eac3to - audio conversion tool

[De_Hollander]

I never see THD 6.1


And can't recommend use libAften to AC3 encode, use ffmpeg instead.

That is not possible encode with ffmpeg, even though ffmpeg is manual installed in windows 11, UsEac3to133 use libAften to AC3 encode.

[tebasuna51]

...UsEac3to133 use libAften to AC3 encode.

With UsEac3to use the 'A/V Recode' button (read the Readme).

[tebasuna51]

From the Readme:
Working
-------
The first time ask for eac3to, but also you can go to 'Settings' to inform about:
...

- 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.
...

UsEac3to don't search in the system PATH, maybe in next version.

I have a folder "C:\Portable\0" where I have all command line encoders/decoders and have it in the system PATH and also in UsEac3to, Foobar2000, and others softs than use them.
When there are new versions only need modify the soft in C:\Portable\0

[FrasierCrane]

Hi!

I've encountered a few bugs of eac3to:

1) Sometimes (seems random), conversion of any audio track to AC3 results in garbled sound. The audio is still somewhat recognizable, but distorted and overlayed with noise, etc. Running exactly the same eac3to command line again, mostly fixes it and the output is fine! Really strange... Logs attached.

2) Extracting PGS subtitles from MKV files results in corrupt SUP files, they are unreadable and are only about 1 third of the size they should be (when correctly extracting them with e.g. MKVExtract).

3) Demuxxing some formats (e.g. SUB subtitles) result in "This conversion is not supported" (or something like this) - should simply be demuxxed without any conversion.

4) Filenames containing ö,ä,ü,ß etc. lead to strange error messages! E.g.:

eac2to.exe "Hören.mkv" -progressnumbers -LOG="C:\log.txt"

Would be really nice if these issues could be fixed! Especially 1-3.

And I have a feature request:

It would be great if eac3to could extract the forced subtitles of each subtitle track into separate tracks (e.g. with the suffix "_FORCED"! The program already detects the forced flag of subtitles, so it should be possible to extract the forced captions. This would save a lot of time (currently I need to run "BDSup2Sub" just for doing that).

Thanks!

[tebasuna51]

1) Use ffmpeg to encode AC3, Aften/libAften is abandoned longtime ago.

2) Know problem, eac3to extract compressed (not corrupt) SUP files only when them was compressed (default) by mkvmerge. If them was muxed without compression eac3to work fine.
Use MkvExtract

3,4,feature request) I doubt than madshi want add this, but who know?

Removed attachements not needed.

[kdantas]

Hello, I'm trying to insert a subtitle in a bluray movie, using TSMuxer/BDEdit, but the m2ts file has LPCM audio. Unfortunately I am not having success. Neither extract the LPCM audio. After demux, the LPCM audio file (wav) gets smaller by 2 minutes. I already tried to do demux/conversion with eac3to or ffmpeg, but the conversion of the LPCM file does not proceed with error, or the extracted file takes a shorter time than the original. It doesn't matter if I use eac3to or ffmpeg with gui, it always gives error. Does anyone have a solution to extract the LPCM file and then convert it to DTS/FLC/AC3?

The mediainfo of the m2ts file is:

General
ID : 0 (0x0)
Complete name : E:\Videodrome 1983 1080p GBR Blu-ray AVC LPCM 1.0-ESiR\BDMV\STREAM\00004.m2ts
Format : BDAV
Format/Info : Blu-ray Video
File size : 24.3 GiB
Duration : 1 h 28 min
Overall bit rate mode : Variable
Overall bit rate : 39.3 Mb/s
Maximum Overall bit rate : 48.0 Mb/s

