eac3to - audio conversion tool

[tebasuna51]

The eac3to normalize make impossible any clip, even if the decoder output have float values over 0dB the normalize function correct these values.

[canuckerfan]

Does anyone know the command to extract a TrueHD stream from an Atmos stream? Excuse me if it's already been mentioned in this thread, I tried searching but couldn't find it and I don't want to sift through so many pages.

[Nico8583]

A little question : is eac3to a good choice to just demux a BD (2D or 3D) ? Thank you.

[LigH]

To mainly obtain the audio streams: Yes. But it won't care much about all other streams, I believe...

[tebasuna51]

Does anyone know the command to extract a TrueHD stream from an Atmos stream?

Atmos or not you can extract the standard TrueHD (without Atmos info) with:

eac3to stream.thd+ac3 stream.thd

if the stream is already extracted, or from the BD, if TrueHD is track 3:

eac3to movie.mpls 3: stream.thd

[tebasuna51]

A little question : is eac3to a good choice to just demux a BD (2D or 3D) ? Thank you.

Yes for 2D BD, I'm never try a 3D if you want recover 3D info (3D planes for 3D sups, etc.)

[Groucho2004]

To mainly obtain the audio streams: Yes. But it won't care much about all other streams, I believe...I use eac3to for de-multiplexing audio and video streams from BDs on a regular basis. I'm surely not alone.

[LigH]

I use eac3to for de-multiplexing audio and video streams from BDs on a regular basis. I'm surely not alone.

^ OK, practical experience. :cool:

[Ripman]

I test, with eac3to, wav files with extra chunks after the data chunk without problems. For instance cue points created with GoldWave editor.

But, maybe, the eac3to behaviour can change with wav files greater than 2 GB, like here. Is know than there are soft than don't support this limit.

I can't understand how eac3to show a duration of 0:06:29 when seems (by size, channels, bitdepth and samplerate) the correct duration is 00:22:09.205 like show MediaInfo.

Thanks for your observations. Regarding the differing durations, please note the info I posted was for two different tracks in the same set (album).

I'll clean up that info for consistency, and post the wav when I get back tomorrow night. The sample is actually about 1.6gb, but I also put a 176 eac3to down sample log that shows the command line, and a pic of the static I get.

Perhaps in this case, the issue is related both to the embedded artwork and expanding to 32bit to apply gain.

[Ripman]

Thanks. I tried SoX 14.4.2 and it works just fine. Here is the sox command line I used to replicate the one I used with eac3to.

sox -V4 352.wav --rate 176400 176.wav gain -3.5 2>sox_log.txt

Just to reiterate that sox approach worked without issue.

And I'll post some additional info as described in my previous post related to a 352-176 down sample issue I had with eac3to. The hotel wifi never lets me upload something that big.

[Nico8583]

Yes for 2D BD, I'm never try a 3D if you want recover 3D info (3D planes for 3D sups, etc.)
Thank you ! BD 3D info is not a problem, I can extract it from MPLS file. I will try if MVC file extracted keeps 3D planes info.

[canuckerfan]

Atmos or not you can extract the standard TrueHD (without Atmos info) with:

eac3to stream.thd+ac3 stream.thd

if the stream is already extracted, or from the BD, if TrueHD is track 3:

eac3to movie.mpls 3: stream.thd

I'm trying something like this but I'm still getting an outputted Atmos track:

eac3to.exe "source.mkv" 2:aud2.thd

[kypec]

I'm trying something like this but I'm still getting an outputted Atmos track:

eac3to.exe "source.mkv" 2:aud2.thd

Please post your full output eac3to log.

[r0lZ]

A little question : is eac3to a good choice to just demux a BD (2D or 3D) ? Thank you.
No for the subtitles of most 3D BDs. If the same subtitles are present in the AVC M2TS and in the MVC M2TS (and it's frequent), eac3to fails completely, and you have to unselect the subtitle streams and retry. AFAIK, for everything else, eac3to works fine as a demuxer.

[canuckerfan]

Please post your full output eac3to log.

Hope this is the one.

eac3to v3.31
command line: eac3to.exe "source.mkv" 2:aud2.thd
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 1 subtitle track, 0:03:40, 24p
1: h264/AVC, Hindi, 1920x804 24p (160:67)
"Mohe Rang Do Laal"
2: TrueHD (Atmos), Hindi, 7.1 channels, 48kHz
"7.1 TrueHD Audio"
3: Subtitle (SRT), English, "English Subtitles"
[v01] The video framerate is correct, but rather unusual. <WARNING>
[a02] Extracting audio track number 2...
[a02] Creating file "aud2.thd"...
Video track 1 contains 5288 frames.
eac3to processing took 2 seconds.
Done.

[tebasuna51]

@canuckerfan

What is the problem?
The <WARNING> is related to 24 fps, but the "aud2.thd" seems extracted fine.

[junior_l3oss]

@canuckerfan

What is the problem?
The <WARNING> is related to 24 fps, but the "aud2.thd" seems extracted fine.

hi...i have netflix dolby digital plus audio.
dd+ audio not detected

eac3to v3.31
command line: "C:\eac3to\eac3to.exe" s1e01.ac3 \s1e01.ac3~n.wavs
------------------------------------------------------------------------------
The format of the source file could not be detected. <ERROR>


Audio #0 : E-AC-3 at 192 Kbps
Infos : 6 kanal, 48.0 KHz

how can i decode wavs and etc.

[tebasuna51]

@junior_l3oss

When your command line don't work can be:

1) eac3to don't support this kind of eac3

2) Or the eac3 stream is buggy because:

2-1) A bad capture of video from Netflix

2-2) Or a bad stream extraction from video

We can't know the problem without a sample of your video.

[TheLastOfUs]

Seems the creator of this program 'madshi' "doesn't have the resources" for this insanely difficult problem I face....which is ridiculous.

90 google results pretty much end up in this question never getting answered. 3 times it has been asked here and never answered..but here goes yet again.

I have Nero 7 installed and I have the HD/BluRay plugin installed and registered yet EAC3TO still seems to crap on me. It keeps saying cannot find Nero Audio Decoder 2..which is BS.

eac3to.exe -test
eac3to (v3.31) is up to date
Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed
http://www.nero.com/eng/store-blu-ray.html
CAUTION: You need Nero 7. Nero 8 won't work with eac3to. (errr wrong Nero 7.8.5.0 ..its installed and registered ALONG with the BLURAY plugin!!)
ArcSoft DTS Decoder doesn't seem to be installed
http://www.arcsoft.com/products/totalmediatheatre
Sonic Audio Decoder (3.31.0.0) doesn't seem to be installed
Haali Matroska Muxer (2013-04-14) is installed
Nero AAC Encoder (1.5.4.0) is installed
Surcode DTS Encoder (1.0.29.0) is installed

https://snag.gy/1sYHTw.jpg

TL;DR

I think program is dead when trying to do AAC to AAC/M4A

[tebasuna51]

I have Nero 7.11.10.0, and Blu-Ray/HD DVD Video Plug-in, and work in W 7 without problems:

eac3to (v3.31) is up to date
Nero Audio Decoder (Nero 7) works fine
ArcSoft DTS Decoder (1.1.0.0) works fine
Sonic Audio Decoder (3.31.0.0) doesn't seem to be installed
Haali Matroska Muxer (2013-04-14) is installed
Nero AAC Encoder (1.5.4.0) is installed
Surcode DTS Encoder doesn't seem to be installed

[TheLastOfUs]

Want to update...the above only happens when using -slowdown for aac...otherwise inputting .aac and then exporting to a .aac works fine...without -slowdown

[TheLastOfUs]

I have Nero 7.11.10.0, and Blu-Ray/HD DVD Video Plug-in, and work in W 7 without problems:

eac3to (v3.31) is up to date
Nero Audio Decoder (Nero 7) works fine
ArcSoft DTS Decoder (1.1.0.0) works fine
Sonic Audio Decoder (3.31.0.0) doesn't seem to be installed
Haali Matroska Muxer (2013-04-14) is installed
Nero AAC Encoder (1.5.4.0) is installed
Surcode DTS Encoder doesn't seem to be installed

I'm using Windows 10.

I have Nero 7 installed and mine definitely is not coming back with that. Can you please show me your product information without the serials? So I can compare what I may be missing. I did a full install

[madshi]

eac3to needs the Nero decoder to be accessible via DirectShow. If it isn't, eac3to can't use it. So something seems to be wrong with the installation. I don't know how to fix it, though.

[TheLastOfUs]

well im installing Nero 7.11.10 now direct from Nero's archive.....so far this is a nightmare.

I extracted eac 3.31 just fine

[TheLastOfUs]

Yup..installed 7.11.10.0

Same crap.

https://snag.gy/Fhdnzx.jpg

% List of all my installs and versions.

I'm more than likely guessing this is happening because I'm using Windows 10 and Nero.

[madshi]

Maybe manually registering "NeAudio2.ax" with regsvr32 helps?

http://forum.doom9.org/showpost.php?p=1238209&postcount=7877

If it doesn't then I'm out of ideas.

[TheLastOfUs]

My nero is not installing NeAudio2.ax

No idea why. Reinstalled cleanly and uninstalled in Safe mode a few times.

[TheLastOfUs]

I don't have this file "C:\Program Files\Common Files\Ahead\DSFilter\NeAudio2.ax"

[TheLastOfUs]

Madhsi thanks for linking me to above. Seems that guy had the EXACT SAME PROBLEM as me. He was also missing NeAudio2.ax but didn't quite mention how he solved it (super annoying).

I will be less annoying and say I solved this easily by installing NERO MICRO/LITE with my Nero 7 key. I then registered the .dlls and it worked minus the .dll C:\Program Files (x86)\Common Files\Ahead\DSFilter\NeEacDec.dll (this didn't register under the elevated command prompt). Didn't need to do ANY of the registry stuff.

Now it works!

[TheLastOfUs]

I went ahead and updated the EAC3TO Wiki FAQ on this question. Should prevent repeats of this :)

[amayra]

I went ahead and updated the EAC3TO Wiki FAQ on this question. Should prevent repeats of this :)
i have some problem and i given up for solved it so thanks for share you experience with us :thanks:

[AYColumbia]

The DS filter issue with Nero 7 goes back to Windows 7 64bit. I found that if you install it on a Win7 32bit and then back up your DSFilter folder from C:\Program Files (x86)\Common Files\Ahead you can later use this to register/re-register the filters. After you install Nero 7, on Win 10 for example, you then copy your backed-up DSFilter folder to the one currently installed (this will also restore the ones that Nero install did NOT include). Then register all the .ax files using elevated privileges. I wrote a command file that I can "Run as Admin..." to do this, i.e.,

@echo off
@cd /d "%~dp0"
for /f "delims=" %%i in ('dir /b *.ax') do regsvr32 "%%i" /s

I think way back I had done an install inside a VM on a Win7 32bit just for this purpose; to grab the DSFilter folder. It was painful.

[filler56789]

For best AAC decoding you need:

(1) Nero 7 (Nero 8 won't work!)
(2) Nero HD DVD / Blu-Ray plugin

It was true several years ago... but ¿is it still true today? :confused:
¿Time for an overdue update?

[torturesauce]

I read on dcadec's github page that the program is now deprecated and its development will now continue through ffmpeg. Will this be implemented in a future edition of eac3to as well? In the meantime, how should I replace dcadec with the latest ffmpeg manually? And where can we keep track of the latest developments on the decoding of DTS formats?

[Sparktank]

I read on dcadec's github page that the program is now deprecated and its development will now continue through ffmpeg.

Hmm. Interesting.

https://github.com/foo86/dcadec/blob/master/README.md
This program is deprecated!

This decoder has been fully integrated into FFmpeg master branch and further development will continue there. Using FFmpeg (https://ffmpeg.org/) for DTS decoding is now recommended.

how should I replace dcadec with the latest ffmpeg manually?

I believe you can't. Madshi uses custom builds of lav/ffmpeg which he would need to update.

Only manual updates that don't cause lots of issues seem to be:
libFLAC.dll
libdcadec.dll
dtsdecoderdll.dll (although should be deprecated now; depending how far ff-dcadec has come)

And where can we keep track of the latest developments on the decoding of DTS formats?

The official git for FFMPEG (https://git.ffmpeg.org/gitweb/ffmpeg.git) shows a lot of activity (from foo86) for dcadec over his github page.

https://git.ffmpeg.org/gitweb/ffmpeg.git?a=search&h=HEAD&st=commit&s=dcadec

Thanks for pointing it out.

Will definitely have to keep my eyes open in the future.

Zeranoe builds (https://ffmpeg.zeranoe.com/builds/) should have all the latest commits, no?
Will have to adjust my workflow to use ffmpeg

[filler56789]

.......
dtsdecoderdll.dll (although should be deprecated now; depending how far ff-dcadec has come)

FWIW: I've been using a not-so-recent x64-build of MPlayer, and thus far, its ffdca has decoded okay all the DTS sample-files and audio tracks I have (principally DTS Express and coreless Master Audio) — so yes, me too thinks the Arcsoft DLL should be used only "as a last resort", from now on... and preferably, without depending on DirectShow anymore...

[stax76]

There is a issue with unicode file names, processing works but output display has a problem:

D:\Projekte\VS\VB\StaxRip\bin\Apps\eac3to\eac3to.exe "D:\Temp\Video\Предприятий_temp\Предприятий ID2 English.flac" "D:\Temp\Video\Предприятий_temp\Предприятий ID2 English_out1.m4a" -quality=1 -normalize -downStereo -progressnumbers

FLAC, 2.0 channels, 0:00:43, 24 bits, 1731kbps, 44.1kHz
Decoding FLAC...
Writing WAV...
Creating file "D:\Temp\Video\???????????_temp\??????????? ID2 English_out1.m4a.pass1.wav"...

[canuckerfan]

@canuckerfan

What is the problem?
The <WARNING> is related to 24 fps, but the "aud2.thd" seems extracted fine.

The problem is that it's still an Atmos track. I'm trying to convert from Atmos to TrueHD.

[Music Fan]

I don't believe it's able to remove the Atmos part keeping only the THD part.

[duedel]

I just tested this:
ffmpeg -i "dtsHD_MA_track.dts" -bsf:a dca_core -c:a copy "dtsCore.dts"

produces a bit-idenctical file to

eac3to "dtsHD_MA_track.dts" "dtsCore.dts" -core

with a recent zeranoe ffmpeg build.


Hmm. Interesting.

https://github.com/foo86/dcadec/blob/master/README.md




I believe you can't. Madshi uses custom builds of lav/ffmpeg which he would need to update.

Only manual updates that don't cause lots of issues seem to be:
libFLAC.dll
libdcadec.dll
dtsdecoderdll.dll (although should be deprecated now; depending how far ff-dcadec has come)



The official git for FFMPEG (https://git.ffmpeg.org/gitweb/ffmpeg.git) shows a lot of activity (from foo86) for dcadec over his github page.

https://git.ffmpeg.org/gitweb/ffmpeg.git?a=search&h=HEAD&st=commit&s=dcadec

Thanks for pointing it out.

Will definitely have to keep my eyes open in the future.

Zeranoe builds (https://ffmpeg.zeranoe.com/builds/) should have all the latest commits, no?
Will have to adjust my workflow to use ffmpeg

[Sparktank]

I just tested this:

Yeah, I found some 5.1 showing up bit-identical, with foobar2000 to bit-compare.

Except I've been converting to FLAC.

But I haven't tested anything over 5.1 yet.
Especially concerning 'strange setup'.

[canuckerfan]

I don't believe it's able to remove the Atmos part keeping only the THD part.

Dang... so there's no free method out there that will convert Atmos to TrueHD?

[nevcairiel]

Dang... so there's no free method out there that will convert Atmos to TrueHD?

Atmos is just additional metadata in the TrueHD stream, most players that don't support Atmos will just ignore it and decode the TrueHD parts. Why do you want to remove it?

[LigH]

Some software was not able to ignore the additional Atmos metadata correctly. But current versions of them should all be fixed.

[Weirdo]

Any suggestions on how we should install/update dcadec?

I read on dcadec's github page that the program is now deprecated and its development will now continue through ffmpeg. Will this be implemented in a future edition of eac3to as well? In the meantime, how should I replace dcadec with the latest ffmpeg manually? And where can we keep track of the latest developments on the decoding of DTS formats?

Hmm. Interesting. I believe you can't. Madshi uses custom builds of lav/ffmpeg which he would need to update. Only manual updates that don't cause lots of issues seem to be:
libFLAC.dll
libdcadec.dll
dtsdecoderdll.dll (although should be deprecated now; depending how far ff-dcadec has come)

The official git for FFMPEG (https://git.ffmpeg.org/gitweb/ffmpeg.git) shows a lot of activity (from foo86) for dcadec over his github page.

https://git.ffmpeg.org/gitweb/ffmpeg.git?a=search&h=HEAD&st=commit&s=dcadec

Thanks for pointing it out. Will definitely have to keep my eyes open in the future. Zeranoe builds (https://ffmpeg.zeranoe.com/builds/) should have all the latest commits, no?
Will have to adjust my workflow to use ffmpeg

[tulala]

Hi,

I have a question. I have been trying to demux the true-hd track from the BD Monster. Inc, and I am curious as to why the logs look this way.

Why does tsmuxer find one overlapping frame at most (but not all) the junctions between m2ts (seamless branching disk) while eac3to only finds 2 of them?
Also, I noticed that with eac3to 3.27, the number of frames processed is one less than when using 3.31. How come? 3.31 gives me the same number of frames with tsmuxer.

tsmuxer log
Network Optix tsMuxeR. Version 2.6.12. www.networkoptix.com
Decoding H264 stream (track 1): Profile: High@4.1 Resolution: 1920:1080p Frame rate: 23.976
H.264 muxing fps is not set. Get fps from stream. Value: 23.976
Decoding TRUE-HD stream (track 2): AC3 core+TRUE-HD. Peak bitrate: 7290Kbps (core 640Kbps) Sample Rate: 48KHz Channels: 7.1
Decoding PGS stream (track 3): Resolution: 1920:1080 Frame rate: 23.976
TRUE-HD stream (track 2): overlapped frame detected at position 00:06:14,879. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 00:07:56,927. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 00:09:56,255. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 00:13:11,679. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 00:17:06,208. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 00:18:05,119. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 00:19:35,455. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 00:25:48,703. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 00:27:40,000. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 00:33:40,223. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 00:39:47,295. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 00:43:27,103. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 00:43:59,327. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 01:22:38,047. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 01:23:07,391. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 01:23:38,880. Remove frame.
TRUE-HD stream (track 2): overlapped frame detected at position 01:31:44,319. Remove frame.
Processed 132553 video frames
Flushing write buffer
Demux complete.
Demuxing time: 20 min 28 sec

eac3to log
eac3to v3.31
command line: "C:\Users\Home\Desktop\BD\Clown_BD_v0.81\eac3to\eac3to.exe" "D:\0000000YAYYYYY\Monsters.Inc.2001.1080p.BluRay.AVC.TrueHD.7.1-FGT\BDMV\PLAYLIST\00800.mpls" 1) 1: "D:\0000000YAYYYYY\demux\Chapters_1.txt" 2: "D:\0000000YAYYYYY\demux\Video_2.*" 3: "D:\0000000YAYYYYY\demux\Audio_3_English.THD+AC3" 9: "D:\0000000YAYYYYY\demux\Subtitles_9_English.sup" -progressnumbers -LOG="D:\0000000YAYYYYY\demux\LOGS\eac3to_PASS3_LOG.LOG"
------------------------------------------------------------------------------
M2TS, 1 video track, 6 audio tracks, 8 subtitle tracks, 1:32:07, 42.809p
1: Chapters, 32 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 7.1 channels, 48kHz
(embedded: AC3 EX, 5.1 channels, 640kbps, 48kHz)
4: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
5: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
6: AC3 EX, Spanish, 5.1 channels, 640kbps, 48kHz
7: E-AC3, French, 7.1 channels, 896kbps, 48kHz
(core: AC3, 5.1 channels, 512kbps, 48kHz)
8: AC3, English, 2.0 channels, 192kbps, 48kHz
9: Subtitle (PGS), English
10: Subtitle (PGS), English
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), French
13: Subtitle (PGS), English
14: Subtitle (PGS), English
15: Subtitle (PGS), Spanish
16: Subtitle (PGS), French
Creating file "D:\0000000YAYYYYY\demux\Chapters_1.txt"...
[a03] AC3 encoding doesn't support back channels. Will mix them into the surround.
[a03] Extracting audio track number 3...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[s09] Extracting subtitle track number 9...
[a03] Extracting TrueHD stream...
[a03] Extracting TrueHD stream...
[a03] Decoding with libav/ffmpeg...
[a03] Mixing surround channels...
[a03] Remapping channels...
[a03] Encoding AC3 <640kbps> with libAften...
[v02] Creating file "D:\0000000YAYYYYY\demux\Video_2.h264"...
[a03] Creating file "D:\0000000YAYYYYY\demux\Audio_3_English.THD+AC3"...
[a03] Skipping identical AC3 frames (seamless branching)...
[a03] Skipping identical AC3 frames (seamless branching)...
[s09] Creating file "D:\0000000YAYYYYY\demux\Subtitles_9_English.sup"...
[a03] Audio overlaps for 5ms at playtime 0:13:12. <WARNING>
[a03] Audio overlaps for 5ms at playtime 0:33:40. <WARNING>
[a03] The audio gaps/overlaps can't be removed from the TrueHD bitstream. <WARNING>
[a03] In order to remove them you'll have to transcode to another format. <WARNING>
[a03] The original audio track has a constant bit depth of 20 bits.
[a03] Processed audio track, L+R+C+LFE: constant bit depth of 20 bits.
[a03] Processed audio track, SL+SR: max 24 bits, average 21 bits.
Video track 2 contains 132553 frames.
Subtitle track 9 contains 1083 captions.
eac3to processing took 16 minutes, 38 seconds.
Done.

I am puzzled.

[tebasuna51]

Why does tsmuxer find one overlapping frame at most (but not all) the junctions between m2ts (seamless branching disk) while eac3to only finds 2 of them?

eac3to v3.31
...
[a03] Skipping identical AC3 frames (seamless branching)...
[a03] Skipping identical AC3 frames (seamless branching)...
[a03] Audio overlaps for 5ms at playtime 0:13:12. <WARNING>
[a03] Audio overlaps for 5ms at playtime 0:33:40. <WARNING>


When eac3to find identical frames (at end of a m2ts and at begin of next m2ts) it safely skip one of them without show a <WARNING>
See http://forum.doom9.org/showthread.php?p=1600694#post1600694

[r0lZ]

Does that mean that both eac3to and tsMuxeR are now correct when dealing with overlapping frames?

