 22:43

I guess he's refering to bass response with AC3 vs DTS, i.e. due to higher attenuation of low-level frequencies before encoding AC3 does not sound as "bass-rich" as DTS. This is especially true with on-the-fly AC3 (Dolby Digital Live) and DTS solutions on mainboards and soundcards. However, concerning overall faithfulness to the master I'd say AC3 is able to achieve much better replication than DTS, even when comparing 640kbps AC3 to 1536kbps DTS.

tebasuna51

24th October 2009, 00:48

I guess he's refering to bass response with AC3 vs DTS, i.e. due to higher attenuation of low-level frequencies before encoding AC3 does not sound as "bass-rich" as DTS.
This is especially true with on-the-fly AC3 (Dolby Digital Live) and DTS solutions on mainboards and soundcards.

Please, can you explain this. I don't understand...

Ryu77

24th October 2009, 08:18

I guess he's refering to bass response with AC3 vs DTS, i.e. due to higher attenuation of low-level frequencies before encoding AC3 does not sound as "bass-rich" as DTS.

This thread is going off topic, but no I wasn't only referring to bass response. I can actually hear finer details and more presence with DTS @ 1536 over Dolby @ 640. I can see that a few of you don't agree here which is fine but I can only go with what my ears tell me.

However, I have noticed bass heavy DTS tracks present on quite a few DVD's but for some reason this seems to be different on Blu-ray discs. Whether it is because most DVD's used 768 Kbit/s instead of 1536 Kbit/s or they used an older encoder, I am not sure.

How hell no? like comparing simple algo (DTS) with some hevy efficient algo (AC3). Like comparing MPEG2 and MPEG4 and say that MPEG2 still better. AC3 @ 640kbps clearly produce higher sound quality than DTS @ 1536kbps, even AC3@448kbps can be comparable to DTS 1536kbps.
I proved to myself several times on different equipment, even now when I had access to the DTS-HD MA Encoding Suite, which mean i encode my samples with up to date DTS encoding tool.

Comparing MPEG2 and MPEG4 (AVC) to Dolby and DTS is nowhere near a similar comparison. There has got to be at least a decade between MPEG2 and MPEG4 (AVC). Computing power and technology has had a quantum leap since then. I believe there was only about a year difference between the first commercial use of Dolby and DTS. I understand what you meant by this in terms of codec efficiency but I really don't think it's as clear cut as that.

Just out of curiousity, what sort of equipment are you comparing these on?

Looks like I am not alone on these thoughts...

http://www.audioholics.com/education/surround-sound/dolby-digital-vs-dts-a-guide-to-the-strengths-of-the-formats
http://www.practical-home-theater-guide.com/dolby-vs-dts.html
http://ezinearticles.com/?Dolby-vs-DTS-Surround-Sound---Which-is-Better?&id=553768

As I mentioned above, all these reference the fact that you need the right equipment to hear the difference.

tebasuna51

24th October 2009, 12:52

@Ryu77
Your links don't say "DTS is better than AC3" only:
"you need a thoroughly refined audio playback setup to bring out that subtle difference in sound quality between these two formats."

And I agree with this. My audio equipment, or my human ears I don't know, can't distinguish between them.

For that I need trust in blind test with professional audio equipment.
My first link say: DTS 1500 is best than AC3 448 but very close
My second link (not sctrict blind test but also with professional audio equipment) conclude:
"The lossless, Dolby Digital Plus, and DTS-HD High Resolution compressed tracks were just a little more open and airy. I hate to say it, but they just sounded more realistic and transparent. The 448 kbps Dolby Digital and standard DTS tracks were less so, a little more closed off. Between the 640 kbps Dolby Digital and the uncompressed, the difference was even less noticeable."

If most of the people can't distinguish between them (by audio equipment or human ears) and some experts prefer AC3 640 Kb/s over DTS 1500, for what we need spend bytes and select a format less compatible?

If you prefer DTS, use it, but I can't accept DTS is better than AC3 without probe it.

raquete

25th October 2009, 22:04

...but I can't accept DTS is better than AC3 without probe it.
me too!
edit: please, read the next post.
tebasuna,
first thank you for the new UsEac3to.

you know, i'm horrible in command lines.
can i ask how to 5.1 96/24 multichannel waves to flac without change sample rate, etc...only compression?
thanks.

raquete

26th October 2009, 01:16

now i know why "was so hard" to encode 5.1 96/24 multichannel waves to flac. lol
in my system(win2000) the down arrows on the right to select formats, parameters, etc are not working, i need to click until the output format(for example) be blue and use the arrows of the keybord to select the desired adjust.(flac, wave, waves,pcm,etc)
i can't say that is a GUI's problem, must be my old system. (?) :confused:

tebasuna51

26th October 2009, 04:10

I think Autoit ( http://www.autoitscript.com/autoit3/ ) support Win2000, but I can test it.
Try download the Autoit software and run the .au3 file instead the .exe.

raquete

26th October 2009, 23:14

I think Autoit ( http://www.autoitscript.com/autoit3/ ) support Win2000, but I can test it.
Try download the Autoit software and run the .au3 file instead the .exe. yes, Autoit works in win2k.
i download, run Autoit and open the UsEac3To.au3.
just the same, i can't chose options in "Track Input and Output format" and others (is "combo"? editing the file i saw "GUICtrlCreateCombo")
but works with the "trick" like i wrote before.

thanks for the link and hints.

raquete

6th November 2009, 02:44

tebasuna51,
i'm lost ...
how can i load lots of wavs with differents names from the same folder to encode as flac as the "+ Sources" buttom don't work in this case?

tebasuna51

6th November 2009, 12:51

The button '+ Sources' is only to concatenate sources like .VOB or .m2ts.
Isn't a batch process.

To convert to flac all the wav in a folder use this con2flac.bat file (put your correct path to eac3to) in the same folder and double click:
@echo off
for %%a in (*.wav) do "z:\path\eac3to.exe" "%%a" "%%a.flac"
pause

raquete

6th November 2009, 13:13

i did a copy of the lesson and is saved. :)

i never saw the word "do" in command lines before.

thanks so much.

raquete

25th November 2009, 17:54

To convert to flac all the wav in a folder use this con2flac.bat file (put your correct path to eac3to) in the same folder and double click:
@echo off
for %%a in (*.wav) do "z:\path\eac3to.exe" "%%a" "%%a.flac"
pause
command line(.bat) absolutely perfect, works from waves to flacs and adjusting the .bat is perfet from flacs to waves too!
i did dozen files. :cool:

regards tebasuna51!

MuLTiTaSK

2nd January 2010, 04:58

@tebasuna51

feliz ano nuevo :D

thank you so much for UsEac3to it sure is the best eac3to GUI i tried so far any chance on making it more awesome with some batch job features thanks anyway if not for sharing you are helping many unleash the powers of eac3to :)

DLynch

11th February 2010, 06:47

Thanks! This program is exactly what I need.:D

tebasuna51

15th February 2010, 01:31

New version to add Chapters and Timecodes extraction using MkvExtract.
Split m2ts in 4GB fragments is now optional.

UsEac3to05.7z (http://www.sendspace.com/file/pauv43)

MuLTiTaSK

15th February 2010, 01:43

@tebasuna51

amigo thank you very much for the update i'am a very happy UsEac3to abuser i always recommend it over all other eac3to GUI's good stuff;)
:thanks::thanks::thanks::thanks::thanks:....

tebasuna51

6th March 2010, 02:38

New version
Changelog v0.6 2010-03-06
- Improved command line parse (external programs 'pipe')
- Decode with NeroAacDec and Lame
- Time change also in XML and CUE
- Workaround to change ANSI chars to OEM ASCII for cmd's

Now work with more than one 'pipe'.
Example for a simple stereo downmix:
stdout.wav | Sox --ignore-length - -t wav - remix -m 1v0.3694,3v0.2612,5v0.3694 2v0.3694,3v0.2612,6v0.3694 norm | Lame -b 128 - %_.mp3

UsEac3to06.7z (http://www.sendspace.com/file/jlsl8j)

MuLTiTaSK

6th March 2010, 03:30

thanks for the update teba;)

chagos

2nd April 2010, 09:13

Which chain of command should I use in useac3to to extract and convert the core of a DTS-THDMA track?
I use -core %_.ac3 -640 -2pass, but give an error Please specify the source and dest files first and then the options..

tebasuna51

2nd April 2010, 10:58

Please put the full log to know the input file (DTS-MA or TrueHD?, audio only or container source?. 5.1 input or 6.1-7.1?, ArcSoft decoder installed or not?).

First dest file (Track Input (if source is a container) and Output format):
%_.ac3

After the parameters, no matter the order:
-640 (not needed because is the default)
-2pass (not needed because is the default if eac3to detect a overflow in first pass)
-core (not recommended if source is lossless)

If source is lossless 6.1 or 7.1 DTS-HD (and you have the ArcSoft decoder, if not -core is the default) or TrueHD is better use:
-down6

chagos

2nd April 2010, 12:26

Arcsoft not installed, but not working when I installed ArcSoft Total Media Theatre.

MediaInfo:

General
Complete name : E:\qqqqq.mkv
Format : Matroska
File size : 6.50 GiB
Duration : 2h 0mn
Overall bit rate : 7 729 Kbps
Encoded date : UTC 2009-09-30 13:38:59
Writing application : mkvmerge v2.6.0 ('Kelly watch the Stars') built on Mar 24 2009 15:23:17
Writing library : libebml v0.7.7 + libmatroska v0.8.1

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L4.1
Format settings, CABAC : Yes
Format settings, ReFrames : 8 frames
Muxing mode : Container profile=Unknown@4.1
Codec ID : V_MPEG4/ISO/AVC
Duration : 2h 0mn
Bit rate : 5 426 Kbps
Nominal bit rate : 5 581 Kbps
Width : 1 280 pixels
Height : 528 pixels
Display aspect ratio : 2.40:1
Frame rate : 23.976 fps
Resolution : 8 bits
Colorimetry : 4:2:0
Scan type : Progressive
Bits/(Pixel*Frame) : 0.335
Stream size : 4.56 GiB (70%)
Title : Eight.Below.2006
Writing library : x264 core 68 r1183M f21daff
Encoding settings : cabac=1 / ref=8 / deblock=1:-3:-3 / analyse=0x3:0x133 / me=umh / subme=9 / psy_rd=1.0:0.0 / mixed_ref=1 / me_range=32 / chroma_me=1 / trellis=2 / 8x8dct=1 / cqm=0 / deadzone=21,11 / chroma_qp_offset=-2 / threads=12 / nr=0 / decimate=0 / mbaff=0 / bframes=4 / b_pyramid=0 / b_adapt=2 / b_bias=0 / direct=3 / wpredb=1 / keyint=250 / keyint_min=25 / scenecut=40 / rc=2pass / bitrate=5581 / ratetol=1.0 / qcomp=0.60 / qpmin=10 / qpmax=51 / qpstep=4 / cplxblur=20.0 / qblur=0.5 / ip_ratio=1.40 / pb_ratio=1.30 / aq=1:0.80
Language : English

Audio #1
ID : 2
Format : DTS
Format/Info : Digital Theater Systems
Codec ID : A_DTS
Duration : 2h 0mn
Bit rate mode : Constant
Bit rate : 1 510 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Surround: L R, LFE
Sampling rate : 48.0 KHz
Resolution : 24 bits
Stream size : 1.27 GiB (20%)
Title : DTS-HDMA core 1536k
Language : English

Full useac3t0 log:

eac3to v3.18
command line: "E:\useac3to\eac3to.exe" "E:\zzzzzzz\ddddd TN 2 DTS-HDMA core 1536k - en-English.dts" -progressnumbers -log="E:\useac3to\UsEac3To.log"
------------------------------------------------------------------------------
DTS, 5.1 channels, 2:00:24, 24 bits, 1510kbps, 48khz

About Arcsoft decoders: they work in Windows XP SP3? I installed ArcSoft Total Media Theatre and not working. I put their folder in environment path, but nothing.

ron spencer

3rd April 2010, 03:57

the arcsoft decoders work fine in xp sp3...only for version two of total media theater though!!!

try:

If EAC3to says it cannot find Arcsoft Decoders, do the following (try in this order):

1. Add to the PATH envirnonment variable in XP C:\Program Files\Common Files\ArcSoft\Bin

2. Goto C:\Program Files\ArcSoft\TotalMedia Theatre\Codec and open DOS window and enter
regsvr32 ASAudioHD.ax

Or goto Start and then Run and copy:

regsvr32 C:\Program Files\ArcSoft\TotalMedia Theatre\Codec\ASAudioHD.ax hit enter!!!

#2 should work in MOST if not ALL cases.

3. do #2 above for arcdemux.ax, asaudio.ax, asudioeffecthd.ax, and asaudiohd.ax of #2 alone does
not work

4. DTS decoding may only work when you copy MagCore.dll, MagPCMac.dll, MagUIEngine.dll and
MagUIInter.dll from Program Files\Common Files\ArcSoft\Bin\ to eac3to\ (last item to try)

NOTE: In DOS window typing EAC3to -test (or see GUI for Test) will let you see available filters.
Choose "none" as decoder option in EAC3to to decode DTS with Arcsoft Decoders.

You can also unregister all the decoders if you want, then register just the DTS one:

after installing TMT, open up a command prompt, cd to the directory that contains the filters and run:

for /F %i in ('dir /B') do@regsvr32 /u /s %i

you then just regsvr32 the ax/dll's you need for DTS decoding.

The names of the dlls are pretty obvious.

tebasuna51

3rd April 2010, 04:12

If the file is already extracted like:

DTS, 5.1 channels, 2:00:24, 24 bits, 1510kbps, 48khz

you only need:
%_.ac3

chagos

3rd April 2010, 13:56

I've found this working solution:
1. Copy the next files from an ArcSoft Total Media Theatre 3.0.1.170 Platinum Retail installation to eac3to folder:
ASAudioHD.ax
DtsDec.dll
dtsdecoderdll.dll
MagCore.dll
MagPCMac.dll
MagUIEngine.dll
MagUIInter.dll
2. Copy checkactivate.dll from a Total Media Theatre 2 trial version installation to eac3to folder
3. Register ASAudioHD.ax

I've notice that in this case eac3to encode in a single pass.
:thanks:

chagos

18th April 2010, 15:52

I've demuxed a pcm audio track from a blu-ray, using tsmuxer. This ended with 2 wav files (3.99 and 2.69 GB). What command line should I use in useac3to to encode to dts ? Or things is much more complicated ?
Thanks.

tebasuna51

18th April 2010, 16:21

The WAVs extracted with tsmuxer are broken, use eac3to instead.

Drag & Drop the folder with BRip over UsEac3to, select feature (if more than 1), select track and ac3 output (with -down6 and/or -resampleTo48000 if necessary)

If you want encode to DTS (not recommended) you can select 'wavs' output to obtain 6 monowavs and encode to DTS with a external encoder.

chagos

18th April 2010, 16:53

The blu-ray has already an ac3 @640 kbps audio stream.
Your concern is about quality of pcm->dts encoding or about the size of the audio stream ? Already read that ac3 @640 kbps is almost good as dts @1536 kbps, but I'm not sure. What is your opinion ?

tebasuna51

18th April 2010, 20:03

Less size, more compatible, and maybe more or at least equal quality.

To distinguish between PCM-source and the DTS-1536/AC3-640 encodes you need:
1) A room acoustic condioned.
2) A very good audio equipament (receiver and speakers).
3) Good ears.

I haven't that room, my audio equipment is not so good (~1000€) and my ears are very old.
Someone say than the AC3-640 is more close to PCM-source than DTS-1536.
I don't know for me is the same.

tebasuna51

25th May 2010, 02:28

Light update (only the GUI): UsEac3to07.7z (http://www.sendspace.com/file/duy3zy)

v0.7 2010-05-25 eac3to Help, Track input filled with only audio, Settings improved.

SeeMoreDigital

18th July 2010, 22:12

I've been meaning to ask...

Can your application be used to split the "core" and "HD" audio streams into separate streams?

ACrowley

19th July 2010, 09:08

I've been meaning to ask...

Can your application be used to split the "core" and "HD" audio streams into separate streams?

eac3to can do it so the Gui too :)

SeeMoreDigital

19th July 2010, 13:09

eac3to can do it so the Gui too :)Really... Please tell me how?

tebasuna51

19th July 2010, 18:05

Really... Please tell me how?
With:
...
1: h264/AVC, 1080i60 /1.001 (16:9)
2: DTS Master Audio, 7.1 channels, 24 bits, 96khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
Select 'Track Input': 2: DTS, 'Output format': dts, 'Add'
Select 'Frequent parameters': core, 'Add'

Now the command line is:
2: %_2.dts -core
Press 'RUN Command line' and you have the dts core, with:
2: %_2.dtshd
you have the full DTSMA

If the Bluray source have a TrueHD track (3)
3: %_3.ac3 extract the included ac3 track
3: %_3.thd extract only the thd part
3: %_3.tdh+ac3 extract the full track

SeeMoreDigital

19th July 2010, 19:26

Thanks for that...

Is it possible to extract the DTS HD audio stream without the core?

Cheers

b66pak

19th July 2010, 19:46

nope...core is mandatory for dts-ma!
_

SeeMoreDigital

19th July 2010, 20:42

nope...core is mandatory for dts-ma!Yes, I understand that the "core" is mandatory for use with the Blu-ray disc format.

But does this also mean the inclusion of the core is mandatory for DTS-HD MA and DTS-HD HR streams as an audio format?

If this is the case, I guess I'll need a method of transcoding the DTS-HD element to say, PCM or FLAC :eek:

tebasuna51

20th July 2010, 09:18

Yes, the HD part is a complement to core part.
You need the ArcSoft DTS-HD decoder http://forum.doom9.org/showthread.php?p=1398577#post1398577

PatchWorKs

20th July 2010, 15:08

Just discovered/tested this software, but seems that don't fit my needs.

Dunno if anyone else could be interested in this, but I decided to switch to Vorbis/MKV for CD-Images with embedded cuesheets.

It will be possible to use UsEac3to for this purpose ?

SeeMoreDigital

20th July 2010, 15:18

Many thanks tebasuna51.

Decoding the HD audio element works a treat now :)

tebasuna51

21st July 2010, 10:28

Dunno if anyone else could be interested in this, but I decided to switch to Vorbis/MKV for CD-Images with embedded cuesheets.
Please explain your sources and destination formats.

UsEac3to in only a GUI for eac3to (and some Auxiliary Tools).

buzzqw

21st July 2010, 11:06

hi tebasuna51

jfyi i made a guide for your gui
it's focused on remuxing ts to mkv
it's on www.divxmania.it , an italian forum about video/audio encoding (where i am mod)
http://forum.divxmania.it/da-ts-a-mkv-senza-ricodifica/t12364.html

thanks for your tool!

BHH

tebasuna51

30th July 2010, 15:35

Update (only the GUI): UsEac3to 0.7.42 (http://www.sendspace.com/file/upomrq)

With mkv's generated with defaults in new versions of MkvMerge, from 4.1.0, the audio tracks (AC3, DTS, MP3 and MP2) can't be managed correctly with eac3to (header compression problem).
Same problem (eac3to now crash) with video track H264, from v4.2.0.

Changes:
- The button 'Run and MkvMux' use now --compression -1:none, all tracks muxed without compression.
- New option in 'MkvExtract/Mux' to remux the problematic mkv input without compression (now compatible with eac3to, tsmuxer, WD TV, ...)

fbd

2nd August 2010, 12:57

Update (only the GUI): UsEac3to 0.7.42 (http://www.sendspace.com/file/upomrq)

Thank you :)

tormento

13th September 2010, 07:58

I am trying to use it with a user defined output path but it always forget to ask me where I want the output files. Ideas?

tebasuna51

14th September 2010, 02:42

Use 'Settings' -> 'Output folder' before.

tormento

14th September 2010, 12:12

Couldn't be easier to ask for a destination folder when custom option is selected?

tebasuna51

14th September 2010, 14:59

Is easy to implement but uncomfortable to use in my opinion.
You have the source, make you your own version.

tebasuna51

30th November 2010, 15:55

New UsEac3to v0.8 (http://www.sendspace.com/file/huipq9)

Changes v0.8 2010-11-29

- Now 'Output format' only show eac3to extract/convert allowed options for 'Track Input'
If show 'A.Tools' only 'Auxiliary Tools' can manage track/file.
Examples:
Xvid in mkv can be extracted with 'MkvExtract'
Vobsub in mkv also can be extracted with 'MkvExtract'
.ogg, .m4a, ... can be decoded with 'Decode ogg/mpx' (external decoders needed).
.srt, .ssa, .ass subs and .xml, .txt chapters can be delayed and/or stretched with 'SRT/.../TXT'.

- Better support for .mpls.
Automatically feature 1) added.

Boardlord

24th December 2010, 16:13

Thanks!

The best eac3to gui IMHO :)

DruidCtba

30th December 2010, 19:40

Thanxs a lot dear friend, very cool your GUI for eac3to, greather.

Druid®

trold666

9th January 2011, 21:16

Is there any chance for, that UsEac3to, could take the file format *.m2ts, so we could rip the original subtitles out in the format (*.srt or *.ass) ?

Best regards
trold666

tebasuna51

10th January 2011, 02:38

To convert SUP subtitles (from m2ts) to SRT you need suprip (or other), UsEac3to can't do the job.

DVDBob

22nd January 2011, 23:00

I can not convert AAC files to AC3 files, as your GUI as it says "Nero Audio Decoder 2 instance failed."

Since I will not have Nero installed, so I downloaded this zip file "NeroAACCodec-1.5.1.zip" from here: http://www.nero.com/eng/downloads-nerodigital-nero-aac-codec.php

And copied exe files into the folder of your GUI and eac3to.

I hope you or someone else can help me.

tebasuna51

23rd January 2011, 00:25

NeroAacDec.exe (from NeroAACCodec-1.5.1.zip) can decode AAC audio inside MP4 container (.mp4 or .m4a).

To decode AAC audio with ADTS headers (.aac) you need Faad.exe (http://www.rarewares.org/)

DVDBob

23rd January 2011, 00:32

Now have i Faad.exe too, but it still does not work.

tebasuna51

23rd January 2011, 15:56

Use 'Decode ogg/mpx' to obtain a wav file and after recode to ac3.

DVDBob

23rd January 2011, 16:45

It works.

DVDBob

29th January 2011, 01:37

I have a problem with delay, see log:

eac3to v3.24
command line: "C:\Users\Pepsi\Downloads\UsEac3to\eac3to.exe" "D:\DVD2\Een.Pige.Og.39.Soemaend.vob" "D:\DVD2\Een.Pige.Og.39.Soemaend.vob_.*" -progressnumbers -log="C:\Users\Pepsi\Downloads\UsEac3to\UsEac3To.log"
------------------------------------------------------------------------------
VOB, 1 video track, 1 audio track, 0:51:27
1: MPEG2, 576i50 (4:3)
2: AC3, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB, -8ms
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] A remaining delay of -8ms could not be fixed.
[v01] Extracting video track number 1...
[v01] Creating file "D:\DVD2\Een.Pige.Og.39.Soemaend.vob_ - 1 - MPEG2, 576i50.m2v"...
[a02] Creating file "D:\DVD2\Een.Pige.Og.39.Soemaend.vob_ - 2 - AC3, 2.0 channels, 192kbps, 48kHz.ac3"...
Video track 1 contains 164177 frames.
eac3to processing took 2 minutes, 44 seconds.
Done.

Can i fix that with add +8ms to the command line ???

b66pak

29th January 2011, 03:27

nope...only multiple of 32ms...
_

Mtz

9th April 2011, 17:37

I'm not satisfied by the center channel from one AC3 5.1 file. In some moments the center si too low and what personages speak is very difficult to hear.
There is any option in useac3to to solve this in the simple way or I need more work with other programs?
Any advice?

Mtz

tebasuna51

9th April 2011, 18:10

There is any option in useac3to to solve this in the simple way or I need more work with other programs?
Sorry, you can't amplify only one channel from a ac3 5.1 using only eac3to.

Mtz

9th April 2011, 22:26

I was afraid of that. :(
Normalize is not good on AC3 5.1 audio?
The problem is when are a lot of ambiental sounds at high levels, them are covering the voices. This is one reason why I don't like AC3 and prefer DTS.

enjoy,
Mtz

tebasuna51

10th April 2011, 02:38

Normalize is not good on AC3 5.1 audio?
Normalize amplify (or not) all the channels with the same value. The relative volume of center channel don't change.

The problem is when are a lot of ambiental sounds at high levels, them are covering the voices. This is one reason why I don't like AC3 and prefer DTS.
?
Sorry, but I can't undesrtand.

Mtz

10th April 2011, 09:20

When I play movies on DVD players (I have more than 10), the AC3 sound is low compared with DTS which is loud and clear. The same thing now with HD mediaplayers (I have 4; 3 with Realtek chipsed and one with Sigma).
I live in Romania. Here the only movies with romanian sound are animations. If the sound is not clear for me, what about the kids?

SeeMoreDigital

10th April 2011, 09:25

Mtz

10th April 2011, 12:04

Yes. Here is a 3 minutes sample. Scenes as chapters, so you can use the NEXT buttons. Also I think the audio quality is bad, but this is more offtopic.

tebasuna51

18th October 2011, 14:27

New version 0.9 (http://www.sendspace.com/file/pnxdso)

- v0.9.0 2011-10-18 'Decode Ogg/mpx' changed to 'Recode Audio'.
Better control of track order in 'MkvExtract/Mux' and 'Run and MkvMux'
The user must close the command windows to can see errors if any.

More details in Readme.txt

kypec

18th October 2011, 16:41

Thanks for the new version! :)

SeeMoreDigital

18th October 2011, 16:49

Thanks from me too :)

nautilus7

22nd October 2011, 13:55

how do i use the srt/.../txt sync option? how do i load input files to sync?

menlvd

22nd October 2011, 16:54

tebasuna51
how about Unicode support?
eac3to v3.22
command line: "D:\minep\me_gui\tools\eac3_to\eac3to.exe" "D:\temp\a?S?Ж?. O??Е? ?аЧЕ?Е\01 ?Ж?S?.mpg" -progressnumbers -log="D:\minep\me_gui\tools\eac3_to\UsEac3to\UsEac3To.log"
------------------------------------------------------------------------------
Source file "D:\temp\a?S?Ж?. O??Е? ?аЧЕ?Е\01 ?Ж?S?.mpg" not found. <ERROR>

tebasuna51

22nd October 2011, 23:50

how do i use the srt/.../txt sync option? how do i load input files to sync?
Input or Drag&Drop the srt/.../txt file, eac3to say:
The format of the source file could not be detected.<ERROR>

Click over the 'SRT/.../TXT' button.

Some 'Auxiliary Tools' don't need eac3to to work, with 'Audio recode' there are others formats not supported by eac3to than can be used: ogg, wma and wmv.

tebasuna51

22nd October 2011, 23:55

tebasuna51
how about Unicode support?
Yes, is a problem.

Many UsEacto functions work with bat/cmd files and I don't know how support Unicode with this.

BTW, you have the source UsEac3To.au3, let me know if you can modify the code to support Unicode.

nautilus7

23rd October 2011, 02:01

Input or Drag&Drop the srt/.../txt file, eac3to say:
The format of the source file could not be detected.<ERROR>

Click over the 'SRT/.../TXT' button.