Video
ID : 4113 (0x1011)
Menu ID : 1 (0x1)
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L4.1
Format settings : CABAC / 3 Ref Frames
Format settings, CABAC : Yes
Format settings, Reference frames : 3 frames
Codec ID : 27
Duration : 1 h 28 min
Bit rate mode : Variable
Bit rate : 35.0 Mb/s
Maximum bit rate : 38.0 Mb/s
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16:9
Frame rate : 23.976 (24000/1001) FPS
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.704
Stream size : 21.8 GiB (90%)
Writing library : x264 core 128 r2216 198a7ea
Encoding settings : cabac=1 / ref=4 / deblock=0:-1:-1 / analyse=0x3:0x133 / me=umh / subme=10 / psy=1 / psy_rd=1.00:0.15 / mixed_ref=1 / me_range=24 / chroma_me=1 / trellis=2 / 8x8dct=1 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-3 / threads=12 / lookahead_threads=2 / sliced_threads=0 / slices=4 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=1 / constrained_intra=0 / bframes=3 / b_pyramid=1 / b_adapt=2 / b_bias=0 / direct=3 / weightb=1 / open_gop=1 / weightp=1 / keyint=24 / keyint_min=1 / scenecut=40 / intra_refresh=0 / rc_lookahead=24 / rc=2pass / mbtree=0 / bitrate=35000 / ratetol=1.0 / qcomp=0.00 / qpmin=0 / qpmax=69 / qpstep=4 / cplxblur=20.0 / qblur=0.5 / vbv_maxrate=38000 / vbv_bufsize=28500 / nal_hrd=vbr / ip_ratio=1.40 / pb_ratio=1.30 / aq=1:1.00
Color range : Limited
Color primaries : BT.709
Transfer characteristics : BT.709
Matrix coefficients : BT.709

Audio #1
ID : 4352 (0x1100)
Menu ID : 1 (0x1)
Format : PCM
Format settings : Big / Signed
Muxing mode : Blu-ray
Codec ID : 128
Duration : 1 h 28 min
Bit rate mode : Constant
Bit rate : 1 152 kb/s
Channel(s) : 1 channel
Channel layout : C
Sampling rate : 48.0 kHz
Bit depth : 24 bits
Stream size : 731 MiB (3%)

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 : 1 h 28 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 : 122 MiB (0%)
Service kind : Complete Main

Text
ID : 4608 (0x1200)
Menu ID : 1 (0x1)
Format : PGS
Codec ID : 144

[tebasuna51]

Here we can't help over downloaded movies.

Claim to ESiR any problem.

Whit well performed BD files that must work fine:

eac3to "...\00004.m2ts" 2: output.flac

[kdantas]

Here we can't help over downloaded movies.

Claim to ESiR any problem.

Whit well performed BD files that must work fine:

eac3to "...\00004.m2ts" 2: output.flac


Hi @tebasuna51, could you take a look at this audio file (url deleted), because the converted file (flac, wav, ...) is 2 minutes shorter than the original?

[tebasuna51]

I don't need check that audio (url deleted by rule 6), it's OK.
The problem is trust in all MediaInfo data, when a audio it's inside a container MediaInfo do some assumptions instead analize the full track

Check the PCM data:
Format : PCM
Duration : 1 h 28 min
Bit rate : 1 152 kb/s
Channel(s) : 1 channel
Sampling rate : 48.0 kHz
Bit depth : 24 bits
Stream size : 731 MiB

The BitRate is correct and without possible error:
24 bits * 1 channel * 48000 samples/sec = 1152000 b/s

But there are some problem whit the stream size:

1152000 b/s * 1 h 28 min = (1152000 * 5280 sec) bits = 726 MB

What is wrong, the Stream size, the Duration or both?

It's know than MediaInfo assume the duration of container when know the exact duration require analize the full audio stream (PCM, DTS-MA, THD, FLAC, AAC,...)