[tulala]

When eac3to find identical frames (at end of a m2ts and at begin of next m2ts) it safely skip one of them without show a <WARNING>
See http://forum.doom9.org/showthread.php?p=1600694#post1600694

I understand. Here though there are warnings are in the log, so the identical frames aren't skipped (I am extracting the true-hd track). The log also says eac3to cannot fix the overlap. So in the output .THD file, the two identical frames are still here. What does that mean exactly? Are the two identical frames "next to each other"? Are they "on top of each other"?

I understand this has been discussed many times before, but I read pretty much everything related to that issue and there was never a clear answer.

Regarding the difference between eac3to and tsmuxer, any idea?

Thank you for your help.

[tebasuna51]

Here though there are warnings are in the log, so the identical frames aren't skipped (I am extracting the true-hd track).
Nope, identical 15 frames are skipped (case C in related thread), but there are 2 overlaps with different frames (case A).

The log also says eac3to cannot fix the overlap.
Without decoding the track we can only preserve both frames (the last frame of one m2ts and the first frame of next m2ts) or delete one.

Regarding the difference between eac3to and tsmuxer, any idea?

tsMuxeR removed 15 frames and:
TRUE-HD stream (track 2): overlapped frame detected at position 00:13:11,679. Remove frame.
...
TRUE-HD stream (track 2): overlapped frame detected at position 00:33:40,223. Remove frame.

eac3to:
[a03] Skipping identical AC3 frames (seamless branching)... (15 frames)
[a03] Audio overlaps for 5ms at playtime 0:13:12. <WARNING>
[a03] Audio overlaps for 5ms at playtime 0:33:40. <WARNING>

eac3to only remove a frame if there are a delay equal to the frame duration. Here we can finish with a +10ms delay.

I don't know the THD frame duration, but with AC3 frames (32 ms) delete the 2 frames for 10 ms of overlap we can finish with a delay of -54 ms.

[madshi]

THD frames are *very* short (IIRC each THD frame has 40 samples for an 48khz track, which means each frame should be slightly less than 1ms), so removing overlaps is not as important for THD as it is for any other (compressed) audio codec. There are "major" and "minor" THD frames, though, and a new m2ts file is likely to start with a "major" frame. Which makes it somewhat unlikely that the last THD frame of a previous m2ts part, and the first THD frame of the following m2ts part are bit-by-bit identical. As a result eac3to might not be able to cleanly solve this situation. With AC3 and DTS, every frame has the same type, so the last/first frame usually match bit-by-bit for seamless branching, which eac3to can resolve very cleanly.

P.S: That said, I'm not even trying to detect if the last THD frame of a previous m2ts part and the first THD frame of the following m2ts part match bit-by-bit. I think it's unlikely that they do, but I can't say with 100% certainty right now.

[tebasuna51]

THD frames are *very* short (IIRC each THD frame has 40 samples for an 48khz track, which means each frame should be slightly less than 1ms)...

Then for THD part of track seems the problem is unnoticeable.

tulala extract the thd+ac3 track, maybe the messages are referred only to the AC3 part of the track?

[madshi]

Yes, eac3to's "Skipping identical AC3 frames" message is only for the AC3 part of the track.

[SeeMoreDigital]

I don't know if anybody has asked this before... Are there any plans to add support for DSD (Direct Stream Digital) decoding?

Once extracted they're usually in the form of .dsf or .dff streams ;)

[tulala]

I understand that tsmuxer removes 15 identical frames and 2 others that aren't identical but that overlap.
I understand that eac3to also removes the 15 identical frames in the ac3 track, and it doesn't remove the frames that overlap.

More questions (for curiosity more than anything else).

- When there are some identical frames in the ac3 track (here 15), are those identical frames also in the true-hd track? Those 15 identical frames in the ac3 are deleted/skipped correctly apparently, but if they are also in the true-hd track, are they also deleted/skipped, or are they still in the output audio file?

- When audio overlaps (here twice), is it in both ac3 and true-hd tracks?

- If the audio overlaps for 5ms in the true-hd track and a true-hd frame has a duration of a little less than 1ms, that means that there are about 5 frames that overlap?

- If decoding a true-hd frame was possible (it isn't if I documented myself correctly), would it become possible to cleanly solve the overlap situation?

Thanks again for clarifying things up.

[tebasuna51]

@tulala
Re-read the madshi post.
Short answer to first 4 questions: no, no, no and no.

When decode is possible to cleanly solve the overlap because we can delete the needed PCM samples 1/48000 sec. = 0.02083 ms of duration.

[Thunderbolt8]

could you please add the option to handle zlib packed subtitles correctly? currently eac3to cant demux those correctly.

[hubblec4]

Hi madshi

Is eac3to able to demux TextST subtitles from Bluray?
If not, could you add such a feature?

[jriker1]

This was probably discussed before but trouble rifling thru 600+ pages of thread. I know eac3to now uses a different default decoder for DTS content. I am used to using ArcSoft. Challenge with Arcsoft is if it's 6.1 content I have to use a certain version dll, if it's labeled as strange setup I use a different one, all other content I use a third. No problem, I can deal with that. Right now keep using the -arcsoft flag when using eac3to. Is the default decoder better than using Arcsoft and does it have any of the same problems with separation of channels based on the content type?

Thanks.

JR

[tebasuna51]

The default decoder decode the "strange setup" without problems. ArcSoft is only needed for DTS Express.

[Snowknight26]

The default decoder decode the "strange setup" without problems. ArcSoft is only needed for DTS Express.

And hopefully not for long either, now that ffmpeg can decode DTS Express.

[tebasuna51]

And hopefully not for long either, now that ffmpeg can decode DTS Express.

You are right, now ffmpeg can decode DTS Express.
I make a test with a sample and the output have same lenght, samplerate, etc., than decoded by ArcSoft.

They aren't bit-identical but we can't expect that with lossy codecs.
Maybe that is not important with these kind of tracks.

But I don't know when madshi can do the changes.

Now eac3to work with special dll's:
10/03/2015 avcodec-54.dll
10/03/2015 avutil-52.dll
14/11/2015 libdcadec.dll

A recent ffmpeg shared work with:
19/09/2016 avcodec-57.dll
19/09/2016 avutil-55.dll

And about libdcadec.dll
This program is deprecated!

This decoder has been fully integrated into FFmpeg master branch and further development will continue there.

Then madshi must make new avcodec-57.dll and avutil-55.dll to add DTS Express support (maybe without libdcadec.dll)

I know than eac3to can't distribute the same dll's than ffmpeg by copyright licenses (for instance to decode AAC) but if we can change the dll's with the included in ffmpeg shared eac3to don't need be actualized so quickly.

madshi ¿Can you make compatible dll's with the ffmpeg shared ones?
Seems than work fine replacing libFLAC.dll with new ones.

You can distribute limited dll's, but the users can replace them.
Is the same than you can't distribute ArcSoft or Nero decoders but users can use dtsdecoderdll.dll, etc.

[filler56789]

For what it's worth...

three weeks ago, I dropped Arcsoft's DTSdecoderdll.dll definitely, after noticing it introduced (through LAV Audio) lots of distortion in certain pure-lossless tracks :scared:

[SeeMoreDigital]

Out of interest...

How mature is FFmpeg's support for decoding 2 channel and multi-channel DSD?


Cheers

[Motenai Yoda]

Hi I have a bluray with a lpcm 7.1 that sports a uncommon layout, it is something like L R C S1 B1 B2 S2 LFE, both the back channel are bit identical, but I don't know how to rearrange to a usual layout
L R C LFE B1 B2 S1 S2 or S2 S1 (hoping only SR and LFE were switched) ?

I also have a DTS 6.1 ES Discrete with a mute LFE and all other channels with low frequencies... exist something like this?

[tebasuna51]

Hi I have a bluray with a lpcm 7.1 that sports a uncommon layout, it is something like L R C S1 B1 B2 S2 LFE
What tool say you this LPCM layout?
If you extract to wavs with eac3to you obtain other than L, R, C, LFE, BL, BR, SL, SR?
Or you think than there are switched? In a english DTS-MA track of "The Revenant" I detect a wrong mapping, the BR must be the SR and the SR the BR.

I also have a DTS 6.1 ES Discrete with a mute LFE and all other channels with low frequencies... exist something like this?
Why not?, this track seems don't have Low Frequency Effects, but can have low frequency music in all channels.

Is not mandatory put all low frequencies in LFE channel, the receivers/amplifiers can extract (by config) all low frequencies from all channels and send them to the subwoofer.

[Motenai Yoda]

What tool say you this LPCM layout?
If you extract to wavs with eac3to you obtain other than L, R, C, LFE, BL, BR, SL, SR?
Or you think than there are switched? In a english DTS-MA track of "The Revenant" I detect a wrong mapping, the BR must be the SR and the SR the BR.

As it sounds strange I open audacity and I get this
http://1.t.imgbox.com/bo0a8xJb.jpg (http://imgbox.com/bo0a8xJb)

UsEac3To require 'A.Tools' to extract the PCM/INT/BIG track, so I extracted it with mkvextractgui2. and extracted to wavs with eac3to, all are labelled as it is in a standard layout order.
But I don't know if only LFE and SR are switched or SR and SL too

Why not?, this track seems don't have Low Frequency Effects, but can have low frequency music in all channels.

Is not mandatory put all low frequencies in LFE channel, the receivers/amplifiers can extract (by config) all low frequencies from all channels and send them to the subwoofer.

nope even the effects have a lot of lf, actually I merged an ac3 lfe into

[tebasuna51]

UsEac3To require 'A.Tools' to extract the PCM/INT/BIG track,
I never see a track labeled by eac3to like PCM/INT/BIG for that I only offer 'A.Tools' ('MkvExtract/Mux' here). What eac3to version have you?

BTW you can force the extraction in COMMAND LINE PARAMETERS:

TRACK: %_.wav (or w64 or wavs)

to see if eac3to output other channel order.

But I don't know if only LFE and SR are switched or SR and SL too
Yep, seems LFE <-> SR changed but is not easy know if there are other change.

nope even the effects have a lot of lf, actually I merged an ac3 lfe into
At your risk. Maybe when receiver mux LF from all channels and new LFE you obtain undesired output.
My reconmendation is preserve the LFE empty.

[Elegant]

Hey guys, I've setup my Windows 10 machine recently and noticed that NeAudio2.ax is not registering with regsvr32. If anyone has had any luck getting these to work, your help would be appreciated!

EDIT: After using a few other tools and some research, I have found the registry keys regsvr32 is adding for NeAudio2.ax. I tested the solution on a Windows 7 machine and had instant success. Unfortunately, eac3to now crashes when accessing NeAudio2.ax on Windows 10. This explains why it couldn't register it in the first place; using force does not solve the issue. This isn't an issue with eac3to as far as I can tell, you simply can't use NeAudio2.ax on Windows 10.

[tebasuna51]

@Elegant
I can't help you with that.

But Nero decoder is only needed to decode AAC, and you have many options in UsEac3to 'A/V Recode':

- If your input is a standalone .aac or .m4a (or the first audio track in mp4 container) you can decode it, or recode to all formats offered, in 1 pass.
- If your input is a track of a mkv container you can decode it, or recode to AC3 with ffmpeg, in 1 pass. You need 2 pass to recode to other format.

[73ChargerFan]

eac3to needs the Nero decoder to be accessible via DirectShow. If it isn't, eac3to can't use it. So something seems to be wrong with the installation. I don't know how to fix it, though.
I want to slowdown an AAC 25fps audio track to AAC 23.97fps, but Nero 7 cannot be found anymore, and I'm on Win 10 x64, so I doubt it would work.

Does anyone have another suggestion? I read neroAacDec.exe doesn't work with eac3to. I'd prefer to stay with eac3to because I need to automate the process by command line.

I should note that the file is in an MKV container, and I'm using eac3to for other conversions. Basically I'm trying to automate revconverting PAL versions of US television shows back to their proper speed.

Thanks!

[Music Fan]

For this kind of operations I use Hybrid which allows to change the pitch or not for slow-down or speed-up but it's a quite big program (using a lot of free tools, including Sox for audio) ;
http://forum.doom9.org/showthread.php?t=153035

[hello_hello]

I'd prefer to stay with eac3to because I need to automate the process by command line.

I should note that the file is in an MKV container, and I'm using eac3to for other conversions. Basically I'm trying to automate revconverting PAL versions of US television shows back to their proper speed.

Thanks!

AVISynth? Something like:

LoadPlugin("C:\Program Files\avisynth_plugin\BassAudio.dll")
BassAudioSource("E:\audio.m4a")
SSRC(Round((AudioRate()*1001.0)/960.0)).AssumeSampleRate(AudioRate())

Or aside from Hybrid (as suggested above), MeGUI also has speed up and slow down options in it's encoder configuration (I borrowed the above script from MeGUI's log file).

[73ChargerFan]

Thanks for the suggestions. I've used MeGUI and hacked some avisynth scripts for Ripbot264, but avisynth is complicated, and I can't use Ripbot264 because I don't want to re-encode the video, just adjust the timing.

But, really, eac3to is PERFECT for this operation, because it will slowdown the video into a new mkv and slowdown the audio into a new container at the same time, then I just use mmg to combine them. 2 commands, plus delete the temp files.

I'll probably use foobar2000 to convert the aac file to flac so that eac3to can do the slowdown. I'll ask on Hydrogenaudio about command line AAC decoders.

[73ChargerFan]

Madshi,

I ask for an update to EAC3TO which will allow decoding of AAC codec without installing Nero 7, which I can't find and I don't know if it'd work in Windows 10 x64.

Thanks.

[LigH]

I agree that if eac3to supported libavcodec as decoder for more input audio formats, it would probably not be too hard to add rather universal conversion features, in addition to the already impressive built-in lossless or low-loss conversions. But that may be a matter of secondary topics, like licenses and maintainability...

As a command line decoder, you may use ffmpeg to convert the audio stream only. And if you build it yourself with "non-free" license (e.g. using jb-alvarado's media-autobuild_suite) and keep your binary private, you may even use the fdk-aac encoder to do it all in one. There is a simple "atempo" filter in ffmpeg (not preserving the pitch), as well as a more complex "rubberband" time stretch filter.

[dvdemon]

Yeah, what I would really love to see is eac3to being able to use the (privatly pre-built) FDK-AAC binary over Nero for (multichannel) AAC encoding if said binary is provided. :cool:

Sometimes Nero's AAC encoder produces very strange output or even complete garbage and I can't figure out why. :confused:

And I also can't wrap my head around piping eac3to's output to the FDK-AAC binary, which only takes WAV files as input...

Only way I got this to work was extracting HD-audio to multichannel WAV file with eac3to and then feeding it to FDK-AAC which works perfectly fine but takes twice as long as if eac3to could do this on its own.

Maybe support for the FDK-AAC encoder could be added to some future incarnation of eac3to.

Best. Tool. Ever.
Keep on! :D

Cheers.

[LigH]

FDK-AAC is an AAC encoder with a quite restrictive license (no binary distribution, DIY from sources ... not hard with the script available in the QAAC "cabinet", but a bit more elaborate); for decoding, libavcodec is more interesting and has a more cooperative license.

[sneaker_ger]

FDK-AAC is an AAC encoder with a quite restrictive license (no binary distribution, DIY from sources ... not hard with the script available in the QAAC "cabinet", but a bit more elaborate)
I doubt script from QAAC will help for FDK-AAC. ;) But since QAAC did better in last test I'd prefer that anyways.

Madshi could probably avoid licensing issues like he did with arcsoft.dll, i.e. users would have to get the required dlls elsewhere. Nothing new, really.

[LigH]

You may not have seen that there is an auto-build script for fdk-aac on the qaac website.

https://sites.google.com/site/qaacpage/cabinet => old: fdkaac_autobuild.zip

[tebasuna51]

And I also can't wrap my head around piping eac3to's output to the FDK-AAC binary, which only takes WAV files as input...

fdkaac support STDIN then you can, for instance:

eac3to INPUT stdout.wav | fdkaac -b 130 -o OUTPUT.m4a -

[sneaker_ger]

You may not have seen that there is an auto-build script for fdk-aac on the qaac website.
Indeed, I did not. My bad.

[dvdemon]

fdkaac support STDIN then you can, for instance:

eac3to INPUT stdout.wav | fdkaac -b 130 -o OUTPUT.m4a -

Yes! :D
That did the trick. I actually missed this tiny " - " at the end, which is obviously very, very important to fdkaac.

I know: rtfm. Shame on me.

But thanks anyway, for clearing this up for me.
:thanks:

I'd still prefer, if eac3to could simply use an existing, pre-provided fdk-aac binary, so this would work without all the piping in the first place.
Just like with Nero's or Arcsoft's DLLs: if they are there, it simply puts them to good use.

But that's just an idea. I'm happy for now.

[LigH]

If there is an fdk-aac DLL with a useful API, that may be possible. In case of doubt, it may require a libav DLL from a dynamically built ffmpeg with "nonfree" license...

[Kempniu]

How exactly does eac3to determine that there are gaps in the source file?

eac3to v3.31 often finds gaps in the AC-3 track of my MPEG-TS DVB captures. The problem is, I cannot tell why. Consider this raw, unprocessed sample (17 MB) straight from the capture device:

https://www.sendspace.com/file/rogha4

eac3to -check claims there is a 21 ms gap in this file at playtime 0:00:03. How is this calculated? PES packets which are part of stream 0x13F0 have continuous PTS, continuity counters in TS packet headers are correct, each PES packet has exactly four AC-3 frames, there are no playback issues. So where is the alleged gap?

The gap warning is displayed regardless of the AC-3 decoder used, -no2ndpass also does not help. What is even more confusing is that if you cut off the last megabyte or so of this sample, eac3to -check no longer displays the message in question.

This looks like a bug in eac3to. Has anyone come across a similar issue?

[MeteorRain]

Any plan to support HEVC stream in TS?
I personally don't rely on this, but I wonder if someone else would be interested in demuxing HEVC from a TS file.

That being said, we do have some HEVC TS video files for testing.

[Music Fan]

Did you try TSmuxer or FFmpeg ?

[szabi]

Hi

I tried this:
stdout.w64 | ffmpeg -i - -c:a eac3 -b:a 1120k %_.eac3

Log:
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
Output #0, eac3, to 'E:\my audio.eac3':
Metadata:
encoder : Lavf57.50.100
Stream #0:0: Audio: eac3, 48000 Hz, 5.1(side), fltp (24 bit), 1120 kb/s
Metadata:
encoder : Lavc57.58.100 eac3

EAC-3 support 8 channels.
Why these 2 channel lost?

bye

[Snowknight26]

You should ask that on the ffmpeg mailing list or IRC channel.. this doesn't relate to eac3to.

[bmcelvan]

Quick question about converting from 24-bit to 16-bit. According to post #10046
http://forum.doom9.org/showthread.php?p=1404223#post1404223

They recommend converting a 24-bit source to .wav first, and then run a second command with the -down16 to convert to 16 bit.

Is this really true? Will this make for a better conversion?

Or is converting from 24bit DTS-HD MA or TrueHD to 16 bit (-down16) and switching formats (FLAC) all in one command yield an identical and just as good conversion?

[Music Fan]

Quick question about converting from 24-bit to 16-bit. According to post #10046
http://forum.doom9.org/showthread.php?p=1404223#post1404223

They recommend converting a 24-bit source to .wav first, and then run a second command with the -down16 to convert to 16 bit.

Is this really true? Will this make for a better conversion?

Or is converting from 24bit DTS-HD MA or TrueHD to 16 bit (-down16) and switching formats (FLAC) all in one command yield an identical and just as good conversion?
It seems to concern only padded 16 bit, thus "fake" 24 bit.

[mini-moose]

I've tried to demux E-AC3 audio on a couple recent discs with eac3to, seems to be failing:

Info:

5: E-AC3, Spanish, 7.1 channels, 768kbps, 48kHz, dialnorm: -27dB
(core: AC3, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB)

command:

eac3to F:\ 1) 5: audio.ac3

output:

a05 Extracting audio track number 5...
a05 Removing AC3 dialog normalization...
a05 Extracting E-AC3 core...
a05 Decoding with libav/ffmpeg...
a05 libav frame CRC mismatch
a05 libav get_buffer() failed
a05 The libav decoder reported error -22 while decoding.
Aborted at file position 1048576.

Demuxing to .eac3 ("eac3to F:\ 1) 5: audio.eac3") :

a05 Extracting audio track number 5...
a05 Removing AC3 dialog normalization...
a05 Applying (E-)AC3 delay failed.
Aborted at file position 1048576.

This format seems to be getting on more and more discs now. TSMuxer is able to handle those.

[LigH]

Is any full MediaInfo analysis of this specific track available? Maybe a short cut-out?

[mini-moose]

Is any full MediaInfo analysis of this specific track available? Maybe a short cut-out?

the disc has a few dubbed audios in E-AC3, non of them will extract. Also get a similar result on another disc with E-AC3 EX

here's the mediainfo for the one I gave as an example:

Audio #3
ID : 4354 (0x1102)
Menu ID : 1 (0x1)
Format : E-AC-3
Format/Info : Audio Coding 3
Format settings, Endianness : Big
Muxing mode : Stream extension
Codec ID : 132
Duration : 1 h 26 min
Bit rate mode : Constant
Bit rate : 500 b/s
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 kHz
Frame rate : 187.500 FPS (256 spf)
Compression mode : Lossy
Stream size : 317 KiB (0%)

short cut out you mean a chunk from the m2ts?

[tebasuna51]

Bug reported to madshi with a sample: http://bugs.madshi.net/view.php?id=450

Seems hapen with all E-AC3 reported by eac3to with 'core'

[SeeMoreDigital]

Bug reported to madshi with a sample: http://bugs.madshi.net/view.php?id=450

Seems hapen with all E-AC3 reported by eac3to with 'core'
Hi Tebasuna51,

Out of interest, I can see from your bug report that the Dolby Plus part of the stream was encoded with 8 (7.1) channels. Aren't such sources quite rare - and getting rarer?


Cheers

[tebasuna51]

I never see that sources. Was a sample uploaded by SquallMX here: http://forum.doom9.org/showthread.php?p=1787122#post1787122

[MeteorRain]

Just a heads up, for TS with HEVC inside it, eac3to cannot detect the audio delay and always assume there's no delay.

[SquallMX]

Hi Tebasuna51,

Out of interest, I can see from your bug report that the Dolby Plus part of the stream was encoded with 8 (7.1) channels. Aren't such sources quite rare - and getting rarer?


Cheers

They are getting more common, E-AC3 is the more practical format for delivering 7.1 dubs tracks, you only need 768 Kbps instead of the 2048 Kbps needed for DTS-HR, some releases have up to 10 dubs, so the savings are huge. Unfortunately most converting software has no proper support for the format.

[SeeMoreDigital]

They are getting more common, E-AC3 is the more practical format for delivering 7.1 dubs tracks...Eh?

HD-DVD predominately used Dolby Digital Plus but none of my disc's offered 7.1Ch audio, they were all 5.1.

