UsEac3to v1.3.3: a different GUI for eac3to

[Hellboy.]

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

[tebasuna51]

eac3to atmos.thd output.thd+ac3 -keepDialnorm

[Hellboy.]

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]

Originally Posted by Hellboy.

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/loss...-1.4.2-x86.zip
And https://github.com/foo86/dcadec/rele....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.]

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]

Originally Posted by Hellboy.

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.]

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

[seagate]

Hello

What is causing this error?

I am attaching a photo:

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]

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]

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]

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]

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]

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]

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/442zs..._info.mht/file

[tebasuna51]

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]

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]

Originally Posted by hypnotize666

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]

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]

Originally Posted by hypnotize666

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.
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]

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