You can't trust in duration and stream size until extract the track outside the container.
Trust in eac3to, ffmpeg than read the full m2ts to extract fine the track

[kdantas]

I don't need check that audio (url deleted by rule 6), it's OK.
The problem is trust in all MediaInfo data, when a audio it's inside a container MediaInfo do some assumptions instead analize the full track

Check the PCM data:


The BitRate is correct and without possible error:
24 bits * 1 channel * 48000 samples/sec = 1152000 b/s

But there are some problem whit the stream size:

1152000 b/s * 1 h 28 min = (1152000 * 5280 sec) bits = 726 MB

What is wrong, the Stream size, the Duration or both?

It's know than MediaInfo assume the duration of container when know the exact duration require analize the full audio stream (PCM, DTS-MA, THD, FLAC, AAC,...)

You can't trust in duration and stream size until extract the track outside the container.
Trust in eac3to, ffmpeg than read the full m2ts to extract fine the track

https://imgur.com/WfiCrPv.jpg

I don't know if MediaInfo is showing wrong information, but the file player shows that it has a time of 01:28:39.

https://imgur.com/zggTDQX.jpg

Extracted PCM File MediaInfo. See Duartion info:

General
Complete name : E:\FFMpeg Temp\DG_output PID 1100 eng DELAY 0ms.w64
Format : Wave64
File size : 716 MiB
Duration : 1 h 26 min
Overall bit rate mode : Constant
Overall bit rate : 1 152 kb/s

Audio
Format : PCM
Format settings : Little / Signed
Codec ID : 1
Duration : 1 h 26 min
Bit rate mode : Constant
Bit rate : 1 152 kb/s
Channel(s) : 1 channel
Sampling rate : 48.0 kHz
Bit depth : 24 bits
Stream size : 716 MiB (100%)

[tebasuna51]

You can see the correct duration (1 h 26 min) and size (716 MiB) of PCM track.

Not all the tracks in a container must have the same duration and the container must show the longest one.
Of course a player must show the longest duration, not problem with a short one: play silence for audio or black screen for video.

You can see (signaled) other wrong data: 2304 kb/s when we know it is 1152 Kb/s

[Taran2L]

Hello. I have used eac3to for a lot of tracks for automatic converting from 7.1/5.1 Atmos or DD to AC3 5.1 640 and I never asked myself if is there something better then default libaften.dll
So now I’m a bit confused about ffmpeg
Which one is better to download from ffmpeg.org:
Windows builds from gyan.dev
Windows builds by BtbN
Or is it some other source to download only little dll?
And I never worried about convert option, just:
eac3to c:\Films\eng.dts c:\Films\eng.ac3
Is there something common for converting from dts file to ac3 with automating downmixing if need and remapping using ffmpeg parameters?

[LigH]

Hi.

The AC3 encoder currently built in ffmpeg/libavcodec (-codec:a ac3) is a lot better than libaften.

Furthermore, in addition to your list, one of the most generally famous Windows builds is that of Zeranoe.

[tebasuna51]

Furthermore, in addition to your list, one of the most generally famous Windows builds is that of Zeranoe.

"ffmpeg.zeranoe.com will close on Sep 18, 2020, and all builds will be removed"

Both gyan.dev or BtbN are good, with more or less components, but to AC3 encode the gyan essentials build is enough.

ffmpeg do automatic downmix from 7.1 to 5.1, if you want a better downmix read https://forum.doom9.org/showthread.php?t=181726

[LigH]

Oof, I completely missed that since I build my own with the media-autobuild suite... :o

[guest]

eac3to v3.36 released

* fixed: UHD Blu-Ray support didn't work with newer releases
* fixed: on some OSs there were crashes
* fixed: libav command line output hooking wasn't working properly
* added special handling for TrueHD seamless branching
* 2nd pass for audio gap fixing is auto-disabled for (UHD) Blu-Rays now
* libDcaDec: updated to 0.2.0

Hi madshi,

So with v3.36, you state that it has "fixed-UHD Blu-Ray support", but there's a couple of users on the RipBot264 thread that are having issues,

eg:-

https://forum.doom9.org/showthread.php?p=1972107#post1972107

and more recently another user posted this :-

https://forum.doom9.org/showthread.php?p=1978527#post1978527

Would be interested in your thoughts / suggestions on this :cool:

[odino]

My fault, I forget the stdout.wav. Corrected in the post.