And I'm yet to find a Blu-ray disc that offers 7.1Ch DD+. In-fact, I think I've only got one Blu-ray disc that offers DD+ audio :scared:

[Snowknight26]

You aren't looking hard enough.

http://www.blu-ray.com/movies/Suicide-Squad-3D-Blu-ray/161265/
Note: BD Info: Eng TrueHD on Ext Cut only. 3D and Theatrical Disks have French TrueHD and Eng DD+ only.

http://www.blu-ray.com/movies/The-Secret-Life-of-Pets-3D-Blu-ray/169188/
Spanish: Dolby Digital Plus 5.1

http://www.blu-ray.com/movies/Brave-3D-Blu-ray/48045/
French: Dolby Digital Plus 7.1

etc.

[73ChargerFan]

IIRC, DD+ / BD is different than DD+ / HD-DVD

[Music Fan]

It's optional on BD but I believe that it's the only difference ;
https://en.wikipedia.org/wiki/Dolby_Digital_Plus#HD_DVD_and_Blu-ray_Disc

edit : I read too quickly, it's probably the same codec but used differently ; DD+ on BD only concerns rear channels as an extension to the ac3 core and is limited to 1.7 Mbit/s (3.0 Mbit/s on HD DVD).

But IIRC, some BD players accept the DD+ from HD DVD rips.

[ismail0100]

Sorry my english.
Can i convert 448kbps 23.976fps ac3 to 448kbps 24.000fps ac3 using nero?

[LigH]

I see no reason to do that. Which player is unable to handle NTSC-FILM correctly?

[tebasuna51]

@LigH
Sometimes we need convert a second language (from a 23.976 movie) to mix with main movie (24 fps).

@ismail0100
You need Nero only to AAC encode. To convert a AC3 to AC3 you can use the included libAften.dll:

eac3to input.ac3 output.ac3 -448 -23.976 -changeTo24.000

Or better, with ffmpeg:

eac3to input.ac3 stdout.w64 -23.976 -changeTo24.000 | ffmpeg -i - -c:a ac3 -b:a 448k -center_mixlev 0.707 output.ac3

[ismail0100]

@LigH
Sometimes we need convert a second language (from a 23.976 movie) to mix with main movie (24 fps).

@ismail0100
You need Nero only to AAC encode. To convert a AC3 to AC3 you can use the included libAften.dll:

eac3to input.ac3 output.ac3 -448 -23.976 -changeTo24.000

Or better, with ffmpeg:

eac3to input.ac3 stdout.w64 -23.976 -changeTo24.000 | ffmpeg -i - -c:a ac3 -b:a 448k -center_mixlev 0.707 output.ac3

Hello thank you for answering.
My internet friend say to me I am using nero 7 and it is don't change audio's wave but libaften change wave so sound damage quality.

[khoahoc0508]

@LigH
Sometimes we need convert a second language (from a 23.976 movie) to mix with main movie (24 fps).

@ismail0100
You need Nero only to AAC encode. To convert a AC3 to AC3 you can use the included libAften.dll:

eac3to input.ac3 output.ac3 -448 -23.976 -changeTo24.000

Or better, with ffmpeg:

eac3to input.ac3 stdout.w64 -23.976 -changeTo24.000 | ffmpeg -i - -c:a ac3 -b:a 448k -center_mixlev 0.707 output.ac3

Why need setting center_mixlev and it should be done for all ac3 audio?
If i want convert another audio format, it still require?

Thanks!

[tebasuna51]

Why need setting center_mixlev and it should be done for all ac3 audio?
If i want convert another audio format, it still require?


The default value is low and sometimes there are problems with dialog volume when is played by a stereo player (like TV's).

I recommend use it encoding to AC3 from any multichanel input.

[tebasuna51]

Hello thank you for answering.
My internet friend say to me I am using nero 7 and it is don't change audio's wave but libaften change wave so sound damage quality.

eac3to can't encode with nero 7, can use NeroAacEnc.exe to obtain AAC, not AC3.

Of course if you can play AAC multichannel is better option than AC3.

If you still want AC3 the best free encoder is ffmpeg.

[ismail0100]

eac3to can't encode with nero 7, can use NeroAacEnc.exe to obtain AAC, not AC3.

Of course if you can play AAC multichannel is better option than AC3.

If you still want AC3 the best free encoder is ffmpeg.
Okay I like aac more than ac3.
Now how can i convert 448kbps 23.976fps ac3 sound to x(how value it is losless)kbps 24fps aac sound.

If you know can u write code?

[tebasuna51]

You need NeroAacEnc.exe in same folder than eac3to (see first post for links).

eac3to input.ac3 output.m4a -23.976 -changeTo24.000

By default the quality is 0.5, more than enough for a input ac3 448 Kb/s

[pcroland]

Hi!

Is it possible to overwrite the first N second of an AC3 sound with another sound? I have a foreign language dub that fits a film but the beginning of that video has a 519 frame intro (24 fps -> 21.625s). If I apply a positive delay to the dub it will be totally fine for the video, but the intro will be silent. How do I keep the intro sound from the English dub?

I was able to get what I want with
ffmpeg -i film.mkv -c:a ac3 -b:a 192k -ac 2 -t 21.625 intro.ac3
ffmpeg -i "concat:intro.ac3|dub.ac3" -c:a copy dubwithintro.ac3
but I don't know whether it's a good solution or not.

[tebasuna51]

To concat two ac3 streams both must have the same: channels, samplerate and bitrate.

If both have the standard samplerate 48000, your first ffmpeg command line obtain a AC3 2.0 192 Kb/s 48000, like must be your dub.ac3, with a duration of 21.632 sec (duration must be a multiple of 32 ms). And yes, ffmpeg is the best option to obtain this intro.ac3.

After that to concat the two streams you can use ffmpeg (never tested by me) or eac3to:

eac3to intro.ac3+dub.ac3 dubwithintro.ac3

Or the simple msdos copy binary:

copy /B intro.ac3 + dub.ac3 dubwithintro.ac3

[pcroland]

The ~21s intro.ac2 has the same channels etc. The concat speed was really fast so I guess it was successful. Thanks for the eac3to and copy command :)

[73ChargerFan]

Can someone give an example of eac3to identifying and fixing errors in m2ts streams when it demuxes them? Something that makes it inherently superior to all other demuxers?

[tebasuna51]

Can someone give an example of eac3to identifying and fixing errors in m2ts streams when it demuxes them? Something that makes it inherently superior to all other demuxers?

Not sure about your question: "identifying and fixing errors"

If is related with audio demux, eac3to is superior to other demuxers because see the frames timestamps, and can detect gaps/overlaps to be corrected to mantain sync with video.

Other demuxers can extract the audio tracks as is, without check gaps/overlaps.

[Nico8583]

Hi, I have some MKV movies with 5.1 AC3 or 5.1 DTS audio track. I would like (for compatibility with some devices : smartphone, tablet, TV...) to add in each MKV an AAC stereo audio track so I need to convert AC3 or DTS to AAC-LC 2.0.
For AC3, is Nero still the best decoder ? For DTS, is Arcsoft still the best decoder ? For AAC encoding I'm using Nero ACC Encoder.
I have tried "eac3to.exe Movie.mkv 1:Audio.m4a -quality=0.35 -downStereo". Is it sufficient ? Or are there some parameters to add (normalize or other) ? I'm not sure for 0.35 quality for stereo, perhaps I'll go to 0.40 or 0.45.
Thank you !

[tebasuna51]

@Nico8583
You don't need Nero or ArcSoft to decode, lets work the decoders included in last eac3to version.

Your command line is ok..., if the desired audio is first track 1: , normally the first track is the video.
Maybe the first audio track is 2:

You can select the desired quality, 0.35 is the low LC quality, the default is 0.5.

[Nico8583]

Thank you :)
I'll adapt track number, it's not a problem, I'm going to try a method to convert and add an audio track to my MKV, perhaps I'll make a soft to do that, I don't know.
And to quality I'll let 0.5, I'll compare size between differents values

[tebasuna51]

... I'm going to try a method to convert and add an audio track to my MKV, perhaps I'll make a soft to do that...
You can try my GUI UsEac3to (read the readme.txt):

1) Extract the UsEac3to.7z in eac3to folder
2) Run UsEac3to.exe and inform 'Settings' -> 'Path to MkvMerge'
3) Load your mkv
4) In 'Track Input -> Output format' extract and/or convert the 2 audio than you want preserve.
5) 'Run and MkvMux' mux your original mkv without audios with the extracted/converted audios

[Nico8583]

Thank you for the tip, I'll try it but I would like to process about 1000 movies so I would like to make an automated process :)

For downmixing to stereo, there is no interest to use Normalize ? I would like to have a correct sound level without putting my TV level to 90%

[tebasuna51]

With -downStereo eac3to add center and surround channels to front channels, most the times the volume reach more than 0 dB and eac3to make a automatic Normalize to obtain peaks at 0 dB only.

You can force the Normalize but normally is not necesary, and don't solve the low dialog volume because this only can be corrected with a specific downmix for each movie.

[Nico8583]

I've tried manually to convert to AAC and mux to MKV. MediaInfo shows a 9ms delay for each AAC track, is it normal ?

[sneaker_ger]

Did you use eac3to to create the .m4a and mkvmerge to mux it? Then: yes, it's probably normal. Mkvmerge compensates for the aac encoder delay.

[Nico8583]

Yes I convert to .m4a with eac3to 3.31 and mux with mkvmerge 9.8.0. Why does the encoder add delay ?

[sneaker_ger]

https://developer.apple.com/library/content/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html
https://en.wikipedia.org/wiki/Modified_discrete_cosine_transform

[Nico8583]

Thank you, I'll look at it :)

[ndjamena]

When a program reads the EAC3To output I'd like a sure way to match the track info to the output of other programs (MediaInfo/MKVMerge). Outputting the PIDs and Matroska Track Numbers would be the surest way of achieving that.

(Sorry, I had noticed there was no "feature request option" on the tracker and was going to post here instead, but I looked through the tickets and one of them was a request to use pre-encoded AC-3 to mux with TrueHD instead of making a new one, which is a feature request and was labeled a feature request and a bug as well.)

[Nico8583]

I've tried this command line "eac3to.exe Movie.mkv 1:Audio.m4a -downStereo" but the sound level is low. Is it possible to increase it ? Is it done by applying DRC (with NicAc3Source/NicDtssource) ?

[LigH]

eac3to does not use AviSynth audio source plugins.

Whether DRC can be applied, may depend on the format of the audio stream in the MKV. 2-pass normalization would always be possible during a complete conversion, but I don't know for sure if eac3to uses a 2-pass method. Probably not.

[tebasuna51]

... but the sound level is low

The whole volume or the dialog volume?
Please put the log.

eac3to never apply DRC when decode.

[Nico8583]

It seems to be the whole volume but mainly dialog volume.
Here is the log :
eac3to v3.31
command line: "D:\eac3to\eac3to.exe" "D:\Test\Test.mkv" 2:"D:\Test\Test.mkv_0.50.m4a" -downStereo
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 1:46:34, 24p /1.001
1: h264/AVC, 1080p24 /1.001
2: AC3 EX, French, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Decoding with libav/ffmpeg...
[a02] Downmixing multi channel audio to stereo...
[a02] Reducing depth from 64 to 32 bits...
[a02] Encoding AAC <0.50> with NeroAacEnc...
[a02] Clipping detected, a 2nd pass will be necessary. <WARNING>
[a02] Starting 2nd pass...
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Decoding with libav/ffmpeg...
[a02] Downmixing multi channel audio to stereo...
[a02] Reducing depth from 64 to 32 bits...
[a02] Encoding AAC <0.50> with NeroAacEnc...
[a02] Applying -2.24dB gain...
Video track 1 contains 153312 frames.
eac3to processing took 6 minutes, 34 seconds.
Done.
Thank you !

[tebasuna51]

The whole volume can't be bigger because eac3to detect clip:

[a02] Clipping detected, a 2nd pass will be necessary. <WARNING>

and need attenuate it:

[a02] Applying -2.24dB gain...

Then your movie maybe need other tool to downmix that audio. Try BeHappy.

[Nico8583]

All tests I've done allways report "Clipping detected"...
Do you have some advices about BeHappy or NicAudio in order to downmix to stereo ?

[73ChargerFan]

How about this:
[a02] Removing AC3 dialog normalization...

Perhaps that step can be disabled? I haven't tried it.

[Music Fan]

You don't need to keep the dialog normalization, it's a flag which actually concerns the level of all channels, not only the center channel. This flag is often used on stereo contents on TV programs, which is rather a nonsense when you think it only concerns the center channel, but it has though something to do with dialogs (when well used) ;
http://hometheaterhifi.com/volume_7_2/feature-article-dialog-normalization-6-2000.html

http://hometheaterhifi.com/volume_7_2/images/feature-article-dialog-normalization-fig-2.gif

http://hometheaterhifi.com/volume_7_2/images/feature-article-dialog-normalization-fig-3.gif

[tebasuna51]

About Dialogue Normalization (DN) and Dynamic Range Compression (DRC)

(That was explained in the first posts in this thread)
Like you can see in the arcticle linked by Music Fan both concepts, used by Dolby Digital (DD) in their encodes, are very ussefull.
What is the problem?

1) Only DD uses them, other sources don't have, or don't use, similar concepts (DTS, MP3, CDA, AAC, ...).
Normally the same source, encoded with DD, sound at low volume than encoded with other encoder, when played.

2) DN and DRC must be applied at play time, the tools (like eac3to) designed to recode between formats, can't apply DN/DRC when decode DD if we don't want lose quality.
If you want preserve DN/DRC you need use a DD certified encoder.
Free AC3 encoders don't claim make DD certified streams, only AC3 streams and, by default, don't use DN/DRC.
Of course DN/DRC can't be translated to other encoders.

Modern players (more powerfull than olds) have "Night mode" to solve the same problem than DRC at play time, and work over any format, not only DD streams.

Downmix problems

We can't send the same sound volume with 2 speakers than with 6 speakers, that is obvious. The -downStereo parameter try to preserve the balance between channels adding 5 input channels to only 2 (FL'-FR') with this method:

FL' = FL + 0.707*FC + SL
FR' = FR + 0.707*FC + SR

Is normal than adding volume like that produce volume sounds over 0 dB (than produce clip), for that the WARNING and the posterior:
Applying -2.24dB gain...
To output the max volume peaks without distort.

But sometimes the FC (many of dialogs) contribution is not enough to listen clearly the dialogs compared with F and S contributions.

There are workarounds to solve this problem, but not with eac3to than try always preserve the best quality.

With BeHappy (or BeLight) you can apply DRC, at decoder time, over DD streams (not with other formats).
Also you can apply Boost (BeLight) or many other options with AviSynth (BeHappy).
But explain them are OT in this thread.

[Nico8583]

Thank you for all your informations, it's very interesting :)
Is it the same "issue" if I convert to AAC multichannel ? Or is it specific at the downstereo and not at the format ?

[tebasuna51]

We don't need mix channels in a multichannel conversion, then don't exist the problem.
Each channel preserve their original volume.

[gzuz]

Hey folks! I apologize if this is the wrong thread for my post.

I have 2 Blu-ray releases of the same movie. They're different editions.

I used eac3to to extract the video track from the first disc, and the audio/subtitle tracks from the other, for the purpose of muxing them together into an .mkv.

Unfortunately, they don't share the same length. The video from disc #1 is 8 seconds longer than the video from disc #2. So the audio track and subtitles from disc #2 are running ahead of the picture.

I calculated the exact delay, and it's 7967 ms. I used mkvmergeGUI to apply the delay in question, but to my surprise the sync was still off for ~half a second. Finally, I set it to 7217 ms and I seem to have hit the spot.. but have I?

How can I be certain that I've achieved the perfect sync? Surely there must be a better way of doing this, rather than relying on my own two eyes and guessing/assuming. Is there a method for doing this with eac3to, or perhaps some other application?

Thanks in advance! :)

[Music Fan]

Is it the same cut and the same framerate for both versions ?

[tebasuna51]

I have 2 Blu-ray releases of the same movie. They're different editions.

Then, probably, have different initial credits (~7217 ms) and maybe the end until 7967 ms.

Don't exist a easy tool to determine the exact amount of delay.
You need found the same video frame (after credits) and the time diference is the exact delay than you need.

[gzuz]

Is it the same cut and the same framerate for both versions ?

Yes, they're the same. There's a Warner Brothers intro that lasts for ~8 seconds in the beginning of my video source. That's what's causing the delay.

@tebasuna51 - You're on point. How do I find the exact video frame, and thus the time difference? Thank you!

Okay (2.5hrs later),

So I managed to find a frame match in both sources:

https://postimg.org/image/lh6y3e68h/

I found the beginning and the ending of this "scene". But I'm confused:

7010 - 7024 - > 6839 - 6852

14 frames - > 13 frames

So this still shot from the movie runs for 14 frames on the first source, while on the second one it does so for 13 frames. Do I pick the first frame of each video source and subtract, or?

For example - 7010 minus 6839? What am I supposed to do? Thanks!

[tebasuna51]

You have already the time (seems the same frame), the delay is:

4:52.375 - 4:45.243 = 7132 ms

You can verify if the delay is the same at end of movie, because if there are differences (14 frames - > 13 frames) along the movie you have a serious problem.

[gzuz]

You have already the time (seems the same frame), the delay is:

4:52.375 - 4:45.243 = 7132 ms

You can verify if the delay is the same at end of movie, because if there are differences (14 frames - > 13 frames) along the movie you have a serious problem.

Thanks. I just tested with another frame and it's way off. Well, I tried.

Thanks for helping me out guys, it's appreciated!

[MeteorRain]

Let me bump this up a little bit:

Any plan to support HEVC so that we can demux the audio with the correct delay?

[Thunderbolt8]

support HEVCyes please

[LigH]

HEVC is a format of video content streams. It has no audio in itself. It may have audio streams in parallel when multiplexed in containers like TS, MP4, MKV. In this case, it should not be very different from AVC video streams. The only difference I would imagine is a decoder delay, due to the higher complexity of the GOP structure and the video encoding algorithms. Would eac3to be able to consider that?

[tebasuna51]

Any plan to support HEVC so that we can demux the audio with the correct delay?

Of course eac3to can't extract HEVC video or manage Ultra HD Blu-ray, and is a desired feature.

BTW extract, and delay, the audio from a mkv can work even with HEVC video. At least with a simple test:

eac3to v3.31
command line: "D:\Programa\eac3to\eac3to.exe" "D:\Temp\t\Test.mkv" 2: "D:\Temp\t\Test.mkv_2spa.ac3"
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 0:00:37, 24p /1.001
1: MPEGH/ISO/HEVC, English, 3840x1568p (120:49)
2: AC3, Spanish, 5.1 channels, 448kbps, 48kHz, 12000ms
Bitstream parsing for track 1 failed. <WARNING>
Demuxing this track may still produce correct results - or not. <WARNING>
[a02] Extracting audio track number 2...
[a02] Applying (E-)AC3 delay...
[a02] Creating file "D:\Temp\t\Test.mkv_2spa.ac3"...
eac3to processing took 1 second.
Done.

The extracted ac3 is OK for me.

[MeteorRain]

eac3to main.ts main.aac
TS, 1 audio track, 0:58:40
1: AAC, 2.0 channels, 198kbps, 48kHz
Track 1 is used for destination file "main.aac".
a01 Extracting audio track number 1...
a01 Creating file "main.aac"...
eac3to processing took 1 minute, 52 seconds.
Done.

It's a file recorded from TV broadcasting with HEVC Main10 video stream and AAC audio stream. It has about -450ms delay, and since eac3to can't recognize the video, it cannot figure out the actually delay.

I believe it only happens when a TS file is given, as with MKV it seems to work fine.

[gzuz]

Hey again guys,

I demuxed a TrueHD stream from a BD, and eac3to reported a delay of 125ms in relation to the video stream.

That means the 125ms audio delay is originally included in the BD, correct? I assume I don't need to "fix" the delay when muxing into an. mkv, because that's how it got out of the studio.

Am I right? Thanks in advance!

[tebasuna51]

I demuxed a TrueHD stream from a BD, and eac3to reported a delay of 125ms in relation to the video stream.
:logfile:

I assume I don't need to "fix" the delay when muxing into an. mkv
You need fix the delay because eac3to can't apply the delay over a TrueHD, like do automatically with other audio track.

[gzuz]

:logfile:


You need fix the delay because eac3to can't apply the delay over a TrueHD, like do automatically with other audio track.


eac3to v3.31
------------------------------------------------------------------------------
M2TS, 2 video tracks, 9 audio tracks, 26 subtitle tracks, 2:20:03, 24p /1.001
1: Chapters, 40 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: VC-1, 480p24 /1.001 (3:2), 83ms
4: AC3, English, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB, 125ms
5: TrueHD/AC3, English, 5.1 channels, 48kHz, dialnorm: -27dB, 125ms
(embedded: AC3, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB)
6: AC3, Czech, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
7: AC3, Hungarian, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
8: AC3, Polish, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
9: AC3, Russian, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
10: AC3, Thai, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
11: AC3, Turkish, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB
12: E-AC3, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB, 83ms
13: Subtitle (PGS), English
14: Subtitle (PGS), Chinese
15: Subtitle (PGS), Arabic
16: Subtitle (PGS), Indonesian
17: Subtitle (PGS), Bulgarian
18: Subtitle (PGS), Croatian
19: Subtitle (PGS), Czech
20: Subtitle (PGS), Estonian
21: Subtitle (PGS), Modern Greek
22: Subtitle (PGS), Hebrew
23: Subtitle (PGS), Hungarian
24: Subtitle (PGS), Icelandic
25: Subtitle (PGS), Latvian
26: Subtitle (PGS), Lithuanian
27: Subtitle (PGS), Polish
28: Subtitle (PGS), Romanian
29: Subtitle (PGS), Russian
30: Subtitle (PGS), Serbian
31: Subtitle (PGS), Slovenian
32: Subtitle (PGS), Thai
33: Subtitle (PGS), Turkish
34: Subtitle (PGS), Czech
35: Subtitle (PGS), Hungarian
36: Subtitle (PGS), Turkish
37: Subtitle (PGS), English
38: Subtitle (PGS), Chinese
Creating file "C:\lol\chap.txt"...
[s13] Extracting subtitle track number 13...
[v02] Extracting video track number 2...
[a05] Extracting audio track number 5...
[a05] Extracting TrueHD stream...
[a05] Removing TrueHD dialog normalization...
[v02] Creating file "C:\lol\mov.vc1"...
[a05] Creating file "C:\lol\sound.thd"...
[s13] Creating file "C:\lol\eng.sup"...
Video track 2 contains 201466 frames.
Video track 3 contains 201465 frames.
Subtitle track 13 contains 1473 captions.
eac3to processing took 5 minutes, 11 seconds.
Done.


Thanks for the reply @tebasuna51. How do I proceed about applying the delay? Do I just do it via eac3to? Thank you.

[tebasuna51]

eac3to can't apply delay to a thd.