Some 'Auxiliary Tools' don't need eac3to to work, with 'Audio recode' there are others formats not supported by eac3to than can be used: ogg, wma and wmv.

I got it thanks.

madhatter300871

11th March 2012, 18:55

nope...only multiple of 32ms...
_

Why only in 32ms multiples ?

tebasuna51

11th March 2012, 19:49

Because, without recode, only can add/cut AC3 frames, and a AC3 (48 KHz) frame duration is 32 ms (a frame have 6 blocks of 256 samples = 1536 samples).

madhatter300871

12th March 2012, 11:29

Understood. Thanks for that answer.

XMEN3

24th April 2012, 14:31

Thanks for this tool.
My question:
Transcoding a trueHD track, the MLP data is going to be used ?
Better... if the trueHD core is ac3@448k, transcoding a trueHD@2500k to ac3@640k is useful?
I've been looking around to find out an answer to this with no success.

tebasuna51

25th April 2012, 01:09

Yes, to both questions.

tebasuna51

9th May 2012, 22:18

New version 0.9.2 (http://www.sendspace.com/file/s6xyb5)

v0.9.2 2012-05-10
Added Queue system and Shutdown.
Support for dcaenc, qaac and fghaacenc.
Added language in track name and some minor improvements.

More details in Readme.txt

To download and use qaac read http://forum.doom9.org/showthread.php?t=154233
For fghaacenc http://forum.doom9.org/showthread.php?t=161806
For dcaenc http://forum.doom9.org/showthread.php?t=163526

RainyDog

16th May 2012, 17:56

Cheers for the new version of this tebasuna51, appreciated.

Very handy to be able to use fghaac with eac3to through a GUI.

Pomegranate

2nd January 2013, 11:47

Hi tebasuna51! I'm sorry to dig up a 7 months old thread, but are you still maintaining this software? It's a wonderful tool, and makes Eac3to very easy to use. In any case, thank you very much for providing this great gui.

tebasuna51

2nd January 2013, 22:35

Maybe I need make some changes with the new eac3to 3.25.

Pomegranate

3rd January 2013, 02:04

If you do make changes, would you consider

1) making wav the default output format or an option to configure the default command,
2) making it possible to have the gui "always on top" like windows task manager?

rahzel

3rd January 2013, 05:49

Maybe I need make some changes with the new eac3to 3.25.
Would be greatly appreciated. :)

tebasuna51

5th January 2013, 18:35

New version 1.0.0 (http://www.sendspace.com/file/sjey8j)

Changelog
=======
- v1.0.0 2013-01-05 Minor changes to support eac3to v3.26

If you do make changes, would you consider

1) making wav the default output format or an option to configure the default command,
I don't like this, but you have the source.
You have the lines 557-583, for each type of audio source you can change the default output format, for instance in:

Case "AC3"
GUICtrlSetData($t2, "|ac3|flac|wav|wavs|pcm|w64|m4a|dts|mp3-ext|mp2-ext|ogg-ext|ac3-ext|m4a-ext|qaa-ext|fgh-ext|dca-ext|pcm-tsm", "ac3")

Replace the final "ac3" by "wav"

2) making it possible to have the gui "always on top" like windows task manager?
I prefer the Command Window over the GUI Window.
But you can comment with ';' the line 88 and uncomment the line 89 to obtain your desired behaviour.

rahzel

5th January 2013, 19:17

thanks tebasuna51.

Pomegranate

5th January 2013, 23:02

Thank you very much for v1.0.0 and for your guidance, it is much appreciated good sir!

VincAlastor

3rd March 2013, 16:39

thank you very much for qaac support!!! coult you support opus too?

tebasuna51

3rd March 2013, 22:22

VincAlastor

3rd March 2013, 23:03

Ok, in next version.

BTW is easy add encoders than support STDIN, is enough to have opusenc.exe in eac3to, UsEac3to or Encoder folder and use:

stdout.wav | opusenc --bitrate 128 - %_.opus

or others parameters.

that would be great! :) but i will try it tomorrow with your instruction, i hope that there is no reason for confusing channel remappings for 2.1, 5.1, 6.1 channel output :)

edit: and i would like to use the FLACCL encoder confortable in your gui, too. this encoder will be compress stronger and faster than the original flac encoder !!! - really cool, isn't it?!

bennynihon

10th March 2013, 23:29

This is a great app being able to combine so many functions. But after running "Run and MKVMux", it didn't mux the subtitles or chapters into the MKV output file. I've chose to extract both chapters and a subtitle SUP stream, yet it's not being muxed into the MKV output. And I know MKVMerge supports chapters and muxing in a SUP file. Is there a reason why this doesn't happen? Thanks!

tebasuna51

11th March 2013, 13:48

Yes, if the source is not a mkv file the 'Run and MkvMux' is limited to one video stream and two audio streams.
That is because I think than mux SUP subtitles in mkv is not a good idea (not supported by all players) and I recommend convert SUP before to SRT.
After you can add chapters and srt's to the generated mkv.

If the source is a mkv and you only convert 1 or 2 audio tracks 'Run and MkvMux' preserve the video, chapters and subtitles from original mkv.

Motenai Yoda

6th June 2013, 21:34

just a question, I'm trying to mux mkv to ts with the button "run and tsmux" but every time it come out with an error about file missing or something.
before the update (useac3to 1.0 and eac3to 3.27) worked

*hided*\Audio\UsEac3to>"*hided*\Audio\UsEac3to\eac3to\tsMu
xeR\tsMuxer.exe" "*hided*.mkv_.meta" "*hided*.mkv_t.m2ts"
Impossibile trovare il percorso specificato.
End job.
Premere un tasto per continuare . . .

tebasuna51

7th June 2013, 02:02

Work for me, without change in this function in last version.

And the eac3to log?
Maybe some conversión crash and not exist a needed file.

Motenai Yoda

7th June 2013, 08:28

ah seems to be a filename tsmuxer's problem.

kypec

7th June 2013, 09:25

ah seems to be a filename tsmuxer's problem.

No wonder if you use asterisks as part of the filename, how can you even expect that it could work? :eek:
Asterisk * is a wildcard placeholder in Windows and can't be used in regular paths or filenames.

Rumbah

9th December 2013, 18:29

Does anyone have a working link? The sendspace link doesn't work, unfortunately.

tebasuna51

9th December 2013, 23:24

My fault, I deleted the previous link, here is now: http://www.sendspace.com/file/sjey8j

Rumbah

10th December 2013, 14:21

Thanks

NikosD

25th August 2014, 14:13

Sorry if I resurrect this thread for wrong reason, but because I really like the app and the ability of automating things like the "Run and MkvMux" button, I would like to ask if it's possible by pressing that particular button to add automatically 2 audio tracks and 1 video track in muxing process.

I know that it's already happening if you select two different audio tracks.

But if , like me, you select the same audio track two times i.e first time to demux DTS-HD and second time to demux the DTS core from the same audio track (DTS-HD) in the same command line, then only one audio track is muxed by "Run and MkvMux", only the DTS core audio stream.

So the final MKV has 1 video and only 1 audio stream (DTS core), but it has demuxed two audio streams (DTS-HD, DTS).

The functionality is there, of muxing together two audio streams with one video with a single button.

Is it difficult to add the new functionality ?

Thanks!

SeeMoreDigital

25th August 2014, 15:02

Sorry if I resurrect this thread for wrong reason, but because I really like the app...
I don't think there's anything to feel sorry about... I use this app most days ;)

tebasuna51

25th August 2014, 18:01

The problem is only if you let the same name for the two audio track like this:

2: %_2eng.dts 2: %_2eng.dts -core

then the second one rewrite the first.
Put some char to have different names like:

2: %_2engHD.dts 2: %_2eng.dts -core

NikosD

25th August 2014, 19:06

Nope.

I tried it and didn't work.

Because from the first time the names were different, because the extensions were different.

The first name was : %_2eng.dtshd and the second 2: %_2eng.dts -core.

Different extension, different name.

But I tried your way renaming the name before the extension just to be sure and I had again just one audio (DTS core).

tebasuna51

25th August 2014, 20:19

Please use dts extension instead dtshd, like I put in my sample.
The dtshd extension isn't a standard extension, must be dts also for DTS-HD.

In next version I make support for dtshd extension.

NikosD

25th August 2014, 21:14

At last! It worked!

I have now the functionality I asked for the button. Perfect.

But to be honest, I just chose the .dtshd extension from the output format list, I didn't put it myself.

It's the first time I use eac3to and your GUI, probably more experienced users never choose .dtshd extension.

It would be better if you remove that extension from the list, or if you do what you said.Fix the app to recognize the .dtshd extension.

One last question.

Is it better/more accurate/ recommended/ faster (?) to extract the video stream from the original .mkv, in .h264 elementary stream or in .mkv muxed format, before pressing the "Run and MkvMux" and mux it again with the new audio streams.

Both formats (.h264) and (.mkv) can be used with MkvMerge.

Does it have any difference ?

thanks!

NikosD

26th August 2014, 05:27

I did another test putting %_2eng.dtshd -core and it worked.

The app (eac3to) extracted the DTS core from the DTS-HD stream.

So I think .dtshd is a recognizable extension, but not fully (for all functions).

IMO, it would be better if you leave it in the list in order to distinguish DTS-HD from DTS and just fix the button.

Thanks again!

tebasuna51

26th August 2014, 08:30

Fix the app to recognize the .dtshd extension.

Yes, I can add this in next version.

Is it better/more accurate/ recommended/ faster (?) to extract the video stream from the original .mkv, in .h264 elementary stream or in .mkv muxed format, before pressing the "Run and MkvMux" and mux it again with the new audio streams.
...
Does it have any difference ?
Without extract .h264 or .mkv is faster (same accurate) because you don't need write a another big file in disk.
And without video in command line you copy all subs/chapters/... to the new mkv.

Extract the video like .h264 is only usefull when you want correct the fps of the h264 stream. Sometimes the mkv container have the correct fps 24/1.001 but the h264 stream have 23.975 (for instance, eac3to send messages about this, the origin is a bad recode). Is not important but you can use:

1: %_1eng.h264 -changeTo23.976

But remember to add subs/chapters/... if you need.

NikosD

26th August 2014, 09:19

Without extract .h264 or .mkv is faster (same accurate) because you don't need write a another big file in disk.
And without video in command line you copy all subs/chapters/... to the new mkv.

Right!

I had an .m2ts file and without video command line, your app complained with the message "At least 1 video and 1 audio must be selected:"

But with .mkv this is not necessary, so I process only the audio part inside the existing .mkv with no subtitles, chapters etc.

I think it's perfect.

One more question:

How can I clear the info messages of eac3to in the main window ?

Is there something like clear screen as in command line (Clear CL)?

I think it would be useful.

tebasuna51

27th August 2014, 01:37

How can I clear the info messages of eac3to in the main window ?
I don't know for what can be usefull.
Show the info of active file or log of last process.
You can edit the info, Select with mouse and Delete.

NikosD

27th August 2014, 07:38

I meant something like a reset/restart button that will bring the status of the app like when you first open it.

Without any file in the input bar saying just "No file selected..."

and without any info in the eac3to main window saying "No File Selected... Drag&Drop is allowed."

After the processing of a file or a bunch of files, I need to clear everything and start from the beginning like the initial start.

I have to close the app and start it again to clear everything.

But it's more a psychological need of me that everything has finished and I'm ready for a new clear start, so never mind.

The app is perfect for my needs.

Thanks.

tebasuna51

11th September 2014, 22:47

Added new version 1.1.2 (https://www.sendspace.com/file/pkw3bx)

Changelog
=========
- v1.1.2 2014-09-12 Now we can extrat attached files from mkv.
ffdcaenc replace to dcaenc.
Some minor bugs solved.
- v1.1.1 2013-12-13 Solved bug XviD -cq mode.
- v1.1.0 2013-12-08 Added Video Recode.

See Readme for more details.

tebasuna51

11th March 2015, 20:32

New version (only the GUI): UsEac3to 1.1.5 (https://www.sendspace.com/file/vxdu2y)

Changelog
=========
- v1.1.5 2015-03-10 Solved bug when press Cancel selecting feature
- v1.1.4 2014-11-09 Support for wavi.exe

NikosD

1st June 2015, 10:28

Hi.

Since MkvMerge version 7.7.0, there is an option to mux TrueHD and AC3 in mkv container as separate tracks, instead of truncating the embedded AC3 stream of a .thd+ac3 file.

Maybe you could add that functionality to "Run and MkvMux", because as it is now it truncates the AC3 stream.

tebasuna51

1st June 2015, 13:30

I can't understand your request, if you want the two tracks yo can do now:

Input file:
M2TS, 1 video track, 2 audio tracks, 0:02:29, 24p
1: h264/AVC, 1080p24 (16:9)
2: TrueHD/AC3 (Atmos), English, 7.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)

UsEac3to COMMAND LINE:
1: %_1eng.h264 2: %_2eng.thd 2: %_2eng.ac3

With "Run and MkvMux" I get
Output file:
MKV, 1 video track, 2 audio tracks, 0:02:29, 24p
1: h264/AVC, 1080p24 (16:9)
2: TrueHD (Atmos), English, 7.1 channels, 48kHz
3: AC3, English, 5.1 channels, 640kbps, 48kHz

NikosD

1st June 2015, 13:36

You could always do that.

I'm talking about the new ability of Mkvmerge to add only one file with two streams inside .thd+ac3 and produce two separate tracks - one TrueHD and one AC3 in the same MKV during muxing.

As it is now your app deletes AC3 if you mux the .thd+ac3 file and keeps only the TrueHD.

tebasuna51

2nd June 2015, 16:11

Like MkvMux is limited to 2 audios I added to next v1.1.7 release your request.

Now with input file:
M2TS, 1 video track, 2 audio tracks, 0:00:27, 24p
1: h264/AVC, 1080p24 (16:9)
2: TrueHD/AC3 (Atmos), English, 7.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
3: AC3, Spanish, 5.1 channels, 640kbps, 48kHz

Whit 'Run and MkvMux' and this command line:

1: %_1eng.mkv 2: %_2eng.thd+ac3 3: %_3spa.ac3

The output is:
MKV, 1 video track, 3 audio tracks, 0:00:27, 24p
1: h264/AVC, 1080p24 (16:9)
2: TrueHD (Atmos), English, 7.1 channels, 48kHz
3: AC3, English, 5.1 channels, 640kbps, 48kHz
4: AC3, Spanish, 5.1 channels, 640kbps, 48kHz

NikosD

2nd June 2015, 16:14

Perfect, but I think next version is 1.16 :)

tebasuna51

3rd June 2015, 15:07

New release 1.1.7 (https://www.sendspace.com/file/ldaj94)

Changelog
=========
- v1.1.7 2015-06-02 MkvMux: if an audio is a thd+ac3 mux the two tracks.
Trim of timecodes.txt for VFR.
- v1.1.6 2015-03-31 Added parameters -dcadec, -arcsoft and OpusEnc

WARNING: -arcsoft parameter don't override the default dcadec for DTSHD 7.1 in eac3to v3.29
-dcadec override the default Arcsoft for others DTS.

NikosD

29th October 2015, 12:59

Hi.

After the new updates of MkvMerge, the button "MkvMergeGUI" doesn't work.
Maybe you could update "MkvMergeGUI" to link to mkvtoolnix-gui.exe ?

Also, what could be the fastest way to run and mux 3 audio files using UsEac3to ?

I have an mkv with 2 audio files (1 with dts-hd and 1 with AC-3) and I want to remux it to 3 audio files (1 with dts-hs, 1 with dts-hd -core and 1 with AC-3)

If I put to the CL parameters the 2:dts-hd, along with 2:dts-hd -core and the 3:AC-3 and press "Run and MkvMux" only 2 audio files (dts and AC-3) are muxed to the final MKV, not 3.

Is it possible to mux 3 audio files with UsEac3to ?

Thanks!

tebasuna51

29th October 2015, 19:30

After the new updates of MkvMerge, the button "MkvMergeGUI" doesn't work.
Maybe you could update "MkvMergeGUI" to link to mkvtoolnix-gui.exe ?
Yep, in next release.

I have an mkv with 2 audio files (1 with dts-hd and 1 with AC-3) and I want to remux it to 3 audio files (1 with dts-hs, 1 with dts-hd -core and 1 with AC-3)

The fastest way is extract the core (eac3to) and mux it to original mkv (mkvtoolnix-gui).

With UsEac3to "Run and MkvMux" you only can mux 2 audios.
Only if there are a track thd+ac3 the ouptput have the tracks thd and ac3 separated because mkv don't support one track thd+ac3.

For me mux the -core and the DTS-HD don't have sense, any player can extract the core from a DTS-HD if not support decode DTS-HD.

NikosD

29th October 2015, 19:42

Yep, in next release.

The fastest way is extract the core (eac3to) and mux it to original mkv (mkvtoolnix-gui).

Thanks for both.

Sneaker_ger told me that I could use mkvtools and select "reduce to core option" but I have to load the same file two times.

I think I have to try both.

For me mux the -core and the DTS-HD don't have sense, any player can extract the core from a DTS-HD if not support decode DTS-HD.

Unfortunately, my BD player - like many other BD players - doesn't agree to that, because decoding MKV rips via USB port is according to BD manufacturers a pirated copy, so it doesn't decode DTS-HD or Dolby EAC3/ TrueHD and it doesn't even decode the core DTS from inside DTS-HD (!)

Using the USB port, BD player can only decode DTS or AC3, so for someone who would like to see the movie and keep the original audio source for quality and for better BD players in the future, I think he has to do what I do.

tebasuna51

2nd November 2015, 00:44

New release 1.1.8 (https://www.sendspace.com/file/xtw6lc)

Changelog
=========
- v1.1.8 2015-11-02 Support for ffmpeg external encoder.
MkvToolNix-GUI instead MkvMergeGUI
Minor bug (features) and some defaults changed.

Remember update eac3to v3.30 (http://madshi.net/eac3to.zip)

NikosD

13th November 2015, 10:14

I have a MKV file with only TrueHD stream (without AC3 track) and tried to convert it to AC3 but I got this error:

[a02] Extracting audio track number 2...
[a02] Removing TrueHD dialog normalization...
[a02] Extracting audio track number 2...
[a02] Decoding with libav/ffmpeg...
[a02] Removing TrueHD dialog normalization...
[a02] This track begins with a non-major frame. <WARNING>
[a02] Remapping channels...
[a02] Encoding AC3 <640kbps> with libAften...
[a02] Initialization of the AC3 encoder failed. <ERROR>
Aborted at file position 262144. <ERROR>

It looks like a problem with the AC3 encoder (libaften) and not the TrueHD decoder.

Any other options/ quality encoders for TrueHD to AC3 conversion ?

nevcairiel

13th November 2015, 10:44

You could try ffmpeg itself, the ffmpeg ac3 encoder is based on libaften, by the same author, so quality should be comparable.

SeeMoreDigital

13th November 2015, 11:59

I have a MKV file with only TrueHD stream (without AC3 track) and tried to convert it to AC3 but I got this error:

After reading your post I thought I try this myself. Using my 'Gravity (Dolby Atmos).mkv' source: -

[a02] thd, 48000, 7.1
[a02] AC3 encoding doesn't support back channels. Will mix them into the surround.
[a02] Extracting audio track number 2...
[a02] Decoding with libav/ffmpeg...
[a02] Mixing surround channels...
[a02] Remapping channels...
[a02] Encoding AC3 <448kbps> with libAften...
[a02] Creating file "\\NAS\My HD Movies\Gravity (Special Edition) @ 91mins (8Ch_TrueHD Atmos+DD).mkv_2eng.ac3"...
[a02] Original audio track: max 24 bits, average 20 bits, most common 20 bits.
[a02] Processed audio track: max 24 bits, average 21 bits, most common 20 bits.
Video track 1 contains 130859 frames.
eac3to processing took 38 minutes, 30 seconds.
Done.

Everything went well for me...

NikosD

13th November 2015, 12:04

Using StaxRip (which uses eac3to) it worked for me too.

I'll try later eac3to directly, without GUIs and we 'll see.

It could be a bug of this GUI in the way it sets the commands to eac3to.

tebasuna51

13th November 2015, 12:35

...
[a02] This track begins with a non-major frame. <WARNING>
...
Any other options/ quality encoders for TrueHD to AC3 conversion?
Yep, I see this <WARNING> one time.

Now the preferred encoder to AC3 is ffmpeg.

1) Dowload the static versión (32 or 64 bits for your OS) from http://ffmpeg.zeranoe.com/builds/

2) Put only ffmpeg.exe in your eac3to folder (or in UsEac3to folder, or in your Encoders folder)

3) Extract the .thd from your .mkv

4) Load the .thd in UsEac3to, click in 'A/V Recode', select 'Recode to AC3' and 'RUN' or 'EnQueue'

ffmpeg recode your .thd to .ac3.

EDIT: same process recommended for DTS

With 7.1 sources you can use also the last BeHappy versión:
[1] Select LWLibavAudioSource like decoder (even from .mkv selecting the correct index)
[3] Downmix - 7.1 -> 5.1 (optimized downmix using AudioLimiter.dll)
[4] AC3 FFmpeg encode

tebasuna51

13th November 2015, 12:42

...
It could be a bug of this GUI in the way it sets the commands to eac3to.

If you put the full log of UsEac3to you can see the command line generated by UsEac3to.

Please tell me the differences with the command line than you use.

Please put always the full log.

NikosD

13th November 2015, 16:12

If you put the full log of UsEac3to you can see the command line generated by UsEac3to.

Please tell me the differences with the command line than you use.

Please put always the full log.

Well, it may sound strange and it definitely is, but there was no difference between StaxRip and UsEac3to commands to eac3to.

So, i did the only test left, i used the previous eac3to version that StaxRip uses v3.29.

And it worked with UsEac3to!

The next test will be to use eac3to v3.30 with StaxRip and test if it works (I doubt it)

Update:

OK, StaxRip worked even with eac3to v3.30 and libaften, so I put those two log files here:

StaxRip log:

eac3to v3.30
command line: "D:\various_progs\Video\Transcoders\Progs\GUI\StaxRip_x64_2015-10-24\Apps\eac3to\eac3to.exe" "D:\utorrent\Finished\...English.thd" "D:\utorrent\Finished\T..._out.ac3" -640 -normalize -progressnumbers
------------------------------------------------------------------------------
TrueHD, 5.1 channels, 48kHz, dialnorm: -27dB
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Writing WAV...
This track begins with a non-major frame. <WARNING>
Creating file "D:\utorrent\Finished\...ac3.pass1.wav"...
The original audio track has a constant bit depth of 16 bits.
Caution: The WAV file is bigger than 4GB. <WARNING>
Some WAV readers might not be able to handle this file correctly. <WARNING>
Starting 2nd pass...
Reading WAV...
Stripping zero bytes...
Remapping channels...
Encoding AC3 <640kbps> with libAften...
Applying -0.09dB gain...
Creating file "D:\utorrent\Finished\..._out.ac3"...
The processed audio track has a constant bit depth of 64 bits.
eac3to processing took 3 minutes, 34 seconds.
Done.

UsEac3to log:

eac3to v3.30
command line: "D:\various_progs\Video\Tools\...\eac3to.exe" "D:\utorrent\Finished\....mkv" 2: "D:\utorrent\Finished\....mkv_2eng.ac3" -progressnumbers -log="D:\various_progs\Video\Tools\UsEac3to118\UsEac3To.log"
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 18 subtitle tracks, 1:40:51, 24p /1.001
1: h264/AVC, English, 1080p24 /1.001
2: TrueHD, English, 5.1 channels, 48kHz, dialnorm: -27dB
3: Subtitle (PGS), English
4: Subtitle (PGS), French
5: Subtitle (PGS), Italian
6: Subtitle (PGS), Italian
7: Subtitle (PGS), Dutch
8: Subtitle (PGS), Spanish
9: Subtitle (PGS), Portuguese
10: Subtitle (PGS), Japanese
11: Subtitle (PGS), English
12: Subtitle (PGS), Japanese
13: Subtitle (PGS), French
14: Subtitle (PGS), Italian
15: Subtitle (PGS), Italian
16: Subtitle (PGS), Dutch
17: Subtitle (PGS), Spanish
18: Subtitle (PGS), Portuguese
19: Subtitle (PGS), Japanese
20: Subtitle (PGS), English
[a02] Extracting audio track number 2...
[a02] Removing TrueHD dialog normalization...
[a02] Decoding with libav/ffmpeg...
[a02] Remapping channels...
[a02] Encoding AC3 <640kbps> with libAften...
[a02] Initialization of the AC3 encoder failed. <ERROR>
Aborted at file position 262144. <ERROR>

tebasuna51

14th November 2015, 21:34

Problem solved with new eac3to v3.31.

RainyDog

1st December 2015, 10:12

Hi tebasuna, I finally succumbed to the Windows 10 upgrade but UsEac3to no longer works unfortunately.

Whenever I open it, an error message pops up stating that it can't find UsEac3to.cmd in the installation folder. If I just click OK I can then change the settings etc. but when I try to add an input source, the same error message pops up and the program freezes and needs killing in task manager.

Any ideas what could be causing it please? Is a fresh install of Windows 10 after doing the upgrade first and activating.

:thanks:

tebasuna51

1st December 2015, 14:34

Well in Autoit main window say:

Compatible with Windows XP / 2003 / Vista / 2008 / Windows 7 / 2008 R2 / Windows 8 / 2012 R2

But last versión say:

3.3.14.0 (10th July, 2015) (Release)
...
Added: @OSVersion supports Windows 10 and Windows Server 2016.

Try this one compiled with last version https://www.sendspace.com/file/g6iq27

Say me if work for you.

SeeMoreDigital

1st December 2015, 16:23

Hi tebasuna, I finally succumbed to the Windows 10 upgrade but UsEac3to no longer works unfortunately.

How strange...

I can confirm that I'm currently using UsEac3to v1.1.8 with eac3to v3.31 on a Windows 10 32-bit laptop and on a Windows 10 64-bit laptop without any issues.

Cheers

RainyDog

2nd December 2015, 21:30

Thanks guys.

Turns out it was Windows UAC that was blocking UsEac3to writing to it's program folder. Changed the folder security privileges and it worked fine. Or did for one encode because it's now stopped working again!

Whenever I try adding any input file I just get what looks to be eac3to's readme file in the main window :-

eac3to v3.31, freeware by madshi.net

- can show information about audio, video, VOB/EVO/(M2)TS and MKV files
- can decode and encode various audio formats
- can remove dialog normalization from AC3, E-AC3, DTS and TrueHD tracks
- can extract AC3 stream from Blu-Ray TrueHD/AC3 tracks
- can extract TrueHD stream from Blu-Ray TrueHD/AC3 tracks
- can extract DTS core from DTS-HD tracks
- can remove DTS zero padding and repair outdated DTS-ES headers
- can apply positive or negative audio delays
- can reduce bitdepth of decoded audio data by using TPDF dithering
- can resample decoded audio data (using SSRC or r8brain)
- can apply/reverse PAL speedup on decoded audio data (SSRC/r8brain)
- can demux video / audio tracks of EVO/VOB/(M2)TS and MKV sources
- can list available titles of Blu-Ray and HD DVD discs
- can extract Blu-Ray and HD DVD chapter information and subtitles
- can mux MPEG2, VC-1 and h264 video tracks to Matroska
- can remove pulldown flags from MPEG2, VC-1 and h264 video tracks

(there's more but I've cut it off for the sake of post length)

Any ideas again please? Thanks again.

tebasuna51

2nd December 2015, 22:51

In the main window?
Below the title "**EAC3TO INFO**"?