Yep, in all my test using 0.34 or less the output is HE, using 0.35 or higer the output is LC.


Then your problem can't be about that.


Of course, QAAC is, for now, considered the best AAC encoder at low bitrates, for high bitrates we can't make test because the output can't be differentiated.


I'm not sure is this can cause the problem and how MkvMerge mux the m4a.

I will make some test about that.

Sorry to bring up this super old topic but I have this issue that every AAC encode results in HE AAC but I want it to be AAC-LC. Forcing quality doesn't change anything. Does it have anything to do with the neroAacEnc.exe I am using or can I force LC in any way?

[tebasuna51]

I can't reproduce your problem, using quality greater than 0.35 always output AAC-LC.

Please show your command line used.

My encoder:
18/02/2010 10:54 868.352 neroAacEnc.exe

[odino]

I have the same encoder version; tried several command lines now but always HE AAC.

C:\Program Files (x86)\eac3to>eac3to.exe d:\test.mkv d:\test.aac -224 -quality=0.5
MKV, 1 video track, 1 audio track, 1 subtitle track, 0:41:47, 48i /1.001
1: h265/HEVC, 1080i
2: AC3, English, 5.1 channels, 448kbps, 48kHz
3: Subtitle (SRT), English
Track 2 is used for destination file "test.aac".
a02 Extracting audio track number 2...
a02 Decoding with libav/ffmpeg...
a02 Reducing depth from 64 to 32 bits...
a02 Encoding AAC <224kbps> with NeroAacEnc...
eac3to processing took 3 minutes, 9 seconds.
Done.

Same if I convert to WAV first and several different -quality= options.

ID : 1
Format : AAC LC SBR
Format/Info : Advanced Audio Codec Low Complexity with Spectral Band Replication
Commercial name : HE-AAC
Format settings : Explicit
Codec ID : mp4a-40-2
Duration : 41 min 47 s
Bit rate mode : Variable
Bit rate : 224 kb/s
Maximum bit rate : 250 kb/s
Channel(s) : 6 channels
Channel layout : C L R Ls Rs LFE
Sampling rate : 48.0 kHz
Frame rate : 23.438 FPS (2048 SPF)
Compression mode : Lossy
Stream size : 66.9 MiB (100%)
Writing library : Nero AAC codec 1.5.4.0
Encoding settings : -cbr 224000

[tebasuna51]

eac3to.exe d:\test.mkv d:\test.aac -224 -quality=0.5
...
a02 Encoding AAC <224kbps> with NeroAacEnc...


You can't encode to a fix average bitrate (-224) and in quality mode.
eac3to send to encoder only the fix bitrate and it is not enough for LC mode.

Please use only -quality=0.35 to obtain the less bitrate possible at LC mode.

The encoder have a parameter (-lc) to force LC mode but eac3to can't use it, you can use the external encoder with:

eac3to "input" stdout.wav | NeroAacEnc -br 224 -lc -ignorelength -if - -of "output.m4a"

to force LC mode but it is not recommended at all.

[LigH]

Quality based AAC encoding will produce a variable audio bitrate. That's fine, the MP4 and MKV containers can handle that.

[weust]

I was ripping some UHD's which are giving me a hard time for some reason, and at first I was using version 3.34 which is working without issues, but I noticed 3.36 was released too.
Normally I use MKVToolnix, since that works fine for me, and eac3to to list the playlists I might need to rip.

But with version 3.36 I am running into some weird issues when trying to rip.
Both video and audio show these messages:
"This video conversion is not supported", "This audio conversion is not supported."
This is for only ripping either the video track (both video and Dolby Vision) or audio track.