When add the .thd in MkvToolNix GUI, select the track and put the Delay (125).

[gzuz]

eac3to can't apply delay to a thd.

When add the .thd in MkvToolNix GUI, select the track and put the Delay (125).

Thanks man. I actually thought I have to apply a negative delay. But I realized it's a positive delay when I muxed the files. :D

Thanks again for the help, it's appreciated!

[Weirdo]

Is the -shutdown parameter confirmed as not working? It has never worked for me (Win 8/10, admin account).

[NanoBot]

Hi,

i just learned that FLAC has recently been updated to version 1.3.2. Does anyone already has compiled a new libflac.dll ( to be used with eac3to ) of the new version ?

[LigH]

Not yet on RareWares; asked in Hydrogenaud.io for it. The media-autobuild_suite unfortunately builds different DLLs.

[heerschop]

Not yet on RareWares; asked in Hydrogenaud.io for it.

Flac.dll 1.3.2 is available on RareWares.
http://www.rarewares.org/files/lossless/

flac_dll-1.3.2-git-20170314-icl.zip
flac_dll-1.3.2-git-20170314-x64-icl.zip

[LigH]

"That escalated quickly." ;) - A pity that their forum notifications seem to be broken for me. Thanks for notifying me (us) here. Plus: Lossless (http://www.rarewares.org/lossless.php) web page.

[tormento]

Just found, dunno if #old:

aften.dll git-20091226-icl11_1_054 (https://github.com/wieslawsoltes/wavtoac3encoder)
dcadec.dll 0.2.0 (https://github.com/foo86/dcadec/releases)

Tested and working.

If somebody has other newer and working dll's to replace, please post.

[hubblec4]

Just found, dunno if #old:

aften.dll git-20091226-icl11_1_054 (https://github.com/wieslawsoltes/wavtoac3encoder)
dcadec.dll 0.2.0 (https://github.com/foo86/dcadec/releases).

Thanks for this info.

[Sparktank]

Just found, dunno if #old:

aften.dll git-20091226-icl11_1_054 (https://github.com/wieslawsoltes/wavtoac3encoder)
dcadec.dll 0.2.0 (https://github.com/foo86/dcadec/releases)

Tested and working.

If somebody has other newer and working dll's to replace, please post.

Might be impossible to get newer dcadec DLL's now that foo86 has moved everything to FFMPEG.

The git for FFMPEG should be more updated than GitHub.

commits from foo86:
https://git.ffmpeg.org/gitweb/ffmpeg.git?a=search&h=HEAD&st=author&s=foo86

commits about avcodec/dca in general:
https://git.ffmpeg.org/gitweb/ffmpeg.git?a=search&h=HEAD&st=commit&s=avcodec%2Fdca

Madshi might have to remove dcadec DLL's in favor of FFMPEG udpates.

Which means, we'd just have to wait for madshi to update his modified libraries whenever FFMPEG goes through major updates.

[tebasuna51]

The same for aften dll's, maybe are fast thanks to wisodev optimizations, but with the same problems than old aften (https://forum.doom9.org/showthread.php?p=1794293#post1794293)

The AC3 encoder included in ffmpeg is the preferred.

[wish_ihadalife]

Hello all,

I have been lurking the doom9 forums for quite some time now, and been experimenting to get eac3to work, but I am a little stuck.

Running a test on my command line, I am getting the following messages:

Microsoft Windows [Version 6.1.7601]
Copyright (c) 2009 Microsoft Corporation. All rights reserved.

C:\Users\User\Documents\Conversions>C:\Users\User\Documents\eac3to\eac3to.exe -t
est
eac3to (v3.31) is up to date
Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed
http://www.nero.com/eng/store-blu-ray.html
CAUTION: You need Nero 7. Nero 8 won't work with eac3to.
ArcSoft DTS Decoder doesn't seem to be installed
http://www.arcsoft.com/products/totalmediatheatre
Sonic Audio Decoder (3.31.0.0) doesn't seem to be installed
Haali Matroska Muxer (2013-04-14) is installed
Nero AAC Encoder (1.5.4.0) is installed
Surcode DTS Encoder (1.0.29.0) is installed

Does anybody know where I can find/buy the Nero Audio decoder? As well as the ArcSoft and Sonic ones? Also, which versions will work best?

Are Libav / ffmpeg the same program?


Thank you for all of your help. I usually use Headac3e and tsMuxer, but I am hearing this is a much better encoder. Would love to get started, thank you for your support!

Sorry for all the questions, just trying to get started!

[LigH]

Welcome.

Are Libav / ffmpeg the same program?

Depends on the desired meaning...

The ffmpeg project contains libraries for the handling of containers (libavformat) and contents (libavcodec).

During some stage of development, some developers split and created a fork of ffmpeg, named libav. It doesn't matter much, probably, from which fork you prefer to use the codec libraries as long as they are up to date, so they can handle modern HD audio formats as well.

Does anybody know where I can find/buy the Nero Audio decoder? As well as the ArcSoft and Sonic ones?

In most recent versions, the libav* libraries should be able to handle most formats with satisfying quality. To use 3rd party decoders, you would possibly have to have bought this software a while ago, or use "codec packs" regardless of software licenses... (see forum rules).

Also, which versions will work best?

Toughest question, regarding the ArcSoft decoder; several versions have different (dis-)advantages. I believe only "fanatic" audiophiles care. Home users won't recognize a difference to the results of libav* decoders while just listening in their living rooms, especially after compressing with another lossy format...

[sneaker_ger]

ArcSoft has pretty much been replaced by DcaDec. The only thing it cannot decode compared to ArcSoft is DTS Express and that's only because the version in eac3to is outdated. The newer one in ffmpeg can decode it.

[tebasuna51]

Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed
ArcSoft DTS Decoder doesn't seem to be installed
Sonic Audio Decoder (3.31.0.0) doesn't seem to be installed
- DirectShow Nero Audio Decoder (not free) is only needed to decode audio AAC.
You can extract the AAC track from a .mkv and use other free tool (ffmpeg,...) to recode, if you need recode, most the times is better preserve the AAC track as is.
Install not free Nero 7 for only this function is not recommended.

- ArcSoft (not free) DTS Decoder is only needed to decode DTS Express (some comentary audio tracks in BD's) like sneaker_ger say.
Also ffmpeg can be used for that.
The included decoder (dcadec) is equal, or best, than ArcSoft for DTS standard and HD.

- Sonic Audio Decoder (not free) is not needed at all.
It was used at first time to compare outputs, but now it is not recommended.

You can work with eac3to without these 3 commercial soft's.

[nevcairiel]

dcadec can only decode the lossy stream "core"

This is wrong. libdcadec can decode the full lossless DTS-HD parts, even the "strange setup" streams that ArcSoft does not decode lossless in all cases.

[gonzalo76]

Just one question,

When I had eac3to v3.27, I could decode DTS Master Audio with ArcSoft Decoder:

eac3to v3.27
command line: "C:\Program Files (x86)\eac3to\eac3to.exe" "F:\BDRips\Bob Marley & The Wailers\Legend\Blu-ray\FullDisc\BD_VOLUME\BDMV\STREAM" 1) 4:"D:\MyBDRips\1_4_audio.flac" -progressnumbers
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 1:07:21, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 2.0 channels, 24 bits, 96kHz
4: DTS Master Audio, English, 5.1 channels, 24 bits, 96kHz
(core: DTS-96/24, 5.1 channels, 24 bits, 1509kbps, 96kHz)
5: TrueHD/AC3, English, 5.1 channels, 96kHz
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
[a04] Extracting audio track number 4...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Encoding FLAC with libFlac...
[a04] Creating file "D:\MyBDRips\1_4_audio.flac"...
[a04] The original audio track has a constant bit depth of 24 bits.
Video track 2 contains 96897 frames.
eac3to processing took 41 minutes, 17 seconds.
Done.

I recently updated to eac3to v3.31 and now, I can not decode DTS Master Audio stream anymore.

[22:11:40] Retrieving features
[22:11:40] Arguments: "F:\BDRips\Bob Marley & The Wailers\Legend \Blu-ray\FullDisc\BD_VOLUME\BDMV"
[22:11:41] Feature Retrieval Completed
[22:11:41] Retrieving streams
[22:11:41] Arguments: "F:\BDRips\Bob Marley & The Wailers\Legend \Blu-ray\FullDisc\BD_VOLUME\BDMV" 1) -progressnumbers
[22:11:46] Stream Retrieval Completed
[22:11:58] Extracting streams
[22:11:59] Arguments: "F:\BDRips\Bob Marley & The Wailers\Legend \Blu-ray\FullDisc\BD_VOLUME\BDMV" 1) 4:"D:\MyBDRips\Bob Marley & The Wailers\Legend\Blu-ray\FullDisc\BD_VOLUME\BDMV\1_4_audio.flac" -progressnumbers
[22:12:04] a04 Extracting audio track number 4...
[22:12:04] a04 Decoding with libDcaDec DTS Decoder...
[22:12:04] a04 libDcaDec reported the warning "XLL output not lossless".
[22:12:04] a04 Encoding FLAC with libFlac...
[22:12:05] a04 Creating file "D:\MyBDRips\Bob Marley & The Wailers\Legend\Blu-ray\FullDisc\BD_VOLUME\BDMV\1_4_audio.flac"...
[22:12:15] Stream Extraction Completed

C:\Program Files (x86)\eac3to>eac3to -test
eac3to (v3.31) is up to date
Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed
http://www.nero.com/eng/store-blu-ray.html
CAUTION: You need Nero 7. Nero 8 won't work with eac3to.
ArcSoft DTS Decoder (1.1.0.0) works fine
Sonic Audio Decoder (3.31.0.0) doesn't seem to be installed
Haali Matroska Muxer doesn't seem to be installed
http://haali.net/mkv
Nero AAC Encoder could not be located
http://www.nero.com/eng/nero-aac-codec.html
Copy NeroAacEnc.exe to the eac3to or to the Windows folder.
Surcode DTS Encoder doesn't seem to be installed
http://www.surcode.com

I read that "libDcaDec is now default for all DTS tracks except XSA / low bitrate"...

Does anybody know what's going on?

Thank you very much

[gonzalo76]

This is wrong. libdcadec can decode the full lossless DTS-HD parts, even the "strange setup" streams that ArcSoft does not decode lossless in all cases.

And do you know why I have this warning message?

[22:12:04] a04 Decoding with libDcaDec DTS Decoder...
[22:12:04] a04 libDcaDec reported the warning "XLL output not lossless".

[Midzuki]

And do you know why I have this warning message?

[22:12:04] a04 Decoding with libDcaDec DTS Decoder...
[22:12:04] a04 libDcaDec reported the warning "XLL output not lossless".

The latest eac3to uses an outdated build of libdcadec.

[wish_ihadalife]

- DirectShow Nero Audio Decoder (not free) is only needed to decode audio AAC.
You can extract the AAC track from a .mkv and use other free tool (ffmpeg,...) to recode, if you need recode, most the times is better preserve the AAC track as is.
Install not free Nero 7 for only this function is not recommended.


Are you confusing Nero Audio Decoder and Nero AAC Encoder? (which I already have) Or are they similar in any way?



- ArcSoft (not free) DTS Decoder is only needed to decode DTS Express (some comentary audio tracks in BD's) like sneaker_ger say.
Also ffmpeg can be used for that.
The included decoder (dcadec) is equal, or best, than ArcSoft for DTS standard and HD.


Yeah, it seems as if you need a very early version of ArcSoft... I tried installing ArcSoft TotalMedia Theatre 6.7 and eac3to doesn't seem to pick up the DTS Decoder. Even when I moved it into System Folders.

[Midzuki]

P.S.:

I recently updated to eac3to v3.31 and now, I can not decode DTS Master Audio stream anymore.

[22:11:40] Retrieving features
[22:11:40] Arguments: "F:\BDRips\Bob Marley & The Wailers\Legend \Blu-ray\FullDisc\BD_VOLUME\BDMV"
[22:11:41] Feature Retrieval Completed
[22:11:41] Retrieving streams
[22:11:41] Arguments: "F:\BDRips\Bob Marley & The Wailers\Legend \Blu-ray\FullDisc\BD_VOLUME\BDMV" 1) -progressnumbers
[22:11:46] Stream Retrieval Completed
[22:11:58] Extracting streams
[22:11:59] Arguments: "F:\BDRips\Bob Marley & The Wailers\Legend \Blu-ray\FullDisc\BD_VOLUME\BDMV" 1) 4:"D:\MyBDRips\Bob Marley & The Wailers\Legend\Blu-ray\FullDisc\BD_VOLUME\BDMV\1_4_audio.flac" -progressnumbers
[22:12:04] a04 Extracting audio track number 4...
[22:12:04] a04 Decoding with libDcaDec DTS Decoder...
[22:12:04] a04 libDcaDec reported the warning "XLL output not lossless".
[22:12:04] a04 Encoding FLAC with libFlac...
[22:12:05] a04 Creating file "D:\MyBDRips\Bob Marley & The Wailers\Legend\Blu-ray\FullDisc\BD_VOLUME\BDMV\1_4_audio.flac"...
[22:12:15] Stream Extraction Completed

IIRC, you may use the -arcsoft switch for forcing eac3to to use the well-known DTSDecoderDll.DLL.

[gonzalo76]

The latest eac3to uses an outdated build of libdcadec.

OK, I downloaded the latest release version 0.2.0 (https://github.com/foo86/dcadec/releases), released on 4 Jan 2016. The warning message "XLL output not lossless" disappeared now!!!

But instead of 24 bits, the decoded file has 16 bits, with libDcaDec DTS Decoder and also with ArcSoft DTS Decoder :confused:

Decoding with libDcaDec DTS Decoder...

eac3to v3.31
command line: "C:\Program Files (x86)\eac3to\eac3to.exe" "D:\BDrips\Def Leppard\
Viva! Hysteria\BDMV" 2) 1:"D:\BDrips\Def Leppard\Viva! Hysteria\2_1_chapter.txt"
3:"D:\BDrips\Def Leppard\Viva! Hysteria\2_3_audio.flac" -progressnumbers
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 0:38:44, 24p /1.001
1: Chapters, 8 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
4: DTS Master Audio, English, 2.0 channels, 24 bits, 48kHz
(core: DTS, 2.0 channels, 1509kbps, 48kHz)
Creating file "D:\BDrips\Def Leppard\Viva! Hysteria\2_1_chapter.txt"...
[a03] Extracting audio track number 3...
[a03] Decoding with libDcaDec DTS Decoder...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "D:\BDrips\Def Leppard\Viva! Hysteria\2_3_audio.flac"...
[a03] The original audio track has a constant bit depth of 16 bits.
[a03] Superfluous zero bytes detected, will be stripped in 2nd pass.
[a03] Starting 2nd pass...
[a03] Decoding FLAC...
[a03] Reducing depth from 24 to 16 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "D:\BDRips\Def Leppard\Viva! Hysteria\2_3_audio.flac"...
[a03] The processed audio track has a constant bit depth of 16 bits.
Video track 2 contains 55725 frames.
eac3to processing took 10 minutes, 50 seconds.
Done.

Decoding with ArcSoft DTS Decoder...

==> ..\BDMV\PLAYLIST\01002.mpls
>> M2TS, 1 video track, 2 audio tracks, 1:25:21, 24p /1.001 (16 chapters)
> 3: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz

> 4: DTS Master Audio, English, 2.0 channels, 24 bits, 48kHz

> eac3to v3.31
command line: "eac3to.exe" "D:\BDrips\Def Leppard\Viva! Hysteria\BDMV\PLAYLIST\01002.mpls"
1) 4:"D:\BDrips\Def Leppard\Viva! Hysteria\Output.wav" -arcsoft -progressnumbers
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 1:25:21, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
4: DTS Master Audio, English, 2.0 channels, 24 bits, 48kHz
(core: DTS, 2.0 channels, 1509kbps, 48kHz)
[a04] Extracting audio track number 4...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Writing WAV...
[a04] Creating file "D:\BDrips\Def Leppard\Viva! Hysteria\Output.wav"...
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] Superfluous zero bytes detected, will be stripped in 2nd pass.
[a04] Starting 2nd pass...
[a04] Reading WAV...
[a04] Stripping zero bytes...
[a04] Writing WAV...
[a04] Creating file "D:\BDrips\Def Leppard\Viva! Hysteria\Output.wav"...
Video track 2 contains 122783 frames.
eac3to processing took 6 minutes, 37 seconds.
Done.



Thank you.

[Sparktank]

IIRC, all decoders for DTS (dcadec, arcsoft) do the same thing with "XLL output no lossless".
Except, they were all silent, so the only difference was that the recent dcadec allowed the message to be seen.

One user did make a custom build that just hid the message again for the placebo users who want that message "out of sigh of mind".

Arcsoft has so many versions, and no one has thoroughly tested the last version (1.09), so we can't really say if it behaves the same as previous versions or has different negative behaviours like some versions of Arcsoft.

For Arcsoft, v1.0.0.0 is the most known version to handle most scenarios.

But Dcadec should be the same and handle all formats (including Strange Setup, which Arcsoft requires a specific version) except for DTS Express.

[Sparktank]

[a03] The original audio track has a constant bit depth of 16 bits.
[a03] Superfluous zero bytes detected, will be stripped in 2nd pass.

Superfluous means "unnecessary", so it was really 16bit stored in a 24bit.

[Midzuki]

.........

But Dcadec should be the same and handle all formats (including Strange Setup, which Arcsoft requires a specific version) except for DTS Express.

dcadec 0.2.0 is really outdated. Since April of 2016, dcadec supports DTS Express, even the multichannel (5.1) streams.

https://github.com/foo86/dcadec/issues/33

[sneaker_ger]

The problem is development of the (external) dcadec library has stopped. It's all in ffmpeg now.

[nevcairiel]

eac3to already uses ffmpeg for other formats, so that wouldn't be a big hurdle - assuming madshi updates to a recent ffmpeg at some point in the future.

[tebasuna51]

Are you confusing Nero Audio Decoder and Nero AAC Encoder? (which I already have) Or are they similar in any way?

The Nero Audio Decoder needed by eac3to are Directshow filters included in Nero 7 package (plus a extra plugin). All not free, but there are some hacks to install the required filters without install the full Nero 7 package.

The Nero AAC Encoder (NeroAacEnc.exe) is a free standalone encoder than can be used by eac3to to output .m4a (AAC in mp4 container).
There are also a free standalone NeroAacDec.exe than can decode AAC inside mp4 container (.mp4 or .m4a), but can't be used by eac3to. You need run it in command line mode.

Yeah, it seems as if you need a very early version of ArcSoft... I tried installing ArcSoft TotalMedia Theatre 6.7 and eac3to doesn't seem to pick up the DTS Decoder. Even when I moved it into System Folders.

The lastest versions of ArcSoft don't have a required (by eac3to) checkactivate.dll. You need include this dll from previous versions to work with latest versions.

There are hacks (in this forum) to install the required (by eac3to) ArcSoft files without install the full ArcSoft TotalMedia Theatre package.

BTW, ArcSoft decoder is only needed to decode XSA / low bitrate (DTS Express in BD's), with the last libdcadec.dll you can decode any other DTS (standard or HD) the same (or better like with Strange Setup 7.1) than with ArcSoft.

Of course a great improvement of eac3to is (if possible):

- Add full dcadec decoder to support XSA / low bitrate, like is already in ffmpeg.

- Add AAC decoder from ffmpeg (maybe need some workaround to avoid license problems, similar to used with ArcSoft, Nero 7, Sonic or Surcode DTS)

- Add AC3 encoder from ffmpeg.

[Sparktank]

commits from foo86:
https://git.ffmpeg.org/gitweb/ffmpeg.git?a=search&h=HEAD&st=author&s=foo86

commits about avcodec/dca in general:
https://git.ffmpeg.org/gitweb/ffmpeg.git?a=search&h=HEAD&st=commit&s=avcodec%2Fdca


Those are the latest commits made for dcadec so far on ffmpeg.

[gonzalo76]

IIRC, you may use the -arcsoft switch for forcing eac3to to use the well-known DTSDecoderDll.DLL.

I did it! :thanks:

dcadec 0.2.0 is really outdated. Since April of 2016, dcadec supports DTS Express, even the multichannel (5.1) streams.

https://github.com/foo86/dcadec/issues/33

So, 0.2.0 is the latest version (and also outdated), right?

What decoder should I use to extract DTS Master Audio 2.0, 5.1, 6.1 and 7.1 now :confused: I was using ArcSoft DTS Decoder (1.1.0.0).

Thank you very much for all your help! :thanks:

[gonzalo76]

IIRC, all decoders for DTS (dcadec, arcsoft) do the same thing with "XLL output no lossless".
Except, they were all silent, so the only difference was that the recent dcadec allowed the message to be seen.

One user did make a custom build that just hid the message again for the placebo users who want that message "out of sigh of mind".

Arcsoft has so many versions, and no one has thoroughly tested the last version (1.09), so we can't really say if it behaves the same as previous versions or has different negative behaviours like some versions of Arcsoft.

For Arcsoft, v1.0.0.0 is the most known version to handle most scenarios.

But Dcadec should be the same and handle all formats (including Strange Setup, which Arcsoft requires a specific version) except for DTS Express.

Yes, Arcsoft, v1.0.0.0 is the version I'm using.

Superfluous means "unnecessary", so it was really 16bit stored in a 24bit.

Oops, I guess I made a newbie question, sorry!

Thank you very much for all your help!

:thanks:

[tebasuna51]

What decoder should I use to extract DTS Master Audio 2.0, 5.1, 6.1 and 7.1 now :confused: I was using ArcSoft DTS Decoder (1.1.0.0)

To extract don't need any decoder.

To decode DTS MA let the eac3to default libdcadec.dll.

[rivera]

I am confused, please help.

Tried with several different files - got same result.
After conversion mpg -> mkv, m2ts -> mkv, mediainfo does not contain any info about audio stream.
But audio stream can be played in MPC-HC.