This content must be at a pop-up window named "Eac3to Help" when you select "? -> Help"

If you close UsEac3to and run again, you can open a new file?

I don't know what happen.

kypec

3rd December 2015, 08:43

Turns out it was Windows UAC that was blocking UsEac3to writing to it's program folder. Changed the folder security privileges and it worked fine. Or did for one encode because it's now stopped working again!
Just make sure that you execute UsEac3To with administrator privileges --> Run as administrator. I've been having similar problems (on Windows 7 actually) until I realized this. You can even set this option under Compatibility settings tab on your desktop or quick launch shortcut which starts UsEac3To.

RainyDog

3rd December 2015, 09:18

Just make sure that you execute UsEac3To with administrator privileges --> Run as administrator. I've been having similar problems (on Windows 7 actually) until I realized this. You can even set this option under Compatibility settings tab on your desktop or quick launch shortcut which starts UsEac3To.

Thanks kypec. Yes, I finally figured this out last night. The issues have all been down to that I put my eac3to folder (with UsEac3to inside) in my program files folder without installing them as such.

So it wasn't letting anything run without admin permissions. But I've set up a shortcut to run as admin now like you say.

tebasuna51

12th December 2015, 17:21

New version: UsEac3to 1.1.9

Changelog
=========
- v1.1.9 2015-12-12 SRT/.../TXT Added option to process all files with the same extension in the folder.

szabi

13th January 2016, 16:52

Hi

I would like to ask, where could any info be found about outputs?

http://abload.de/img/useac3to-outlkau2.png

Specially meaning of "ac3-ext" carry my interest.

bye

tebasuna51

13th January 2016, 23:04

What info?

ac3-ext is for use external encoder Aften, that allow use all parameters of encoder.

For instance -pad 0 cancel the standard 5 ms delay of AC3 encoders.
-exp 32 -s 1 maybe done better quality (more slow encode)

BTW the recommended encoder to AC3 is ffmpeg, for instance:
stdout.w64 | ffmpeg -i - -c:a ac3 -b:a 640k -center_mixlev 0.707 %_.ac3

szabi

15th January 2016, 08:55

Hi

Thanks fot this info.
If ffmpeg is recommended it means the first AC3 choice is selected from output list, it going to use ffmpeg, or "ac3-ffm" has to be selected?

bye

tebasuna51

15th January 2016, 12:50

If ffmpeg is recommended it means the first AC3 choice is selected from output list, it going to use ffmpeg, or "ac3-ffm" has to be selected?

The default eac3to is use libAften.dll.
If you want use a external encoder you need select it, "ac3-ffm" in this case.

Remember than external encoder (standalone ffmpeg.exe here) are searched by UsEac3to in this order:
1) In UsEac3to folder
2) In "Encoders folder" defined in Settings
3) In eac3to folder

szabi

16th January 2016, 08:58

Ok, thanks for all. :)

szabi

16th January 2016, 17:38

Hi

Now I checked it, this is the auto created command line.
stdout.w64 | ffmpeg -i - -c:a ac3 -b:a 640k -center_mixlev 0.707 -surround_mixlev 0.707 %_.ac3
My source is 5.1 DTS-HD MA.
Why center and sourrond mixlev are added to it?
How can ffmpeg be used for Dolby Digital Plus encode?

bye
szabi

tebasuna51

16th January 2016, 19:17

Why center and sourrond mixlev are added to it?

http://forum.doom9.org/showthread.php?p=1752561#post1752561

How can ffmpeg be used for Dolby Digital Plus encode?
Modify the line to:

stdout.w64 | ffmpeg -i - -c:a eac3 -b:a 448k %_.eac3

szabi

19th January 2016, 17:42

Hi

Thanks for this too. ;)
I was confused by this "mixlev" switch, believing that center and surround channel will be encoded by this gain.
Now I see gain info only stored in AC3 header as a message to the decoder "apply it in case of stereo downmix".

Concerning Dolby Digital Plus, I will check whether 7.1 channel from DTS-HD MA can be encoded to lossy EAC3 correctly.

EDIT: Now It was tested.
Input 4887kbps@7.1ch.
This switch used:
stdout.w64 | ffmpeg -i - -c:a eac3 -b:a 1344k %_.eac3
Output 1344kbps@5.1ch.
I lost the extra channels.

bye
szabi

szabi

2nd February 2016, 21:35

Hi

Are there any way to keep extra channels?

bye

tebasuna51

2nd February 2016, 22:19

Seems than ffmpeg EAC3 encoder don't support 7.1

szabi

3rd February 2016, 17:09

Thanks for the info.
Seems you have answer all of my question. :)
Any suggestion of my existence. :D
Thanks again.

bye
szabi

szabi

23rd June 2016, 07:48

Hi

Nowadays I get the following alert for the usage of "UsEac3to.exe".
EID_autoit_heur is a potential threat (http://www.avgthreatlabs.com/ww-en/virus-and-malware-information/content/generic-virus/?name=@EID_autoit_heur&utm_source=TDPU&utm_medium=RS&PRTYPE=PROT)
Do u have any idea for reason?

bye

tebasuna51

23rd June 2016, 08:33

Well, any .exe file can contain virus, for that I upload also the source file (.au3) that can be interpreted or compiled with the free tools suplied in https://www.autoitscript.com/site/autoit/

You can read the source code to see than not exists malicious code inside, and you need trust than the autoit compiler don't introduce malicious code also (don't exist warnings about that).

This antivirus message is a joke because can be applied to any .exe file in your system.

szabi

25th June 2016, 13:37

Hi

I believe in you.
It is added now to the exception list.

bye

romanturbo

11th July 2016, 22:01

Hi, I got a new problem this time.
It seems like useac3to only works if I encode on my C: disk, if i want to encode on my external disk L, it got stuck after processing verything and starting the conversion job, at 1%.

I think it may be something related with administration rights, is there anyway to fix it?

All the programms are in C disk

kypec

12th July 2016, 07:09

I think it may be something related with administration rights, is there anyway to fix it?

Try to run UsEac3to.exe as administrator and you'll see if it helps.

romanturbo

12th July 2016, 20:16

Try to run UsEac3to.exe as administrator and you'll see if it helps.

didnt work, because I think that the other programs like the muxer (useac3to just sends the command to the program) gotta be running in admin mode too, the only thing that runs in admin mode is the interface.

Its not a critical bug, because i can still work extracting the audios with mkvextract and then converting with useac3to, but i just wanted to comment it.

BTW: I hate the random question lol

locotus

4th August 2016, 17:50

Just run into this running UsEac3to v.1.19:

Source file "C:\Downloads\En proceso\DC Audio espa-ñol.ac3" not found. <ERROR>

Just after changing ñ for n:

eac3to v3.27
command line: "C:\MeGui\eac3to\eac3to.exe" "C:\Downloads\En proceso\DC Audio espanol.ac3" -progressnumbers -log="C:\MeGui\tools\eac3to\UsEac3To.log"
------------------------------------------------------------------------------
AC3, 1.0 channels, 1:39:46, 192kbps, 48kHz, dialnorm: -27dB

tebasuna51

5th August 2016, 00:13

Source file "C:\Downloads\En proceso\DC Audio espa-ñol.ac3" not found. <ERROR>
Without problems here:

eac3to v3.31
command line: "D:\Programa\eac3to\eac3to.exe" "D:\tmp\Audio español.ac3" -progressnumbers -log="D:\Programa\eac3to\UsEac3to\UsEac3To.log"
------------------------------------------------------------------------------
AC3, 5.1 channels, 0:00:20, 640kbps, 48kHz

Maybe you have a extrange character, showed here like '-ñ'
I can't guarantee than UsEac3to work fine with all languages but yes with spanish, my native language.

Try rename the original file changing the character by a typed 'ñ' and say me if work now.

locotus

5th August 2016, 00:28

Without problems here:

Maybe you have a extrange character, showed here like '-ñ'
I can't guarantee than UsEac3to work fine with all languages but yes with spanish, my native language.

Yeah, I know that, mine too. But since then do some tests changing audionames including and ñ to see what happens and get the same result.

Seems a silly bug, maybe in Eac3to, but is good to know so nobody break his head with it.

Thanks anyhow.

tebasuna51

5th August 2016, 02:41

New version v1.2.0 (https://www.sendspace.com/file/w5gt5n)

Changelog
=========
- v1.2.0 2016-08-05 Aften.exe deprecated by ffmpeg.exe. See Audio recode.
Added Opus encoder and BeSplit for dtswav.
Option to enqueue Run and MkvMux
Update MkvExtract options (all ATT or SUP)

szabi

11th August 2016, 20:59

Hi

I got this error message:
v02 Muxing video to Matroska...
v02 Getting "Haali Matroska Muxer" instance failed.

Do I need to install Haali Media Splitter (https://haali.su/mkv/)?

bye

tebasuna51

11th August 2016, 22:30

Is a eac3to requirement:

For video muxing you need:
(1) Haali Matroska Muxer

BTW you can extract elementary video stream: h264, vc1,m2v

szabi

24th August 2016, 15:40

Hi

You are right.
I was just little confused to install 3 years old Haali Media Splitter (https://haali.su/mkv/).
Nowadays LAV splitter is preferred.
Thanks.

Bye

tebasuna51

15th September 2016, 19:55

New version v1.2.1 (https://www.sendspace.com/file/9nukog)

Changelog
=========
- v1.2.1 2016-09-15 Improved 'A/V Recode':
Added ffmpeg MAP to select track in container.
Added option to override encoder parameters.

More info in Readme, for instance now we can do:

You have a mkv with the second audio track:

3: EAC3, English, 5.1 channels, 48kHz, 678ms

and you want recode to AC3 and apply the delay

1) Put in COMMAND LINE PARAMETERS: -af "adelay=678|678|678|678|678|678"
2) Click in 'A/V Recode'
3) Select the AC3 desired bitrate
4) Select the 'ffmpeg map' to 2
5) Select 'Use Commad Line Parameters' to 'Add to def'
6) 'Run' or 'EnQueue' the job

SeeMoreDigital

15th September 2016, 20:55

Many thanks for your continuing support :)

tebasuna51

16th September 2016, 04:02

Sorry, a little bug, reupload the new version.

MaxitoHP

23rd September 2016, 19:20

Thank you very much, Tebasuna. Great job, as always.

szabi

27th September 2016, 20:26

Hi

I want to join two AC3.
I open the 1st one and after click "+Sources".
Instead of adding 2nd AC3 the following failure message received "Don't exist a file"
What am I doing wrong?

EDIT: Ok, I found it: CLICK (http://forum.doom9.org/showpost.php?p=1713375&postcount=20http://forum.doom9.org/showpost.php?p=1713375&postcount=20)
I forget this naming requirement.
Again. :)

bye

tebasuna51

6th October 2016, 11:58

New version v1.2.2 (https://www.sendspace.com/file/1n7428)

Changelog
=========
- v1.2.2 2016-10-02 Added -drc_scale 0 to ffmpeg (to decode AC3 or E-AC3)

Now ffmpeg AC3/EAC3 decoder don't apply DRC like eac3to do.

If track input is not a AC3/EAC3 you can safely ignore the yelow (WARNING) ffmpeg message:
Codec AVOption drc_scale (percentage of dynamic range compression to apply) specified for input file #0 (D:\Temp\test.mkv) 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.

romanturbo

18th November 2016, 15:24

ive seen u implemented the queue , as well now truehd encoding is also working, awesome work!! thx for your effort

SquallMX

23rd November 2016, 21:20

I´m trying to remove DN from an E-AC3 file, but I keep getting error codes:

E-AC3, 7.1 channels, 1:26:28, 768kbps, 48kHz, dialnorm: -27dB
(core: AC3, 5.1 channels, 1:26:28, 448kbps, 48kHz, dialnorm: -27dB)
Removing AC3 dialog normalization...
Applying (E-)AC3 delay failed. <ERROR>
Aborted at file position 262144. <ERROR>

tebasuna51

23rd November 2016, 21:32

Hello SquallMX!

What is the command line?
Because the problem seems related with delay:

Applying (E-)AC3 delay failed.

and the info not show any delay.

Can you upload a sample of a few MB?

SquallMX

23rd November 2016, 21:52

Hello SquallMX!

What is the command line?
Because the problem seems related with delay:

Applying (E-)AC3 delay failed.

and the info not show any delay.

Can you upload a sample of a few MB?

Of course:

eac3to v3.31
command line: "D:\Archivos de Programa\EAC32AC3\eac3to.exe" "D:\Spanish.ac3" "D:\Spanish.ac3_.eac3" -progressnumbers -log="D:\Archivos de Programa\EAC32AC3\UsEac3To.log"

https://1drv.ms/u/s!AiCgDguh7E_OxTzONBGLTcmHnCop

I had to extract the audio using TSMuxerGUI "Demux" option, since eac3to failed to do so.

tebasuna51

24th November 2016, 03:35

Yep, I can reproduce the problem.

Reported to madshi http://bugs.madshi.net/view.php?id=450

tebasuna51

21st May 2017, 18:40

New version 1.2.3 (https://www.sendspace.com/file/sq9gae)

Changelog
=========
- v1.2.3 2017-05-21 The exe (or link) support Drag & Drop.
Extract Info from mkv's.
Audio Recode use ffmpeg if don't found a proper decoder.
Audios cut with Trim's of .avs
Offset in Delays of srt's, ass, etc.

SeeMoreDigital

21st May 2017, 19:16

Bloody marvellous :)

tebasuna51

10th June 2017, 20:33

New version 1.2.4 (https://www.sendspace.com/file/6n8isk)

Changelog
=========
- v1.2.4 2017-06-10 Video recode: x265 support using avs4x26x.exe
Audio recode: Decode to W64 (always ffmpeg) or WAV.
Encode to EAC3 (ffmpeg), bitrate with prefix 'E'
Merge mono-channels WAV with sufix L,R,C,LFE,...

SeeMoreDigital

11th June 2017, 16:13

Thanks again :)

SquallMX

13th June 2017, 22:25

SeeMoreDigital

14th June 2017, 09:45

I've just run the newest build through Windows Defender and it's reported this: -

http://i64.tinypic.com/2qvsf48.png

tebasuna51

14th June 2017, 11:46

@SquallMX
@SeeMoreDigital

Thanks for your reports.

I have here W 7 and Windows Defender without problems.

Please check if that exe run without problems: https://www.sendspace.com/file/t75su0

If not, you can try compile yourself the source (.au3) using the free tool Autoit3 from here (recommended Download Zip without install): https://www.autoitscript.com/site/autoit/downloads/

Like you can see Autoit3 seems stop their development in September 2015 and don't mention W 10 like OS supported.

- The source is free of trojan/virus like you can see.
- The Autoit3 compiler seems also free of trojan/virus (I don't see any report about that since 2015)
- My machine seems also free of trojan/virus (I hope, but if your compile work please report me!)

After that only rest a false positive or a W 10 issue.

SeeMoreDigital

14th June 2017, 15:52

tebasuna51

14th June 2017, 20:45

Sorry, I was using a previous version to compile, with the last AutoIt v3.3.14.2 seems there are some changes to avoid malware false positives.

@SquallMX
Work for you?

tebasuna51

15th June 2017, 03:02

New version 1.2.5 (https://www.sendspace.com/file/fzs1ly)

Changelog
=========
- v1.2.5 2017-06-15 Audio recode: decoder admit parameters change (for ffmpeg).
INF: ffmpeg report of tracks in any container.
Can extract audio tracks.

I hope than this new version don't have malware false positives.

SquallMX

15th June 2017, 05:18

Sorry, I was using a previous version to compile, with the last AutoIt v3.3.14.2 seems there are some changes to avoid malware false positives.

@SquallMX
Work for you?

Everything working fine, thank you!

tebasuna51

19th June 2017, 14:23

New version v1.2.6 (https://www.sendspace.com/file/kx39ro)

- v1.2.6 2017-06-19 Bug fix: Autoit v3.3.14.2 write files by default like UTF-8
Before are writed like ANSI. Corrected.

Filenames with spanish characters (ñ's, accents, etc.) are accepted now.

Also affect to modified srt's.

SeeMoreDigital

19th June 2017, 15:18

Hi tebasuna51,

Windows defender is reporting a malware issue with this new version ;)

tebasuna51

20th June 2017, 03:39

I used the same compiler than 1.2.5 changing only the mode to write files to ANSI.

sk2316

11th July 2017, 20:17

Hi tebasuna51,
I try to add core ac3 to thd but error
what wrong?
http://imageshack.com/a/img922/4397/k59Pzg.png

tebasuna51

11th July 2017, 22:09

Maybe is the samplerate (192 KHz).

I don't have a sample like this to test.
But work fine with a:

TrueHD, 7.1 channels, 96kHz
AC3 encoding doesn't support back channels. Will mix them into the surround.
AC3 encoding doesn't support this samplerate. Will resample to 48kHz.
Decoding with libav/ffmpeg...
Resampling to 48kHz...
Mixing surround channels...
Remapping channels...
Encoding AC3 <640kbps> with libAften...
Creating file "D:\tmp\test_96.thd_.thd+ac3"...

Try adding -resampleTo48000

EDIT: I deleted your post in eac3to thread, please remember the forum rule :

8) No cross posting. Post your message once, to the appropriate forum and nowhere else or it will be locked or deleted without warning.

tebasuna51

12th October 2017, 13:28

New version v1.2.7 (https://www.sendspace.com/file/ivk136) to support eac3to v3.32

Changelog
=========
- v1.2.7 2017-10-12
Support eac3to v3.32 (UHD BD's y HEVC)
Force ANSI to read text files.
TwoLame replaced by ffmpeg to encode MP2.

SeeMoreDigital

12th October 2017, 14:06

Thanks :D

Taurus

12th October 2017, 15:17

:thanks:

tebasuna51

16th October 2017, 11:58

New version 1.2.8 (https://www.sendspace.com/file/l0nkez) to support MkvToolnix v17.0.0

Changelog
=========
- v1.2.8 2017-10-16 Support MkvToolNix v17.0.0. New MkvExtractGUI.

Don't work with previous MkvToolNix versions because the mkvextract.exe have different syntax, before:

mkvextract {mode} {source-filename} options extraction-spec

Now:

mkvextract {source-filename} {mode1} options extraction-spec1 mode2 options extraction-spec2 …

With mode: tracks, timestamps_v2, chapters, attachments
Now we can use a unique job for all modes.

SeeMoreDigital

30th October 2017, 20:30

New version 1.2.8 (https://www.sendspace.com/file/l0nkez) to support MkvToolnix v17.0.0

Is it just me or is there an issue opening .mp4 contained files?

Cheers

tebasuna51

30th October 2017, 23:41

eac3to don't support mp4 container.

You can try 'MkvExtract/INF' to extract tracks with ffmpeg or 'A/V Recode' to decode/recode audio.

SeeMoreDigital

31st October 2017, 09:54

eac3to don't support mp4 container...Well that explains it then...

I could have swore I've used it before to de-mux streams out of the .mp4 container....

Alzheimer's :scared:

staygroovy

19th November 2017, 19:28

Just wanted to say thank you to tebasuna for this GUI, I just started using it and it makes things much simpler compared to having to enter everything on the command line.

SeeMoreDigital

19th November 2017, 19:31

Just wanted to say thank you to tebasuna for this GUI, I just started using it and it makes things much simpler compared to having to enter everything on the command line.Agreed... I love it :)

ripoyet

3rd December 2017, 22:04

Thanks tebasuna. I hope that it runs with the last update of eac3to. Thanks for all.

tebasuna51

4th December 2017, 02:02

Yes, and with MkvToolNix v17 or greater.

fAy01

3rd February 2018, 20:59

Thank you!

deltec

5th February 2018, 20:47

Thank for this easy to use gui I use it to change an RF64 24/192khz file to w64 but at the end I have an RF64 file, and I can see it on prompt windows when program start to demux, why?

tebasuna51

5th February 2018, 22:47

I can't reproduce your problem:

eac3to v3.34
command line: "D:\Programa\eac3to\eac3to.exe" "D:\Test\test.rf64" "D:\Test\test.rf64_.w64"
------------------------------------------------------------------------------
RF64, 5.1 channels, 0:00:20, 24 bits, 27648kbps, 192kHz
Reading RF64...
Writing W64...
Creating file "D:\Test\test.rf64_.w64"...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 1 second.
Done.

command line: "D:\Programa\eac3to\eac3to.exe" "D:\Test\test.rf64_.w64"
------------------------------------------------------------------------------
W64, 5.1 channels, 0:00:20, 24 bits, 27648kbps, 192kHz

Put your log file.

tebasuna51

12th June 2018, 01:37

New versión 1.2.9 (https://www.sendspace.com/file/6hln3c)

Some bug corrected and improved 'A/V Recode' to make easy filter and recode audio with ffmpeg.

Some samples of audio filters:

Downmix: -af "pan=stereo|FL<FC+0.70*FL+0.30*SL|FR<FC+0.70*FR+0.30*SR" [or other]
Resample: -af aresample=48000
Speedup: -af atempo=1.042708 [23.976->25], 1.041667 [24->25], 1.001 [23.976->24]
Slowdown: -af atempo=0.959041 [25->23.976], 0.96 [25->24], 0.999001 [24->23.976]
Delay: -af adelay=1500|0|500|0|500|500 [one delay for each channel]
Trim: -filter_complex "[0]atrim=2:8[a];[0]atrim=12:18[b];[a][b]concat=n=2:v=0:a=1"

And a sample of use in the image:

1) Load a mp4 in UsEac3to, ignore the non supported message.
2) Click at 'MkvExtract/INF' and ffmpeg inform:
3) The mp4 tracklist. We need recode the aac 44.1 KHz to 48 KHz and half volume.
4) We can extract the aac track, but is not necesary and close the dialog.
5) We put the ffmpeg filter: -af "volume=0.5, aresample=48000"
6) Click at 'A/V Recode' to open the dialog.
7) Select track 1 in 'map:X' or other like we see in 3)
8) Select 'Add to DEC' to add the filter.
9) Select ENCODER and quality.
10) 'Run' or 'EnQueue'

SeeMoreDigital

12th June 2018, 09:08

Thank-you very much indeed :)

Taurus

12th June 2018, 11:33

:thanks:Thanks a lot dear tebasuna51.:thanks:

szabi

23rd July 2018, 19:41

Hi

I tried to encode 8ch audio to E-AC3 format: CLICK (https://forum.doom9.org/showpost.php?p=1785382&postcount=14090)
It was almost two years ago.
Since that some version update occured I did try it again.
UsEac3To (v1.2.9)
eac3to (v3.34)
ffmpeg (ffmpeg-20180723-d134b8d-win64-static)
Command line parameter:
stdout.w64 | ffmpeg -i - -c:a eac3 -b:a 1120k %_.eac3

Encoder
Input #0, w64, from 'pipe:':
Duration: N/A, bitrate: 9216 kb/s
Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 7.1, s32 (24 bit), 9216 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (pcm_s24le (native) -> eac3 (native))
Output #0, eac3, to 'E:\my audio.eac3':
Metadata:
encoder : Lavf58.17.101
Stream #0:0: Audio: eac3, 48000 Hz, 5.1(side), fltp (24 bit), 1120 kb/s
Metadata:
encoder : Lavc58.21.106 eac3

I still get 5.1 output, loosing extra channels.
What am I doing wrong?

bye

sneaker_ger

23rd July 2018, 20:01

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

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

szabi

23rd July 2018, 20:07

Linked topic almost 1 year old.
Seems no improvement since. :(

SeeMoreDigital

23rd July 2018, 20:40

Linked topic almost 1 year old.
Seems no improvement since. :(
As far as I know, nobody has been able to create a 'freeware' E-AC3 encoder yet.

szabi

25th July 2018, 10:01

I did not know that.

gonca

11th November 2018, 15:53

Sorry to open an old thread
If I use a playlist from a BD (say 800.mpls) as input and then demux the streams, their names all start as 800xxx.avc, etc
Is there a way to have the names be that of the parent folder, I guess I need to somehow edit the cmd line that USeac3to creates for eac3to

tebasuna51

11th November 2018, 18:20

I'm not sure what do you want. Maybe...

When you put in COMMAND LINE PARAMETERS:

%_BDNAME.*

and click in 'RUN CL' button you obtain:

eac3to v3.34
command line: "D:\Programa\eac3to\eac3to.exe" "D:\Temp\bd_name\BDMV\PLAYLIST\00011.mpls" 1) "D:\tmp\00011.mpls_BDNAME.*" -progressnumbers -log="D:\Programa\eac3to\UsEac3to\UsEac3To.log"
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 0:00:26, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: AC3, English, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[v01] Creating file "D:\tmp\00011.mpls_BDNAME - 1 - h264, 1080p24.h264"...
[a02] Creating file "D:\tmp\00011.mpls_BDNAME - 2 - AC3, English, 5.1 channels, 640kbps, 48kHz.ac3"...
Video track 1 contains 616 frames.
eac3to processing took 1 second.
Done.

gonca

11th November 2018, 18:25

Creating file "D:\tmp\00011.mpls_BDNAME - 1 - h264, 1080p24.h264"...

make it read
Creating file "D:\tmp\BDNAME - 1 - h264, 1080p24.h264"...
but your example might work well, thank you

richardpl

11th November 2018, 19:10

As far as I know, nobody has been able to create a 'freeware' E-AC3 encoder yet.

I can, but not for free.

SquallMX

14th November 2018, 22:10

Any workaround for removing diagnorm on EAC3 files?

tebasuna51

14th November 2018, 22:42

You can remove DN to 5.1 eac3:

E-AC3, 5.1 channels, 0:00:34, 1536kbps, 48kHz, dialnorm: -27dB

eac3to 51chan_dn.eac3 51chan.eac3

E-AC3, 5.1 channels, 0:00:34, 1536kbps, 48kHz

But to 7.1.eac3 only can renove the DN from the core:

E-AC3, 7.1 channels, 0:01:23, 768kbps, 48kHz, dialnorm: -27dB
(core: AC3, 5.1 channels, 0:01:23, 448kbps, 48kHz, dialnorm: -27dB)

eac3to 71chan_dn.eac3 71chan.eac3 -> abort
eac3to 71chan_dn.eac3 71chan.eac3 -core

AC3, 5.1 channels, 0:01:23, 448kbps, 48kHz

SeeMoreDigital

19th November 2018, 22:14

Hi tebasuna51,

Windows Defender has just reported the following: -

https://i.imgur.com/CYLrk2p.png

Cheers

Overdrive80

8th February 2019, 00:46

Hi, I need save audio files on 32 bitdepth, I think that only ALAC audio encoder supported that bit depth (compression audio). How I could using ALAC with Useac3to? Thanks in advance, Teba.