Going back to version 3.34 and it works fine as before.

I don't have any DLL's replaced, so pure unzip and use.

Am I doing something wrong?

Example:
D:\Tools\eac3to>eac3to.exe f: 2) 2:d:\videos\02video.hevc
M2TS, 2 video tracks, 3 audio tracks, 12 subtitle tracks, 0:54:12, 24p /1.001
1: Chapters, 8 chapters
2: h265/HEVC, 2160p24 /1.001 (16:9), 10 bits
3: h265/HEVC, 1080p24 /1.001 (16:9), 10 bits
4: TrueHD/AC3 (Atmos), English, 7.1 channels, 48kHz
(embedded: AC3, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB)
5: AC3, French, 5.1 channels, 640kbps, 48kHz
6: AC3, German, 5.1 channels, 640kbps, 48kHz
7: Subtitle (PGS), English
8: Subtitle (PGS), French
9: Subtitle (PGS), German
10: Subtitle (PGS), Dutch
11: Subtitle (PGS), Chinese
12: Subtitle (PGS), Korean
13: Subtitle (PGS), Danish
14: Subtitle (PGS), Finnish
15: Subtitle (PGS), Norwegian
16: Subtitle (PGS), Swedish
17: Subtitle (PGS), French
18: Subtitle (PGS), German
This video conversion is not supported.

[Boulder]

Put a whitespace after 2: :)

[Ex0rcist]

I'm trying to demux .mkv file with positive audio delay using eac3to. I've noticed that after delay is fixed distortion audio is added/occurs at the beginning, why is that and how to avoid it ?

[tebasuna51]

Only silence frames are added at the begining, if there are distortion must be present in the original audio.

Use mkvextract to extract the audio without apply the delay.

[Ex0rcist]

There's no distortion noticeable if i extract audio with mkvextract, so it's definitely eac3to issue. It should add silence frames where audio is missing but instead it adds buzzing sound. Does it make any difference if positive delay isnt fixed when demuxing/remuxing files ?

Edit: It doesnt work well if delay isnt fixed, because it shifts the audio for a few frames which doesnt make it synced as original.

[tebasuna51]

Please upload the log of eac3to when extract/demux the orginal file.

[light5out]

Hello, I have a remux of a UHD bluray of a live concert. I am converting the audio to flac. I used MKVtoolnix to extract the audio to MKA, exported by chapters. Out of the 16 "tracks" that I was left with two of them fail to convert to flac. I am using the UsEac3To gui running a simple command of DTS to FLAC. This is the error I am getting on the failed files.

MKA, 1 audio track, 0:05:15
1: DTS Master Audio, English, 5.1 channels, 24 bits, 96kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
"DTS-HD Master Audio / 5.1 / 96 kHz / 6853 kbps / 24-bit"
[a01] Extracting audio track number 1...
[a01] Decoding with libDcaDec DTS Decoder...
[a01] libDcaDec reported the warning "Failed to parse XLL". <WARNING>
[a01] libDcaDec output changed from 6 channels, 96kHz to 6 channels, 48kHz. <ERROR>
Aborted at file position 262144. <ERROR>

Can anyone help a guy out?

[tebasuna51]

I used MKVtoolnix to extract the audio to MKA, exported by chapters.
Maybe there are problems when mkvextract (designed to cut by video frames) cut audio by chapters.

Or there are some corrupt data, or not supported by libDcaDec.

With UsEac3to you can try the conversion with ffmpeg decoder:

Load the problematic file in UsEac3to and click in 'A/V Recode', select the first option 'Decode,Lossless.Wavdts' and 'flac' like output.
Run or Enqueue
(with Enqueue a .log file is created to attach here)

[light5out]

Maybe there are problems when mkvextract (designed to cut by video frames) cut audio by chapters.

Or there are some corrupt data, or not supported by libDcaDec.