Here is an example.
Input file:
General
ID : 1 (0x1)
Complete name : E:\Temp\pana (ac3).m2ts
Format : BDAV
Format/Info : Blu-ray Video
File size : 265 MiB
Duration : 4 min 6 s
Overall bit rate mode : Variable
Overall bit rate : 9 036 kb/s
Maximum Overall bit rate : 35.5 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 : Yes
Format settings, ReFrames : 3 frames
Codec ID : 27
Duration : 4 min 6 s
Bit rate mode : Variable
Bit rate : 8 327 kb/s
Maximum bit rate : 24.0 Mb/s
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16:9
Frame rate : 29.970 (29970/1000) FPS
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.134
Stream size : 247 MiB (93%)
Writing library : x264 core 61
Encoding settings : cabac=1 / ref=3 / deblock=1:0:0 / analyse=0x3:0x133 / me=umh / subme=6 / psy_rd=1.000000 / brdo=1 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=0 / 8x8dct=1 / cqm=0 / deadzone=21,11 / chroma_qp_offset=0 / threads=3 / nr=0 / decimate=1 / mbaff=0 / bframes=3 / b_pyramid=1 / b_adapt=1 / b_bias=0 / direct=3 / wpredb=1 / bime=1 / keyint=250 / keyint_min=25 / scenecut=40(pre) / rc=2pass / bitrate=8327 / ratetol=1.0 / rceq='blurCplx^(1-qComp)' / qcomp=1.00 / qpmin=10 / qpmax=51 / qpstep=4 / cplxblur=20.0 / qblur=0.5 / vbv_maxrate=24000 / vbv_bufsize=9000 / ip_ratio=1.40 / pb_ratio=1.30 / aq=2:1.00

Audio
ID : 4352 (0x1100)
Menu ID : 1 (0x1)
Format : AC-3
Format/Info : Audio Coding 3
Format settings, Endianness : Big
Codec ID : 129
Duration : 4 min 6 s
Bit rate mode : Constant
Bit rate : 256 kb/s
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 kHz
Frame rate : 31.250 FPS (1536 spf)
Bit depth : 16 bits
Compression mode : Lossy
Stream size : 7.51 MiB (3%)
Service kind : Complete Main

Usage:
eac3to "pana (ac3).m2ts" "pana (ac3).mkv"

Output file:
General
Unique ID : 47257860322188542931018185783133792477 (0x238D88678FA5573A5F64972DBB12E8DD)
Complete name : E:\Temp\pana (ac3).mkv
Format : Matroska
Format version : Version 1
File size : 244 MiB
Duration : 4 min 6 s
Overall bit rate mode : Variable
Overall bit rate : 8 325 kb/s
Encoded date : UTC 2017-04-03 08:27:59
Writing application : eac3to
Writing library : Haali DirectShow Matroska Muxer 1.13.138.14
ErrorDetectionType : Per level 1
CRC_Error_Pos : 231

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L4.1
Format settings, CABAC : Yes
Format settings, ReFrames : 3 frames
Muxing mode : Container profile=@0.0
Codec ID : V_MPEG4/ISO/AVC
Duration : 4 min 6 s
Bit rate mode : Variable
Bit rate : 8 327 kb/s
Maximum bit rate : 24.0 Mb/s
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 29.970 (29970/1000) FPS
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.134
Stream size : 240 MiB (98%)
Writing library : x264 core 61
Encoding settings : cabac=1 / ref=3 / deblock=1:0:0 / analyse=0x3:0x133 / me=umh / subme=6 / psy_rd=1.000000 / brdo=1 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=0 / 8x8dct=1 / cqm=0 / deadzone=21,11 / chroma_qp_offset=0 / threads=3 / nr=0 / decimate=1 / mbaff=0 / bframes=3 / b_pyramid=1 / b_adapt=1 / b_bias=0 / direct=3 / wpredb=1 / bime=1 / keyint=250 / keyint_min=25 / scenecut=40(pre) / rc=2pass / bitrate=8327 / ratetol=1.0 / rceq='blurCplx^(1-qComp)' / qcomp=1.00 / qpmin=10 / qpmax=51 / qpstep=4 / cplxblur=20.0 / qblur=0.5 / vbv_maxrate=24000 / vbv_bufsize=9000 / ip_ratio=1.40 / pb_ratio=1.30 / aq=2:1.00
Default : No
Forced : No

[LigH]

Is your MPC-HC set up to search external audio sources with the same file name? Check the filters submenu whether the audio source is the loaded file or an additional file.

[rivera]

Is your MPC-HC set up to search external audio sources with the same file name? Check the filters submenu whether the audio source is the loaded file or an additional file.
Audio source is INSIDE of the loaded file.

[tebasuna51]

Please put the eac3to log.

I can't reproduce your problem, my log:

eac3to v3.31
command line: eac3to "test.m2ts" "test.mkv"
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 0:00:21, 25p
1: h264/AVC, 720p25
2: AC3, 2.0 channels, 192kbps, 48kHz
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[v01] Muxing video to Matroska...
[a02] Creating file "test - 2 - AC3, 2.0 channels, 192kbps, 48kHz.ac3"...
Added fps value (25) to MKV header.
Video track 1 contains 525 frames.
eac3to processing took 1 second.
Done.

In my folder:
03/04/2017 11:29 1.050.624 test.m2ts
03/04/2017 11:30 528.384 test - 2 - AC3, 2.0 channels, 192kbps, 48kHz.ac3
03/04/2017 11:30 160.886 test.mkv
there are 2 files the .ac3 and the .mkv (by size is impossible than contain the audio also)

[rivera]

there are 2 files the .ac3 and the .mkv (by size is impossible than contain the audio also)
OMG, seems I got the point!
The output file does not contain audio stream, this stream is demuxed into .ac3 file!
I thought that "eac3to input.m2ts output.mkv" generates ONE file which includes both video & audio streams...

Now I understand why I was asked about external audio tracks in MPC-HC.

[Bandits]

Rogue One original disc M2TS shows 72p /1.001 and video track shows 24p/1.001

M2TS, 1 video track, 4 audio tracks, 5 subtitle tracks, 2:13:56, 72p /1.001
1: Chapters, 49 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
4: AC3, English, 2.0 channels, 320kbps, 48kHz
5: AC3, French, 5.1 channels, 640kbps, 48kHz
6: AC3, Spanish, 5.1 channels, 640kbps, 48kHz
7: Subtitle (PGS), English
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), French
11: Subtitle (PGS), Spanish
v02 The video bitstream framerate field doesn't seem to match the timestamps.

Is there going to be any issues with extracting these tracks? Or anything special I should be doing?

[tebasuna51]

Extract as always and put the log. Seems a eac3to mistake.

[XadoX]

Is there a way while using windows PowerShell to output the eac3to options as list, like usualy with cmd?

[LigH]

You mean something similar to "eac3to > eac3to.txt"? This never really worked well, due to the strange output formatting used. I never understood why it is necessary to prepend every output line with 80 backspace chars. As if a CRLF is not enough to start at the left margin.

[XadoX]

I only ask because in Windows 10 1703 CMD is defaulty replaced by PowerShell.
While using Powershell with eac3to the output is not shown as list. Instead every output is shown only in the first line.
So I have to use a pipe like you explained and thats not realy comfortable.

[Brazil2]

I only ask because in Windows 10 1703 CMD is defaulty replaced by PowerShell.
https://www.tenforums.com/tutorials/22882-show-command-prompt-windows-powershell-win-x-menu-windows-10-a.html

;)

[LigH]

Additionally, "Take Command" by JP Software (former 4DOS / 4NT) is an enhanced CMD-like command interpreter (Lite version is free to use).

It saved my day by avoiding surprises when comparing both LFN and SFN against wildcard patterns, matching more files than desired...

[Brazil2]

Additionally, "Take Command" by JP Software (former 4DOS / 4NT) is an enhanced CMD-like command interpreter (Lite version is free to use).
While you're at it, I like ColorConsole:
http://www.softwareok.com/?seite=Freeware/ColorConsole

[r0lZ]

I recommend to redirect the eac3to help to a file, and load the file in a good text editor to remove the BS characters. Due to the extremely bizarre formatting options of eac3to, it's the only method that gives good results.

[LigH]

@ Brazil2: But this still lets CMD interpret the commands.

[Brazil2]

@ Brazil2: But this still lets CMD interpret the commands.
I had the feeling that's what XadoX wanted: to go back to CMD.

[Reino]

Is there a way while using windows PowerShell to output the eac3to options as list, like usualy with cmd?I have very little experience with PowerShell, but I believe it can on its own, yes.
Cmd on the other hand doesn't have a replace-tool, so you'd have to use 3rd party software.
My favorite (http://videlibri.sourceforge.net/xidel.html):
eac3to.exe | xidel.exe -s - -e "replace($raw,'[\b]','','!')" | clip
or
eac3to.exe | xidel.exe -s - -e "replace($raw,'\x08','','!')" | clip
Or Cygwin's sed.exe (http://cygwin.org):
eac3to.exe | sed.exe "s/\x08//g" | clip

[iSeries]

Hi, I was under the impression that when converting a 7.1 track to 5.1, eac3to would automatically run a second pass and reduce the volume if clipping was detected. So I just ran this conversion and eac3to didn't generate a second pass. As a test, I ran the same command with -normalize added, and eac3to applied a -0.09db gain, so in other words, there was clipping. So am I supposed to add -normalize to every 7.1 to 5.1 downmix?

[LigH]

I can't tell you for sure, but I guess there may be circumstances where eac3to will only extract a 5.1 core and simply omit additional channels. Or it was just matrix encoded, no discrete 7.1 source anyway. You will have to specify more precisely which source format you have and which parameter set you used.

[iSeries]

I can't tell you for sure, but I guess there may be circumstances where eac3to will only extract a 5.1 core and simply omit additional channels. Or it was just matrix encoded, no discrete 7.1 source anyway. You will have to specify more precisely which source format you have and which parameter set you used.

Source was DTS-HD MA 7.1 (Zootopia). I just used the stream extractor gui for eac3to, selected the audio track and selected AC3 for output (which would default to a 5.1 640kbps output). The log in both cases say the additional channels were mixed into the surround channels.

[LigH]

In this case, I guess a conservative fixed downmix gain was used. Ensures no clipping, but may not utilize the whole amplitude range. Correct me if I'm wrong...

[iSeries]

In this case, I guess a conservative fixed downmix gain was used. Ensures no clipping, but may not utilize the whole amplitude range. Correct me if I'm wrong...

I just ran the same tests on a completely different movie (but still DTS-HD MA 7.1), and again no messages about clipping. The second test with -normalize added, eac3to again applied a -0.09 gain.

So I guess eac3to just applies a standard -0.09 gain when downmixing 7.1 to 5.1 when the -normalize switch is used, and no clipping is detected?

So now of course I'm wondering, is using the -normalize switch good practice, or should I just let eac3to mix the surrounds and only automatically normalize if there is detected clipping?

[tebasuna51]

Hi, I was under the impression that when converting a 7.1 track to 5.1, eac3to would automatically run a second pass and reduce the volume if clipping was detected.
Correct.
So I just ran this conversion and eac3to didn't generate a second pass. As a test, I ran the same command with -normalize added, and eac3to applied a -0.09db gain, so in other words, there was clipping.

A -0.09db gain can be considered like clipping, because LigH is right when say:
In this case, I guess a conservative fixed downmix gain was used. Ensures no clipping, but may not utilize the whole amplitude range.

So am I supposed to add -normalize to every 7.1 to 5.1 downmix?

Is not recommended.

Even, if the automatic normalize show a big attenuation value, you can consider use a special downmix (MeGUI or BeHappy) than preserve volume in FL,FR,FC and LFE, and only apply attenuation to mix SL+BL, SR+BR.

[iSeries]

Correct.


A -0.09db gain can be considered like clipping, because LigH is right when say:




Is not recommended.

Even, if the automatic normalize show a big attenuation value, you can consider use a special downmix (MeGUI or BeHappy) than preserve volume in FL,FR,FC and LFE, and only apply attenuation to mix SL+BL, SR+BR.

Perfect, thanks guys.

[SquallMX]

Is there a way to avoid the re-encoding of the AC3 embedded track when processing a segmented movie?

[a03]eac3to v3.31
command line: "D:\Archivos de Programa\EAC32AC3\eac3to.exe" "I:\TestTHD\BDMV\PLAYLIST\00620.mpls" 1) 3: "D:\\00620.mpls_3eng.thd+ac3" -progressnumbers -log="D:\Archivos de Programa\EAC32AC3\UsEac3To.log"
------------------------------------------------------------------------------
M2TS, 1 video track, 6 audio tracks, 4 subtitle tracks, 2:07:47, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3 (Atmos), English, 7.1 channels, 48kHz
(embedded: AC3 EX, 5.1 channels, 640kbps, 48kHz)
4: AC3, Spanish, 5.1 channels, 640kbps, 48kHz, -9ms
5: AC3, French, 5.1 channels, 640kbps, 48kHz, -9ms
6: AC3 Surround, English, 2.0 channels, 224kbps, 48kHz, -9ms
7: AC3 Surround, English, 2.0 channels, 224kbps, 48kHz, -9ms
8: AC3 Surround, English, 2.0 channels, 224kbps, 48kHz, -9ms
9: Subtitle (PGS), English
10: Subtitle (PGS), English
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), French
[a03] AC3 encoding doesn't support back channels. Will mix them into the surround.
[a03] Extracting audio track number 3...
[a03] Extracting audio track number 3...
[a03] Extracting TrueHD stream...
[a03] Extracting TrueHD stream...
[a03] Decoding with libav/ffmpeg...
[a03] Mixing surround channels...
[a03] Remapping channels...
[a03] Encoding AC3 <640kbps> with libAften...
[a03] Creating file "D:\\00620.mpls_3eng.thd+ac3"...
[a03] Audio overlaps for 5ms at playtime 0:15:21. <WARNING>
[a03] Audio overlaps for 5ms at playtime 0:48:26. <WARNING>
[a03] The audio gaps/overlaps can't be removed from the TrueHD bitstream. <WARNING>
[a03] In order to remove them you'll have to transcode to another format. <WARNING>
[a03] Original audio track, L: max 23 bits, average 19 bits.
[a03] Original audio track, R+BL+BR: max 24 bits, average 18 bits.
[a03] Original audio track, C: max 22 bits, average 19 bits.
[a03] Original audio track, LFE: constant bit depth of 18 bits.
[a03] Original audio track, SL: max 21 bits, average 18 bits.
[a03] Original audio track, SR: max 20 bits, average 18 bits.
[a03] Processed audio track, L: max 23 bits, average 19 bits.
[a03] Processed audio track, R+SL+SR: max 24 bits, average 19 bits.
[a03] Processed audio track, C: max 22 bits, average 19 bits.
[a03] Processed audio track, LFE: constant bit depth of 18 bits.
Video track 2 contains 183851 frames.
eac3to processing took 6 minutes, 58 seconds.
Done.

[rhaz]

Hi, what's the command to disable the notification 'boink' sound? If you run some batch file with 10 encodes, it boinks 10 times for each which drives me crazy while wearing headphones.

[Sparktank]

Hi, what's the command to disable the notification 'boink' sound? If you run some batch file with 10 encodes, it boinks 10 times for each which drives me crazy while wearing headphones.

You can delete the WAV file.
it doesn't crash if there's no WAV file.

[rhaz]

Hi, what's the command to disable the notification 'boink' sound? If you run some batch file with 10 encodes, it boinks 10 times for each which drives me crazy while wearing headphones.

You can delete the WAV file.
it doesn't crash if there's no WAV file.

Uhhh okay...

[Sparktank]

Uhhh okay...

That is, I don't believe there is a "command" to disable the notification. It's just designed that way.

Hence: Delete the WAV file in the eac3to folder.
No WAV file, no sound.
It doesn't break anything if you delete it.

Try looking inside the folder with eac3to, or whatever GUI/AiO you are using and delete the WAV files.
They're not really needed.

[Asmodian]

I do that right away, after the first success sound reminds me I rename the wavs to nos. It is actually a nice and easy way to configure that feature. :)

[stax76]

Looks like eac3to writes chapters to m4a or is it created by NeroAacEnc?

-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Audio encoding using eac3to 3.31 x86
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

D:\Projekte\VS\VB\StaxRip\bin\Apps\eac3to\eac3to.exe "D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2 44ms.flac" "D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a" -quality=0.35 -normalize +44ms -progressnumbers

FLAC, 2.0 channels, 0:00:41, 24 bits, 827kbps, 44.1kHz
Decoding FLAC...
Applying RAW/PCM delay...
Writing WAV...
Creating file "D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a.pass1.wav"...
The original audio track has a constant bit depth of 24 bits.
Starting 2nd pass...
Reading WAV...
Reducing depth from 64 to 32 bits...
Encoding AAC <0.35> with NeroAacEnc...
Applying 0.05dB gain...
The processed audio track has a constant bit depth of 32 bits.
eac3to processing took 1 second.
Done.

Start: 21:53:24
End: 21:53:25
Duration: 00:00:01

General
Complete name : D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a
Format : MPEG-4
Format profile : Base Media / Version 2
Codec ID : mp42 (M4A /mp42/isom)
File size : 318 KiB
Duration : 41 s 215 ms
Overall bit rate mode : Variable
Overall bit rate : 63.2 kb/s
Encoded date : UTC 2017-05-12 19:53:25
Tagged date : UTC 2017-05-12 19:53:25

Audio
ID : 1
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : LC
Codec ID : 40
Duration : 41 s 215 ms
Bit rate mode : Variable
Bit rate : 61.4 kb/s
Maximum bit rate : 70.0 kb/s
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 44.1 kHz
Frame rate : 43.066 FPS (1024 spf)
Compression mode : Lossy
Stream size : 309 KiB (97%)
Writing library : Nero AAC codec 1.5.4.0
Encoding settings : -q 0.35
Encoded date : UTC 2017-05-12 19:53:25
Tagged date : UTC 2017-05-12 19:53:25

Menu

-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Muxing using mkvmerge 11.0.0 pre x64
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

D:\Projekte\VS\VB\StaxRip\bin\Apps\MKVToolNix\mkvmerge.exe -o "D:\Video\Samples\MKV\720p SBR_new.mkv" "D:\Video\Samples\MKV\720p SBR_temp\720p SBR_new_out.h264" --audio-tracks 0 --language 0:ger --default-track 0:0 "D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a" --global-tags "D:\Video\Samples\MKV\720p SBR_temp\720p SBR_new_tags.xml" --ui-language en

mkvmerge v11.0.0 ('Alive') 64bit
'D:\Video\Samples\MKV\720p SBR_temp\720p SBR_new_out.h264': Using the demultiplexer for the format 'AVC/h.264'.
'D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a': Using the demultiplexer for the format 'QuickTime/MP4'.
'D:\Video\Samples\MKV\720p SBR_temp\720p SBR_new_out.h264' track 0: Using the output module for the format 'AVC/h.264 (unframed)'.
'D:\Video\Samples\MKV\720p SBR_temp\720p SBR ID2_out1.m4a' track 0: Using the output module for the format 'AAC'.
The file 'D:\Video\Samples\MKV\720p SBR_new.mkv' has been opened for writing.
The cue entries (the index) are being written...
Multiplexing took 0 seconds.

Start: 21:53:36
End: 21:53:37
Duration: 00:00:00

General
Complete name : D:\Video\Samples\MKV\720p SBR_new.mkv
Format : Matroska
Format version : Version 4 / Version 2
File size : 11.4 MiB
Duration : 41 s 156 ms
Overall bit rate : 2 324 kb/s
Encoded date : UTC 2017-05-12 19:53:36
Writing application : mkvmerge v11.0.0 ('Alive') 64bit
Writing library : libebml v1.3.4 + libmatroska v1.4.5

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : Baseline@L2.1
Format settings, CABAC : No
Format settings, ReFrames : 1 frame
Codec ID : V_MPEG4/ISO/AVC
Duration : 41 s 66 ms
Bit rate : 2 264 kb/s
Width : 480 pixels
Height : 272 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 30.000 FPS
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.578
Stream size : 11.1 MiB (97%)
Writing library : x264 core 148 r2762 90a61ec
Default : Yes
Forced : No

Audio
ID : 2
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : LC
Codec ID : A_AAC
Duration : 41 s 146 ms
Bit rate : 61.5 kb/s
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 44.1 kHz
Frame rate : 43.066 FPS (1024 spf)
Compression mode : Lossy
Delay relative to video : 10 ms
Stream size : 309 KiB (3%)
Language : German
Default : No
Forced : No

Menu
00 : 00:00.059 : en:00:00:00.059

[sneaker_ger]

Written by NeroAacEnc to signal AAC encoder delay.

[stax76]

And mkvmerge accounts this delay? It's shown by a player as chapters (Editions):

https://cloud.githubusercontent.com/assets/10564090/25983746/3bb263ce-36ac-11e7-921d-ba76906c2343.png

[sneaker_ger]

NeroAacEnc writes the delay info 2 times:
A) as chapter
B) as proprietary iTunSMPB tag

Mkvmerge reads and applies delay from B automatically. A is treated like a regular chapter. So probably best to set --no-chapters for these m4a files.

[stax76]

Thanks for the explanation, I'll use --no-chapters.

[robena]

When I use eac3 on Windows 7, every time it completes an operation, it makes a zipping sound.

I could not find it, is there a switch to turn that sound off?

[sneaker_ger]

Delete the .wav file(s) in the eac3to folder.

[robena]

Thanks, very neat!

[Thunderbolt8]

sometimes (or often) extracting .sup subtitles from remuxes results in subtitles which are comparably small in size and unusable when trying to open them with the usual programs to edit/ocr subtitles. when extracting these subs with gMKVExtractGUI they works fine though (and are bigger in size). it would be nice if this could be fixed.

[mkver]

sometimes (or often) extracting .sup subtitles from remuxes results in subtitles which are comparably small in size and unusable when trying to open them with the usual programs to edit/ocr subtitles. when extracting these subs with gMKVExtractGUI they works fine though (and are bigger in size). it would be nice if this could be fixed.
Seems that during extraction eac3to doesn't take into account whether the subtitles in the Matroska file are compressed (with zlib/deflate) and treats them as if they were uncompressed.
[Edit]: Looking at the result in a hex editor, I can confirm this.

[tebasuna51]

Yep, this is the problem.

When you mux the sup with MkvToolNix the default option (automatically) is compress it with zlib, if you select uncompressed after eac3to extract the sup correctly.

[Thunderbolt8]

so I guess this shouldnt be too hard to implement then...? or is it more difficult than it sounds?

[stax76]

It's probably not difficult but since there is mkvextract it probably has low priority...

[Thunderbolt8]

its not that there isnt another possibility to do this, its just very inconvenient when already working with it on a file and remuxing other streams from it and then having to use another tool afterwards again because the subs wont work otherwise.

e.g. with HEVC streams not working as well, when remuxing a file which contains HEVC video, audio with dialnorm and compressed subs, currently youd need either tsmuxer or mkvmerge for the HEVC video, eac3to to remove dialnorm from the audio stream and mkvextract for the compressed subtitle file, while ideally you could do all this just with one step using just eac3to.

[stax76]

Maybe there there are situations, I just can't think of any, I don't know which tools and formats you use.

[stax76]

It seems eac3to uses always 24 bit depth for flac output, could this be improved to automatically use only 16 bit depth if the input has only 16 bit? The resulting files are larger then I would decode to w64.

In staxrip currently I convert to flac using eac3to in case qaac cannot read the input format, is there a way I can make eac3to output 16 bit flac? If I remember right then -down16 don't help here, does it even make sense to convert to flac or should rather be w64 used?

The included flac lib is from 2007, I tried to replace it with a newer but it didn't work.