EDIT: Maybe?--> stdout.wav | refalac - -o audio.m4a

tebasuna51

8th February 2019, 02:57

If refalac.exe is in the eac3to/UsEac3to/defined path the sintax is correct.

But I don't know the source, and by default eac3to output stdout.wav in 24 bits int, you can try

stdout.wav -down32 | refalac - -o audio.m4a

To obtain a 32 bits int.
From lossy decoders you can obtain until 64 bits float with the parameter -full, for instance:

eac3to input.ac3 output64float.wav -full

I don't know how obtain 32 bits float.

Overdrive80

8th February 2019, 19:49

The source is 32 bit float but, refalac convert directly to 24 bit integer. I think that ALAC does not support 32 bit depth. Any other lossless encoder with support for 32 bit? Thanks.

tebasuna51

9th February 2019, 11:02

The float audio samples are only usefull to operate with them (mix, amplify, etc.), after that must be normalized and converted to int samples before to be played (to avoid overflow).

The precission of 32 float (24 bits of mantissa) is, more or less, the same than 24 int. A float format must be converted always to int (normalize first) to lossless encode.

I never see a source of 32 bit float lossless, maybe is a decode of a lossy format.

Overdrive80

9th February 2019, 14:29

Eleonore

17th April 2019, 16:42

hello , sorry my english...

Windows 10 Pro, user Paweł (Polish letters)
UsEac3To choose the test and show me:
"C:\Users\Paweü\Desktop\eac3to.exe" -test -progressnumbers -log="C:\Users\Paweü\Desktop\UsEac3to\UsEac3To.log"
Processing Info... Please wait or Ctrl+C over CMD window.

The problem is that instead of Paweł, I have Paweü
Can you fix it somehow? so that the program would support Polish letters?

Now, when I open UsEac3 it does not work, the program does nothing ...

regards Eleonore

tebasuna51

17th April 2019, 21:50

Sorry Eleonore I can't do anything about that.
The unique workaound is create folders to install and work UsEac3to without special letters.
C:\Portable\eac3to\UsEac3to...

orisvl

7th July 2019, 17:17

My os drive is C:. When I demux to mkv (using Haali Media Splitter) and select output folder not on c drive (D:\... for ex) it creates a few mb mkv file and the process gets stuck.
If i choose C:\ it creates files in appdata\local\VirtualStore. If i choose C:\...\ it works correctly.
I'm using eac3to 3.34 with UsEac3to GUI 1.2.9
Any thoughts?

tebasuna51

7th July 2019, 21:56

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

8th July 2019, 00:12

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

1st November 2019, 17:07

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

1st November 2019, 17:15

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

manolito

1st November 2019, 18:50

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

2nd November 2019, 10:28

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

3rd November 2019, 21:26

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

4th November 2019, 12:15

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

29th November 2019, 13:55

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

29th November 2019, 22:26

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

22nd December 2019, 23:18

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

23rd December 2019, 11:28

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

Natto

23rd December 2019, 11:44

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

23rd December 2019, 11:57

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

23rd December 2019, 12:20

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

23rd December 2019, 23:11

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

28th December 2019, 11:57

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

28th December 2019, 20:50

... 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

29th December 2019, 11:57

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

29th December 2019, 20:42

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

mgutt

13th February 2020, 02:58

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

13th February 2020, 10:25

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

13th February 2020, 11:29

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

13th February 2020, 23:08

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

14th February 2020, 01:58

... 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

31st March 2020, 18:44

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

1st April 2020, 09:31

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

1st April 2020, 10:02

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

1st April 2020, 11:08

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

1st April 2020, 11:32

tebasuna51

1st April 2020, 11:37

SeeMoreDigital

1st April 2020, 13:18

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

1st April 2020, 16:54

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

1st April 2020, 19:03

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

1st April 2020, 19:29

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

5th April 2020, 18:14

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

SeeMoreDigital

5th April 2020, 18:25

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

jlw_4049

5th April 2020, 18:29

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

6th April 2020, 11:11

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

6th April 2020, 17:17

tormento

19th April 2020, 10:04

tebasuna51

19th April 2020, 11:23

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

20th April 2020, 14:26

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

2nd May 2020, 10:02

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

tebasuna51

2nd May 2020, 11:19

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

10th May 2020, 00:44

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

tebasuna51

10th May 2020, 00:52

kdantas

10th May 2020, 02:18

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

10th May 2020, 10:14

kdantas

10th May 2020, 13:06

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?

tebasuna51

10th May 2020, 16:58

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

szabi

2nd June 2020, 17:28

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

3rd June 2020, 00:16

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

szabi

4th June 2020, 19:35

SeeMoreDigital

4th June 2020, 21:15

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

8th June 2020, 07:15

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

SeeMoreDigital

8th June 2020, 09:57

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

8th June 2020, 11:45

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

8th June 2020, 20:50

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

9th June 2020, 09:48

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

tormento

9th June 2020, 17:06

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

9th June 2020, 17:31

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

16th June 2020, 20:21

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

17th June 2020, 00:44

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

17th June 2020, 09:12

@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

3rd August 2020, 11:36

SeeMoreDigital

10th August 2020, 21:05

Has anyone been able to download this new version?

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

tebasuna51

11th August 2020, 00:03

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

LUCHOO

11th August 2020, 13:16

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

11th August 2020, 13:58

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

11th August 2020, 18:05

thanks tebasuna ...

You are welcome.

How are you, all ok?

LUCHOO

12th August 2020, 12:53

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

12th August 2020, 17:45

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

12th August 2020, 18:50

Samsung UE40NU7125 (not very new, from 2018).

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

roisiano

19th November 2020, 16:09

Thanks again for this excellent gui, tebasuna51.

tebasuna51

26th November 2020, 15:01

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

30th November 2020, 23:28

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

1st December 2020, 01:03

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

1st December 2020, 02:28

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

1st December 2020, 20:27

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

tebasuna51

1st December 2020, 21:46

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

1st December 2020, 21:59

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

2-perf

1st December 2020, 22:22

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

1st December 2020, 22:28

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

2nd December 2020, 16:54

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

18th January 2021, 05:07

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

18th January 2021, 19:55

@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

11th February 2021, 19:44

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

21st February 2021, 00:51

tebasuna51

21st February 2021, 12:50

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

26th May 2021, 15:39

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

26th May 2021, 21:07

Hi

Excellent tebasuna .. Thanks!

tebasuna51

26th May 2021, 23:20

You are welcome LUCHOO, nice to read you here.

wolflop

27th May 2021, 11:18

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

tebasuna51

27th May 2021, 18:22

@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

9th August 2021, 08:37

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

9th August 2021, 09:36

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

9th August 2021, 09:44

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

9th August 2021, 10:29

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

9th August 2021, 11:32

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

how can i do this please?

Axeldook

9th August 2021, 13:39

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

9th August 2021, 15:32

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

tebasuna51

9th August 2021, 23:31

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

10th August 2021, 08:06

Axeldook

10th August 2021, 09:03

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

10th August 2021, 21:13

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

gonca

11th August 2021, 00:57

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

11th August 2021, 09:33

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

Axeldook

18th August 2021, 09:37

Hi,

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

tebasuna51

18th August 2021, 12:20

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

18th October 2021, 18:31

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

18th October 2021, 22:35

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

31st October 2021, 13:13

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

9th November 2021, 21:38

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

10th November 2021, 11:14

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

karbre

25th November 2021, 23:26

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

26th November 2021, 00:44

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

26th November 2021, 03:36

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

26th November 2021, 11:45

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

18th December 2021, 20:10

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

30th December 2021, 11:05

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

30th December 2021, 22:50

If you put in COMMAND LINE PARAMETERS:

%_.*

You can EnQueue the demux job.

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

DanDare1983

3rd January 2022, 14:24

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

5th January 2022, 00:19

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

6th January 2022, 22:11

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

7th January 2022, 09:09

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

10th January 2022, 21:34

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

14th January 2022, 13:50

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

8th February 2022, 14:26

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

8th February 2022, 14:35

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

DanDare1983

8th February 2022, 16:05

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

8th February 2022, 16:11

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

8th February 2022, 16:26

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

8th February 2022, 21:26

1sec 1ms = 1001 ms

DanDare1983

16th February 2022, 12:36

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

16th February 2022, 19:46

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

DanDare1983

24th February 2022, 12:27

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

24th February 2022, 20:02

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

24th February 2022, 23:42

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

25th February 2022, 03:22

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

DanDare1983

26th February 2022, 21:36

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

26th February 2022, 22:09

... 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

4th March 2022, 19:26

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

4th March 2022, 20:57

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

5th March 2022, 09:58

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

8th March 2022, 14:31

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

8th March 2022, 22:46

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

9th March 2022, 13:04

Thanks for the thdmerge support.

von Suppé

16th March 2022, 07:44

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

16th March 2022, 10:27

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]
...

arestarh

17th May 2022, 13:44

Hi tebasuna51,

First of all thanks for great tool LeeAudBi.

I've tried to execute latest compiled binary from https://forum.doom9.org/showthread.p...067#post424067 and got next error:
https://www.imagebam.com/view/MEAIFYP

I've tried to fix that error and changed only 2 lines in LeeAudBi7.au3 and uploaded all content as one package (in case you need it):
https://www.mediafire.com/file/q0myv...dified.7z/file

The content of the package is:
old - directory with src files provided by you
patch - just diff file
LeeAudBi7.au3 - modified script
diff.htm - just diff in .html
LeeAudBi7.ico - icon extracted from compiled binary LeeAudBi7.exe

I've commented line related to LeeAudBi7.jpg, because when I've tried to compile the LeeAudBi7.au3 I've got an error regarding absence of LeeAudBi7.jpg

I hope that my little fix is correct one.
If it is the wrong thread to post this massage then my apologies.

tebasuna51

18th May 2022, 11:38

Thanks arestarth

You are right.
The first one (the jpg) is unnecesary and only can cause troubles when compile. Can be commented (or deleted) from source.

Both changes will make in sources in next release LeeAudBi8

writemaster

18th May 2022, 23:19

tebasuna51

19th May 2022, 13:40

Will E-AC3 support come in LeeAudi8?

? -> About and Help

DanDare1983

31st May 2022, 14:51

Hi,

I was just wondering if you could clear a couple of things up for me.

I've read that eac3to doesn't support atmos tracks, would that be atmos within uhd disks or atmos within bluray and uhd?

Also I was queueing up some movies for demux and one of them which had an E-AC3 file refused to demux. All other movies without the E-AC3 demuxed without problem.

Regards

Dan

tebasuna51

31st May 2022, 19:08

I've read that eac3to doesn't support atmos tracks, would that be atmos within uhd disks or atmos within bluray and uhd?
What is the difference?

Also I was queueing up some movies for demux and one of them which had an E-AC3 file refused to demux. All other movies without the E-AC3 demuxed without problem.

It is a know problem. Demux track by track and the eac3 track with the -keepDialnorm parameter.

DanDare1983

1st June 2022, 16:26

Hey Tebasuna51,

Thankyou for for your quick response, in regards to the atmos track I was just asking if theirs any problems demuxing with eac3to as I've read that atmos tracks can have problem when demuxing and that makemkv is the only option when it comes to demuxing.

Also in regards to the other question, so basically I can't do a straight -demux at all when an EAC-3 exists and I have to manually select what files to demux opposed to doing a straight demux? And if I am to select the eac-3 track I need to select eac-3 with -keepdialnorm?

Regards

Dan

tebasuna51

1st June 2022, 21:48

..in regards to the atmos track I was just asking if theirs any problems demuxing with eac3to as I've read that atmos tracks can have problem when demuxing and that makemkv is the only option when it comes to demuxing.
I don't see problems with Atmos tracks, use always -keepdialnorm to extract eac3 tracks.

Also in regards to the other question, so basically I can't do a straight -demux at all when an EAC-3 exists and I have to manually select what files to demux opposed to doing a straight demux? And if I am to select the eac-3 track I need to select eac-3 with -keepdialnorm?

Yes.

WSC4

10th June 2022, 09:53

This is a very good GUI for eac3to, and I have been using for a month and it work well thanks.

In the bottom blue window, there is this: (% Character Replaced By Input Name)

Is it possible to have the % character only using a file name without the extension? For example, if an audio file is called Audio.mka and my parameters are:

1: %_1eng.wav -down16

The output file will be named: Audio_1eng.wav

This is instead of Audio.mka_1eng.wav

tebasuna51

10th June 2022, 18:52

I like to know from what file was obtained.

BTW you have the source .au3 to change it.

WSC4

11th June 2022, 10:34

I do not understand what you mean by "I like to know from what file was obtained."

the source .au3 to change it.

Do you mean UsEac3To.au3? I looked in that and it was too complex. Can you do it?

tebasuna51

12th June 2022, 11:31

Do you mean UsEac3To.au3? I looked in that and it was too complex. Can you do it?

In line 1507 added:

; patch
$ruta_filo = StringTrimRight($ruta_filo, 4)
if StringRight($ruta_filo, 1) = "." Then $ruta_filo = StringTrimRight($ruta_filo, 1)
; end patch

The first line delete 4 chars like: .ac3,...,.mkv
The second is for 5 chars like: .eac3,...,.m2ts

WSC4

12th June 2022, 14:01

I am sorry but this does not work.

I downloaded your UsEac3To.au3 patch and replaced the original UsEac3To.au3 with it. I ran UsEac3To.exe and added the input file called Film.mka.

eac3to v3.36
command line: "C:\Program Files\eac3to_GUI\eac3to.exe" "C:\Film.mka" -progressnumbers -log="C:\Program Files\eac3to_GUI\UsEac3To.log"
------------------------------------------------------------------------------
MKA, 1 audio track, 0:54:22
1: AC3, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB

In the track input output format it is showing ac3. I added a "wav" format and also the down 16 parameter. In the **COMMAND LINE PARAMETERS (% char replaced by input name)**

1: %_1eng.wav -down16

It is still showing that %_1eng.wav thing. I ran the CL and the file ends up with "film.mka_1eng.wav"

tebasuna51

12th June 2022, 14:37

You need compile the .au3 to obtain the .exe (to big to be attached here) or install Autoit (https://www.autoitscript.com/site/) and execute the new .au3 to be interpreted by Autoit3.exe

outgoing

6th August 2022, 17:44

Hi, I'm trying to convert some .wavs (DTS-HD Master Audio) 1080i/29,970 fps to 24 fps but I always get an unknown or unsupported format error. For slowdown/speedup there is no problem but for the 29,970 format it is just not possible. Please can someone tell me the parameters, if any, suitable to do this conversion or is it not a viable case.

SeeMoreDigital

6th August 2022, 22:08

Hi, I'm trying to convert some .wavs (DTS-HD Master Audio) 1080i/29,970 fps...Just so you know.... DTS-HD MA audio streams are never placed within the .wav container! What is your source?

outgoing

7th August 2022, 07:24

Just so you know.... DTS-HD MA audio streams are never placed within the .wav container! What is your source?

It is from a bd, passing a dts-hdma to .wavs is a step that I always do to make the fps conversion, it does not involve re-encoding it at all and then reassembling it in dts-hdma. In this case, as I said before, the conversion step of the extracted .wavs always returns an error because it does not recognise 29.97 fps as valid.

I'm sure I'm missing something and it won't be such a simple or straightforward process to make this conversion.

von Suppé

7th August 2022, 14:12

Changing DTS-HD speed while keeping DTS-HD format would require DTS-HD encoding software. It is possible to change speed if outputting in a different format.

Probably your DTS-HD audio has to be sped-up with factor 1.001 to speed-match with your 24 fps video. Load the DTS-MA in USEac3to, choose desired outputformat (W64 or FLAC to be as lossless as possible). Under "Frequent parameters" both choose and add "23.976" and "changeTo23.976". Edit the last parameter so in the end, it looks like this (for flac)

%_.flac -23.976 -changeTo24.000

Running this will give you a multichannel FLAC which is sped-up with 1.001. Don't worry about 29.97 not showing anywhere; audio doesn't really carry "video framerate". It's just the 23.976 : 24.000 ratio that's important here.

Should it be necessary to keep the native DTS-HD track, another widely applied workflow is to instead slow down 24 fps video to 23.976. For now, I won't go further into how. Just sayin'.

outgoing

7th August 2022, 14:48

@von Suppé. Thanks so much for his detailed instructions. When I try it out, i'll let you know the results.

von Suppé

7th August 2022, 15:10

Please do. I did forget to mention that - once audio and video are speed-synced - you may have to apply a (positive or negative) audio delay in mux. The two videos may be offset.

outgoing

7th August 2022, 20:23

@von Suppé. Yeah, i have done the test with the settings you have told me about and indeed everything has synchronised perfectly once the right delay have been applied as you said, didn't know that a video at that frequency was actually equal to a 23.976. More than grateful!

von Suppé

8th August 2022, 22:01

You're welcome, good you managed.

Do note that you can also do the opposite. Should you need to slowdown audio with factor 1.001, use

%_.flac -24.000 -changeTo23.976

...didn't know that a video at that frequency was actually equal to a 23.976.

Yeah, 23.976 fps source is usually "telecined" for interlaced NTSC 29.97. In short, the source frames are interlaced and the fields are blended, where some fields are used twice to make up for NTSC's higher framerate.
Because this process blends exactly every 4 source-frames into exactly 5 NTSC frames, there is no runtime difference between 23.976 video and it's telecined 29.97 counterpart.

outgoing

10th August 2022, 08:01

@von Suppé. Thanks again man! I take note of all the good information you have given me, the truth is that in a bluray film i have never come across one with this frequency. More than grateful!

bananenmann

23rd October 2022, 15:10

Hello, I have a small problem: I think I made a big mistake:
I upgraded to Win 11 Pro and after I started UsEac3to Windows suddenly asked me with which file I want to open the app/program (???). I chose "eac3to.exe", because I was not sure what was meant and also (2nd mistake) klicked on "always open with it". But that seems to be wrong. I don't understand what windows wanted from me - and I can't corret/reverse this. Now, always I try to import a file, nothing happens and in the log is written, that the file could not be opened, because it wasn't found - alhough its there. There is only a message about eac3to 3.34 (but in the folder I have the new version 3.36 (???) ... very confusing... Please help !!! :(

tebasuna51

23rd October 2022, 16:03

Configure your explorer to show always the file extensions. You need open UsEac3to.exe and not the source file UsEac3to.au3.

If you try open the .au3 is when windows ask you how open it. If you install Autoit3 (https://www.autoitscript.com/site/) you can open it with AutoIt3_x64.exe

bananenmann

23rd October 2022, 16:25

Thanx. But I always have "show always the file extensions" aktivated. Surely I opened UsEac3to.exe and NOT UsEac3to.au3. Therefore it seemed strange for me, that Windows asked me with which program "UsEac3to "should be opened because the Windows of UsEac3to was already shown there. Despite that Windows opened the Window where you can choose apps and aksed me that irritating question. I use UsEac3to since a lot of years and did not change anything. The only change was my upgrade to Win 11. But with Win 11 it also worked for month without a problem. Since a few days I updated to Win11 22H2 ...and then this error appeared...

Now everytime I start UsEac3to.exe my Windows Text Editor also automatically starts at the same time and opens the "UsEac3To2.cmd" - file with (only) this inside:

"C:\Sonstiges\UsEac3to\eac3to.exe" >"C:\Sonstiges\UsEac3to\UsEac3To.log"

(This is the folder where the program is located in)

Despite the UsEac3to.exe - Window is open, I can't import any file, because nothing happens...

Edit:
-----

I found the Problem: Notepad++ was associated with *cmd Files (my Fault :) ) ... I removed that and now everything works fine again. Sorry to bother you. I thought it had something to to with Win11 22h2 and the new Terminal app. ;)

tebasuna51

24th October 2022, 10:28

Then it is still partially solved, now run fine the GUI but Win 11 22h2 ask you something when open UsEac3to?

BigPines

7th January 2023, 21:14

The first post said source code was available but I don't see it. Could someone please point me in the right direction?

tebasuna51

8th January 2023, 03:11

The source code of de GUI UsEac3to.exe (not for eac3to.exe) is UsEac3to.au3 attached in the first post.

BigPines

18th January 2023, 18:12

Thank you sir!

DanDare1983

23rd January 2023, 00:18

Hi,

The fact you can queue up playlists to demux is fantastic, really like the gui. Can anyone tell me if its possible to automatically keep dialnorm for all tracks that eac3to removes it from? And also whether there's an option to automatically put all HD tracks into ac3 dolby digital as ac3 is widely supported and plays on literally everything? Thankyou

DanDare1983

23rd March 2023, 00:12

Hi, could anybody tell me if you can have audio and subtitle tracks name set as ita.flac or fre.ac3 when demuxed with eac3to as when they're labelled like that mkvmerge automatically tags with the right language. When i use the -demux option mkvmerge can never recognise the language and is set to my default english.

tebasuna51

23rd March 2023, 11:20

That is a mkvmerge feature (https://forum.doom9.org/showthread.php?t=175593), eac3to can't do anything.

The mkvmerge parameter --language 0:fre still work, but I don't know if mkvmerge can assign the language from the track name.

felguera10

1st April 2023, 16:20

Hello,

One question, is there a way to extract forced subtitles from a Blu-ray PGS subtitle with this tool?
I mean extract the forced subtitle track from the full ones. There are subtitles on Blu-ray that come like this.

Thanks.

tebasuna51

2nd April 2023, 21:39

Short answer: nope

Emulgator

2nd April 2023, 22:17

DGDemux, then SubtitleEdit.

tebasuna51

3rd April 2023, 09:12

eac3to can extract fine pgs from bd's (not from compressed in mkv's), after you can use SubtitleEdit, like say Emulgator, or BDSup2Sub to select the forced ones.

felguera10

4th April 2023, 12:17

Until now it did it with BDSup2Sub 5.1.2, but with some subtitles it modifies the color palette and always modifies the timecodes.
I have discovered BDSup2Sub++1.0.2, which is not java based. I have done some tests and it does not modify the color palette or the timecodes.

Thanks!

Hellboy.

22nd April 2023, 00:20

I want to merge Dolby TrueHD 7.1 Atmos + ac3 core like here:
https://forum.doom9.org/showthread.php?t=183566

But when i open the UsEac3To it show an error:
The application was unable to start correctly (0xc000007b).
Click OK to close the application.

What can i do? Thanks

tebasuna51

22nd April 2023, 09:58

I don't know what is the problem.

Download and extract the last UsEac3to 1.3.3 (https://www.mediafire.com/file/s3xwf8g9h67u232/UsEac3to133.7z/file), and execute UsEac3to.exe

If the problem persist maybe is a problem with your OS.

Hellboy.

22nd April 2023, 16:10

I try but same error. Can you give me the command line arguments for eac3to needed for this. Thanks.

tebasuna51

23rd April 2023, 17:24

eac3to atmos.thd output.thd+ac3 -keepDialnorm

Hellboy.

1st May 2023, 19:19

Worked great, thanks.

The problem was that i copied libFLAC.dll x64 and libdcadec.dll x64, and eac3to.exe don't work with neithers of those. Need to be x32.

Just for learning, this can be done with ffmpeg? (eac3to atmos.thd -keepDialnorm output.thd+ac3 )

tebasuna51

2nd May 2023, 12:18

The problem was that i copied libFLAC.dll x64 and libdcadec.dll x64, and eac3to.exe don't work with neithers of those. Need to be x32.
Of course, eac3to.exe is 32 bits then the dll's must be 32 bits.
You can download it from https://www.rarewares.org/files/lossless/flac_dll-1.4.2-x86.zip
And https://github.com/foo86/dcadec/releases/download/v0.2.0/dcadec-0.2.0-win32.zip

Or you can use the pipe method with:
stdout.wav | flac -o %_.flac -5 -
with flac.exe 64 bits in eac3to or UsEac3to or Encoders folder.

To decode DTS-HD (libdcadec.dll) you can always use ffmpeg with the 'A/V Recode' button.

Just for learning, this can be done with ffmpeg? (eac3to atmos.thd -keepDialnorm output.thd+ac3 )

Nope with my knowledge, but maybe...

Hellboy.

3rd May 2023, 20:25

Forgot to say, for this to work -keepDialnorm need to be at the end: eac3to atmos.thd output.thd+ac3 -keepDialnorm. Or "eac3to" throw an error.

But looks like this is badly modifying the file looking at the log file.
Move SR to SL and remove BL and BR completely.

eac3to v3.36
command line: "C:\Users\Admin\Documents\eac3to\eac3to.exe" "D:\Downloads\Insurgent.2015.2160p.REMUX\Insurgent.2015.PROPER.2160p.BluRay.REMUX.thd" "D:\Downloads\Insurgent.2015.2160p.REMUX\Insurgent.2015.PROPER.2160p.BluRay.REMUX.thd+ac3" "-keepDialnorm"
------------------------------------------------------------------------------
TrueHD (Atmos), 7.1 channels, 48kHz
AC3 encoding doesn't support back channels. Will mix them into the surround.
Decoding with libav/ffmpeg...
Mixing surround channels...
Remapping channels...
Encoding AC3 <640kbps> with libAften...
Creating file "D:\Downloads\Insurgent.2015.2160p.REMUX\Insurgent.2015.PROPER.2160p.BluRay.REMUX.thd+ac3"...
Original audio track, L: max 21 bits, average 18 bits.
Original audio track, R+BL: max 23 bits, average 18 bits.
Original audio track, C+BR+SR: constant bit depth of 19 bits.
Original audio track, LFE: constant bit depth of 18 bits.
Original audio track, SL: max 20 bits, average 18 bits.
Processed audio track, L: max 21 bits, average 18 bits.
Processed audio track, R: max 23 bits, average 18 bits.
Processed audio track, C: constant bit depth of 19 bits.
Processed audio track, LFE: constant bit depth of 18 bits.
Processed audio track, SL+SR: max 24 bits, average 20 bits.
eac3to processing took 6 minutes, 7 seconds.
Done.

tebasuna51

4th May 2023, 08:54

Forgot to say, for this to work -keepDialnorm need to be at the end: eac3to atmos.thd output.thd+ac3 -keepDialnorm. Or "eac3to" throw an error.
You are right, sorry. [post edited]

But looks like this is badly modifying the file looking at the log file.
Move SR to SL and remove BL and BR completely.

Of course, AC3 only support 5.1 like say the message:
AC3 encoding doesn't support back channels.

All BD tracks thd+ac3 have the ac3 5.1 max.

Hellboy.

4th May 2023, 17:51

I was confused. I was thinking "Processed audio" was for both thd+ac3, and that is only for the ac3. Thanks.

seagate

27th May 2023, 17:36

Hello

What is causing this error?