With UsEac3to you can try the conversion with ffmpeg decoder:

Load the problematic file in UsEac3to and click in 'A/V Recode', select the first option 'Decode,Lossless.Wavdts' and 'flac' like output.
Run or Enqueue
(with Enqueue a .log file is created to attach here)

Ok so that worked, but now it is messing with the file. We went from 96khz to 48khz and the bitrate dropped from 6,000 to 3,000.

I also tried making a wav, but the same thing happened. Strange. I might just manually extract the section with MKVtoolnix and see if I can then use the method I did with the original files?

[tebasuna51]

Ok so that worked, but now it is messing with the file. We went from 96khz to 48khz

ffmpeg log show something like the:?
[a01] libDcaDec output changed from 6 channels, 96kHz to 6 channels, 48kHz. <ERROR>

I also tried making a wav, but the same thing happened.
The 96 to 48 KHz change?
Try decoding to w64 the full file without split by chapters.

[light5out]

ffmpeg log show something like the:?
[a01] libDcaDec output changed from 6 channels, 96kHz to 6 channels, 48kHz. <ERROR>


The 96 to 48 KHz change?
Try decoding to w64 the full file without split by chapters.

I was able to decode the entire file to w64. Not sure what to do with that. Manual splitting?

[tebasuna51]

Then the problem is when mkvextract cut the full DTS.
I don't know how cut DTS/W64 by chapters then only method I can suggest you is manual split.

Sems there are only 2 parts with problems make the new flacs with:

[light5out]

Then the problem is when mkvextract cut the full DTS.
I don't know how cut DTS/W64 by chapters then only method I can suggest you is manual split.

Sems there are only 2 parts with problems make the new flacs with:

Thanks for the help, I was also able to rip a full quality flac file of the entire film. So I cam mess with that as well.

[DreckSoft]

I'm currently trying to sync a second audio track to a series. Movie is 23.976 fps, Audio source 25fps. Usually it should just be -slowdown or -25.000 -changeto23.976. This worked for previous seasons but now the resulting audio file is too short, resulting in async audio. It seems to be drifting continuously over the playtime. So changing the source or target framerate should fix this. Is there any way to make eac3to use non-standard framerates? It does not seem to accept 25.002 as source or 23.970 as target.

[tebasuna51]

It is not possible with eac3to.

Try with ffmpeg -i "INPUT" -af "atempo=0.959041" ...
with (slowdown) 23,976/25 = 0.959041 (change as you need)

[DreckSoft]

Thanks, that worked. I tried BeSweet before but that produces unpredictable output.

[ErazorTT]

It is apparently not possible export the dolby video stream from a uhd using version 3.36 anymore. This was working in 3.34. I am using eac3to in conjunction with megui, which it will now throw an error.

[tebasuna51]

It is apparently not possible export the dolby video stream from a uhd using version 3.36 anymore. This was working in 3.34. I am using eac3to in conjunction with megui, which it will now throw an error.

Then use v3.34 if the changes in the v3.36 don't work for you:

v3.36 2022-05-08
* fixed: UHD Blu-Ray support didn't work with newer releases
* fixed: on some OSs there were crashes
* fixed: libav command line output hooking wasn't working properly
* added special handling for TrueHD seamless branching
* 2nd pass for audio gap fixing is auto-disabled for (UHD) Blu-Rays now
* libDcaDec: updated to 0.2.0

(Preserve the libDcaDec.dll 0.2.0)

[ErazorTT]

Then use v3.34 if the changes in the v3.36 don't work for you:

Yes, thats exactly what I'm doing. I thought perhaps it would be nice to give feedback, so that it is known that there was a regression.

[guest]

What is the quickest & easiest way to convert THD 7.1 (or higher) to AAC 7.1 ??

Thanks

EDIT:- I think I may have found how to, BUT if there IS an easy way.