Which formats by extension typically use not more then 16 bit and typically work well with ffmpeg? I believe for many HD formats eac3to works better, for instance with atmos files ffmpeg often showed various errors and defaulted to 6 instead of 7 or 8 channels.

[sneaker_ger]

It seems eac3to uses always 24 bit depth for flac output, could this be improved to automatically use only 16 bit depth if the input has only 16 bit?
I believe eac3to already does that. :confused:
sample + log?

The resulting files are larger then I would decode to w64.
sample + log?

In staxrip currently I convert to flac using eac3to in case qaac cannot read the input format, is there a way I can make eac3to output 16 bit flac? If I remember right then -down16 don't help here
Why? That's the only purpose of the option.

does it even make sense to convert to flac or should rather be w64 used?
Do you want to save space? That's your answer.
Extra-note: eac3to always uses slowest FLAC compression. Probably a waste of CPU/time for intermediate files.

Note that many lossy formats do not have an inherent bitdepth. You can decode to 16, 24, 32, ... bit however you like. Decoding is not as 100% defined as is with e.g. H.264.

[stax76]

According to MediaInfo my AC3 test file is 16 bit, eac3to converts it to 24 bit FLAC, eac3to writes: Reducing depth from 64 to 24 bits...

-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Audio Source File MediaInfo
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

General
Complete name : D:\test.ac3
Format : AC-3
Format/Info : Audio Coding 3
File size : 5.33 MiB
Duration : 3 min 19 s
Overall bit rate mode : Constant
Overall bit rate : 224 kb/s

Audio
Format : AC-3
Format/Info : Audio Coding 3
Format settings, Endianness : Big
Duration : 3 min 19 s
Bit rate mode : Constant
Bit rate : 224 kb/s
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 kHz
Frame rate : 31.250 FPS (1536 spf)
Bit depth : 16 bits
Compression mode : Lossy
Stream size : 5.33 MiB (100%)
Service kind : Complete Main

-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Convert to FLAC using eac3to 3.31 x86
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

eac3to.exe D:\test.ac3 D:\test.flac -down2 -simple -progressnumbers

AC3, 2.0 channels, 0:03:19, 224kbps, 48kHz, dialnorm: -27dB
Removing AC3 dialog normalization...
Decoding with libav/ffmpeg...
Reducing depth from 64 to 24 bits...
Encoding FLAC with libFlac...
Creating file "D:\test.flac"...
eac3to processing took 7 seconds.
Done.

Start: 17:44:09
End: 17:44:17
Duration: 00:00:07

General
Complete name : D:\test.flac
Format : FLAC
Format/Info : Free Lossless Audio Codec
File size : 36.0 MiB
Duration : 3 min 19 s
Overall bit rate mode : Variable
Overall bit rate : 1 516 kb/s

Audio
Format : FLAC
Format/Info : Free Lossless Audio Codec
Duration : 3 min 19 s
Bit rate mode : Variable
Bit rate : 1 516 kb/s
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 kHz
Bit depth : 24 bits
Stream size : 36.0 MiB (100%)
Writing library : libFLAC 1.2.1 (UTC 2007-09-17)

[sneaker_ger]

I don't know why MediaInfo says so. eac3to is working as designed.
https://forum.doom9.org/showthread.php?t=145096

[stax76]

Thanks, it's clear now.

[tebasuna51]

I make a request to MediaInfo to don't show Bit depth in lossy formats, without sucess.

[stax76]

I'll hide it then in case CompressionMode returns Lossy.

[Thunderbolt8]

eac3to also doesnt work with UHD playlists. has there been a slight change of code with UHD BDs so that these playlist cannot be detected at all currently?

[73ChargerFan]

v3.31 is now 20 months old.

[mkver]

eac3to also doesnt work with UHD playlists. has there been a slight change of code with UHD BDs so that these playlist cannot be detected at all currently?
It seems that these playlists contain a version field that differs for UHD-BDs from the version used on ordinary BDs. Therefore it might possibly be a simple version check failure.
Disclaimer: I am just reporting what I read here (https://github.com/mbunkus/mkvtoolnix/issues/2010) and have no experience with UHD-BDs. It is also said that tsmuxer is not strict regarding the version check.

[stax76]

Hi, these are two errors I've seen lately:

Error Demux M2TS using eac3to 3.31 x86

Demux M2TS using eac3to 3.31 x86 failed with exit code: 1 (0x1)

The exit code might be a system error code: STATUS_WAIT_1

The exit code might be a system error code: 函数不正确。

M2TS, 6 audio tracks, 4 subtitle tracks, 2:07:56
1: TrueHD/AC3 (Atmos), 7.1 channels, 48kHz
(embedded: AC3 EX, 5.1 channels, 640kbps, 48kHz)
2: AC3, 5.1 channels, 640kbps, 48kHz, 1ms
3: AC3, 5.1 channels, 640kbps, 48kHz, 1ms
4: AC3 Surround, 2.0 channels, 224kbps, 48kHz, 1ms
5: AC3 Surround, 2.0 channels, 224kbps, 48kHz, 1ms
6: AC3 Surround, 2.0 channels, 224kbps, 48kHz, 1ms
7: Subtitle (PGS)
8: Subtitle (PGS)
9: Subtitle (PGS)
10: Subtitle (PGS)
Track 1 is used for destination file "00000 ID1.thd+ac3 -progressnumbers".
This audio conversion is not supported.

eac3to v3.31
command line: "D:\Projekte\VS\VB\StaxRip\bin\Apps\eac3to\eac3to.exe" "E:\BDMV\PLAYLIST" 1) 2: D:\Temp\StaxRip\BD\eac3to\eac3to.mkv 3: "D:\Temp\StaxRip\BD\eac3to\eac3to ID3 English.eac3" 4: "D:\Temp\StaxRip\BD\eac3to\eac3to ID4 English.ac3" 5: "D:\Temp\StaxRip\BD\eac3to\eac3to ID5 German.thd+ac3" 6: "D:\Temp\StaxRip\BD\eac3to\eac3to ID6 German.dtsma" 7: "D:\Temp\StaxRip\BD\eac3to\eac3to ID7 English.ac3" 1: "D:\Temp\StaxRip\BD\eac3to\eac3to_chapters.txt" 8: "D:\Temp\StaxRip\BD\eac3to\eac3to ID8 English.sup" 9: "D:\Temp\StaxRip\BD\eac3to\eac3to ID9 German.sup" 10: "D:\Temp\StaxRip\BD\eac3to\eac3to ID10 German.sup" -progressnumbers
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 3 subtitle tracks, 2:08:52, 24p /1.001
1: Chapters, 14 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: E-AC3 EX, English, 7.1 channels, 1512kbps, 48kHz, dialnorm: -27dB
(core: AC3 EX, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB)
4: AC3, English, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
5: TrueHD/AC3 (Atmos), German, 7.1 channels, 48kHz
(embedded: AC3 EX, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB)
6: DTS Master Audio, German, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
7: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
8: Subtitle (PGS), English
9: Subtitle (PGS), German
10: Subtitle (PGS), German
Creating file "D:\Temp\StaxRip\BD\eac3to\eac3to_chapters.txt"...
[s10] Extracting subtitle track number 10...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Removing AC3 dialog normalization...
[a03] Applying (E-)AC3 delay failed. <ERROR>
[a05] Extracting audio track number 5...
[a06] Extracting audio track number 6...
[a05] Removing AC3 dialog normalization...
[a04] Extracting audio track number 4...
[a04] Removing AC3 dialog normalization...
[s08] Extracting subtitle track number 8...
[s09] Extracting subtitle track number 9...
[a07] Extracting audio track number 7...
[a07] Removing AC3 dialog normalization...
[v02] Muxing video to Matroska...
Aborted at file position 1048576. <ERROR>

Replacing Haali with ffmpeg and supporting mp4 input would be great.

General

ID/String : 0 (0x0)
CompleteName : E:\BDMV\STREAM\00084.m2ts
Format : BDAV
Format/Info : Blu-ray Video
FileSize/String : 32.7 GiB
Duration/String : 2 h 8 min
OverallBitRate_Mode/String: Variable
OverallBitRate/String : 36.3 Mb/s
OverallBitRate_Maximum/String: 48.0 Mb/s

Video

ID/String : 4113 (0x1011)
MenuID/String : 1 (0x1)
Format : AVC
Format/Info : Advanced Video Codec
Format_Profile : High@L4.1
Format_Settings_CABAC/String: Yes
Format_Settings_RefFrames/String: 4 frames
CodecID : 27
Duration/String : 2 h 8 min
BitRate_Mode/String : Variable
BitRate_Maximum/String : 30.0 Mb/s
Width/String : 1 920 pixels
Height/String : 1 080 pixels
DisplayAspectRatio/String : 16:9
FrameRate/String : 23.976 (24000/1001) FPS
ColorSpace : YUV
ChromaSubsampling/String : 4:2:0
BitDepth/String : 8 bits
ScanType/String : Progressive

Audio #1

ID/String : 4352 (0x1100)
MenuID/String : 1 (0x1)
Format : E-AC-3
Format/Info : Enhanced Audio Coding 3
Format_Profile : E-AC-3+Atmos / E-AC-3
Format_Settings_Endianness: Big
MuxingMode : Stream extension
CodecID : 132
Duration/String : 2 h 8 min
BitRate_Mode/String : Constant
BitRate/String : 500 b/s
Channel(s)/String : 11 objects / 6 channels
ChannelPositions : 11 objects / Front: L C R, Side: L R, LFE
SamplingRate/String : 48.0 kHz
FrameRate/String : 187.500 FPS (256 spf)
Compression_Mode/String : Lossy
StreamSize/String : 472 KiB (0%)
ServiceKind/String : Complete Main

Audio #2

ID/String : 4353 (0x1101)
MenuID/String : 1 (0x1)
Format : AC-3
Format/Info : Audio Coding 3
Format_Settings_Endianness: Big
CodecID : 129
Duration/String : 2 h 8 min
BitRate_Mode/String : Constant
BitRate/String : 448 kb/s
Channel(s)/String : 6 channels
ChannelPositions : Front: L C R, Side: L R, LFE
SamplingRate/String : 48.0 kHz
FrameRate/String : 31.250 FPS (1536 spf)
BitDepth/String : 16 bits
Compression_Mode/String : Lossy
StreamSize/String : 413 MiB (1%)
ServiceKind/String : Complete Main

Audio #3

ID/String : 4354 (0x1102)
MenuID/String : 1 (0x1)
Format : TrueHD / AC-3
Format_Profile : TrueHD+Atmos / TrueHD / AC-3
Format_Settings_Endianness: Big
MuxingMode : Stream extension
CodecID : 131
Duration/String : 2 h 8 min
BitRate_Mode/String : Variable / Constant
BitRate/String : 448 kb/s
BitRate_Maximum/String : 7 737 kb/s
Channel(s)/String : Object Based / 8 channels / 6 channels
ChannelPositions : Object Based / Front: L C R, Side: L R, Back: L R, LFE / Front: L C R, Side: L R, LFE
SamplingRate/String : / 48.0 kHz / 48.0 kHz
FrameRate/String : 31.250 FPS (1536 spf)
BitDepth/String : 16 bits
StreamSize/String : 413 MiB (1%)
ServiceKind/String : Complete Main

Audio #4

ID/String : 4355 (0x1103)
MenuID/String : 1 (0x1)
Format : DTS
Format/Info : Digital Theater Systems
Format_Profile : MA / Core
Format_Settings_Mode : 16
Format_Settings_Endianness: Big
MuxingMode : Stream extension
CodecID : 134
Duration/String : 2 h 8 min
BitRate_Mode/String : Variable / Constant
BitRate/String : Unknown / 1 509 kb/s
Channel(s)/String : 6 channels
ChannelPositions : Front: L C R, Side: L R, LFE
SamplingRate/String : 48.0 kHz
FrameRate/String : 93.750 FPS (512 spf)
BitDepth/String : 16 bits
Compression_Mode/String : Lossless / Lossy

Audio #5

ID/String : 4356 (0x1104)
MenuID/String : 1 (0x1)
Format : AC-3
Format/Info : Audio Coding 3
Format_Settings_Mode : Dolby Digital
Format_Settings_Endianness: Big
CodecID : 129
Duration/String : 2 h 8 min
BitRate_Mode/String : Constant
BitRate/String : 192 kb/s
Channel(s)/String : 2 channels
ChannelPositions : Front: L R
SamplingRate/String : 48.0 kHz
FrameRate/String : 31.250 FPS (1536 spf)
BitDepth/String : 16 bits
Compression_Mode/String : Lossy
StreamSize/String : 177 MiB (1%)
ServiceKind/String : Complete Main

Text #1

ID/String : 4608 (0x1200)
MenuID/String : 1 (0x1)
Format : PGS
CodecID : 144

Text #2

ID/String : 4609 (0x1201)
MenuID/String : 1 (0x1)
Format : PGS
CodecID : 144

Text #3

ID/String : 4610 (0x1202)
MenuID/String : 1 (0x1)
Format : PGS
CodecID : 144

[madshi]

For the first problem it might make sense to create a bug report here with a little (zipped) sample file:

http://eac3to.bugs.madshi.net

[hubblec4]

Hi madshi

Could you imagine update eac3to to support UHD-Bluray?

[madshi]

I would like to, but I have no UHD Blu-Ray specs. I'm not sure if the playlist format or anything else has changed. If anybody has UHD Blu-Ray specs available, please PM me, of course I'll keep everything confidential.

[tebasuna51]

Maybe here (https://forum.doom9.org/showthread.php?p=1806590#post1806590)?

[madshi]

Already have that one. It's just a white paper with 49 pages. The full 1080p Blu-Ray spec with all the technical detail has almost 1500 pages.

[hubblec4]

I'm not sure if the playlist format or anything else has changed.

Yes somethings has changed. The version number is 0300 now.
And the language position seems different (eac3to don't find language names for the streams), the video stream is not recognized, so far my tests).


If anybody has UHD Blu-Ray specs available, please PM me, of course I'll keep everything confidential.

Me too, please.

[MrVideo]

I'm trying to extract the AC3 core from a MKV video that has an H.264 video, the EAC3 audio stream and a subtitle stream.

But, when I do:
eac3to inputvideo.mkv outputaudio.ac3
eac3to identifies the 2nd stream to be E-AC3, but errors out saying that "This audio conversion is not supported." But it is implied that it can be done because E-AC3 is the input source and AC3 is the output.

Am I doing something wrong?

[LigH]

There are possibly some variants of the E-AC3 format. I guess not all of them have a legacy AC3 core to be extracted losslessly. It may also require the -core option. To help you better: Always quote your full console output / logfile when any error message appears (best in a CODE block). In addition, a MediaInfo analysis can't be wrong.

[MrVideo]

You asked, you shall receive (and I know how to use code blocks :D )
eac3to v3.31
command line: C:\cygwin\home\Vidiot\bin\eac3to\eac3to.exe input_eac3_video.mkv output.ac3 -core
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 1 subtitle track, 0:43:13, 24p /1.001
1: h264/AVC, English, 1080p24 /1.001 (16:9)
2: EAC3, English, 5.1 channels, 48kHz
3: Subtitle (SRT), English
Bitstream parsing for track 2 failed. <WARNING>
Demuxing this track may still produce correct results - or not. <WARNING>
Track 2 is used for destination file "output.ac3".
This audio conversion is not supported. <ERROR>
General
Complete name : F:\input_eac3_video.mkv
Format : Matroska
File size : 4.05 GiB
Duration : 43mn 13s
Overall bit rate : 13.4 Mbps
Encoded date : UTC 2016-08-29 13:09:36
Writing application : mkvmerge v7.9.0 ('Birds') 32bit
Writing library : libebml v1.3.1 + libmatroska v1.4.2

Video
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L4.0
Format settings, CABAC : Yes
Format settings, ReFrames : 4 frames
Muxing mode : Container profile=Unknown@4.0
Codec ID : V_MPEG4/ISO/AVC
Duration : 43mn 10s
Bit rate : 12.2 Mbps
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16/9
Frame rate : 23.976 fps
Resolution : 24 bits
Colorimetry : 4:2:0
Scan type : Progressive
Bits/(Pixel*Frame) : 0.246
Writing library : x264 core 148 r2623 d5b2374
Encoding settings : cabac=1 / ref=4 / deblock=1:-1:-1 /
analyse=0x3:0x133 / me=umh / subme=9 / 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=34 / lookahead_threads=5 /
sliced_threads=0 / nr=150 / decimate=1 / interlaced=0 /
bluray_compat=0 / constrained_intra=0 / bframes=8 / b_pyramid=2 /
b_adapt=2 / b_bias=0 / direct=3 / weightb=1 / open_gop=0 /
weightp=2 / keyint=120 / keyint_min=61 / scenecut=0 /
intra_refresh=0 / rc_lookahead=120 / rc=crf / mbtree=1 / crf=18.0 /
qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 /
vbv_maxrate=15000 / vbv_bufsize=31250 / crf_max=0.0 /
nal_hrd=vbr / filler=0 / ip_ratio=1.40 / aq=3:1.00

Audio
Format : E-AC-3
Format/Info : Audio Coding 3
Codec ID : A_EAC3
Duration : 43mn 13s
Bit rate mode : Constant
Bit rate : 640 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Surround: L R, LFE
Sampling rate : 48.0 KHz

Text
Format : UTF-8
Codec ID : S_TEXT/UTF8
Codec ID/Info : UTF-8 Plain Text

[LigH]

This appears like legacy AC3 to me, with 640 kbps and typical 5.1 channel layout. I would recommend a dedicated MKV demultiplexer here (MKVToolnixGUI / mkvextract + optional UI). Extracting from MKV may not be the primary feature of eac3to, in relation to other source media formats.

But I may be wrong. Other users may have more experience here.

BTW, is there the same error without the "-core" parameter? Reading the help output again, it only mentions DTS...

[MrVideo]

Oops, I think you have misunderstood what I am trying to do. I have no problem extracting the video or audio streams from the MKV file. What I am trying to do is extract the AC3 core from the EAC3 audio stream. I'm trying to keep from having to recode the EAC3 to AC3. Recoding is not good.

[Q-the-STORM]

There is no ac3 core in this eac3...

[MrVideo]

There is no ac3 core in this eac3...
Then by definition, it can't be EAC3. I read the paper that Dolby Labs released as a PDF and EAC3 is supposed to be for adding 2 surround channels, making it 7.1. For compatibility, there is supposed to be a core of the 5.1. When played back, if the amp doesn't know EAC3, the packets for the added 2 channels are ignored and only the core 5.1 is played. If the amp knows EAC3, then the 2 surround channels of the core are thrown out and the 4 channels of the EAC3 packets are added to the L/C/R of the core and played.

The paper that I read makes it sound like EAC3 with only 5.1 is not supposed to be possible, yet here it is. Streaming services like Amazon and Netflix are expanding EAC3 use, but only seem to have 5.1. Advantage unknown.

The paper says nothing about having the L/C/R also in the EAC3 added packets.

Color me totally confused.

I used tsmuxer to extract the EAC3 stream from the MKV file. I then fed that single stream to eac3to and that worked, sortof. I mean that eac3to no longer complains about it not being supported, but it recoded the audio with libaften, which I did not want to happen.

I want to get the EAC3 5.1 turned into AC3 5.1, without recoding. Seems like an impossible task.

[LigH]

AFAIR, there are different formats of E-AC3: for Blu-ray, it's used to add extra channels to a legacy AC3 core; but there may also be "native E-AC3" which contains all channels in E-AC3 packets, without any legacy AC3 packets to add to (just E-AC3 packets having a different header than legacy AC3 packets, but probably the same data content?). If this is true, then I guess it could be repacked to legacy AC3 format if there was any designated packet converter.

[nevcairiel]

Then by definition, it can't be EAC3. I read the paper that Dolby Labs released as a PDF and EAC3 is supposed to be for adding 2 surround channels, making it 7.1. For compatibility, there is supposed to be a core of the 5.1.

This only applies to Blu-ray streams. You can also encode pure EAC3, which is used in broadcast in some countries, and probably various other sources as well (streaming, I guess?).

The specification of EAC3 technically includes a method for converting EAC3 to AC3 (for legacy SPDIF transmission), however I do not know if any software implements that.

[Q-the-STORM]

It's more complicated than that...
There is no AC3 core in this EAC3, but legacy decoders can decode parts of EAC3...

EAC3 is build in a way that that legacy decoders can decode some of the data to 640kbps AC3, which is what e.g. a old BD player or PS4 does when you are streaming netflix. While audio you get from netflix is 192kbps eac3, what your receiver gets is 640kbps ac3, which is converted by the AC3 decoder...

the way I understood it (from reading the white papers), is that it isn't actually decoding and reencoding eac3 (which obviously it can't), it converts the encoded eac3 stream directly to AC3 and losing some data (like the 2 extra channels maybe some frequencies depending on source bitrate, etc.) I have never come across any other codec that works like that, which is why it's so confusing... But your file doesn't have a core like DTS does...

there is no way to get AC3 from this file without reencoding...

[LigH]

So there are also differences in the format of the "payload" between an E-AC3 and a legacy AC3 packet? ... Well, then you will probably have to add a parameter like "-640" to convert with recoding.

[MrVideo]

there is no way to get AC3 from this file without reencoding...
Crap, that sucks.

Thanks to all for posting this morning (for me anyway). I've learned a lot this morning.

[tebasuna51]

Even eac3to have problems with eac3 in mkv container (not in m2ts).
And default Aften encoder is deprecated (https://forum.doom9.org/showthread.php?p=1794293#post1794293).

The best way I know to recode is:

FFMPEG -drc_scale 0 -i "input.mkv" -vn -acodec ac3 -ac 6 -ab 640k -center_mixlev 0.707 "output.ac3"

or the desired bitrate
or -map 0:2 instead -vn if is the second audio track in mkv, or nothing if input.eac3

[Ripman]

Is there an option that can be passed on the command line to tell whether a particular constant 24bit input audio file contains only 16bits of audio data - that there are "superfluous zero bytes"?

eac3to will remove these "superfluous zero bytes" using a second pass - a great feature. And I am wondering if there is an option to tell whether there are "superfluous zero bytes", but without taking the action of actually removing the "superfluous zero bytes".

Here is an example log:

eac3to v3.31
command line: eac3to "02_Schumann-Violin-Sonatas-Op-105-and-Op-121.flac" "..\wav\02_Schumann-Violin-Sonatas-Op-105-and-Op-121.wav"
------------------------------------------------------------------------------
FLAC, 2.0 channels, 0:03:26, 24 bits, 2640kbps, 352.8kHz
Decoding FLAC...
Writing WAV...
Creating file "..\wav\02_Schumann-Violin-Sonatas-Op-105-and-Op-121.wav"...
The original audio track has a constant bit depth of 16 bits.
Superfluous zero bytes detected, will be stripped in 2nd pass.
Starting 2nd pass...
Reading WAV...
Stripping zero bytes...
Writing WAV...
Creating file "..\wav\02_Schumann-Violin-Sonatas-Op-105-and-Op-121.wav"...
eac3to processing took 23 seconds.
Done.