I am attaching a photo: https://i.postimg.cc/TPgXjqtm/11.png

https://postimg.cc/4KfMstkN

I have uploaded the file in question here: https://gofile.io/d/JNdr5R

File Information:

General
Complete name : DB.mp4
Format : MPEG-4
Format profile : Base Media
Codec ID : isom (isom/iso2/avc1/mp41)
File size : 615 MB
Duration : 35 min 30 s
Overall bit rate : 2 422 kb/s
Frame rate : 25,000 FPS
Writing application : Lavf58.20.100

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L4
Format settings : CABAC / 3 Ref Frames
Format settings, CABAC : Yes
Format settings, Reference : 3 frames
Codec ID : avc1
Codec ID/Info : Advanced Video Coding
Duration : 35 min 29 s
Bit rate : 2 289 kb/s
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 25,000 FPS
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0,044
Stream size : 581 MB (94%)
Title : ISO Media file produced by Google Inc.
Color range : Limited
Color primaries : BT.709
Transfer characteristics : BT.709
Matrix coefficients : BT.709
Codec configuration box : avcC

Audio
ID : 2
Format : AAC LC
Format/Info : Advanced Audio Codec Low Complexity
Codec ID : mp4a-40-2
Duration : 35 min 30 s
Bit rate mode : Constant
Bit rate : 128 kb/s
Channel(s) : 2 channels
Channel layout : L R
Sampling rate : 44,1 kHz
Frame rate : 43,066 FPS (1024 SPF)
Compression mode : Lossy
Stream size : 32,5 MB (5%)
Title : ISO Media file produced by Google Inc.
Language : English
Default : Yes
Alternate group : 1

I intended to extract the audio, but it turns out that the program can't even parse the file. I didn't have this problem with mp4 files before.

tebasuna51

28th May 2023, 11:17

eac3to never can manage mp4 files, but UsEac3to can manage it with ffmpeg.

To extract tracks use the 'MkvExt/Thd/INF' button, to de/recode the audio use the 'A/V Recode' button.

hypnotize666

7th July 2023, 20:57

Hello everyone i need your help;could someone upload a version of useac3to gui that would allow to have as output file in aac because every time I get this error:[a02] Extracting audio track number 2...
[a02] Decoding with DirectShow (Nero Audio Decoder 2)...
[a02] Getting "Nero Audio Decoder 2" instance failed. <ERROR>
Aborted at file position 262144. <ERROR> On another post someone said to install Nero so I installed nero start smart V.7.5.9.0 but despite all that my process never ends because I still have the same error .THANKS

tebasuna51

8th July 2023, 09:44

The problem is with eac3to, not with the GUI UsEac3to version.

I can't help you with commercial directshow Nero Audio Decoder filters but the GUI UsEac3to can help you using ffmpeg.exe (must be in eac3to/UsEac3to/Encoders folder):

1) Load your file in UsEac3to GUI and click in 'A/V Recode'

2) Select the map desired (the ffmpeg map order begin in '0' for video, the the map '1' is the first audio track normally), here '1'

3) Select the output format and bitrate

4) Run or Enqueue the job

hypnotize666

8th July 2023, 18:58

Hi i forgot to mentinned i want to downmix the track to stereo for the instance i just try to do like u said but now ive got this problem what can i do thx?
http://imgur.com/m5Euxsd

tebasuna51

8th July 2023, 23:20

I can't understand your error, we need more info about your input file (put the full eac3to log) and your desired output (format and bitrate)

hypnotize666

9th July 2023, 01:36

the error i got up there its when i try to proceed via a/v recode without touched audio this is the file i want to edit then downmix if it possible in the same way AAC CBR at 448Kbps thx

https://www.mediafire.com/file/442zsjaq2vrqo8k/Halloween_media_info.mht/file

tebasuna51

9th July 2023, 10:21

Sometimes 5.1 have empty the FC,SL,SR,LFE channels. And instead downmix you need select only the FL,FR channels.
To know this the best option is decode the audio track to wav (select Decode to wav in step 3 from my image)

If doing this you obtain a error, like in your imgur image, maybe your input source is corrupt.
Try to extract the aac track with the 'MkvEx/Thds/Info' (GUI for MkvExtract) and after decode the aac track with 'A/V Recode'.
If you can't obtain now the 5.1 wav sorry but I can't help you anymore, your input source is corrupt.

If you obtain the 5.1 wav you can open it in a wav editor like Audacity and see if you need discard empty channels or make a downmix using the wav like input file in UsEac3to.
Or extract all mono wavs (Output Format: wavs) and listen each channel to know if there are empty.

If you want use only the x.L.wav and x.R.wav delete the rest of empty channels and load the x.L.wav in UsEac3to and click in 'A/V Recode'.
Select Recode to AAC/M4A at desired encoder bitrate or quality and Run
A window ask you to merge the monochannels detected, answer yes of course.

hypnotize666

10th July 2023, 22:21

Hi im still trying to encode an audio from a .ts its an .ac3 at 5.1 i dont know why in useac3 dont recognize my file it write this:
2: AC3, ???, 5.1 channels, 640kbps, 48kHz then this is my log file when i try via the commands line parameters:
eac3to v3.36
command line: "C:\Installation\Encodeurs\eac3to336-UsEac3to133 (Portable)\UsEac3to133\eac3to.exe" "H:\HDPVR 2 Capture\Manual Recordings\Bait (2012) VF L'Appat French Hdtv x265 Hypnotize.ts" 2: stdout.wav -normalize -progressnumbers -log="H:\HDPVR 2 Capture\Manual Recordings\Bait (2012) VF L'Appat French Hdtv x265 Hypnotize.ts_Job_1.log"
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 1:33:10, 30p /1.001
1: h264/AVC, 1080p30 /1.001 (16:9)
2: AC3, ???, 5.1 channels, 640kbps, 48kHz
[a02] Extracting audio track number 2...
[a02] Decoding with libav/ffmpeg...
[a02] Writing WAV...
[a02] Creating file "stdout.pass1.wav"...
[a02] Starting 2nd pass...
[a02] Reading WAV...
[a02] Reducing depth from 64 to 24 bits...
[a02] Writing WAV...
[a02] Applying 5.5dB gain...
[a02] Creating file "stdout.wav"...
[a02] Writing the destination file failed. <ERROR>
Aborted at file position 2883584. <ERROR>
if i try via the button a/v recode it show me error 193 coreaudiotoolbox.dll ;I'm lost what can i do thx?

tebasuna51

11th July 2023, 09:22

Hi im still trying to encode an audio from a .ts its an .ac3 at 5.1 i dont know why in useac3 dont recognize my file it write this:
2: AC3, ???, 5.1 channels, 640kbps, 48kHz
Only the languaje is not recognized, seems it is not present in your .ts capture,the rest seems fine.

2: stdout.wav -normalize
...
[a02] Creating file "stdout.wav"...
[a02] Writing the destination file failed. <ERROR>

You must use a real filename to write the wav it in the disk, the name stdout.wav (or .w64) is reserved to send the output to the STDOUT device to pipe it to an encoder, for instance to recode it to aac the COMMAND LINE PARAMETERS must be:

2: stdout.wav | qaac -V 99 --ignorelength --adts --no-delay -o %_2.aac -

hypnotize666

13th July 2023, 03:30

Ok Thx im gonna check this but its weird cuz i dont type these commands by myself im not so familiar with commands-line ,it appears when i push the buttons on the gui but does the quality -V 99 exist really i thought i see to a doc its an increments of 9 like -V 73,-V 82,-V 91,-V 100,-V 109,-V 118,-V 125

tebasuna51

13th July 2023, 08:37

Ok Thx im gonna check this but its weird cuz i dont type these commands by myself im not so familiar with commands-line ,it appears when i push the buttons on the gui
Yeah, that appears when you select 'qaa-ext' in output format and push 'Add'.
You can modify any parameter but only if you know what are doing.

but does the quality -V 99 exist really i thought i see to a doc its an increments of 9 like -V 73,-V 82,-V 91,-V 100,-V 109,-V 118,-V 125
You are right, but any value exist, and the quality is the same, read this FAQ (https://forum.doom9.org/showthread.php?p=424070#post424070).
Any value in the range 104-96 output the same quality.

By pm hypnotize666 ask me about the folder tree for eac3to/UsEac3to, this is my recommended one:

hypnotize666

14th July 2023, 05:01

https://imgur.com/opT06yl
Normally with a ac-3 i can do more this 1 command with qaac then i got no error;but when the source are in aac-lc obviously in this example here I don't have downstereo because it is already in 2.0 but i had always errors then the process failed.Like in this example:command line: "C:\Installation\Encodeurs\eac3to336-UsEac3to133 (Portable)\UsEac3to133\eac3to.exe" "K:\FFOutput\MKV-Chapterizer-v2.4\Les.Gags.S22e01.example de 3min.mkv" 2: stdout.wav -normalize -progressnumbers -log="K:\FFOutput\MKV-Chapterizer-v2.4\Les.Gags.S22e01.example de 3min.mkv_Job_1.log"
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 0:03:03, 30p /1.001
1: h265/HEVC, 720p30 /1.001, 10 bits
2: AAC, French, 2.0 channels, 48kHz, 4ms
"Francais 2.0 (QAAC) Cvbr @224k"
[a02] Extracting audio track number 2...
[a02] Decoding with DirectShow (Nero Audio Decoder 2)...
[a02] Getting "Nero Audio Decoder 2" instance failed. <ERROR>
Aborted at file position 262144. <ERROR>
lately you were suggesting to go through the a/v recode button of course it works in one step but if I want to downstereo and normalize or other is there a way to do that and if so how please?Thanks for the qaac witch settings is recommended Constrained 192 vbr,low,med or high? or nothing of these lol

tebasuna51

14th July 2023, 10:26

Like I say you before:
The problem is with eac3to, not with the GUI UsEac3to version.

I can't help you with commercial directshow Nero Audio Decoder filters but the GUI UsEac3to can help you using ffmpeg.exe (must be in eac3to/UsEac3to/Encoders folder)

If you want downmix 5.1 to 2.0 and/or normalize an aac you need two steps (eac3to or AviSynth needs also 2 steps always):

Step 1:
1) Load your file and click in 'Add' FFMPEG More parameters. See first attached image.

2) In the new window search and select the ffmpeg filter '51-20v'. If not exist in your Vid_enc.par copy the text in the image or other downmix metod.
If you only want normalize leave only:
-af "volumedetect"

3) Close the window with 'Done' and the ffmpeg filter are now in the COMMAND LINE PARAMETERS

4) See the image 2. Click in 'A/V Recode'

5) Select the track desired in the 'map:X'

6) Select 'Add to DEC' to include the ffmpeg filter.

7) Select 'Decode...' to 'null'. When this run don't create files then is very fast.

8) Run and see:

...
video:0kB audio:485504kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
[Parsed_volumedetect_1 @ 000002308e4f3400] n_samples: 248578048
[Parsed_volumedetect_1 @ 000002308e4f3400] mean_volume: -36.9 dB
[Parsed_volumedetect_1 @ 000002308e4f3400] max_volume: -9.7 dB
...

Copy the max_volume value.

Step 2:

1) Edit the COMMAND LINE PARAMETERS replacing 'detect' with '=9.7dB' to obtain:
-af "pan=stereo|FL=.3694c0+.2612c2+.3694c4|FR=.3694c1+.2612c2+.3694c5, volume=9.7dB"

2) Repeat the previous 4), 5) and 6)

3) Now, instead 7) 'decode to null', select the desired output format and quality or bitrate.

About the recommended qaac settings:

-V, --tvbr <n> AAC True VBR mode / quality [0-127]
-v, --cvbr <bitrate> AAC Constrained VBR mode / bitrate

Use only the --cvbr when you need a fixed size, if not use your desired quality (for me -V 99 is transparent, 80 is good to save space)
Talking about standard stereo movie tracks (many dialogs and silences, without many music of effects) -v 192 is waste space.
For instance using -V 99 over a movie stereo track I obtain: VBR min./avg./max. : 4 / 143 / 406 Kb/s.
Go until 406 Kb/s when need it, but the average remain at 143.

I'm not sure about "low,med or high?", maybe the quality?

-q, --quality <n> AAC encoding Quality [0-2]

Use always the default 'high', the process is very fast you don't need save time with less quality.

hypnotize666

15th July 2023, 05:09

Cool thanks for all and it was cleared with comments then pictures too.Good work.

seagate

29th September 2023, 14:12

I don't know what this strange inscription is due to, but I show it with a photo. The same error appears with StaxRip v.2.29.0 x64 (https://forum.doom9.org/showthread.php?p=1992122#post1992122).

https://i.postimg.cc/PxsQzns2/Us-Eac3to-v-1-3-3.png

Complete name : 00004.m2ts
Format : BDAV
Format/Info : Blu-Ray Video
File size : 91,6 GB
Duration : 2 h 13 min
Overall bit rate mode : Variable
Overall bit rate : 98,6 Mb/s
Maximum Overall bit rate : 109 Mb/s
Frame rate : 24,000 FPS

Video #1
ID : 4113 (0x1011)
Menu ID : 1 (0x1)
Format : HEVC
Format/Info : High Efficiency Video Coding
Format profile : Main 10@L5.1@High
HDR format : SMPTE ST 2086, HDR10 compatible
Codec ID : 36
Duration : 2 h 13 min
Width : 3 840 pixels
Height : 2 160 pixels
Display aspect ratio : 16:9
Frame rate : 24,000 FPS
Color space : YUV
Chroma subsampling : 4:2:0 (Type 2)
Bit depth : 10 bits
Color range : Limited
Color primaries : BT.2020
Transfer characteristics : PQ
Matrix coefficients : BT.2020 non-constant
Mastering display color pri : BT.2020
Mastering display luminance : min: 0.0001 cd/m2, max: 1000 cd/m2
Maximum Content Light Level : 1015 cd/m2
Maximum Frame-Average Light : 154 cd/m2

Video #2
ID : 4117 (0x1015)
Menu ID : 1 (0x1)
Format : HEVC
Format/Info : High Efficiency Video Coding
Format profile : Main 10@L5.1@High
HDR format : SMPTE ST 2086, HDR10 compatible
Codec ID : 36
Duration : 2 h 13 min
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16:9
Frame rate : 24,000 FPS
Color space : YUV
Chroma subsampling : 4:2:0 (Type 2)
Bit depth : 10 bits
Color range : Limited
Color primaries : BT.2020
Transfer characteristics : PQ
Matrix coefficients : BT.2020 non-constant
Mastering display color pri : BT.2020
Mastering display luminance : min: 0.0001 cd/m2, max: 1000 cd/m2

Audio #1
ID : 4352 (0x1100)
Menu ID : 1 (0x1)
Format : DTS XLL
Format/Info : Digital Theater Systems
Commercial name : DTS-HD Master Audio
Muxing mode : Stream extension
Codec ID : 134
Duration : 2 h 13 min
Bit rate mode : Variable
Channel(s) : 2 channels
Channel layout : L R
Sampling rate : 48,0 kHz
Frame rate : 93,750 FPS (512 SPF)
Bit depth : 18 bits
Compression mode : Lossless

Audio #2
ID : 4353 (0x1101)
Menu ID : 1 (0x1)
Format : AC-3
Format/Info : Audio Coding 3
Commercial name : Dolby Digital
Codec ID : 129
Duration : 2 h 13 min
Bit rate mode : Constant
Bit rate : 384 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 : 365 MB (0%)
Service kind : Complete Main

Audio #3
ID : 4354 (0x1102)
Menu ID : 1 (0x1)
Format : AC-3
Format/Info : Audio Coding 3
Commercial name : Dolby Digital
Codec ID : 129
Duration : 2 h 13 min
Bit rate mode : Constant
Bit rate : 128 kb/s
Channel(s) : 2 channels
Channel layout : L R
Sampling rate : 48,0 kHz
Frame rate : 31,250 FPS (1536 SPF)
Compression mode : Lossy
Stream size : 122 MB (0%)
Service kind : Complete Main

Audio #4
ID : 4355 (0x1103)
Menu ID : 1 (0x1)
Format : AC-3
Format/Info : Audio Coding 3
Commercial name : Dolby Digital
Codec ID : 129
Duration : 2 h 13 min
Bit rate mode : Constant
Bit rate : 128 kb/s
Channel(s) : 2 channels
Channel layout : L R
Sampling rate : 48,0 kHz
Frame rate : 31,250 FPS (1536 SPF)
Compression mode : Lossy
Stream size : 122 MB (0%)
Service kind : Complete Main

Audio #5
ID : 4356 (0x1104)
Menu ID : 1 (0x1)
Format : AC-3
Format/Info : Audio Coding 3
Commercial name : Dolby Digital
Codec ID : 129
Duration : 2 h 13 min
Bit rate mode : Constant
Bit rate : 192 kb/s
Channel(s) : 2 channels
Channel layout : L R
Sampling rate : 48,0 kHz
Frame rate : 31,250 FPS (1536 SPF)
Compression mode : Lossy
Stream size : 183 MB (0%)
Service kind : Complete Main

Text
ID : 4768 (0x12A0)
Menu ID : 1 (0x1)
Format : PGS
Codec ID : 144

tebasuna51

29th September 2023, 18:04

Like I say to madshi longtime ago that WARNING, not a ERROR, is useless, forget it.

24 fps is a normal fps.

DanDare1983

8th October 2023, 17:49

Hi, hopefully someone can help me. When demuxing files with Japanese in the filename i get the language code 'Jap' I didn't think this would be a problem but when remuxing the files with mkvtoolnix I have found a problem - when deriving the language 'Jap' isn't recognised. The only code recognised is 'Ja' and 'Jpn'. Can eac3to demux to ja or jpn? I'm using the eac3to gui. Thankyou

tebasuna51

9th October 2023, 02:46

oniiz86

15th December 2023, 11:19

@tebasuna51 There seems to be an issue with the latest eac3to 3.42 & 3.43 test builds when dropping the eac3to.exe into UsEac3to v1.3.3, it throws an error dialog, eac3to 3.41 is fine
https://i.imgur.com/d5VXiqa.png

tebasuna51

15th December 2023, 12:52

No problem here:
eac3to v3.43
command line: "C:\Portable\eac3to\eac3to.exe" "C:\tmp\test51.mp3" -progressnumbers -log="C:\Portable\eac3to\UsEac3to\UsEac3To.log"
------------------------------------------------------------------------------
Running in normal mode
Removing dialnorm
MP3, 2.0 channels, 0:00:12, 320kbps, 48kHz, dialnorm: 0dB

Try run it from command line.

oniiz86

15th December 2023, 15:22

Sorry not well versed with command line, I discovered the issue though, for some strange reason the "eac3to.ini" file is required with eac3to v3.42 & the latest v3.43 test build within the UsEac3to v1.3.3 folder, it wasn't necessary with v3.41 & previous versions, is that Configuration Settings "INI" file a requirement now as I don't wish to utilise those commands?

tebasuna51

15th December 2023, 19:41

I use a .ini file with only -progressnumbers.
BTW in my GUI I always add that parameter also, like you can see in my previous post.

seagate

7th January 2024, 15:28

@tebasuna51

I have the following case study.

I have a video file with the extension ".ts".

Running in fast mode
Keeping dialnorm
TS, 1 video track, 1 audio track, 01:35:08, 50p
1: h264/AVC, 1080p50 (16:9)
2: AAC, 2.0 channels, 16 kbps, 48 kHz, dialnorm: 0 dB, -19 ms

Frame rate: 50,000 fps.

With this nice tool, I was able to convert the audio file to AC3 - 96,0 kb/s - the same size as the previously demuxed AAC.

(I don't know why UsEac3to v.1.3.3 identifies it as "16 kbps" because it is impossible to be this audio bitrate for a whole movie. When I used the "A/V recode" button this audio in AC3 in size to become the same what is AAC = 96,0 kb/s. Maybe it's some bug in UsEac3to v.1.3.3.)

Why am I writing all this...

I want to encode this video with Megui to cut the black bars, but I don't know how to change the fps from 50 to 25 with UsEac3to v.1.3.3?!

My thought is - is there a way to UsEac3to v.1.3.3 with the demuxed video file and my reworked AC3 - one at a time or how do I change their frame rate from 50 to 25 and then encode them with Megui?

/Just a note, @tebasuna51, I have a recollection that UsEac3to v.1.3.3 does not support files that are written in Cyrillic, I don't know if this will ever be added as support./

SeeMoreDigital

7th January 2024, 15:58

@seagate,

What does MediaInfo report about your .ts source?

seagate

7th January 2024, 16:38

@seagate,

What does MediaInfo report about your .ts source?

Format : MPEG-TS
File size : 3,66 GB
Duration : 1 h 35 min
Overall bit rate mode : Variable
Overall bit rate : 5 513 kb/s
Frame rate : 50,000 FPS

Video
ID : 256 (0x100)
Menu ID : 1 (0x1)
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L5
Format settings : CABAC / 4 Ref Frames
Format settings, CABAC : Yes
Format settings, Reference : 4 frames
Format settings, GOP : M=2, N=100
Codec ID : 27
Duration : 1 h 35 min
Bit rate mode : Variable
Maximum bit rate : 5 000 kb/s
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16:9
Active Format Description : Full frame 16:9 image
Frame rate : 50,000 FPS
Standard : Component
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Color range : Limited
Color primaries : BT.709
Transfer characteristics : BT.709
Matrix coefficients : BT.709

Audio
ID : 257 (0x101)
Menu ID : 1 (0x1)
Format : AAC LC
Format/Info : Advanced Audio Codec Low Complexity
Format version : Version 4
Muxing mode : ADTS
Codec ID : 15-2
Duration : 1 h 35 min
Bit rate mode : Variable
Channel(s) : 2 channels
Channel layout : L R
Sampling rate : 48,0 kHz
Frame rate : 46,875 FPS (1024 SPF)
Compression mode : Lossy
Delay relative to video : -19 ms

Menu
ID : 4096 (0x1000)
Menu ID : 1 (0x1)
Format : AVC / AAC
Duration : 1 h 35 min
List : 256 (0x100) (AVC) / 257 (0x101) (AAC)
Service name : Service01
Service provider : FFmpeg
Service type : digital television

tebasuna51

8th January 2024, 11:45

With this nice tool, I was able to convert the audio file to AC3 - 96,0 kb/s - the same size as the previously demuxed AAC.
For what? With the same size you obtain less quality than the VBR AAC, you need at least 192 Kb/s for a decent AC3. My recommendation is preserve the AAC, it is compatible with all modern devices.

(I don't know why UsEac3to v.1.3.3 identifies it as "16 kbps" because it is impossible to be this audio bitrate for a whole movie. When I used the "A/V recode" button this audio in AC3 in size to become the same what is AAC = 96,0 kb/s. Maybe it's some bug in UsEac3to v.1.3.3.)

The message is generated by eac3to.exe not by UsEac3to v.1.3.3, and I don't know for what show that bitrate, the average bitrate only can be know with tools like LeeAudBi than read the full stream not only the headers like eac3to or MediaInfo do.

I want to encode this video with Megui to cut the black bars, but I don't know how to change the fps from 50 to 25 with UsEac3to v.1.3.3?!

If you change the fps from 50 to 25 with eac3to you obtain a video of 3h 10m (like slow motion) without sync with the audio.
I make .ts video captures with 50 fps with frames duplicated, I use AviSynth SelectEven() to convert it to 25 without lose info. If the frames aren't duplicated that can work but you lose smoth motion (true 50 fps recommended for sport captures)

/Just a note, @tebasuna51, I have a recollection that UsEac3to v.1.3.3 does not support files that are written in Cyrillic, I don't know if this will ever be added as support./

Sorry I can't test other languages, in the first post you have the source code, you are free to make your desired changes.

seagate

8th January 2024, 13:25

If you change the fps from 50 to 25 with eac3to you obtain a video of 3h 10m (like slow motion) without sync with the audio.
I make .ts video captures with 50 fps with frames duplicated, I use AviSynth SelectEven() to convert it to 25 without lose info. If the frames aren't duplicated that can work but you lose smoth motion (true 50 fps recommended for sport captures).

It's about a movie with Steven Seagal.

Even if I successfully or experimentally convert the video/film from 50 fps to 25 fps - I have no idea how to do the same, but with the audio...

you are free to make your desired changes.

Limited freedom is this - knowledge is lacking for such a fundamental change/adding Cyrillic support.

SeeMoreDigital

8th January 2024, 13:51

It's about a movie with Steven Seagal.

Even if I successfully or experimentally convert the video/film from 50 fps to 25 fps - I have no idea how to do the same, but with the audio...Lol... I can't imagine any film featuring that weirdo was shot at 50.000 fps (or even 25.000 fps). :eek:

Emulgator

8th January 2024, 14:35

Could have been restored, interpolated and re-aired for PAL regions.
I remember to have seen something along "Once upon a time in the west" or "Alamo" with smooth 50fps framerate.

SeeMoreDigital

8th January 2024, 14:49

Could have been restored, interpolated and re-aired for PAL regions.
I remember to have seen something along "Once upon a time in the west" or "Alamo" with smooth 50fps framerate.I doubt it...

A short sample cut out of the .ts source might help...

tebasuna51

9th January 2024, 10:11

It's about a movie with Steven Seagal.
Then maybe have duplicated frames, load it in VirtualDub2 and check that.

Even if I successfully or experimentally convert the video/film from 50 fps to 25 fps - I have no idea how to do the same, but with the audio...

You don't need change the audio at all, if it's in sync with a 50 fps also whith a 25 fps with half frames (SelectEven).

Limited freedom is this - knowledge is lacking for such a fundamental change/adding Cyrillic support.

UsEac3to create .bat files than need a specific character set to work properly.
In line 1043 there are a Func Param_page to convert my ANSI spanish (áéíóúñÑ...) characters but I don't know how make this for other languages, sorry.

DanDare1983

25th January 2024, 00:52

Hi, i wonder if you can help me...I'm basically trying to -demux to the folder i'm demuxing from. I've read some notes throughout the forum but i'm having no luck. I've typed %-demux and i get the message come up as 'audio is unsupportd'. All i want is to simply use the -demux in the parameters and save to the folder i'm demuxing from. I know there must be a way but i'm struggling to find it.

tebasuna51

25th January 2024, 10:06

DanDare1983

25th January 2024, 18:05

Like you can see in the EAC3TO To-Do List (https://www.rationalqm.us/board/viewtopic.php?f=18&t=1275) you can't specify the output directory when you use the -demux parameter and the files are created in the ...\UsEac3to folder (default when execute the command line).

For that I put the option in the Global Parameters of UsEac3to GUI and use the following sintax (in upercase full path of files or partial names):

EAC3TO INPUT FOLDER\FILE_.*

with the same demuxed output files but in the desired folder (the original or the User defined)
Your sintax try execute:

EAC3TO INPUT FOLDER\FILE-demux

without sense for eac3to

Thankyou for the quick reply. I've been using %_.* as you explained before and its been great. However I found out last night that using eac3to command line with -demux I get a W.64 file when PCM tracks are present. When using your eac3togui with %_.* I only get raw PCM without W.64 extension.

tebasuna51

25th January 2024, 21:23

... When using your eac3togui with %_.* I only get raw PCM without W.64 extension.
Of course that is demux. If you want specific format you can extract track by track to the desired output.

abkjr

26th January 2024, 06:01

Hey all,

I am currently using a receiver which only supports Dolby digital and DTS. I do have a blurayplayer which sends dolby digital signal for EAC3 JOC file. however I noticed that sound is very hollow with uneven bass. Is it possible to just extract the ac3 core from the Eac3 joc file. I do have eac3to which I use to extract the ac3 core from truhd files.