[tebasuna51]

For thd input I recommend ffmpeg decoder (avutil 57, avcodec 59), faster and more actualized than eac3to (avutil 52, avcodec 54), and for aac 7.1 output only qaac encoder:

"FFMPEG" -i "INPUT71.thd" -acodec pcm_s24le -f wav - | "QAAC" -V QUALITY --ignorelength --adts --no-delay -o "OUTPUT71.aac"

Others encoders, for instance using ffmpeg internal, convert standar thd 7.1 Channel layout: C L R Ls Rs Lb Rb LFE, to aac 7.1 Channel layout: C L R Ls Rs Lw Rw LFE

For easy way you can use UsEac3toGUI (load input and button A/V Recode)

[guest]

For thd input I recommend ffmpeg decoder, faster and more actualized than eac3to, and for aac 7.1 output only qaac encoder:

"FFMPEG" -i "INPUT71.thd" -acodec pcm_s24le -f wav - | "QAAC" -V QUALITY --ignorelength --adts --no-delay -o "OUTPUT71.aac"

Others encoders, for instance using ffmpeg internal, convert standar thd 7.1 Channel layout: C L R Ls Rs Lb Rb LFE, to aac 7.1 Channel layout: C L R Ls Rs Lw Rw LFE

For easy way you can use UsEac3toGUI (load input and button A/V Recode)

Hi, I really appreciate the info, but I am totally lost :(

I don't know how to use that command line you provided, and converting the channel layout's....:confused::confused:

So you're saying that eac3to isn't the best way to convert ??

I have very recently started using "cleverFFmpeg", and it's got so many options, it's crazy, but I do use eac3to quite bit for e-ac3 conversions.

[tebasuna51]

UsEac3toGUI can work like eac3to and ffmpeg GUI. If you want some eac3to audio management use it and encode with qaac.

cleverFFmpeg can decode with ffmpeg and encode with qaac, remember the discussion in that thread (https://forum.doom9.org/showthread.php?p=1977780#post1977780)

[Balling]

The message is clear: this track is a mix of 1.0 and 2.0
Try with:
ffmpeg -i japanese.thd -ac 2 -acodec pcm_s24le -f wav japanese.wav

Hi, what does it mean? THD 1.0? Is that version of THD or mono? We still have this ticket and it futher regressed. Do you have full japanese.thd?

https://trac.ffmpeg.org/ticket/667#comment:10


* added special handling for TrueHD seamless branching


is that as good as https://github.com/domyd/mlp or worse as in DGDemux? Or even more perfect as in MakeMKV?

[tebasuna51]

Hi, what does it mean? THD 1.0? Is that version of THD or mono?
A mix of mono and stereo audio.
We still have this ticket and it futher regressed. Do you have full japanese.thd?
Nope.
The track uploaded in comment 15 is a correct mono track without problems to decode, after extract thd and ac3 streams.

* added special handling for TrueHD seamless branching

is that as good as https://github.com/domyd/mlp or worse as in DGDemux? Or even more perfect as in MakeMKV?

Sorry, currently I haven't TrueHD seamless branching BD's to check it.

[Balling]

>A mix of mono and stereo audio.

The author gave the full sample from another disk, it appears to be just mono.

[tebasuna51]

I'm trying to demux .mkv file with positive audio delay using eac3to. I've noticed that after delay is fixed distortion audio is added/occurs at the beginning, why is that and how to avoid it ?

Seems the method used by eac3to to delay aac streams is repeat the first frame, if it isn't silence an annoying sound can be produced.

Use mkvextract or ffmpeg to extract them.

For ac3/dts streams I think eac3to use methods like the old DelayCut with really ac3/dts silence frames.

[Ex0rcist]

mkvextract ignores delay info and just shifts audio to first frame, so that's pretty much useless because audio won't be in sync as it should be. Is it possible to report this bug to eac3to dev for possible fix in the future ?

[tebasuna51]

In the first post you have the link to the bug tracker (http://eac3to.bugs.madshi.net/).