[Q-the-STORM]

-24 should encode with 24 bitdepth, but I don't know if the "superfluous zero bytes" message comes up, you'd need to try it yourself...

[Ripman]

-24 should encode with 24 bitdepth, but I don't know if the "superfluous zero bytes" message comes up, you'd need to try it yourself...

It'd be more of a "stat" type of function ... the implication being that you need to go through the file anyway to determine that it doesn't have anything over 16its.

I can process something like the log I posted, and then use an re to match the string with the message. -- I was looking for something more precise.

Btw, here is an example of such a file - 24bit/352.8/2ch flac. I did a post maybe a year ago where I had a problem with 352khz files.
https://www.dropbox.com/s/uuhv97hzo25feq1/11_Schumann-Violin-Sonatas-Op-105-and-Op-121_2chDXD.flac?dl=1

[Snowknight26]

Is there an option that can be passed on the command line to tell whether a particular constant 24bit input audio file contains only 16bits of audio data - that there are "superfluous zero bytes"?
You have to decode the whole audio track, just like you did.

eac3to will remove these "superfluous zero bytes" using a second pass - a great feature. And I am wondering if there is an option to tell whether there are "superfluous zero bytes", but without taking the action of actually removing the "superfluous zero bytes".


-no2ndpass, stops the 2nd pass from occurring, but what's your goal behind all this anyway?


-24 should encode with 24 bitdepth, but I don't know if the "superfluous zero bytes" message comes up, you'd need to try it yourself...

-24 defines a PCM input track as 24-bit in case eac3to can't automatically detect it. You're thinking of -down24, which reduces the output bit-depth to 24.

[Q-the-STORM]

-24 defines a PCM input track as 24-bit in case eac3to can't automatically detect it. You're thinking of -down24, which reduces the output bit-depth to 24.

yeah, I meant -down24

[Music Fan]

@ MrVideo : try to keep your eac3 track as is before to recode it, it will perhaps be compliant with your decoder, even if you wanna put it on a dvd-video.
I know one example of commercial dvd with ac3 @ 640 kbps : Pink Floyd, Pulse (at least the european edition). It contains 2 ac3 tracks, one @ 640 kbps, the other @ 448 kbps. And it's decoded by my equipment.
But this is probably a different kind of encoding, a "simple" ac3. Yours was maybe encoded with an eac3 encoder, unless there's a header misinterpretation of your file wich is maybe also a "simple" ac3.

[MrVideo]

The best way I know to recode is:

FFMPEG -drc_scale 0 -i "input.mkv" -vn -acodec ac3 -ac 6 -ab 640k - 0.707 "output.ac3"
Thanks. I will put that into my scripts.

Why use -drc_scale if it is set to 0? Isn't that the default?

If the audio being recoded is from EAC3, should the -center_mixlev not be used, as it should already be that from the time it was initially encoded?

[tebasuna51]

eac3to will remove these "superfluous zero bytes" using a second pass - a great feature. And I am wondering if there is an option to tell whether there are "superfluous zero bytes", but without taking the action of actually removing the "superfluous zero bytes".


That is the correct answer:


-no2ndpass, stops the 2nd pass from occurring, but what's your goal behind all this anyway?

And the correct question also.

[tebasuna51]

Why use -drc_scale if it is set to 0? Isn't that the default?

The default decoding is apply the Dynamic Range Compression (DRC) found in the stream (attenuate volume), set to 0 cancel that action.

If the audio being recoded is from EAC3, should the -center_mixlev not be used, as it should already be that from the time it was initially encoded?

The center-mixlevel is a metadata present in AC3 5.1 header to be used when the 5.1 is downmixed to 2.0.

The ffmpeg default put -4.5 dB instead -3 dB and I recommend use the big value to avoid low dialog volume problem. Other encoders use -3 dB by default, I don't know for what ffmpeg put -4.5.

[Ripman]

-no2ndpass, stops the 2nd pass from occurring, but what's your goal behind all this anyway?

I didn't know about the -no2ndpass -- it doesn't show on the options screen (or even page 1 of this post). Anyway, using that opt prevents the 2nd pass.

Goal? --- Get my money back from the site that sold the file as 24bit. :devil: (The idea would be to use eac3to to confirm bit-depth before processing any files I purchase online. In the case of the 16bit-352.8khz flac I posted, I imagine that the producing firm had an encoder option set incorrectly.)

[Music Fan]

Thanks. I will put that into my scripts.
Did you see my post ?

[MrVideo]

(snip)
Thanks for the info. Mucho appreciated.

[MrVideo]

@ MrVideo : try to keep your eac3 track as is before to recode it, it will perhaps be compliant with your decoder, even if you wanna put it on a dvd-video.
Sorry, missed this post at first. I don't do DVD-videos. I do Blu-ray. My amp has no issue with the variant EAC3 stream.

The program I use is BDRebuilder and jdobbs is a stickler for files being Blu-ray compliant. His latest release is supposed to now handle EAC3, but he wants the compliant version, so he tried to make it so and it fails. I've asked to use the stream as is, with an option to recode if the user so desires.
Yours was maybe encoded with an eac3 encoder, unless there's a header misinterpretation of your file wich is maybe also a "simple" ac3.
It is EAC3, but not the Blu-ray compliant kind.

[Music Fan]

Ok. Your player could maybe play it anyway, even on a Blu-ray. You can try with a short extract that you export with TSMuxer in Blu-ray mode (and burn it with ImgBurn for instance), it will keep sound as is.

[MrVideo]

Ok. Your player could maybe play it anyway, even on a Blu-ray. You can try with a short extract that you export with TSMuxer in Blu-ray mode (and burn it with ImgBurn for instance), it will keep sound as is.
Yes, it will, but it won't be an authored disc with a menu and multiple files in the menu to select for playback, i.e., BDRebuilder.

[Q-the-STORM]

If I remember correctly, you should be able to simply replace the m2ts file... so use your regular software to create the BD, but then remux the m2ts with tsmuxer....

[iSeries]

Hi, I'm extracting a DTS-HD MA 7.1 audio track to 7.1 WAVs, but eac3to is getting about 10% of the way in and then saying that clipping is detected and that a second pass will be necessary. Then at the end it does the second pass and applies a gain of -0.31db. I've never seen this before, is this normal?

[tebasuna51]

Is not normal with a DTS-MA, but I see some DTS-MA bad encoded with this problem.

You can let the -0.31 dB (little atenuation) or use the parameter -no2ndpass to obtain the full volume with a little clip (unnoticeable most the times).

[iSeries]

Thanks, actually I got it wrong, it's a DTS-X soundtrack (Jason Bourne). Are there any known issues decoding this format?

[MrVideo]

If I remember correctly, you should be able to simply replace the m2ts file... so use your regular software to create the BD, but then remux the m2ts with tsmuxer....
Nope. Trying to replace a M2TS file with a different one will result in the M2TS file not being played. The reason is that two other supporting files in the Blu-ray structure must also be updated to support the changed M2TS file.

I tried it once and it failed. Recently I found out why.

[MrVideo]

FFMPEG -drc_scale 0 -i "input.mkv" -vn -acodec ac3 -ac 6 -ab 640k -center_mixlev 0.707 "output.ac3"
Is this the best way to encode ac3 from 6 separate wav files:
ffmpeg -drc_scale 0 -i "${WINPWD}\\${NAME} 00 (Left).wav" \
-i "${WINPWD}\\${NAME} 01 (Right).wav" \
-i "${WINPWD}\\${NAME} 02 (Center).wav" \
-i "${WINPWD}\\${NAME} 03 (LFE).wav" \
-i "${WINPWD}\\${NAME} 04 (Left Surround).wav" \
-i "${WINPWD}\\${NAME} 05 (Right Surround).wav" \
-filter_complex "[0:a][1:a][2:a][3:a][4:a][5:a]amerge=inputs=6[aout]" \
-map "[aout]" -acodec ac3 -ab ${BITRATE}k -center_mixlev 0.707 \
${WINPWD}\\${NAME}.ac3
The command is being done within cygwin. Technically it is being done within a Zshell script. Gotta have my Linux, even when running on M$. :D

I found the command line on the net (I've added your two options). If there is a simplier CLI for doing this, I'd appreciate an example.

Thanks.

[tebasuna51]

Talking about ffmpeg is off topic here BTW...

Seems work fine your command with 5.1, I never try the -map "[aout]" parameter and, to avoid map order problems, I finish using a more complex syntax in my UsEac3to GUI:

ffmpeg -drc_scale 0 -i "x.L.wav" -i "x.R.wav" -i "x.C.wav" -i "x.LFE.wav" -i "x.SL.wav" -i "x.SR.wav" -filter_complex "join=inputs=6:channel_layout=FL+FR+FC+LFE+SL+SR:map=0.0-FL|1.0-FR|2.0-FC|3.0-LFE|4.0-SL|5.0-SR" -acodec ac3 -center_mixlev 0.707 -ab 640k "x.L.wav_.ac3"

I abandoned the filter amerge because can't manage non-standard channel_layout.

[MrVideo]

Talking about ffmpeg is off topic here BTW...
I went looking for a ffmpeg thread, but didn't find one.
ffmpeg -drc_scale 0 -i "x.L.wav" -i "x.R.wav" -i "x.C.wav" -i "x.LFE.wav" -i "x.SL.wav" -i "x.SR.wav" \
-filter_complex "join=inputs=6:channel_layout=FL+FR+FC+LFE+SL+SR:map=0.0-FL|1.0-FR|2.0-FC|3.0-LFE|4.0-SL|5.0-SR" \
-acodec ac3 -center_mixlev 0.707 -ab 640k "x.L.wav_.ac3"
Thanks, I'll convert to this one.

[73ChargerFan]

Back to the EAC3 bit, it is a different stream on HD-DVDs than on Blu-Rays. But it's been too long and I don't remember what...

[LigH]

As already mentioned: 7.1 E-AC3 on Blu-ray consists of 5.1 legacy AC3 + 2 channels E-AC3, to be decoded optionally. On HD DVD, decoding E-AC3 is mandatory. See: Wikipedia – Dolby Digital Plus: HD DVD and Blu-ray Disc (https://en.wikipedia.org/wiki/Dolby_Digital_Plus#HD_DVD_and_Blu-ray_Disc)

[MrVideo]

As already mentioned: 7.1 E-AC3 on Blu-ray consists of 5.1 legacy AC3 + 2 channels E-AC3, to be decoded optionally. On HD DVD, decoding E-AC3 is mandatory. See: Wikipedia – Dolby Digital Plus: HD DVD and Blu-ray Disc (https://en.wikipedia.org/wiki/Dolby_Digital_Plus#HD_DVD_and_Blu-ray_Disc)
As I read somewhere (been doing a lot of reading lately), EAC3 on Blu-ray is 5.1 core plus 4 channels in the subsystem. When played back, the amp takes the L/C/R from the core and mixes it with the four surround channels for correct placement of the surround channels.

[73ChargerFan]

I think on HD-DVD, DD+ didn't have an AC3 core.

[LigH]

Yes, we already mentioned that difference several times now. Just above.

[mstrong]

I've been doing a lot of video editing lately. Most of the tools handle FLAC easily but not DTS HD 7.1. I have been using eac3to to do my conversion but only just noticed that my transcoded files only have FLAC 5.1 (I'm now at this). I would like to have FALC 7.1.

I've been doing some reading but I haven't found my answers yet. I read that I could use the codecs from some external programs to transcode from DTS HD 7.1 to FALC 7.1 but it appears that neither one is available anymore (like ArCSoft). I thought I read somewhere that Libav now supports the conversion of DTS HD 7.1 to FLAC 7.1. I downloaded what I think is the most recent version and I haven't had much success. So I have two questions.

1) Can I convert DTS HD 7.1 to FLAC 7.1?
2) Am I on the right track and can someone help me troubleshoot?

I am using MKV containers.

I have a follow-on question. If I start with DTS HD 7.1 and end with FLAC 5.1, what happened to the other two channels? Do I lose the side channels but keep the back? Do I lose the back but keep the sides? Or are they somehow combined? Using MediaInfo, I've noticed that the DTS HD 7.1 tracks are listed as both 8 and 6 channel. Does this mean I simply get the 6 channel 'core' of the 8 channel track?

[Sparktank]

1) Can I convert DTS HD 7.1 to FLAC 7.1?

Yes.

If you use the latest verison of eac3to, which, will default to DCAdec library for decoding DTS-HD streams.
It's the most-updated library to use since Sonic and Arcsoft have stopped development are now obslete.

FFMPEG has a few more udpates for the DCAdec library, but it's probably for blurays that don't break the video file into segments (multiple cuts, obfusticated playlists, etc).
I have no idea how differently FFMPEG works compared to eac3to in terms of gaps, clipping, etc or even bit-depths (where there are averages, etc).


I downloaded what I think is the most recent version

The most recent version is linked on the first post of this thread.
And the changelog right underneath the link.
This thread is up to date when new releases come out.


and I haven't had much success

That's because the default uses DCAdec, which is superior to the other codecs.
Only issue seems to be with DTSExpress formats.
To use LAV instead, you have to invoke LAV manually with -lav switch (or something similar; I don't use other codecs for DTS streams).



As for the 7.1 to 5.1, that really depends on what you do.

Since you don't share your commands or process, we can only guess what you do.

If you use -down6 on a 7.1 source, the back channels get mixed into the side channels.
It will show it in the log.

MediaInfo shows 8 channels and 6 channels on a 7.1 DTS-HD stream because it's showing the lossless extension (7.1) and the internal core (5.1).
It has no real bearing on what eac3to will do when you do something. If we know what you are doing (commands/logs).
Relying on MediaInfo is a pitall of many users. It just reports information but it even has its own flaws the developers won't address (giving lossy audio a bit depth,etc).

It sounds like you should start from the ground up (and not use MediaInfo).
simply running eac3to with the filename will give you all the info you need on a file:
eac3to.exe audio.dts
And it will show the file properties.

[sneaker_ger]

Post the eac3to log. Like Sparktank said eac3to can convert DTS-HD 7.1 to FLAC 7.1 out-of-the-box. No additional software is required. Command is simple:
eac3to "input.dts" "output.flac"

Latest eac3to version is 3.31.

That's because the default uses DCAdec, which is superior to the other codecs.
Only issue seems to be with DTSExpress formats.
To use LAV instead, you have to invoke LAV manually with -lav switch (or something similar; I don't use other codecs for DTS streams).
It would be -libav but DO NOT use that for DTS with extensions. It is outdated. It will neither fully decode DTS-HD MA nor DTS Express.

(Latest ffmpeg can do DTS Express and also DTS-HD MA).

[mstrong]

Thank you Sparktank and sneaker_ger. This makes sense. I just downloaded version 3.31 (I was using 3.27 before). From time to time I have used HdBrStreamExtractor because it is a very clean and simple interface and then remuxing the streams using MKVToolNix GUI. However, HdBrStreamExtractor appears to be limited in what it can do. But I really likeMKVToolNix GUI for remixing, forcing subtitles, etc. BTW, is HdBrStreamExtractor a GUI for eac3to or is it its own separate program?

Up until now I have been using YR_eac3to_more_GUI. I clearly don't understand how it works because I constantly get errors and only seam to get it to work once in a while (and when it works I only get 5.1 output). Is there some good documentation on how to use it? Is there a better GUI?

I have a movie that is in an MKV container. The video is H264, the audio is DTS - HD 7.1 and AC3, and there are 3 subtitle tracks. I need the output to be an MKV container with H264, with two audio streams FLAC 7.1 and AC3, and the 3 subtitles. I would post a log file but I can't seam to get YR_eac3to_more_GUI to run at the moment. I can't seam to select the right options.

Should I just use the command line?

[stax76]

Up until now I have been using YR_eac3to_more_GUI. I clearly don't understand how it works because I constantly get errors and only seam to get it to work once in a while (and when it works I only get 5.1 output). Is there some good documentation on how to use it? Is there a better GUI?

I don't know if it fits your requirements, staxrip has GUIs for many tools. I've always recommended to use MakeMKV instead of cracked AnyDVD and eac3to, in the end it might not make a great difference which route you take.

eac3to GUI screenshot:

https://github.com/stax76/staxrip/wiki/Screenshots#eac3to

[tebasuna51]

From time to time I have used HdBrStreamExtractor because it is a very clean and simple interface ...
BTW, is HdBrStreamExtractor a GUI for eac3to or is it its own separate program?

Is a GUI and you can replace the new eac3to version.
But HdBrStreamExtractor is abandoned and can't use some news of eac3to.
You can find a actualiced version in MeGUI -> Tools -> HD Streams Extractor.
BTW you need know the syntax parameters [+Options] to obtain all eac3to functions.

Up until now I have been using YR_eac3to_more_GUI. I clearly don't understand how it works because I constantly get errors and only seam to get it to work once in a while (and when it works I only get 5.1 output). Is there some good documentation on how to use it?

I never understand this GUI (abandoned also) and I can't recomend it.

Is there a better GUI?

I wrote a GUI than is the better for me: UsEac3to
But I know it is not so easy than HdBrStreamExtractor.
There are a Readme.

stax64 show you other option.

Should I just use the command line?

Why not?

[stax76]

Should I just use the command line?

CMD.exe fine, just don't forget it's probably as old as me and there is something much younger called PowerShell.exe.

[mstrong]

Where is the most complete eac3to documentation. I have found this one but it is old enough that it still talks about ArcSoft and Sonic Cinemaster.

https://en.wikibooks.org/wiki/Eac3to/How_to_Use

There has to be something more recent.

[mstrong]

I think I am beginning to understand things better. So now I have detailed questions. For example, I just did a DTS Master Audio, English, 7.1 channels, 24 bits, 48kHz (core: DTS, 5.1 channels, 1509kbps, 48kHz) to FLAC Demux and conversion. The result was FLAC, 7.1 channels, 2:00:51, 24 bits, 3919kbps, 48kHz.

Questions:

1) When the process started I was warned that "libDcaDec reported the warning "XLL output not lossless"." Why is this not a lossless process and what can I do to make it lossless?
2) Where did the core 5.1 go? Can I preserve the core 5.1? What happens if I try to play this FLAC file with on 5.1 core on a 5.1 home theatre? Can I extract just the core 5.1? If so, how?

Here is the Log Text

eac3to v3.31
command line: eac3to.exe 2: c:\users\mstrong\desktop\mkvs\GOTG.BluRay.mkv c:\users\mstrong\desktop\GOTG.BluRay.FLAC
------------------------------------------------------------------------------
MKV, 1 video track, 3 audio tracks, 2 subtitle tracks, 2:00:51, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: DTS Master Audio, English, 7.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
3: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
4: AC3, English, 2.0 channels, 192kbps, 48kHz
5: Subtitle (PGS), English
6: Subtitle (PGS), English
[a02] Extracting audio track number 2...
[a02] Decoding with libDcaDec DTS Decoder...
[a02] libDcaDec reported the warning "XLL output not lossless". <WARNING>
[a02] Encoding FLAC with libFlac...
[a02] Creating file "c:\users\mstrong\desktop\GOTG.BluRay.FLAC"...
[a02] The original audio track has a constant bit depth of 24 bits.
Video track 1 contains 347707 frames.
eac3to processing took 15 minutes, 22 seconds.
Done.

[sneaker_ger]

1) Consensus seems to be that this warning can be safely ignored. It should still be lossless. Truth is: no one really knows. Format is reverse engineered. It was discussed before.
2) FLAC doesn't have any core. If you want DTS core and FLAC you have to create two separate files. (You can encode FLAC to AC3 or DTS anytime, though. But it will not be an exact copy of the core on the original Blu-ray.)
To extract just core use the -core switch, e.g.:
eac3to GOTG.BluRay.mkv 2:output.dts -core

[slick1109]

This might have already been mentioned by someone, but in case it hasn't I wanted to share some information.

eac3to does not appear to be updated yet to handle UltraHD Blu-ray Discs. The problem seems to be that the .mpls/.bdmv/.clpi/.bdjo files have changed their version number from "0200" to "0300".

I was able to get around this problem by modifying the files in a Hex Editor. I simply changed the header info, for an mpls file, to "mpls0200" and that allowed eac3to to read the file and extract the audio/video/chapters/subtitles as usual.

Hopefully whomever is in charge of updating this program can do so to accept these new 0300 files.

[hubblec4]

There are more differences. Eac3to will not really work with UHD-BD at the moment.
https://forum.doom9.org/showthread.php?p=1810285#post1810285

[colpoz]

Is there any way to extract only the subtitles flagged as forced instead of the entire track??

Example if the English track contains "1000 normal and 50 forced captions" I want to extract only the 50 forced ones.

Is it possible using only eac3to command line?

[LigH]

Probably not (at least I see no option mentioned in its help output). The purpose of eac3to is mainly to handle audio. You may need a tool specialized in handling subtitles instead.

[stax76]

Is there any way to extract only the subtitles flagged as forced instead of the entire track??

Example if the English track contains "1000 normal and 50 forced captions" I want to extract only the 50 forced ones.

Is it possible using only eac3to command line?

As far as I know such disks are extremely rare, what's the name of the title? Tools you can try are BDSup2Sub(++) and Subtitle Edit.

[Q-the-STORM]

As far as I know such disks are extremely rare, what's the name of the title? Tools you can try are BDSup2Sub(++) and Subtitle Edit.

I've encountered a lot of those, it's not that rare... you can extract them with BDSub2Sub... just drop the file in there, navigate to "File" -> "Save/Export" -> tick "Export only Forced" -> click on "save"...

If you prefer working in CLI, use --forced-only

[stax76]

I've encountered a lot of those, it's not that rare... you can extract them with BDSub2Sub... just drop the file in there, navigate to "File" -> "Save/Export" -> tick "Export only Forced" -> click on "save"

Can you name same titles, I'm always looking to improve my sample collection.

[von Suppé]

Take into account that forced subtitles can be flagged as "forced" in a total subtitle stream,
but they can also be muxed / authored as a separate stream.

[stax76]

I'm interested in titles that have it flagged as forced, ideally with spoken dialog and small file size.

[nevcairiel]

Most of the Avatar Blu-ray releases use flagged forced frames for alien dialog, sometimes even with normal frames active at the same time, which is a good test if the separation works properly.

[stax76]

It's a bit long, at least the price is low.

[nevcairiel]

Blu-rays are the only sources of commercial PGS subtitles, what do you expect? :p

[stax76]

Probably unrealistic to find but something smaller like 100 min, 30 GB and decent movie I've not already seen.

[Q-the-STORM]

I don't remember which ones, it's not like I wrote it down... Definitely a few seasons of True Blood have them (not all, some have separate forced subs), same goes for Game of Thrones...

[thomaz909]

When i use "-normalize" it creates wav files only to get the maxiumum gain value of the source.
Isn't it possible to just read the gain value of the source instead of doing a complete decode to wav (some ac3 640 sources result in very huge 64 bit float wavs)?