Audio Info:

Audio #1
ID : 2
Format : E-AC-3 JOC
Format/Info : Enhanced AC-3 with Joint Object Coding
Commercial name : Dolby Digital Plus with Dolby Atmos
Codec ID : A_EAC3
Duration : 2 h 30 min

tebasuna51

26th January 2024, 10:32

The same with new versions of eac3to, if eac3 is from a BD:

%_.ac3 -core

The log like:

eac3to v3.47
command line: "eac3to.exe" "atmosBD.eac3" "core.ac3" -core
------------------------------------------------------------------------------
E-AC3 EX, 7.1 channels, 0:03:08, 1664kbps, 48kHz, dialnorm: -31dB
(core: AC3 EX, 5.1 channels, 0:03:08, 640kbps, 48kHz, dialnorm: -31dB)
Extracting E-AC3 core...
Creating file "core.ac3"...
Done.

abkjr

28th January 2024, 04:44

The audio if currently from a streaming source and has the JOC enabled.

Audio
ID : 2
Format : E-AC-3 JOC
Format/Info : Enhanced AC-3 with Joint Object Coding
Commercial name : Dolby Digital Plus with Dolby Atmos
Codec ID : A_EAC3
Duration : 2 h 37 min
Bit rate mode : Constant
Bit rate : 768 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
Delay relative to video : 24 s 169 ms
Stream size : 866 MiB (8%)
Language : English
Service kind : Complete Main
Default : Yes
Forced : No
Complexity index : 16
Number of dynamic objects : 15
Bed channel count : 1 channel
Bed channel configuration : LFE

Would using the -core option still work or do we also need to use any additional options such as dialognorm

tebasuna51

28th January 2024, 10:41

That EAC3 5.1 from streaming don't have AC3 core, if your player don't support Atmos only the simple EAC3 (core) is played.

If you want recode it to AC3 to save space, and lose quality, you can use ffmpeg with the button 'A/V Recode' in UsEac3to.

I always recommend remove the Dialog Normalization and disable the Dynamic Range Compression, but everyone can do it or not.

DanDare1983

29th January 2024, 17:33

Hi Tebasuna51, I've just noticed that the DG team have released a test version of eac3to regarding the -demux problem involving where the demux is saved. With syntax I can save to any where I want but obviously with -demux I cannot so i'm very much hoping the issue is sorted. Thank again

eac3to_mod

29th January 2024, 17:38

With syntax I can save to any where I want but obviously with -demux I cannot Sure you can. The -destpath option applies to everything. Maybe I misunderstood you.

DanDare1983

29th January 2024, 17:49

Sure you can. The -destpath option applies to everything. Maybe I misunderstood you.

When using %_.* i was able to have the full demux saved to the source. However due to using the * syntax I was getting the raw demux streams and not getting the W.64 for PCM, only the raw PCM. W.64 only works with -demux and not with the * syntax. I would like to enter %_.-demux instead of using the *

eac3to_mod

29th January 2024, 17:53

When using %_.* i was able to have the full demux saved to the source. However due to using the * syntax I was getting the raw demux streams and not getting the W.64 for PCM, only the raw PCM. W.64 only works with -demux and not with the * syntax. I would like to enter %_.-demux instead of using the * I don't follow you at all. What the heck is "%_.-demux"? Maybe the best bet is to let Tebasuna work with me over at DG land. I simply have no idea what you are talking about and I believe Rocky implemented what he asked for.

DanDare1983

29th January 2024, 18:02

I don't follow you at all. What the heck is "%_.-demux"? Maybe the best bet is to let Tebasuna work with me over at DG land. I simply have no idea what you are talking about and I implemented what he asked for.

Sorry I didn't explain myself clearly. I use Tebasuna gui as I like to queue my movies for demux to save time. I like to have all the PCM tracks demuxed as W.64. The '*' represents -demux but only demuxes the PCM as raw PCM and not W.64 which I where the problem lies. So to cut a long story short there's no option using UsEac3to to save -demux to the source folder whereas using the '*' there is. I hope that sounded better. I think Tebasuna understood which is why he made the request.

eac3to_mod

29th January 2024, 18:07

Just set -destpath to the directory where the source is. Or just cd to that directory to run eac3to from there and don't set -destpath. Did I miss something?

DanDare1983

29th January 2024, 18:17

Just set -destpath to the directory where the source is. Or just cd to that directory to run eac3to from there and don't set -destpath. Did I miss something?

This would work perfect using cmd. However I'm not sure how it would work using UsEac3to as that's the problem I have atm, using * to demux enables me to save to where I want whereas the -demux saves to UsEac3to folder by default.

eac3to_mod

29th January 2024, 18:22

This would work perfect using cmd. However I'm not sure how it would work using UsEac3to as that's the problem I have atm, using * to demux enables me to save to where I want whereas the -demux saves to UsEac3to folder by default. I understand that Teba will change UsEac3to to work with the new functionality. As I said, please let he and I work it out.

DanDare1983

29th January 2024, 18:24

I understand that Teba will change UsEac3to to work with the new functionality. As I said, please let he and I work it out.

Sure thing thankyou.

tebasuna51

30th January 2024, 11:29

The 3.48_test still don't work properly: https://www.rationalqm.us/board/viewtopic.php?f=18&t=1278

You can try the 3.48_test and put in the command line:

-demux -destpath=C:\Temp\

But you obtain a ERROR because the added parameters -progressnumbers -log="logfile"

Without these parameters (enqueue the job, edit the zzJob_1.cmd) the output folder is always d:\tmp\ ignoring the -destpath=C:\Temp\

eac3to_mod

30th January 2024, 17:40

I fixed that stuff but Teba hasn't seen it yet. You can grab my latest test version if you like. With it you should be able to have spaces in the path and options after the destpath one. But you'll need quotes around the path like -log.

-demux -destpath="C:\My folder" -progressnumbers -log="logfile"

Lemme know if it's not OK.

tebasuna51

31st January 2024, 04:14

Thanks Curly seems work fine now, new GUI version 1.3.4 in first post.

Now detect the eac3to version if is greater 3.47 the Global parameter -demux use the -despath= parameter and the pcm tracks output with w64 header.

eac3to_mod

31st January 2024, 08:31

Great, thank you, I'll make a release this morning.

Will DanDare1983's requirements expressed above be satisfied?

DanDare1983

31st January 2024, 09:03

Great, thank you, I'll make a release this morning.

Will DanDare1983's requirements expressed above be satisfied?

First of all, thankyou Teba,Curly/Rocky. I understand it takes time to implement changes/new features so a big thankyou from me. I'm not home to check but will be home very soon so will try and report back with results. Thank again guys.

eac3to_mod

31st January 2024, 09:08

Your wish is my command.

DanDare1983

31st January 2024, 10:45

I'm now using UsEac3To v1.3.4 and can confirm when using the Global Parameter (Run) the demux is now saved to the source folder with W.64 instead of PCM. I'm happy with that so thankyou. I'm still not sure what i have to enter in the command line box in order for me to queue up my movies for demux? Before i used %_.* and that got me the full demux with PCM. Now everything has been sorted i'll now be able to get full demux with W.64 for PCM tracks but i don't know what i have to enter for this to work.

DanDare1983

31st January 2024, 11:52

I've just used -demux -destpath="C:\My folder" -progressnumbers -log="logfile" and understand that i just enter the name and destination and everything works fine. I'm looking for a way where i don't need to enter that info and the files automatically demux to source folder like the %_.* syntax.

tebasuna51

31st January 2024, 13:03

eac3to_mod

31st January 2024, 13:06

eac3to 3.48 was released. I'll leave it to you guys to work out UsEac3to stuff. If anything affects eac3to, please let me know.

eac3to_mod

31st January 2024, 13:07

I've just used -demux -destpath="C:\My folder" -progressnumbers -log="logfile". That's wrong syntax because you need the trailing \ on the path:

-demux -destpath="C:\My folder\"

Probably you have it but you got bit by the forum software removing it in your posting. I had to put \\ there inside the quotes to get it to display. Every time I edit the post it gets changed back and I have to put \\ again.

DanDare1983

31st January 2024, 13:54

Just to remember how work demux in UsEac3to:

1) In Settings -> Output folder you can select your User defined Output folder.
That selection is remembered between sesions in the UsEac3to.ini file but you can change it at your taste.

2) Load or drag and drop any supported container (tested mkv, m2ts and mpls)

3) Select Output Folder between the Input folder and the User Defined
If select 'Same as input' the files are created in the input folder: "D:\Ñu 2" without problems.

4) Just RUN the button below Global Parameters.
Of course you can use the Command Line Parameters with -demux -destpath="C:\My folder" but is not needed.
Now to obtain a demux with .pcm files you can put in CML just: %_.*, but always is better the w64 file.

5) You can see how w64 file is created in my User defined folder C:\tmp\

Remember for the To-do List than PGS subtitles are stored in mkv compressed with zlib by default (in MkvToolNix) and eac3to can't extract them.
From m2ts or in mkv without compression eac3to can do it.

Thankyou for getting back to me so quick :-)

I understand that using the Global Parameter switch 'Run' now puts out .W64 instead of PCM. It also puts all demuxed files in the Original Source folder which is great. However if I was to add to queue with -demux using CML all tracks go to chosen folder. Could you tell me if there is any way all -demux via CML is saved to Original Source folder? This saves time typing destination folder everytime. The %_.* option was great as it was very quick and saved to the Source location without doing anything. Only problem was PCM. Also one more thing, when I use Global Parameter to -demux all files go the same folder apart from log file which seems to go to the UsEac3to folder.

eac3to_mod

31st January 2024, 14:29

Also one more thing, when I use Global Parameter to -demux all files go the same folder apart from log file which seems to go to the UsEac3to folder. I didn't change that because there is a separate option -log="path" for that.

tebasuna51

31st January 2024, 16:27

Then you need a button to EnQueue the demux instead RUN inmediatly.

Maybe like this:

I will add that button this night.
The log go also to the selected folder (Original or User defined)

DanDare1983

31st January 2024, 17:08

Yes this would be great. If you could add a queue button it would make things alot easier. Thanks again Teba.

tebasuna51

31st January 2024, 20:53

Reupload the GUI with the new button 'Q', now you can 'R'un or en'Q'ueue the demux with only one click.

DanDare1983

31st January 2024, 21:16

Thankyou so much, I'll try when I get home.

DanDare1983

1st February 2024, 09:37

Everything works perfect in regards to the new queue button. This will save a lot of time so a big thankyou.

eac3to_mod

1st February 2024, 10:19

A little OT but are you aware that "thank you" is two words?

tebasuna51

4th February 2024, 15:35

Warning: eac3to 3.49 can't be used in UsEac3to.

UsEac3to read the UsEac3to.log of input files to show the tracks, etc. but v3.49 create a file with a wrong name UsEac3to.loglog.txt

[EDIT] Solved, redownload the eac3to 3.49 (https://www.rationalqm.us/eac3to/)

DanDare1983

7th February 2024, 00:16

Whilst using the latest eac3to 3.49 and the latest UsEac3To i've found a problem when using the Global Parameter -demux option. When all tracks are demuxed, there's a problem with the subtitle naming. The name given is 00000 - 5 - Subtitle (PGS), English.sup, when it should be 00000 - 5 - Subtitle (PGS), English, 1416 captions.sup. The number of captions is missing. When using eac3to with cmd theres no problem and all works fine and if i use the * syntax with UsEac3To everything works fine so its just using the Global Parameter -demux option.

tebasuna51

7th February 2024, 00:42

...When using eac3to with cmd theres no problem and all works fine...
UsEac3to only construct and execute the cmd, and can't change at all the filenames created by eac3to.exe.

Y can't do anything about this.

eac3to_mod

7th February 2024, 00:52

@DanDare1983

I've never seen captions count in a subtitle filename. What command line do you give that does that?

DanDare1983

7th February 2024, 01:00

I only use -demux and then the demuxed files come out like this - 00000 - 5 - Subtitle (PGS), English, 1416 captions. That's how they have always come out with -demux.

eac3to_mod

7th February 2024, 01:31

OK, I see that. eac3to re-names the sup files at the end. Learn something new every day!

If -destpath is given the renamer doesn't look there but only in the current directory. I'll fix it. Thank you for the report.

eac3to_mod

7th February 2024, 01:33

The counts are available in the log file, BTW.

DanDare1983

7th February 2024, 09:04

Yes I see that the log file contains the counts, that was the first thing I checked. It's nice to see the sup filename with the amount of captions. That's one of the things I've always liked about eac3to.

tebasuna51

7th February 2024, 10:19

OK, I see that. eac3to re-names the sup files at the end. Learn something new every day!

The same for me.

If -destpath is given the renamer doesn't look there but only in the current directory. I'll fix it. Thank you for the report.

Then I can do it with UsEac3to if is dificult to fix.
When create the .cmd file I can run it in the folder desired adding drive and folder like this:

c:
cd "C:\tmp"
"C:\Portable\eac3to\eac3to.exe" "D:\Ñu 2\0d3.m2ts" -demux -destpath="C:\tmp\" -progressnumbers -log="C:\tmp\0d3.m2ts_Job_1.log"

Tested and work fine.

DanDare1983

7th February 2024, 10:34

Teba, is the demuxed filenames showing for sup showing the number of captions now?

tebasuna51

7th February 2024, 10:48

Of course, before:

0d3 - 3 - Subtitle (PGS), Spanish.sup

After:

0d3 - 3 - Subtitle (PGS), Spanish, 6 captions.sup

But we can wait to Curly before make a new UsEac3to version

DanDare1983

7th February 2024, 11:38

OK that's great news, thanks Teba

eac3to_mod

7th February 2024, 13:24

Please re-download 3.49 for the fix. Sorry, not going to make a new release just for this.

DanDare1983

7th February 2024, 14:35

OK thanks Curly, will re-download.

oniiz86

10th February 2024, 11:37

Of course, before:

0d3 - 3 - Subtitle (PGS), Spanish.sup

After:

0d3 - 3 - Subtitle (PGS), Spanish, 6 captions.sup

But we can wait to Curly before make a new UsEac3to version
Hey @tebasuna51 I was just wondering if the new version has been released as Curly had released a quick slipstream fix for eac3to v3.49, will UsEac3to have a new version number v1.3.5?

tebasuna51

11th February 2024, 10:57

With the new eac3to v3.49 from 2024-02-07 (https://www.rationalqm.us/eac3to/eac3to_3.49.rar) don't need new UsEac3to version the 1.2.4 work without problems.

IntelHEVC

25th February 2024, 13:53

My virus program shows this message:
Trojan:Script/Phonzy.A!ml :devil:

gonca

25th February 2024, 20:07

False positive
According to virustotal 1/61 antivirus scanners flag it as a possible trojan and 1/61 as a possible Potentially Unwanted Application.
The other 59 detect nothing

mannequin80

31st March 2024, 18:25

See the attached image to explain the method.

is it still the best method to downmix 7.1 (TrueHD) to 5.1 (eac3)? is it possible to add DynAudNorm or LoudNorm through command line while using this method?

i did a few quick tests and it feels like the downmix conversion with the simple "-ac 6" command cuts too much from the LFE track so the overall bass becomes audibly less impactful. does this method address that?

if i want to fully preserve the LFE track while downmixing, would using something like this make more "sense":

-filter_complex "asplit [f][s]; [f] pan=5.1|FL=FL|FR=FR|FC=FC|LFE=LFE|BL=BL+SL|BR=BR+SR [a1]; [s] pan=stereo|c0=0.5*c4+0.5*c6|c1=0.5*c5+0.5*c7, compand=attacks=0:decays=0:points=-90/-84|-8/-2|-6/-1|-0/-0.1, aformat=channel_layouts=stereo [d]; [a1][d] amix=inputs=2, dynaudnorm [out]" -map "[out]"

(i left dynaudnorm in there, assuming that would be the correct syntax)?

unless, there's a better way to improve dialogue volume while still maintaining some dynamic range without introducing any volume pumping.

thanks!

tebasuna51

1st April 2024, 10:23

is it possible to add DynAudNorm or LoudNorm through command line while using this method?
Maybe, but I can't recommend that, try yourself.
My method is only a downmix preserving the full front-center-LFE volume.

i did a few quick tests and it feels like the downmix conversion with the simple "-ac 6" command cuts too much from the LFE track so the overall bass becomes audibly less impactful. does this method address that?

if i want to fully preserve the LFE track while downmixing, would using something like this make more "sense":

Nope, my method preserve the full LFE channel, and front, without changes in the first pan=3.1|c0=c0|c1=c1|c2=c2|c3=c3

Your method:
pan=5.1|FL=FL|FR=FR|FC=FC|LFE=LFE|BL=BL+SL|BR=BR+SR and after amix the pan=stereo|c0=0.5*c4+0.5*c6|c1=0.5*c5+0.5*c7
don't have sense at all, sorry.

unless, there's a better way to improve dialogue volume while still maintaining some dynamic range without introducing any volume pumping.

Like I say before I want only downmix preserving the original source, improve it is out of my scope because each source and each user taste can have different methods. Try yourself.

richardpl

2nd April 2024, 10:28

First find out from which channel(s) dialogue comes. For example with mpv, by using showvolume filter.

Hellboy.

7th April 2024, 03:37

I understand that if i convert dts-hd ma 5.1 to w64 don't lose any quality. But if i convert using "-23.976 -changeTo24.000", the audio lose quality?
Thanks

tebasuna51

7th April 2024, 08:52

Of course because there are arithmetic operations involved, and you never can recover the original source exactly.

Lucky38

8th April 2024, 11:27

hi,

is there any option to trim/cut part of AC3 file without re-encoding using eac3to.

Please provide example :)

Hellboy.

8th April 2024, 20:14

Need help with audio frame rate(from what i read audio don't have frame rate). I read a lot and now i am more confused.
I want to synchronize the audio with the video.

video = US uhd 4K 24fps
audio = GER BD 1080p 23fps (original eng audio DTS-HD MA 5.1, 3732 kbps, 48 kHz, 24 bits) (DTS Core: 5.1 / 48 kHz / 1509 kbps / 24-bit)

I was going to use (but not sure this is correct): eac3to.exe input.dts output.w64 -23.976 -changeTo24.000

Some people mention "speedup", "pitch correction". Others mention ffmpeg atempo. But don't understand any of that.

I hope someone can share the correct command line for this, for both, eac3to and ffmpeg.
Thanks.

tebasuna51

8th April 2024, 21:47

is there any option to trim/cut part of AC3 file without re-encoding using eac3to.
From the help:
+/-100ms apply a positive or negative audio delay
-edit=0:00:00,0ms loops or removes audio data at the specified runtime
-silence/-loop forces usage of silence (or looping) for audio edits

For other options you can use DelayCut or BeSplit.

tebasuna51

8th April 2024, 21:59

Need help with audio frame rate(from what i read audio don't have frame rate). I read a lot and now i am more confused.
I want to synchronize the audio with the video.

video = US uhd 4K 24fps
audio = GER BD 1080p 23fps
Like you read righ the audio can't have fps. Even don't exist standard videos with that 23 fps. Maybe 23.976?

If that "audio" dts is sincronized with that "video" played at 23 fps, play it at that speed. Or 23.976 if is the correct fps.

Hellboy.

8th April 2024, 22:22

Yes, is in 23.976 fps.

I did it with the command mentioned before "eac3to.exe input.dts output.w64 -23.976 -changeTo24.000" and it worked. The audio from the 23.976 fps video is fully synchronized with the 24 fps video.

But the audio sound is lowered. There is a better way to do it?

tebasuna51

9th April 2024, 01:04

The audio volume is not changed in the operation.

Hellboy.

9th April 2024, 03:32

Probably is the vlc and my no so good speakers.

Axeldook

30th May 2024, 13:54

Hi,

is this possible using this tool please?

thanks

tebasuna51

30th May 2024, 21:55

https://professionalsupport.dolby.com/s/question/0D54u00008FGd6YCAT/converting-atmos-to-dd-with-joint-object-coding?language=en_US

Guest

31st May 2024, 11:00

Hi,

is this possible using this tool please?

thanks

Not too sure about the JOC part, but DoVi_Scripts has an option to convert to E-AC3

tebasuna51

31st May 2024, 13:43

The EasyAudioEncoder used in DoVi_Scripts can't output EAC3 Atmos data (JOC).

Axeldook

2nd June 2024, 16:59

Thank you

Hellboy.

14th June 2024, 22:59

tebasuna51
When i use UsEac3To to convert a DTS-HD 5.1 to ac3 5.1 using ffmpeg-A/V Recode, the command line use -center_mixlev 0.707.
What is the benefit of using this? Because the ffmpeg documentation say that is used "when downmixing to stereo".
Thanks.

tebasuna51

15th June 2024, 08:32

That parameter, included in the BSI info of each AC3 frame, is only used when it is downmixing automatically to stereo, yes.

Like the default used by ffmpeg AC3 encoder is -center_mixlev 0.500 (I don't know for what, in old Aften encoder was 0.707) I recommend put it to 0.707 to avoid the know problem of low volume dialogs when downmix.

Of course playing it at 5.1 do nothing.

hypnotize666

20th June 2024, 01:01

tebasuna51

20th June 2024, 10:05

The CLP you use is:

stdout.wav -normalize | qaac -v 99 --he --ignorelength --adts --no-delay -o %_.aac -

output a error:
--num-priming is only applicable for AAC LC.

In the qaac help show:
Applicable only for AAC LC.
--num-priming=0 is the same as --no-delay.
Without the --no-delay parameter the output work fine:
AAC-HE Encoder, CVBR 96kbps,...

Using qaac --formats you can see the allowed bitrates for HE CVBR 48KHz 5.1 sources:
...
HE 48000Hz 5.1 (C L R Ls Rs LFE) -- 80,96,112,128,160,192
...
If you put 99 take 96 Kb/s

Without the --no-delay I recommend output in mp4 container:

stdout.wav -normalize | qaac -v 96 --he --ignorelength -o %_.m4a -

Emulgator

20th June 2024, 13:47

Like the default used by ffmpeg AC3 encoder is -center_mixlev 0.500
(I don't know for what, in old Aften encoder was 0.707)
I recommend put it to 0.707 to avoid the know problem of low volume dialogs when downmix.
Just as I came across audio 2.0 tracks consisting of the original mono copied to L and to R, so phase correlation = 1:
If those in-phase tracks would happen to reach 0dB (what they rarely do) -center_mixlev 0.500 would be only safe way to stay below clipping.
-center_mixlev 0.707 would allow 1.41 FS for that corner case, so I guess ffmpeg just wanted to stay away from clipping.
If user could rely that each such fully correlated track would never exceed -6dB then -center_mixlev 0.500 would be safe.
In the end at my place it would boil down to inspect the waveform first, then decide how to continue.

Hellboy.

21st June 2024, 01:53

I know that to preserve the Atmos data you must extract it with -keepDialnorm (eac3to).
But if i have an Atmos that i don't know if -keepDialnorm was used, how i can check?
Thanks.

tebasuna51

21st June 2024, 10:33

...-center_mixlev 0.707 would allow 1.41 FS for that corner case, so I guess ffmpeg just wanted to stay away from clipping.
Well, I prefer some clip than don't listen the dialogs, with the 0.707 coeficient the volume of center channel is preserved, with low coeficients means attenuate the dialogs.
In the end at my place it would boil down to inspect the waveform first, then decide how to continue.
Of course the best option when that audio must be played with a stereo speakers is make a proper downmix, even with greater coeficient for center channel, and Normalize it like is explained here (https://forum.doom9.org/showthread.php?p=2003223#post2003223)

Emulgator

21st June 2024, 11:39

Yep, normalize before is a good solution.

tebasuna51

21st June 2024, 11:55

I know that to preserve the Atmos data you must extract it with -keepDialnorm (eac3to).
The data are preserved (same size and MediaInfo show both like JOC) but with Dialnorm changed (if show -31 dB maybe it is changed) seems not compliant.

Extracted with -keepDialnorm my Denon player show DD Atmos and decoded with Dolby Reference Player seems fine.

Without -keepDialnorm (or converted to mkv from m2ts with MkvToolnix) my Denon player show DD+ and decoded with Dolby Reference Player we obtain only silence.

Like I say MediaInfo show JOC for both and Cavernize decode both with the same output (seems more permissive).

hypnotize666

21st June 2024, 19:28

Hi everyone, I would like to know the step to follow to have an aac type file at the output He is normalized so I imagine that it must be cvbr (aac lc sbr) I tried with this type of parameter but without success.Thanks
http:////imgur.com/Y5qGdNM
http://imgur.com/yelyyjJ
Does the info of the quality 5.1 then stereo are good if not can we have more info to understand how to configure it plz?thx
Hello again, here is my structure of my folder with the codecs I have chosen the qaac codec but why in the log does it indicate a nero direct show codec error?This is the log file:eac3to v3.36
command line: "C:\Installation\Encodeurs\eac3to336-UsEac3to133\eac3to.exe" "K:\00-Classics\Teleseries\MED (2015) {tvdb-304283}\s01\MED s01e01 French Hdtv x265 Hypnotize.mkv" 2: stdout.wav -normalize -progressnumbers -log="K:\00-Classics\Teleseries\MED (2015) {tvdb-304283}\s01\MED s01e01 French Hdtv x265 Hypnotize.mkv_Job_1.log"
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 0:22:22, 30p /1.001
1: h265/HEVC, 720p30 /1.001 (16:9), 10 bits
2: AAC, French, 5.1 channels, 24kHz, 40ms
"Francais 5.1 AAC@192Kbps HE"
[a02] Extracting audio track number 2...
[a02] Decoding with DirectShow (Nero Audio Decoder 2)...
[a02] Getting "Nero Audio Decoder 2" instance failed. <ERROR>
Aborted at file position 262144. <ERROR>
http://imgur.com/h4hmNKg
http://imgur.com/Z38eAp8

hypnotize666

21st June 2024, 19:46

tebasuna51

22nd June 2024, 09:35

... but why in the log does it indicate a nero direct show codec error?

Because eac3to need Nero 7 installed to decode AAC, NeroAacDec is not enough. Decode it before to wav with ffmpeg (A/V Recode) and after normalize and recode to AAC.

tebasuna51

22nd June 2024, 09:41

I found it a bit fishy but I still tried with this command: stdout.wav -normalize | qaac -v 192 --he --ignorelength -o %_.m4a - but the strange thing is that by clicking on the menu at the end it does not mark a .m4a type file but rather an .aac type file as I wasn't sure I tried 2 ways but without success 2: stdout.wav -normalize | qaac -v 192 --ignorelength -o %_2fre.aac -

What menu?
Without success?

Both commands works fine for me (of course over a source than can be decoded by eac3to), the first create a file_.m4a (aac HE in mp4 container) and the second a file_2fre.aac LC

mannequin80

23rd June 2024, 19:06

Well, I prefer some clip than don't listen the dialogs, with the 0.707 coeficient the volume of center channel is preserved, with low coeficients means attenuate the dialogs.

is there a universal way to enhance the dialogue (center channel) without applying additional filters (like dynaudnorm)?

tebasuna51

24th June 2024, 09:54

@mannequin80
Read the thread convert 5.1 to stereo (https://forum.doom9.org/showthread.php?t=185520)

Players can be instructed to enhance the FC in the downmix, for instance with mpc-hc:

mannequin80

24th June 2024, 22:02

@mannequin80
Read the thread convert 5.1 to stereo (https://forum.doom9.org/showthread.php?t=185520)

i'd love to keep it at 5.1 and make it playable across various devices with good sounding dialogue while maintaining the dynamic range as much possible and not using any hardware methods like DRC or Night Mode. after looking at multiple methods, i can't find a better solution than "volume=x.xdB (if needed), dynaudnorm=p=0.9:m=2:g=31:f=300". there's no easily detectable volume pumping even in complicated scenes BUT the drawback of this method is the loud parts (like music, etc) become even louder so it's difficult to preserve the balance. wondering about any additional tweaks to try.

DanDare1983

26th June 2024, 23:39

tebasuna51

27th June 2024, 08:57

Do you have to -keepdialnorm when extracting atmos? I remember using the old eac3to and it would automatically remove dialnorm as it was supposed to help in some way. I'm using -keepdialnorm from now on as I think it's the best thing to do especially as dgdemux and makemkv keeps dialnorm too.

Just a few days before:

The data are preserved (same size and MediaInfo show both like JOC) but with Dialnorm changed (if show -31 dB maybe it is changed) seems not compliant.

Extracted with -keepDialnorm my Denon player show DD Atmos and decoded with Dolby Reference Player seems fine.

Without -keepDialnorm (or converted to mkv from m2ts with MkvToolnix) my Denon player show DD+ and decoded with Dolby Reference Player we obtain only silence.

Like I say MediaInfo show JOC for both and Cavernize decode both with the same output (seems more permissive).

spinkeln

2nd August 2024, 13:44

Deleted

MrVideo

5th September 2024, 07:31

Do you have to -keepdialnorm when extracting atmos?
Using eac3to v3.52, I've discovered that using -keepdialnorm wasn't necessary as eac3to indicates, when it starts to operate on the source, that it is "Keeping dialnorm."

The job that was doing was extracting the thd and ac3 streams from a mkv file so that I can THD merge the two.

MrVideo

5th September 2024, 08:08

Is it possible with (Us)Eac3to to create an eac3 file with an empty core, like what can be done with a THD stream within a MKV file?

tebasuna51

5th September 2024, 09:14

Using eac3to v3.52, I've discovered that using -keepdialnorm wasn't necessary as eac3to indicates, when it starts to operate on the source, that it is "Keeping dialnorm."
It is not necesary if you have the -keepdialnorm in the eac3to.ini file.

Is it possible with (Us)Eac3to to create an eac3 file with an empty core, like what can be done with a THD stream within a MKV file?

UsEac3to can create eac3 files using ffmpeg (only to 5.1 and without ac3 core, only eac3 frames) but I can't understand the relation with THD.

To create a THD track BD compliant need a ac3 file, not eac3.
To create a eac3 track BD compliant you can't use ffmpeg (maybe Dolby Encoder Engine).

MrVideo

6th September 2024, 03:53

It is not necesary if you have the -keepdialnorm in the eac3to.ini file.
It is there, but I never put it there. Must be part if the distro.
UsEac3to can create eac3 files using ffmpeg (only to 5.1 and without ac3 core, only eac3 frames) but I can't understand the relation with THD.

THD was only mentioned as an example.

tebasuna51

6th September 2024, 08:20

Must be part if the distro.
Yes, it is. I don't like that option and I delete that option from the .ini file.

MrVideo

6th September 2024, 22:23

Yes, it is. I don't like that option and I delete that option from the .ini file.
Why don't you like the option?

tebasuna51

7th September 2024, 10:27

The Dialog Normalization different than -31 dB force to compliant decoders to attenuate the E-AC3 tracks all the sound.
Thats is a problem to recode, or listen, compared with others codecs.

For instance, with old DVD's with AC3 and DTS tracks all the people prefer the DTS track because sound high, normally 4 dB high with standard DN at -27 dB, than the same AC3 track.
The DTS encoder is not better, at same bitrate, than the AC3 only it is decoded without attenuation.
The same source encoded with GUIDE: How To Properly Encode Dolby Digital Audio (AC3) (https://forum.doom9.org/showthread.php?t=56020) recomendations sound with less volume than encoded with AAC, DTS,...

The same for the Dolby recomendation about Dynamic Range Compression, when we need recode a E-AC3 track we need force decoders (ffmpeg, avs decoders) to use drc=0.
When ffmpeg encode AC3 or EAC3 by default always use DN -31 dB and DRC=none to offer the same behaviour than others codecs.

If all the audio was encoded with Dolby defaults DN/DRC all can be ok, but if you listen Dolby compliants tracks and TV comercials or CD audio (Loudness war (https://en.wikipedia.org/wiki/Loudness_war)) or the same track encoded with other codec, you need modify the volume button every time.

MrVideo

8th September 2024, 08:13

Thanks for the info. I certainly do not want DRC. I'll remove the keepdialnorm option from the ini.

SeeMoreDigital

8th September 2024, 10:06

With regard to Dolby audio streams, is Dialnorm applied to all the available channels or just the centre channel?

tebasuna51

8th September 2024, 22:11

With regard to Dolby audio streams, is Dialnorm applied to all the available channels or just the centre channel?

To all. Just a part of GUIDE: How To Properly Encode Dolby Digital Audio (AC3) (https://forum.doom9.org/showthread.php?t=56020):
The decoder will perform an attenuation of (31 + dialnorm) dB to the program material when played back. So, in this case, the decoder will attenuate by (31 + -18) = 13 dB. This will bring the average sound level of the material to (-17.6 - 13) = -30.6 dBFS. The program is now played back at approximately -31 dBFS, the reference level.

-31 dBFS is a lower average volume level than what is typical from other sources. It will be noticeable that you will have to turn the volume up on your system when playing a DVD versus playing broadcast, tape, or other non-Dolby Digital program material.

SeeMoreDigital

9th September 2024, 09:27

To all. Just a part of GUIDE: How To Properly Encode Dolby Digital Audio (AC3) (https://forum.doom9.org/showthread.php?t=56020):
Thanks ;)

tormento

15th October 2024, 22:17

Does it work with last eac3to?

I have tried to read a mkv video and the eac3to about appears, instead of the track list.

Yes, I have set the exe paths.

tebasuna51

16th October 2024, 00:21

No problem here:

eac3to v3.52
command line: "C:\Portable\eac3to\eac3to.exe" "D:\Temp\t\00_HD.mkv" -progressnumbers -log="C:\Portable\eac3to\UsEac3to\UsEac3To.log"
------------------------------------------------------------------------------
Running in fast mode
Removing dialnorm
MKV, 1 video track, 1 audio track, 0:00:21, 25p
1: h264/AVC, English, 720p25
2: DTS Hi-Res, 7.1 channels, 2814kbps, 48kHz, dialnorm: 0dB
(core: DTS-ES, 5.1 channels, 1509kbps, 48kHz, dialnorm: 0dB)

tormento

16th October 2024, 08:46

No problem here
I did a clean installation of last version and no way to make it work, both drag and dropping and opening. Tried to rename file to a single character but nothing helped.

It opens a cmd window telling "analyze" and then the useac3to shows eac3to help page.

Would you please post here your eac3to.ini file?

Is there any way to trace the commands sent to eac3to? The log doesn't contain them.

tebasuna51

16th October 2024, 11:02

UsEac3to not need any installation other than fill the Settings options (if can't obtain them automatically):
Path to eac3to
Path to MkvMerge
Path to TsMuxer
Encoders Folder
Output Folder

You don't need eac3to.ini at all because the -progressnumbers -log= is added by UsEac3to (or can be added manually), but I have:
-fast
-progressnumbers

And the command line is show in the log like you can see in my previous post.

tormento

16th October 2024, 16:14

You don't need eac3to.ini at all because the -progressnumbers -log= is added by UsEac3to (or can be added manually), but I have
Ok, found the problem, I had '-nolog' in eac3to.ini.

Axeldook

19th October 2024, 11:33

Hi All,
when trying to encode to true hd. i lose channels.
5.1 come out 5.0
7.1 come out 5.0

how do i fix this ?

Thank you

tebasuna51

19th October 2024, 12:13

5.1 come out 5.0
Work fine here:
"C:\Portable\0\ffmpeg.exe" -i "C:\tmp\6p321.wav" -vn -strict -2 -acodec truehd "C:\tmp\6p321.wav_.thd"
ffmpeg version N-116720-g5c1c0325cd-20240817 Copyright (c) 2000-2024 the FFmpeg developers
Input #0, wav, from 'C:\tmp\6p321.wav':
Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 5.1(side), s32 (24 bit), 6912 kb/s
Output #0, truehd, to 'C:\tmp\6p321.wav_.thd':
Stream #0:0: Audio: truehd, 48000 Hz, 5.1(side), s32p (24 bit), 128 kb/s

7.1 come out 5.0
You are right here:
"C:\Portable\0\ffmpeg.exe" -i "C:\tmp\8w341.wav" -vn -strict -2 -acodec truehd "C:\tmp\8w341.wav_.thd"
Input #0, wav, from 'C:\tmp\8w341.wav':
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 7.1, s16, 6144 kb/s
Output #0, truehd, to 'C:\tmp\8w341.wav_.thd':
Stream #0:0: Audio: truehd, 48000 Hz, 5.0(side), s16p, 128 kb/s

The ffmpeg thd encoder only can encode to 5.1, then need downmix previously 7.1 -> 5.1 and the last versions have a bug (already reported but ignored (https://trac.ffmpeg.org/ticket/10971)) when do so.

To recode to thd 7.1 you need any commercial encoder.

Axeldook

19th October 2024, 12:42

thank you !

DanDare1983

21st October 2024, 16:10

I've come across a DTS-HD MA 5.1 track which has a 1000ms DELAY. The Log is telling me that a remaining delay of -2ms could not be fixed. I understand that -2ms is nothing to worry about but could someone explain how eac3to has got the delay down to -2ms? I've also seen an ATMOS track with a delay of 3003ms too. Also if i was to just convert the track to ac3 would it sort the problem? I'll also add that the video is 2: h264/AVC, 1080i50 (16:9).

tebasuna51

21st October 2024, 21:58

To fix delay without recompress eac3to add silent frames (or remove frames).
Only multiples of 10.666 ms (512 samples in each frame at 48 KHz) can be fixed exactly.

DanDare1983

21st October 2024, 23:19

To fix delay without recompress eac3to add silent frames (or remove frames).
Only multiples of 10.666 ms (512 samples in each frame at 48 KHz) can be fixed exactly.

Thanks for the explanation. So there's nothing to worry about and eac3to is doing its job?

tebasuna51

22nd October 2024, 08:12

You preserve the quality (without decode/recode) and the remaining delay is always less than 6 ms., far than the duration of a video frame.

DanDare1983

23rd October 2024, 00:36

You preserve the quality (without decode/recode) and the remaining delay is always less than 6 ms., far than the duration of a video frame.

I'm currently trying to convert DTS-HD 7.1 to AC3 5.1 640kbps. Can I just choose the ac3 option or do I have to enter -down6 too?

tebasuna51

23rd October 2024, 00:47

You can use the automatic ffmpeg downmix, the -down6 eac3to method or the recommended downmix 71-51o using F-FFMPEG functions

SeeMoreDigital

23rd October 2024, 10:00

I'm currently trying to convert DTS-HD 7.1 to AC3 5.1 640kbps. Can I just choose the ac3 option or do I have to enter -down6 too?
Out of interest... Is there any reason why you don't want to use the 5.1 DTS 'core'?

tebasuna51

24th October 2024, 00:19

The core is a lossy encoded, for what use it if you have the losless source?

SeeMoreDigital

24th October 2024, 10:13

The core is a lossy encoded, for what use it if you have the losless source?DanDare1983 is encoding from lossless 7.1 DTS-HD MA to lossy 5.1 Dolby Digital. So my question is, why not extract 'and play' the lossy 5.1 DTS stream instead of encoding and playing a lossy 5.1 Dolby Digital stream?

tebasuna51

24th October 2024, 13:29

Of course, but some players (like some TV's, for instance the mine) can't support even the lossy core DTS, without problem with AC3 (my TV can send by SPDIF the AC3, also EAC3 5.1.2 to my Denon 5.1.2)

There are also the size to store, the AC3 640 Kb/s have equivalent quality than DTS core 1510 Kb/s (size x 2.3)

DanDare1983

13th January 2025, 13:46

Hi just a quick question. I have a playlist where the DTS-HD MA track has a delay of -1001ms. eac3to gets it down to -2ms which i understand is perfectly fine. Can I ask if the DTS core also has the same delay? If so, does that also get fixed?

tebasuna51

14th January 2025, 10:17

In fact when eac3to do the delay correction only add silence frames at the begining (delay +) , and that silence frames are only core frames because don't need MA extensions to codify silence.
To delay - delete first full frames (core+MA extensions)
Then the core is fixed also.

DanDare1983

14th January 2025, 13:59

tebasuna51

15th January 2025, 06:45

Only with TrueHD can't fix the delay + because can't create silent thd frames.

mockingjay

9th February 2025, 03:24

free Autoit3 (https://www.autoitscript.com/site/autoit/downloads/) to be sure than is virus free.

.

Iam for 99% sure it's safe.. but there's always that 0,1% :↓↓...

https://i.postimg.cc/hPt9JpKp/Screenshot-2025-02-09-032259-Virustotal.png (https://postimg.cc/Cd9ZXCBq)

Windefender flags it as HIGH risk too !!! So high, i can't even restore the file from AV vault .. ¯\_(ツ)_/¯ !?

Just want the OP from this gui know that ..

ta ta

tebasuna51

9th February 2025, 09:01

Here you have the source in plain text, compile it yourself or use Autoit3 to interpret the source (the GUI not need many CPU only decoders/encoders need it).

mockingjay

10th February 2025, 10:21

compile it yourself or use Autoit3 to

No need for that tone buddy !! And yes, iam refering to the GUI part.. （︶^︶）

mindals

16th February 2025, 01:16

Hello, the program only gives me 00:03:42 from a movie audio of 02:02:44 length. Why?
This is CL: 2: %_2eng.wav -25.000 -changeTo23.976 -down16
I'm using eac3to 3.52
Can you help me, please?

tebasuna51

16th February 2025, 07:36

Hello, the program only gives me 00:03:42
What program say you that duration? Of course not is eac3to.

The %_2eng.wav output from your CL must have more than 4 GB and a duration of more than 2h

Try with CL: 2: %_2eng.w64 -25.000 -changeTo23.976 -down16

mindals

17th February 2025, 02:27

What program say you that duration? Of course not is eac3to.

The %_2eng.wav output from your CL must have more than 4 GB and a duration of more than 2h

Try with CL: 2: %_2eng.w64 -25.000 -changeTo23.976 -down16

Solved. I used .flac instead of .wav.
I think this could be due to .wav size limitations.
Thanks, tebasuna51.

Senior72

12th March 2025, 14:24

Hello, with the latest version I get a virus message
Windowsdefender:

Trojan:Win32/Predator!MTB

This trojan is used in multiple spear-phishing campaigns. In these campaigns, we have seen HTML smuggling being utilized to assemble a JavaScript or MSI file, which contains an embedded script. The script downloads the main malicious payload, which is a DLL file executed via DLL sideloading. The DLL steals credentials as well as data related to banking. Attackers typically distribute the trojan as an email attachment.

What to do now

tebasuna51

12th March 2025, 15:33

What to do now

Nothing, it is a false positive.

SeeMoreDigital

10th April 2025, 12:22

tebasuna51

10th April 2025, 13:10

I'm in W10, I can't know how you can disable your antivirus in W11.

BTW see first post link to download the free autoit3 package and encode the exe yourself.
Also the source .au3 can be interpreted with autoit3 (only the drag & drop don't work).

Encore

10th April 2025, 13:14

I'm struggling to install the version 1.3.4. Everytime I try to unzip the file Windows 11 throws a fit a automatically deletes it from my downloads folder!

What's the procedure to stop this from happening?

Add the folder of you are extracting to (or create a new folder) to windows security (or antivirus you have installed) scan settings exclusion (Virus & threat protection -> Virus & threat protection settings (click manage settings) -> Exclusions (scroll down if windows scaled).

Encore

10th April 2025, 13:15

@tebasuna51 is eac3to 3.5.x still in developement? where is that thread for it?

Edit : I found curly's forum...

SeeMoreDigital

10th April 2025, 13:36

BTW see first post link to download the free autoit3 package and encode the exe yourself.
Also the source .au3 can be interpreted with autoit3 (only the drag & drop don't work).
This is 'dark magic' to me ;)

Add the folder of you are extracting to (or create a new folder) to windows security (or antivirus you have installed) scan settings exclusion (Virus & threat protection -> Virus & threat protection settings (click manage settings) -> Exclusions (scroll down if windows scaled).I don't understand the proceedure you mentioned. I can however confirm that I don't have any third-party anti-virus software installed. It's just what comes with Windows 11, which is a real pain!

Encore

10th April 2025, 14:02

This is 'dark magic' to me ;)

I don't understand the proceedure you mentioned. I can however confirm that I don't have any third-party anti-virus software installed. It's just what comes with Windows 11, which is a real pain!

If you don't have any third party antivirus installed, windows 11 comes Windows Securty bundled. Bottom taskbar right corner (up arrow) will show you a bluish/green icon with a tick - clicking on it will open it up. Then follow the above steps...:)

SeeMoreDigital

10th April 2025, 21:28

If you don't have any third party antivirus installed, windows 11 comes Windows Securty bundled. Bottom taskbar right corner (up arrow) will show you a bluish/green icon with a tick - clicking on it will open it up. Then follow the above steps...:)Many thanks...

After visiting: 'Windows Security' -> 'Virus & Threat Protection' and clicking on 'Allowed Threats' I was able to permit the following: -

https://i.ibb.co/V0Qt1KWj/Screenshot-2025-04-10-173139.png

Now I'm able to see, access and demux the eac3 'core'... Amazing!

hardkid

14th April 2025, 11:04

tebasuna51

14th April 2025, 11:35

The .exe, even in .7z compression, don't fit in attached size.
The .exe is in sendspace link still active.

hardkid

14th April 2025, 11:49

The .exe, even in .7z compression, don't fit in attached size.
The .exe is in sendspace link still active.

So, I can just inslall autoit and start UsEac3To.au3 with autoit every time? Is this normal?

tebasuna51

14th April 2025, 12:54

You can asociate extension .au3 to AutoIt3_x64.exe, after that double click over .au3 run it interpreted.

hardkid

14th April 2025, 15:52

You can asociate extension .au3 to AutoIt3_x64.exe, after that double click over .au3 run it interpreted.

Done. Thank you.

oniiz86

3rd October 2025, 16:04

@tebasuna51 I'm not sure if you're aware but eac3to development has resumed & a new build v3.53 has just been released, I was just curious how you determined the EAC3 7.1 streams decoded as 5.1 ignoring the additional channels issue?

I demuxed a EAC3 7.1 track & MediaInfo reports it fine below, will the issue present itself if I select the A/V Recode button & then select "Recode to E/AC3, ffmpeg" which I noticed it doesn't have a bitrate greater than 640k to select & then mux it into a mkv container & play it back via my AVR, will it decode it as only EAC3 5.1 & simply discard the additional channel information?

Format : E-AC-3
Format/Info : Enhanced AC-3
Commercial name : Dolby Digital Plus
Format profile : Blu-ray Disc
File size : 737 MiB
Duration : 1 h 54 min
Overall bit rate mode : Constant
Overall bit rate : 896 kb/s

Audio
Format : E-AC-3
Format/Info : Enhanced AC-3
Commercial name : Dolby Digital Plus
Format profile : Blu-ray Disc
Duration : 1 h 54 min
Bit rate mode : Constant
Bit rate : 896 kb/s
Channel(s) : 8 channels
Channel layout : L R C LFE Ls Rs Lb Rb
Sampling rate : 48.0 kHz
Frame rate : 31.250 FPS (1536 SPF)
Compression mode : Lossy
Stream size : 737 MiB (100%)
Service kind : Complete Main
Dialog Normalization : -31 dB
compr : -0.28 dB
cmixlev : -3.0 dB
surmixlev : -6 dB

tebasuna51

4th October 2025, 09:19

Due to an accident, I'm wheelchair-bound and unable to access my usual equipment to test the new version of eac3to.
However, I don't think it'll modify the previous version for what you're asking for.

Indeed, ffmpeg doesn't encode eac3 7.1, only 5.1, unless you use the method suggested in the thread Encode to 7.1 eac3 for free and volume adjustment (https://forum.doom9.org/showthread.php?p=2016678#post2016678).
In any case, I don't see the point of re-encoding an eac3 7.1 version back to eac3 7.1.

SeeMoreDigital

4th October 2025, 10:05

Due to an accident, I'm wheelchair-bound and unable to access my usual equipment to test the new version of eac3to...So sorry to hear this. I wish you well...

oniiz86

4th October 2025, 18:59

Due to an accident, I'm wheelchair-bound and unable to access my usual equipment to test the new version of eac3to.
@tebasuna51 Oh I'm so sorry to hear this, hoping for a speedy recovery & wish you well.

GeoffreyA

4th October 2025, 20:41

Best wishes, tebasuna. Take care.

tebasuna51

4th October 2025, 20:47

Thanks for the good wishes, I hope it's just a month of discomfort.

infoxtreme

15th October 2025, 19:35

Hi tebasuna51 how are you ? I hope you are better, I wish you a speedy recovery.

after the new eac3 version and a clean install of windows 11 25h2 i'm having this problem, any ideas?

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

chainring

15th October 2025, 22:27

@infoextreme,

https://www.microsoft.com/en-us/download/details.aspx?id=26999

infoxtreme

15th October 2025, 22:39

@infoextreme,

https://www.microsoft.com/en-us/download/details.aspx?id=26999

Thanks very much, install x64 x86 all ok, works !! :thanks:

oniiz86

24th November 2025, 08:55

The names generated by the eac3to demux put the full language names and comments, for instance:
...Japanese, 'Japanese'.srt
...Korean, 'Korean'.srt
...Spanish, 'Latin America Spanish [SDH]'.srt
...Norwegian, 'Norwegian'.srt
...Polish, 'Polish'.srt

My UsEac3to GUI put only the first 3 caracters to identify the track and can't convert it to the exact language codes.

@tebasuna51 Hey Teba, I had sent you a PM recently about an eac3to bug regarding language code syntax & just noticed this post about how UsEac3to manages language codes, as I had mentioned in my PM, I've been in touch with the eac3to developer & he will release a test version shortly, this is what was mentioned "So, I have to revise eac3to to always use filenames that match the expected
mkvtoolnix filename parsing rules. I will do that. Probably, I will always use the 3-letter
code included in square brackets, e.g., '[eng]'.", is this something that you could update in UsEac3to or is it a permanent limitation that you can't get around?

Columbo

24th November 2025, 09:02

tebasuna51

24th November 2025, 10:46

... I had sent you a PM recently...
My answer to your PM:

UsEac3to is only a GUI for eac3to and the track names are generated with the eac3to info, for instance if eac3to show:
MKV, 1 video track, 5 audio tracks, 0:00:21, 25p
1: h264/AVC, English, 720p25
2: AAC, Spanish, 5.1 channels, 48kHz, dialnorm: 0dB
3: AC3, English, 5.1 channels, 448kbps, 48kHz, dialnorm: -31dB
4: E-AC3, French, 5.1 channels, 448kbps, 48kHz, dialnorm: -31dB
5: FLAC, German, 5.1 channels, 0:00:20, 16 bits, 48kHz, dialnorm: 0dB
6: OPUS, Portuguese, 5.1 channels, 48kHz, dialnorm: 0dB

the track names are generated like 2spa, 3eng, ...
the track number and the 3 first characters of the languaje, sometimes can be recognized by MkvToolNix (but not japonese for instance) if you follow the mbunkus Guide (https://help.mkvtoolnix.download/t/deriving-language-names/181/1)

I can insert a default boundary character to obtain 2-spa, 3-eng, ... but still you need config the 'Recognized languajes' not only add 1234567890 to boundary characters.

BTW when eac3to make the new version I can test it, but I hope not need changes to support it.
Maybe if instead languaje info like 'japonese' is changed by the 3 character code 'jpn' all is ok now.

Remember, when you want use the eac3to demux you must use the demux in Global Parameters.

... Probably, I will always use the 3-letter
code included in square brackets, e.g., '[eng]'.[/B]", is this something that you could update in UsEac3to or is it a permanent limitation that you can't get around?

If that info is included in the log I can modify UsEac3to, waiting for the update. BTW you can use the strict eac3to demux using Global Parameters inside UsEac3To.

oniiz86

24th November 2025, 11:27

"My UsEac3to GUI put only the first 3 characters to identify the track and can't convert it to the exact language codes."

When eac3to is revised to use the [eng] format then UsEac3to can presumably pick that up and use it.

@Columbo Unfortunately that doesn't appear to be the case, only if one uses the -demux option then it will be okay, the eac3to developer has mentioned the following "UsEac3to must be revised to make proper filenames when specifying tracks. If it simply uses -demux then things will be fine when using the revised eac3to."

If that info is included in the log I can modify UsEac3to, waiting for the update. BTW you can use the strict eac3to demux using Global Parameters inside UsEac3To.

@tebasuna51 Thanks for that info Teba, it's very much appreciated, yes I was advised that -demux will be fine but I prefer to use the Track Input and Output Format functionality. :)

Columbo

24th November 2025, 14:06

@Columbo Unfortunately that doesn't appear to be the case Sure it does. As I mentioned and teba apparently confirms he can pick it up from the eac3to log file and it works even if you are not demuxing anything at all.

Anyway, Curly just made a test version for y'all:

https://rationalqm.us/misc/eac3to_oniiz86.exe

Note that mkvtoolnix requires the square brackets so don't strip them off.

oniiz86

24th November 2025, 17:30

Sure it does. As I mentioned and teba apparently confirms he can pick it up from the eac3to log file and it works even if you are not demuxing anything at all.

Anyway, Curly just made a test version for y'all:

https://rationalqm.us/misc/eac3to_oniiz86.exe

Note that mkvtoolnix requires the square brackets so don't strip them off.

I've just tested this, I can confirm that everything is now fine when using the Global Parameter -demux option but not when using the Track Input and Output Format function, I have a Spanish PGS subtitle track that when demuxed individually the output filename is with "eng.sup" not "spa.sup" as it had been with the eac3to release versions.

Columbo

24th November 2025, 17:34

I've just tested this, I can confirm that everything is now fine when using the Global Parameter -demux option but not when using the Track Input and Output Format function
Yes, UsEac3to must be modified for using the eac3to test build.

I have a Spanish PGS subtitle track that when demuxed individually the output filename is with "eng.sup" not "spa.sup" as it had been with the eac3to release versions. Using the revised eac3to with the not-yet-revised UsEac3to will give erroneous or undefined results.

tebasuna51

24th November 2025, 18:58

With the new version eac3to show for instance:
MKV, 1 video track, 5 audio tracks, 0:00:21, 25p
1: h264/AVC, [eng], 720p25
2: AAC, [spa], 5.1 channels, 48kHz, dialnorm: 0dB
3: AC3, [jpn], 5.1 channels, 448kbps, 48kHz, dialnorm: -31dB
4: E-AC3, [fre], 5.1 channels, 448kbps, 48kHz, dialnorm: -31dB
5: FLAC, [ger], 5.1 channels, 0:00:20, 16 bits, 48kHz, dialnorm: 0dB
6: OPUS, [por], 5.1 channels, 48kHz, dialnorm: 0dB

And the new UsEac3to extract can be:

3: %_3-jpn.ac3

The - and the . are default boundary characters and can be recognized by MkvToolNix (but still you need configure it to accept 'japonese' like Recognized Languages, by default only eng, fre, ger,ita,por and spa)

EDIT: UsEac3to final v1.3.5 in first post

tebasuna51

24th November 2025, 19:09

Using the revised eac3to with the not-yet-revised UsEac3to will give erroneous or undefined results.

Use 'eng' by default because the character [ is not accepted.

oniiz86

25th November 2025, 05:05

With the new version eac3to show for instance:

And the new UsEac3to extract can be:

3: %_3-jpn.ac3

The - and the . are default boundary characters and can be recognized by MkvToolNix (but still you need configure it to accept 'japonese' like Recognized Languages, by default only eng, fre, ger,ita,por and spa)

UsEac3To_test.exe here:

UsEac3To_test.au3 attached here:

Hey Teba, thanks so much for this release, I can happily confirm that the Track Input and Output Format function works wonderfully, one curious observation, is that I noticed two additional buttons "R" & "Q" next to Global Parameters in this v1.3.5 test release, I recall you adding that in the previous v1.3.4 official release and yet those "R" & "Q" buttons weren't present at all, I only have "RUN" as you can see below,

https://i.imgur.com/pufZBP3.png

https://i.imgur.com/oAsd9Zz.png

Use 'eng' by default because the character [ is not accepted.
I'm a bit confused, I thought you changed it to use "-eng." instead not 'eng', is there any particular reason why UsEac3To can't accept [] characters, will there be some incompatibility concerns when using the latest eac3to revised test build as it has changed to utilise [] characters?

Will there be an official v1.3.5 release eventually that includes both 32-bit & 64-bit variants as you've done in the past updating the first post?

tebasuna51

25th November 2025, 09:02

...I noticed two additional buttons "R" & "Q" next to Global Parameters in this v1.3.5 test release, I recall you adding that in the previous v1.3.4 official release and yet those "R" & "Q" buttons weren't present at all, I only have "RUN" as you can see below,
Added to v1.3.4 without change version (https://forum.doom9.org/showthread.php?p=1996814#post1996814):
Changelog
=========
- v1.3.4 2024-01-31 If eac3to version is > 3.47 -demux with -destpath=
Added enqueue button 'Q' for demux

is there any particular reason why UsEac3To can't accept [] characters, will there be some incompatibility concerns when using the latest eac3to revised test build as it has changed to utilise [] characters?

Because the '[' (second character after the ',' in the log line) can't be part of a language code, then 'eng' is assigned by default.
Now the test version 1.3.5 read the third character (and next 2) to obtain the language code.

Like the default Boundary characters in MkvToolNix are: [](){}.+-=#
And the '.' is already compliant I use only '-' :
...3-jpn.
(similar to previous version, more readable and compatible with the option 'Run and MkvMux' that also uses language codes)

Instead:
...3[jpn.

Will there be an official v1.3.5 release eventually that includes both 32-bit & 64-bit variants as you've done in the past updating the first post?
When eac3to make that test in official release.

Columbo

25th November 2025, 17:52

Curly released 3.55 with the language string change.

oniiz86

26th November 2025, 06:27

Added to v1.3.4 without change version (https://forum.doom9.org/showthread.php?p=1996814#post1996814):
Changelog
=========
- v1.3.4 2024-01-31 If eac3to version is > 3.47 -demux with -destpath=
Added enqueue button 'Q' for demux

Because the '[' (second character after the ',' in the log line) can't be part of a language code, then 'eng' is assigned by default.
Now the test version 1.3.5 read the third character (and next 2) to obtain the language code.

Like the default Boundary characters in MkvToolNix are: [](){}.+-=#
And the '.' is already compliant I use only '-' :
...3-jpn.
(similar to previous version, more readable and compatible with the option 'Run and MkvMux' that also uses language codes)

Instead:
...3[jpn.

When eac3to make that test in official release.

Thanks for that, I'm sorry you're absolutely right, I'm a goof I was using the initial release v1.3.4 with a January 31st 2024 1:57pm build date unaware that it was updated the following day with a February 1st 2024 6:33am build date as soon as I replaced the UsEac3To/UsEac3To32.exe files with the current builds the "R" & "Q" buttons are present. :D

redford1974

26th November 2025, 08:00

Hello, I wanted to mix four tracks in tsMuxer 2.7.0. and I get this error : unsupported codec A_SHD

tebasuna51

26th November 2025, 16:49

New version v1.3.5 in first post (reupload, better than test)

Changelog
=========
- v1.3.5 2025-11-26 Support eac3to v3.55 language codes.

Now the extracted tracks have the correct language codes in the name.
Now the language tracks can be recognized by MkvToolNixGUI if they are between Recognized languages (by default only: eng,fre,ger,ita,por,spa)
The button 'Run and MkvMux' work with all languages because the code is send in the command line to mkvmerge.

tebasuna51

26th November 2025, 17:14

Hello, I wanted to mix four tracks in tsMuxer 2.7.0. and I get this error : unsupported codec A_SHD
tsMuxer 2.7.0 is the final version, we can help you in this thread.
Try in https://forum.doom9.org/showthread.php?t=176668

oniiz86

26th November 2025, 17:17

Hellboy.

3rd December 2025, 17:05

I think I knew the answer to this, but i forgot.
If I cut a part from the beginning of a "TrueHD Atmos" track, the Atmos part get broken?
Thanks.

tebasuna51

3rd December 2025, 19:32

If you can cut a exact number of thd frames no problem.
But eac3to can't cut thd tracks, copy it with a name including something like '...- DELAY -XXXms.thd'

oniiz86

4th December 2025, 15:10

@tebasuna51 I was just curious as to why there is a difference concerning H264 & H265 streams when selecting the Output Format, the H264 stream has the option to output the stream into the MKV container or as an H264 Elementary Stream but with a HEVC/H265 stream it only offers to output as an H265 Elementary Stream, I may be overlooking something so simple. :)

https://i.imgur.com/gYvKLEc.jpeg

https://i.imgur.com/EDJ8ElW.png

tebasuna51

4th December 2025, 18:59

Because gdsmux (part of old Haali Media Splitter used by eac3to) does not support h265 (https://haali.net/mkv/codecs.pdf).

oniiz86

8th December 2025, 08:04

Because gdsmux (part of old Haali Media Splitter used by eac3to) does not support h265 (https://haali.net/mkv/codecs.pdf).
@tebasuna51 Thanks for that, I find it surprising though that UsEac3to will allow a H265/HEVC stream from an MKV file to be extracted as a H265 Elementary Stream.

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

If I import the H265 Elementary Stream into UsEac3to, the Output Format options are only H264 or MKV.

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

I know you said it's a gdsmux limitation with the haali media splitter but I was surprised that it can be extracted fine from the MKV container as H265.

Columbo

8th December 2025, 12:49

Why do you find it surprising? Muxing and demuxing are different things.

oniiz86

9th December 2025, 06:32

I understand they're different things but was curious why you can extract a HEVC stream from an MKV file but if a H265/HEVC stream is imported into UsEac3to it has H264 or MKV options instead for a H265/HEVC stream, I get it because @tebasuna51 mentioned it's a gdsmux limitation with the haali media splitter not offering H265 support but it appears you can still get around the issue if a MKV file is input.

tebasuna51

9th December 2025, 09:24

If I import the H265 Elementary Stream into UsEac3to, the Output Format options are only H264 or MKV.

Of course eac3to can't recode h265 to h264 and show:
This video conversion is not supported. <ERROR>

Also can't mux h265 to mkv and show a error:
h265/HEVC video track muxing to MKV is currently not supported. <ERROR>

It's a clear bug in UsEac3to when it offers those options (I'm only looking at the first three characters, 'h26' for h264).

I'll fix it in the future; it's obvious that eac3to can't do anything with an h265 track.

DanDare1983

23rd December 2025, 17:44

Hey teba, whilst trying to convert a DTS-HD MA 2.0 track to Flac i encountered an error/crash which was something yo do with flac first pass. I'm using the latest eac3to version 3.57 with the latest UsEac3to. Also I have quite alot of atmo tracks/7.1 tracks and would it be ok to simply encode to ac3? Would i have to use the -down6 option or can I just leave it as just ac3?

tebasuna51

24th December 2025, 09:25

Hey teba, whilst trying to convert a DTS-HD MA 2.0 track to Flac i encountered an error/crash which was something yo do with flac first pass. I'm using the latest eac3to version 3.57 with the latest UsEac3to.
No problem here:
eac3to v3.57
DTS Master Audio, 2.0 channels, 16 bits, 48kHz, dialnorm: 0dB
(core: DTS, 2.0 channels, 1509kbps, 48kHz, dialnorm: 0dB)
Decoding with libDcaDec DTS Decoder...
Encoding FLAC with libFlac...
Creating file "D:\Test\AudioN\DtsHD\20\stream2MA.dts_.flac"...
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 1 second.
Done.
Please upload the eac3to log or a sample of your DTSMA

[EDIT]Try also ffmpeg recode with UsEac3to option 'A/V Recode' selecting 'flac' in 'Decode, Lossless,...' options.

Also I have quite alot of atmo tracks/7.1 tracks and would it be ok to simply encode to ac3? Would i have to use the -down6 option or can I just leave it as just ac3?
Never use the old Aften encoder included with eac3to, use always ffmpeg like ac3 encoder. You have 3 options:

1) Decode with old eac3to libav decoder and downmix with eac3to -down6 using the ac3-ffm Output format, like this:
-down6 stdout.w64 | ffmpeg -i - -c:a ac3 -b:a ...

2) Recode with ffmpeg automatic downmix with UsEac3to option 'A/V Recode' selecting only the bitrate.

3) Like 2) but do the recommended downmix 71-51o like this:

Wile_E_Coyote

29th December 2025, 11:20

tebasuna51

30th December 2025, 06:45

Like you say that not gain anything.

seagate

30th December 2025, 20:01

tebasuna51, hello

Today I downloaded Hawaii Five-0 - Brian Tyler 2023 OST in flac files.

Only a few out of 41 flac files could not be converted to ac3.

I am using UsEac3to v.1.3.5, eac3to v.3.57. I am using Windows 10 22H2 (19045.2728) education.

I am attaching a picture so you can see the error.

"I DonТt Ride Shotgun.flac" not found. <ERROR>"

Specifically this file (flac) is not recognized, and most of the 41 files are successfully recognized by UsEac3to v.1.3.5.

I am also attaching only one of the several (5) problematic flac files for you to test and/or determine why there is a problem.

If you need it - I will also upload a flac file that was successfully recognized by UsEac3to v.1.3.5 and I converted it to AC3 without any problems.

https://i.postimg.cc/sgC5fD6v/haa.png

39 I Don’t Ride Shotgun.flac (https://seyarabata.com/695420bcb0e66)

Complete name : 39 I Don’t Ride Shotgun.flac
Format : FLAC
Format/Info : Free Lossless Audio Codec
File size : 11,9 MB
Duration : 55 s 693 ms
Overall bit rate mode : Variable
Overall bit rate : 1 793 kb/s
Album : Hawaii Five-0 (Original Series Soundtrack)
Album/Performer : Brian Tyler
Part : 1
Part/Total : 1
Track name : I Don’t Ride Shotgun
Track name/Position : 39
Track name/Total : 41
Performer : Brian Tyler
Genre : Score
Recorded date : 2023
ISRC : USLS52228539
Copyright : (C) 2023 Lakeshore Records (P) 2023 Lakeshore Records
Cover : Yes
Cover type : Cover (front)
Cover MIME : image/jpeg
UPC : 0780163628524
MEDIATYPE : album
ITUNESADVISORY : 0

Audio
Format : FLAC
Format/Info : Free Lossless Audio Codec
Duration : 55 s 693 ms
Bit rate mode : Variable
Bit rate : 1 549 kb/s
Channel(s) : 2 channels
Channel layout : L R
Sampling rate : 48.0 kHz
Bit depth : 24 bits
Compression mode : Lossless
Stream size : 10.3 MiB (86%)
Writing library : libFLAC 1.1.0 (2003-01-26)
MD5 of the unencoded conten : 00000000000000000000000000000000

Addendum:

Oh my god. I figured out why this problem was occurring.

The problem was due to the apostrophe in the name: "Don’t".

Can’t this be corrected somehow in the software so that it recognizes correctly?!

Columbo

31st December 2025, 01:46

if that is an issue with eac3to itself I can look at it. The ' character is legal in a filename.

tebasuna51

31st December 2025, 05:55

The apostrophe (') character ascii code 39 is legal without problem, but in your name "39 I DonÆt Ride Shotgun.flac" there are the ascii char 149 show in my codepage like other than ('), in your image is showed like T.

Edit the name and replace that character with the (') in your keyboard (ascii code 39).

SeeMoreDigital

31st December 2025, 10:58

There is a world of difference between don’t and don't...

tebasuna51

1st January 2026, 06:18

Today I downloaded ...

And remember rule:

6) No warez, cracks, serials or illegally obtained copyrighted content! Links to content of a questionable nature (e.g. anything you don't own and/or have downloaded), asking for, offering, or asking for help/helping to process such content in any way or form is not tolerated.

This problem hapen with names created in other countries or OS, not with names created by yourself.

tebasuna51

2nd January 2026, 10:17

Hi
Is there a way to output RF64 (WAV) files with eac3to?
I can change the audio FPS without any problem, but I can’t find an option to force RF64 as the output format.
It looks like W64 is the only similar format available.
I’m currently automating the process with a batch script using eac3to + Dolby Encode Engine.

This is a private message received; please use the standard forum for technical questions like this.

In the help you have the answer, use .rf64 like extension.

In my next new GUI version 1.3.6 I offer that output and other new parameters.

tebasuna51

3rd January 2026, 06:38

New version at first post with minor changes

Changelog
=========
- v1.3.6 2026-01-03 Add some options (-forced,...)

oniiz86

3rd January 2026, 08:19

Thanks so very much for releasing v1.3.6 so swiftly Teba, a wonderful start to the New Year, it's so greatly appreciated, a very Happy New Year to you my good sir!! :thanks:

sadpunk

3rd January 2026, 08:44

Thank you @tebasuna51

I’m changing the FPS of AAC audio files.
When decoding AAC, eac3to uses the Nero Audio Decoder. Would it make more sense to use FFmpeg instead of Nero, or is it sufficient to rely only on eac3to?

Currently, I decode AAC with FFmpeg using pcm_s32le and -rf64 always, then I change the FPS of the resulting RF64 audio file and finally convert it to AC-3 or E-AC3 using Dolby Encode Engine (DEE).

I’m using this workflow for AC-3 as well, since I don’t want to use libaften for AC-3 encoding and therefore prefer encoding AC-3 via DEE.

tebasuna51

4th January 2026, 13:15

eac3to decode aac with the old Nero7 decoder if it is installed.
I use always ffmpeg, also like ac3/eac3 encoder, then eac3to is not used at all.

If you want use DEE like encoder can do the first part with UsEac3to (select the fps change in the atempo parameter) like I show in the image:

DanDare1983

28th January 2026, 09:53

Hey teba, just a quick question. Is .w64 audio the same as .wav? I've seen some people converting .w64 audio to .wav and I wonder why? If I was to make a remux would it be ok to just leave the PCM audio as .w64?

tebasuna51

28th January 2026, 13:10

The audio data is the same, but the w64 header (also the rf64 format) have fields to support filesize greater than 4 GB (multichannel audio movie tracks needs that sizes).
It is safe convert w64 to wav if < 4 GB.
Some software can support also wavs > 4 GB with fake header but it is not recommended.

IntelHEVC

17th March 2026, 18:37

“The Dolby Vision data is being interpreted incorrectly. For example, DV 7.6 is physically present, but the GUI always shows it as DV 8.”

tebasuna51

18th March 2026, 10:25

The GUI only show the eac3to.exe output, seems it is a problem to be reported at https://www.rationalqm.us

Columbo

18th March 2026, 12:56

Or you can provide a source sample here and tell how you know it is 7.1 and DG Tools will look into it. Aternatively, provide a link to purchase that exact disk together with telling how you know it is 7.1.

Columbo

18th March 2026, 13:44

Please test this:

https://rationalqm.us/misc/eac3to.exe

Without a sample I can't prove the fix so your help will be appreciated.

Kyaneos

20th March 2026, 11:18

I hadn't used UsEac3to for a while, and after installing the latest version, I see that the `keep dialnorm` option is enabled by default. In previous versions, if I remember correctly, the `remove dialnorm` option was enabled by default.

I understood that dialnorm was obsolete and that it was recommended to remove it from Dolby tracks. Has the recommendation changed? Is it now recommended to keep it?

Thank you very much.

tebasuna51

21st March 2026, 09:03

It is optional, by default eac3to.exe remove dialnorm but in last versions there are a eac3to.ini with:

-fast
-keepDialnorm
-progressnumbers

and change the behaviour.
You can edit that eac3to.ini and remove the line -keepDialnorm like I have.

tebasuna51

9th May 2026, 07:35

There is a test version v.3.62 (https://www.rationalqm.us/eac3to/) of eac3to that allows AAC decoding using a limited version of ffmpeg.exe included in the package.

It works without apparent problems in UsEac3toGUI 1.3.6.

It does not yet support .m4a files, although ffmpeg should handle it without issue.

oniiz86

9th May 2026, 08:52

@tebasuna51 I don't believe it is working with UsEac3to v1.3.6 as unless I'm misunderstanding how it works, can you please show how aac decoding works with the latest eac3to v3.62 test build with ffmpeg invocation, the use of Nero no longer seems possible according to this post https://rationalqm.us/board/viewtopic.php?p=23820#p23820

I can get 7.1 EAC3 decoding to 7.1 WAV using the Output Format "wav" & "wavs" with the latest v3.62 test build & UsEac3to has no issues at all there but I'm stumped with how it handles AAC decoding.

Columbo

9th May 2026, 09:17

@tebasuna51 I don't believe it is working with UsEac3to v1.3.6 as unless I'm misunderstanding how it works, can you please show how aac decoding works with the latest eac3to v3.62 test build with ffmpeg invocation, the use of Nero no longer seems possible according to this post https://rationalqm.us/board/viewtopic.php?p=23820#p23820
AAC decoding works fine in the 3.62 test build. What issue are you having specifically? If you insist on Nero and you have a working Nero install, you can remove ffmpeg.exe.

tebasuna51

9th May 2026, 09:46

Without problem with my GUI but the eac3to support is still limited, works this simple decode to wav (or wavs):

eac3to v3.62
command line: "C:\Portable\eac3to\eac3to.exe" "D:\Test\AudioD\Samples\aac\entrada.aac" "D:\Test\AudioD\Samples\aac\entrada.aac_-.wav" -progressnumbers -log="C:\Portable\eac3to\UsEac3to\UsEac3To.log"
------------------------------------------------------------------------------
Fast mode enabled
Dialnorm removal enabled
AAC, 5.1 channels, 24kHz, dialnorm: 0dB
Decoding AAC via ffmpeg...
eac3to processing took 1 second.
Done.

But not this recode to flac (or w64, ac3, pcm, rf64, etc.):
eac3to v3.62
command line: "C:\Portable\eac3to\eac3to.exe" "D:\Test\AudioD\Samples\aac\entrada.aac" "D:\Test\AudioD\Samples\aac\entrada.aac_-.flac" -progressnumbers -log="C:\Portable\eac3to\UsEac3to\UsEac3To.log"
------------------------------------------------------------------------------
Fast mode enabled
Dialnorm removal enabled
AAC, 5.1 channels, 24kHz, dialnorm: 0dB
Decoding with DirectShow (Nero Audio Decoder 2)...
Getting "Nero Audio Decoder 2" instance failed. <ERROR>
Aborted at file position 147018. <ERROR>

Also not all .aac are supported: mono, 7.1 or in m4a container

EDIT: also functions like slowdown, edit, ... are ignored

Columbo

9th May 2026, 12:11

oniiz86

9th May 2026, 15:23

@tebasuna51 Thanks so much for that, I'm sorry you're right, I'm a complete idiot for some silly reason I was confusing encoding with decoding, AAC decoding does work fine except for this AAC LC SBR test file that eac3to is reporting as 22kHz when it should be 44.1kHz found here https://www2.iis.fraunhofer.de/AAC/multichannel.html, eac3to has no trouble with the ffmpeg decoding the AAC LC variant at the link & reports correctly as 44.1kHz.

oniiz86

9th May 2026, 15:35

Currently, ffmpeg is invoked only for decoding to wav/wavs. For all other scenarios the previous functionality with Nero/Sonic is invoked. You need a functioning Nero/Sonic install for that. In the future transcoding scenarios may be supported. The ffmpeg decoding was added to at least allow decoding to wav/wavs as few people have a functioning Nero/Sonic setup these days. We'll add a note about this to the 3.62 notes file.

If you can provide samples for the unsupported files we can look into supporting them.

@Columbo Thanks for that, I have an issue with the ffmpeg decoding to WAV with this AAC LC SBR test file that eac3to reports as 22kHz https://transfer.it/t/PoelYd0RWDbZ

eac3to v3.62
command line: "C:\Users\MASTER\Desktop\UsEac3to136\eac3to.exe" "C:\Users\MASTER\Desktop\ChID-BLITS-EBU-Narration.mka" 1: "C:\Users\MASTER\Desktop\UsEac3to136\ChID-BLITS-EBU-Narration.mka_1-eng.wav" -progressnumbers -log="C:\Users\MASTER\Desktop\UsEac3to136\UsEac3To.log"
------------------------------------------------------------------------------
Running in normal mode
Dialnorm removal enabled
MKA, 1 audio track, 0:00:47
1: AAC, 5.1 channels, 22kHz, dialnorm: 0dB
[a01] Extracting audio track number 1...
[a01] Decoding AAC via ffmpeg...
[a01] ffmpeg failed decoding AAC. <ERROR>
Aborted at file position 942754. <ERROR>

Columbo

9th May 2026, 17:14

Thank you for the report, oniiz86.

We were using fdkaac but as you have shown it turns out to be problematic. fdkaac as a decoder is poorly supported in ffmpeg. It was primarily designed as an encoder. For HE-AAC/SBR streams specifically, it doesn't properly set the output channel layout on decoded frames, causing the filter graph to fail. The native ffmpeg AAC decoder has no such limitation. For decoding, ffmpeg's native AAC decoder is fine. It handles LC, HE-AAC, HE-AAC v2, and SBR correctly. For decoding, the output is PCM either way and the difference is imperceptible.

Therefore, we've reverted the use of fdkaac. Please re-download the test build and update eac3to.exe. Report your results.

It's still reporting 22kHz. Classic HE-AAC/SBR issue. The ADTS header encodes the base layer sample rate (22050Hz). The SBR extension doubles it to 44100Hz during decoding. eac3to reads the rate from the ADTS header so it reports 22kHz. MediaInfo knows about SBR and reports the actual decoded rate. For the ffmpeg path this doesn't affect the output as ffmpeg decodes to the full 44.1kHz and the WAV(s) will be correct. It's purely a display issue in eac3to's reported format info. Fixing it properly would require parsing the SBR signaling in the AudioSpecificConfig or ADTS extension elements, which is non-trivial.

The ffmpeg trick is a hack to give at least correct decoding. We'll have to live with its limitations and document them well.

To avoid cluttering this thread we'd prefer further issues be addressed at DG forum.

oniiz86

9th May 2026, 18:27

@Columbo Thanks for that, I confirmed all is good now with the AAC LC SBR test file over at the DG forums & will address further issues there.

eac3to v3.62
command line: "C:\Users\MASTER\Desktop\UsEac3to136\eac3to.exe" "C:\Users\MASTER\Desktop\ChID-BLITS-EBU-Narration.mka" 1: "C:\Users\MASTER\Desktop\UsEac3to136\ChID-BLITS-EBU-Narration.mka_1-eng.wav" -progressnumbers -log="C:\Users\MASTER\Desktop\UsEac3to136\UsEac3To.log"
------------------------------------------------------------------------------
Running in normal mode
Dialnorm removal enabled
MKA, 1 audio track, 0:00:47
1: AAC, 5.1 channels, 22kHz, dialnorm: 0dB
[a01] Extracting audio track number 1...
[a01] Decoding AAC via ffmpeg...
eac3to processing took 1 second.
Done.

tebasuna51

10th May 2026, 11:36

We were using fdkaac but as you have shown it turns out to be problematic. fdkaac as a decoder is poorly supported in ffmpeg. It was primarily designed as an encoder...
The native ffmpeg AAC decoder has no such limitation. For decoding, ffmpeg's native AAC decoder is fine...

Therefore, we've reverted the use of fdkaac. Please re-download the test build and update eac3to.exe. Report your results.

1) Of course the native ffmpeg decoder is better and without problems.
Only the default encoder is worse than qaac or fdkaac.

2) There are a legal problem: you can't include in your pack a ffmpeg binary with:
--enable-libfdk-aac

3) Other question is how eac3to recognize or decode the aac streams.
In the attached logs there are some issues.

To avoid cluttering this thread we'd prefer further issues be addressed at DG forum.

Yes, UsEac3to users don't need, by the moment, that new eac3to feature because the A/V Recode functions work better, like the logs attached show.

BTW feel free to ask here some questions because I can't use the DG forum like you know.

tormento

13th May 2026, 13:05

Well, there is always a trick: point to https://github.com/MartinEesmaa/FFmpeg-Builds ;)

Columbo

13th May 2026, 13:46

Exactly. It's only a patent issue. Many of the patents have run out and the rest will be soon. Fraunhofer has never acted against any freeware/open source projects and there are many nonfree builds available for download across the internet.

eac3to's AAC parsing for format detection/reporting is primitive and we have no plans to rewrite it.

Anyone having issues with our site should contact our admin.

tormento

14th May 2026, 12:36

Anyone having issues with our site should contact our admin.
If only he would reply to e-mails.

If I recall correctly, I sent twice some years ago and I am still waiting for a reply.

tebasuna51

3rd July 2026, 08:38

New eac3to v3.63 (https://forum.doom9.org/showthread.php?p=2032723#post2032723)
With a limited ffmpeg version to better decode/manage AC3/EAC3 codecs.
But the AAC support is reverted.

Works fine with UsEac3to GUI

kzkz

12th July 2026, 08:59

Quick question: is the attached .au3 the latest version? When I open it, the window title shows UsEac3To v1.3.5, while the compiled .exe shows v1.3.6. I believe an older version may have been attached by mistake.

Also, would you consider improving the program's Unicode support? Having to rename or move files and folders containing non-ASCII characters can be a bit of a hassle. I managed to get a working version with the help of AI and ran some tests using files named like this: Test_ÃÃ€Ã‚ÃƒÃ„Ã…Ã†Ã‡_Ã˜ÃÃž_ÃŸ_日本語_한국어 (and similar variations). UsEac3to was able to correctly process the file paths and generate/execute the command line without issues.

I can send you the modified script if you would like to review it and possibly incorporate some of the changes or solutions. Ideally, I would need to apply these changes to the latest version of the script.

I would also like to thank you for the GUI. I've been using it for well over a decade now, and I really appreciate the effort you've put into it.