[LigH]

Gain values in headers are no guarantee to avoid an integer overflow or a suboptimal volume.

[Boulder]

A good option would be to output the first pass to NUL if normalization is forced by the user.

[tebasuna51]

Maybe, but with a wav file the second pass don't need re-decode source.

[thomaz909]

isn't encoding the bottleneck in the whole process?

i would really love to have an option to -normalize "onthefly".
ssd wear, small ramdisks.
~107mins ac3 640 produced a 14gb 64bit wav 5.1 file, only to get the 4-5 digit db value.

[Asmodian]

Maybe, but with a wav file the second pass don't need re-decode source.

Or even worse, re-extract if from the BD!

isn't encoding the bottleneck in the whole process?

i would really love to have an option to -normalize "onthefly".
ssd wear, small ramdisks.
~107mins ac3 640 produced a 14gb 64bit wav 5.1 file, only to get the 4-5 digit db value.

To do normalizing well you need to read the entire file before writing anything. Maybe it could be cached in memory (assuming there is enough free memory) or recompressed to flac for the intermediate file, to reduce IO, but true on the fly normalizing is a bad idea.

[sneaker_ger]

To do normalizing well you need to read the entire file before writing anything.
But you don't need to store the decoded data of the first pass. Only store the peak volume. That's just one value that gets updated when you find a new peak. O(1).

[tebasuna51]

~107mins ac3 640 produced a 14gb 64bit wav 5.1 file, only to get the 4-5 digit db value.

BTW, -normalize is only recomended when downmix, don't use normalize when recode 5.1 -> 5.1

[Boulder]

BTW, -normalize is only recomended when downmix, don't use normalize when recode 5.1 -> 5.1Why not? Is it due to the dynamics between the channels?

[mstrong]

I have learned how to use the command line version of eac3to. The process creates a gap file. How do I mux my files back together using the gap file?

How do I do it using eac3to? I am also wondering if it can be done using MKVToolNix GUI.

[tebasuna51]

Why not?

Well, you can do it if you listen low volume, but most the times the input source (from BD's) have a correct volume level without maximize.

Is it due to the dynamics between the channels?

If you use eac3to to decode (ignoring any Dynamic Range Compression in source) and a free AC3 encoder (Aften or ffmpeg) than don't add DRC to output by default, the dynamics between channels are preserved, because the same gain is applied over all channels.

Only if you use a Dolby Digital encoder, than create DRC, the normalize can distort the initial balance between channels.

[tebasuna51]

I have learned how to use the command line version of eac3to. The process creates a gap file. How do I mux my files back together using the gap file?

How do I do it using eac3to? I am also wondering if it can be done using MKVToolNix GUI.

When eac3to create a gap file it do, automatically, a second pass to correct the detected problems, and the corrected files are ready to mux (if all work fine of course).

AFAIK only eac3to can use that gap file.

[Boulder]

Well, you can do it if you listen low volume, but most the times the input source (from BD's) have a correct volume level without maximize.
What I've noticed is that many older movies have a quite big headroom (several dB's) even in the 5.1-channel lossless track. The mono and stereo tracks are all quite low in volume, sometimes eac3to shows that you need to amplify by almost 10dB to normalize. That's why I've always enabled normalization apart from encoding the original audio track to FLAC to save some space.

[Q-the-STORM]

How do I mux my files back together using the gap file?
Just run the same command again. When eac3to sees that there is a gap file, it will use it.

[hello_hello]

isn't encoding the bottleneck in the whole process?

i would really love to have an option to -normalize "onthefly".
ssd wear, small ramdisks.
~107mins ac3 640 produced a 14gb 64bit wav 5.1 file, only to get the 4-5 digit db value.

You could effectively do it after a ReplayGain or EBU R128 scan. The audio still needs to be scanned first, but it should be faster than converting to wave. Once you have the peak values, you know how to adjust the volume when converting.

My old quad-core can scan a 2 hour, 5.1ch AC3 file in around 40 seconds (oversampling disabled), but it can also scan four of them simultaneously in about 40 seconds. I use foobar2000 for that sort of thing as it's ReplayGain scanner is quite fast.

It can also adjust the level to the ReplayGain target volume when converting, but you can specify an additional increase or decrease, so by enabling the "apply gain and prevent clipping" option with a 20dB increase, it'll limit the increase to prevent clipping according to the ReplayGain info, and the end result is peak normalising.
Adjusting to the ReplayGain target volume minus 5dB gives you the standard EBU R128 target volume, which is how I'd normalise after downmixing related files to stereo (episodes of a TV show etc), although it doesn't hurt to scan the output files and check the peaks just to be sure.

Why has ReplayGain/R128 scanning been almost completely ignored by programs related to video conversion?

Avisynth runs two passes, but does it write a wave file? If not that'd be another way to normalise without writing an intermediate file.

What I've noticed is that many older movies have a quite big headroom (several dB's) even in the 5.1-channel lossless track. The mono and stereo tracks are all quite low in volume, sometimes eac3to shows that you need to amplify by almost 10dB to normalize. That's why I've always enabled normalization apart from encoding the original audio track to FLAC to save some space.

That's no doubt true but it probably also means they have less dynamic range than newer movies, but the "average" volume, or at least the dialogue volume, would probably be very similar. Well that's my theory....

[MrVideo]

For the life of me, I have never been able to understand why anyone would want to screw with the volume of a movie/TV show soundtrack. The mix is what the original director/producers/audio engineers wanted it to be in their product. Why do some think that they know better than those who put it together in the first place?

[nevcairiel]

For the life of me, I have never been able to understand why anyone would want to screw with the volume of a movie/TV show soundtrack. The mix is what the original director/producers/audio engineers wanted it to be in their product. Why do some think that they know better than those who put it together in the first place?

Because between the producers and you getting it, there were so many different people messing with it, that any original intent is already lost, especially when handling broadcast, to a lesser extend with optical media - and most people would otherwise just reach for the remote to adjust the volume.
These days most broadcast content is already volume leveled with EBUR128 though, since a lot of stations adopted this standard.

Note that volume leveling is only an overall attenuation of the volume. It does not change its dynamic range, or anything like that, it just changes the overall volume by a constant factor, just like you would change the volume on your speakers.
Thats why a 2-pass process is needed, analyze the full track to find its volume level, and then attenuate by that level.

[tebasuna51]

From the beginning eac3to was designed to transcode audio without lose quality of source.

And only a Peak Normalization (https://en.wikipedia.org/wiki/Audio_normalization) is allowed with -normalize.
Of course you can limit the peaks to avoid excesive volume, but the dynamic range is preserved.

A Loudness Normalization based in RMS values instead peak values, like ReplayGain and EBU R128 do, is not always possible without lose dynamic range.
Of course any user can do this with other tools than eac3to.

[hello_hello]

For the life of me, I have never been able to understand why anyone would want to screw with the volume of a movie/TV show soundtrack. The mix is what the original director/producers/audio engineers wanted it to be in their product. Why do some think that they know better than those who put it together in the first place?

If there's dynamic range reduction info saved to AC3 audio, it would have been the director/producers/audio engineers who put it there. You can decode without using that information to reduce the dynamic range, but it's part of the AC3 spec.

I generally compress audio for movies/TV shows I copy to the hard drive connected to the TV, but that's only because I can do it better than the TVs "night mode", and it's only a temporary copy for viewing in the small hours.

I don't think the recent posts referred to compressing the audio though, only adjusting the level up or down as a whole, or adjusting it so the peaks are near maximum when encoding.

[Ripman]

As far as I know such disks are extremely rare, what's the name of the title? Tools you can try are BDSup2Sub(++) and Subtitle Edit.

MakeMKV shows when sub are forced, and if they're not empty, the forced subs are left in the resultant mkv.

Examples of BDs with forced subs that MakeMKV sees, and that can be extracted with mkvextract or Subtitle Edit or AegisSub etc.: Star Wars VI Return of the Jedi; Patton; Kill Bill BDs; Tora Tora Tora; The Walk; The Martian.

[mstrong]

Just run the same command again. When eac3to sees that there is a gap file, it will use it.

I am assuming the gap file needs to be in the same folder as the video and audio source files?

Could you give me an example syntax for muxing a video and audio file into the same MKV?

Thank you.

[Q-the-STORM]

I am assuming the gap file needs to be in the same folder as the video and audio source files?
Yes, the gap file is being created in the correct folder anyways, so there's no need to move anything, just run the command a second time.

Could you give me an example syntax for muxing a video and audio file into the same MKV?
Can't be done with eac3to, you gotta use mkvmerge for that...

[thomaz909]

i just want to "normalize" to 0db.
that means, it scans all channels, takes the highest overall gain found and then amplifies all channels by this value.
for example L=-5db, R=-6db, C=-2db, SL=-12db, SR=-13db, then the center channel has the highest gain with -2db and finally all channels get an amplification by +2db.
no drc, compressor, loudness maximizer, limiter, etc.
and that all in the most efficient, resource saving way. :)

[tebasuna51]

i just want to "normalize" to 0db.
that means, it scans all channels, takes the highest overall gain found and then amplifies all channels by this value.
for example L=-5db, R=-6db, C=-2db, SL=-12db, SR=-13db, then the center channel has the highest gain with -2db and finally all channels get an amplification by +2db.
no drc, compressor, loudness maximizer, limiter, etc.
and that all in the most efficient, resource saving way. :)

That is just like eac3to -normalize work.

To be clear is a "Peak Normalization" than don't change dynamics, and not a "Loudness Normalization (RMS)"

[MrVideo]

i just want to "normalize" to 0db.
that means, it scans all channels, takes the highest overall gain found and then amplifies all channels by this value.
for example L=-5db, R=-6db, C=-2db, SL=-12db, SR=-13db, then the center channel has the highest gain with -2db and finally all channels get an amplification by +2db.
no drc, compressor, loudness maximizer, limiter, etc.
and that all in the most efficient, resource saving way. :)
I hate to say this, but this a a bad example, because I dare you, in a blind test, that you will not be able to notice that 2dB difference.

[thomaz909]

it would be nice if eac3to ignores -downXX switch when source is lower than the specified bitrate.
for example decoding a 64bit ac3 with -down32 then it "downbits" to 32bit.
when decoding a 24bit dts it ignores the switch and decodes to 24bit instead of "upbitting" to 32bit.

or add -downXXXX (for example -down6432)
then it only downs 64bit to 32bit and leaves all other input bitrates untouched.

[sneaker_ger]

It already works like this, I believe. Can you provide any example (sample file + log) where it doesn't?

[tebasuna51]

it would be nice if eac3to ignores -downXX switch when source is lower than the specified bitrate.

Maybe you mistake "bitrate" with "bitdepth"?

-downXX fix the bitdepth of decoded PCM data.

for example decoding a 64bit ac3 with -down32 then it "downbits" to 32bit.

Is possible to have a 64 Kb/s AC3 (I never see), but AC3 don't have any bitdepth then 64bit AC3 don't have sense.

when decoding a 24bit dts it ignores the switch and decodes to 24bit instead of "upbitting" to 32bit.

- If you are talking about a DTS-MA 24 bits, for what you want 32 bits with the last 8 bits all to "0"?

eac3to try to be smart and ignore usseless parametres, even check if last 8 significant bits are all to "0" to output only 16 bits.

- If you are talking about a standard DTS, like AC3, don't have bitdepth. Does not exist a standard DTS 24 bits.

When eac3to decode a lossy format (AC3, DTS, MP3, ...) internally work with 64 bits float samples and, at output by default, downsample to 24 bits int because is enough precission.

If you want preserve the 64 bit float you can add the parameter:

-full

[stax76]

@madshi

I would like to request 0.1 intervals for the percentage status display/output.

Some years ago I suggested to use an alternative or additional mkv muxer because many people don't like to install directshow filters, configuring DirectShow and in general configuring and repairing Windows requires expertise many people don't have.

This summer I received a email with following content:

Shit don’t work and it sucks.

I was mostly amused about it, two days ago I received something more concrete from the same kind person:

So explain to me why is it you made 100 versions crappy with the haali filter, then I am telling everyone not to use it because of that stupid haali filter and now I see version 101 does not have it? You made me look like a fool, you are a pile of shit and I will love you track and hunt you down for literally being the shittiest developer on planet earth.

I don't know who this person is and these two mails is the only thing I ever heard from him, I do not reply such mails.

In staxrip Haali is optional even for eac3to usage and I didn't remove or change anything. For VC1 I think eac3to users need Haali because I don't think there is a avs/vs source filter that can handle raw VC1.

While staxrip supports many different tools and workflows eac3to is still popular and I always ensured it has the best possible support.

[Music Fan]

Is there a way to select the output folder when using the -demux function ?

[tebasuna51]

You can try:

eac3to "input" "D:\tmp\output.*"

[Music Fan]

Thanks, good trick !

[hubblec4]

Hi madshi

Exists a hidden switch to get the PID's for the streams? The order you display the streams is different to the mpls and the m2ts.

[Snowknight26]

-logpids

[hubblec4]

many thanks for the info

[stax76]

@madshi

There seem to be a problem with a AC3 file, it was reported by a staxrip user so I don't have the source but if you want to work on it I might be able to get the file for you.

-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Source file MediaInfo
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

F:\Movie.mkv

General
Complete name : F:\Movie.mkv
Format : Matroska
Format version : Version 1
File size : 1.46 GiB
Duration : 56 min 46 s
Overall bit rate : 3 677 kb/s
Encoded date : UTC 2009-06-22 03:44:10
Writing application : mkvmerge v2.4.0 ('Fumbling Towards Ecstasy') built on Nov 7 2008 17:24:23
Writing library : libebml v0.7.8 + libmatroska v0.8.1

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L3.1
Format, CABAC : Yes
Format, ReFrames : 3 frames
Codec ID : V_MPEG4/ISO/AVC
Duration : 56 min 46 s
Bit rate : 3 289 kb/s
Width : 1 280 pixels
Height : 720 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
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) : 0.149
Stream size : 1.28 GiB (88%)
Writing library : x264 core 67 r1162M f7bfcfa
Language : English
Default : Yes
Forced : No

Audio
ID : 2
Format : AC-3
Format/Info : Audio Coding 3
Format, Endianness : Big
Codec ID : A_AC3
Duration : 56 min 46 s
Bit rate mode : Constant
Bit rate : 384 kb/s
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 kHz
Frame rate : 31.250 FPS (1536 spf)
Bit depth : 16 bits
Compression mode : Lossy
Stream size : 156 MiB (10%)
Service kind : Complete Main
Default : Yes
Forced : No

-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Demux mkv using mkvextract 13.0.0 x64
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

C:\StaxRip\Apps\MKVToolNix\mkvextract.exe tracks F:\Movie.mkv 1:"F:\Movie_temp\Movie ID1.ac3" --ui-language en

Extracting track 1 with the CodecID 'A_AC3' to the file 'F:\Movie_temp\Movie ID1.ac3'. Container format: Dolby Digital (AC-3)

Start: 9:08:04 PM
End: 9:08:07 PM
Duration: 00:00:02

General
Complete name : F:\Movie_temp\Movie ID1.ac3
Format : AC-3
Format/Info : Audio Coding 3
File size : 156 MiB
Duration : 56 min 46 s
Overall bit rate mode : Constant
Overall bit rate : 384 kb/s

Audio
Format : AC-3
Format/Info : Audio Coding 3
Format, Endianness : Big
Duration : 56 min 46 s
Bit rate mode : Constant
Bit rate : 384 kb/s
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 kHz
Frame rate : 31.250 FPS (1536 spf)
Bit depth : 16 bits
Compression mode : Lossy
Stream size : 156 MiB (100%)
Service kind : Complete Main

-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Audio Source File MediaInfo
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

General
Complete name : F:\Movie_temp\Movie ID1.ac3
Format : AC-3
Format/Info : Audio Coding 3
File size : 156 MiB
Duration : 56 min 46 s
Overall bit rate mode : Constant
Overall bit rate : 384 kb/s

Audio
Format : AC-3
Format/Info : Audio Coding 3
Format, Endianness : Big
Duration : 56 min 46 s
Bit rate mode : Constant
Bit rate : 384 kb/s
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 kHz
Frame rate : 31.250 FPS (1536 spf)
Bit depth : 16 bits
Compression mode : Lossy
Stream size : 156 MiB (100%)
Service kind : Complete Main

-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-
Audio encoding using eac3to 3.31 x86
-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

C:\StaxRip\Apps\eac3to\eac3to.exe "F:\Movie_temp\Movie ID1.ac3" "F:\Movie_temp\Movie ID1_a1.m4a"
-quality=0.23 -normalize -downStereo -progressnumbers

AC3, 2.0 channels, 0:56:46, 384kbps, 48kHz, dialnorm: -27dB
Removing AC3 dialog normalization...
Decoding with libav/ffmpeg...
Writing WAV...
Creating file "F:\Movie_temp\Movie ID1_a1.m4a.pass1.wav"...
Caution: The WAV file is bigger than 4GB.
Some WAV readers might not be able to handle this file correctly.
Starting 2nd pass...
Reading WAV...
Reducing depth from 64 to 32 bits...
Encoding AAC <0.23> with NeroAacEnc...
Applying 0.95dB gain...
The original audio track has a constant bit depth of 64 bits.
The processed audio track has a constant bit depth of 32 bits.
eac3to processing took 3 minutes, 34 seconds.
Done.

Start: 9:08:08 PM
End: 9:11:43 PM
Duration: 00:03:34

General
Complete name : F:\Movie_temp\Movie ID1_a1.m4a
Format : MPEG-4
Format profile : Base Media / Version 2
Codec ID : mp42 (M4A /mp42/isom)
File size : 76.3 MiB
Duration : 2 h 48 min
Overall bit rate mode : Variable
Overall bit rate : 63.4 kb/s
Encoded date : UTC 2017-09-22 01:08:31
Tagged date : UTC 2017-09-22 01:11:42

Audio
ID : 1
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : HE-AAC / LC
Codec ID : 40
Duration : 2 h 48 min
Bit rate mode : Variable
Bit rate : 62.6 kb/s
Maximum bit rate : 77.7 kb/s
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 kHz / 24.0 kHz
Frame rate : 23.438 FPS (1024 spf)
Compression mode : Lossy
Stream size : 75.3 MiB (99%)
Writing library : Nero AAC codec 1.5.4.0
Encoded date : UTC 2017-09-22 01:08:31
Tagged date : UTC 2017-09-22 01:11:42

[tebasuna51]

That is a typical problem with captures from TV.

When MediaInfo/eac3to read the first header of AC3 show:

Format : AC-3
Duration : 56 min 46 s
Bit rate : 384 kb/s
Channel(s) : 2 channels
Stream size : 156 MiB

All seems correct and we can expect a decoded wav (64 bits) size of:

3406 sec. x 2 ch. x 8 bytes/ch x 48000 samples/sec = 2495 MB

But the decoded wav is:

Caution: The WAV file is bigger than 4GB.

because only the first's AC3 are 2.0 and the rest are 5.1, then the decoded wav have 3 times (6ch/2ch) size and are buggy (noise) recoded to a m4a with a duration of:

3406 sec x 3 = 2h 50m

near the duration 2 h 48 min of m4a (seems there are a few of AC3 2.0 frames, maybe commercials)

An AC3 with a mix of 2.0 and 5.1 frames are invalid and can't be recoded properly, you need obtain valid AC3 from it.

You can use DelayCut or a tool I wrote: SplitAc3 (https://forum.doom9.org/showthread.php?p=1447695#post1447695)

Other related threads:
https://forum.doom9.org/showthread.php?p=1679327#post1679327
https://forum.doom9.org/showthread.php?p=1768231#post1768231

[stax76]

Oh Thanks, I didn't notice or expect it.

[hello_hello]

If you have ffdshow installed and don't mind decoding/encoding via DirectShow you can usually fix those problematic AC3 re-encodes by decoding with ffdshow and enabling it's mixer filter and setting it to output 5.1ch.

[pstn]

Hello,

When using the -demux command, is it possible extract the DTS core along with the DTS-HD without having to do 2 runs? (both track separately)

thanks

[sneaker_ger]

eac3to "source.m2ts" 3:"core.dts" -core 3:"both.dts"

[tebasuna51]

But not with -demux

[stax76]

If you have ffdshow installed and don't mind decoding/encoding via DirectShow you can usually fix those problematic AC3 re-encodes by decoding with ffdshow and enabling it's mixer filter and setting it to output 5.1ch.

I could handle the file with ffmpeg.

[hubblec4]

Hi madshi

I found in the Bluray The Great Wall(sample (https://forum.videohelp.com/attachments/43277-1506985180/The%20Great%20Wall.7z)), two streams with E-AC3 EX (I have never seen this before), 7.1 channels and a core with 5.1 channels.
Mkvmerge and ffprobe detect this streams only with 5.1 channels and a bitrate of 576kbps, but it seems to me these is only the core.

4: E-AC3 EX, French, 7.1 channels, 1152kbps, 48kHz
(core: AC3 EX, 5.1 channels, 576kbps, 48kHz)
5: E-AC3 EX, German, 7.1 channels, 1152kbps, 48kHz
(core: AC3 EX, 5.1 channels, 576kbps, 48kHz)

LAV Audio plays the 5.1 channel part only, also MediaInfo displays this stream with the 576kbps value.

So what is wrong, eac3to or all other tools?

best regards

[nevcairiel]

Blu-ray EAC3 is rather peculiar, and ffmpeg doesn't support decoding it (or any other open-source decoders i'm aware of). Its not a proper EAC3 stream but a AC3 core with a EAC3 extension.

[mkver]

FYI: mkvmerge does not strip an extension away. If I mux the streams with mkvmerge id 3 or 4* alone I get files with size 2.8 MiB. MediaInfo 0.7.99 has the following to say about the file with the German track:

General
Unique ID : 245046857353978534520836163522779116025 (0xB85A4EDB6BD25BCBB0D8010ED44371F9)
Complete name : I:\The Great Wall\STREAM\00245_0.mka
Format : Matroska
Format version : Version 4 / Version 2
File size : 2.81 MiB
Duration : 20 s 384 ms
Overall bit rate mode : Constant
Overall bit rate : 1 155 kb/s
Encoded date : UTC 2017-10-03 16:43:44
Writing application : mkvmerge v16.0.0 ('Protest') 64-bit
Writing library : libebml v1.3.5 + libmatroska v1.4.8

Audio
ID : 1
Format : E-AC-3
Format/Info : Enhanced Audio Coding 3
Format settings, Endianness : Big
Codec ID : A_EAC3
Duration : 20 s 384 ms
Bit rate mode : Constant
Bit rate : 500 b/s
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 kHz
Frame rate : 187.500 FPS (256 SPF)
Compression mode : Lossy
Stream size : 2.80 MiB (100%)
Service kind : Complete Main
Default : Yes
Forced : No

So eac3to was right about the bitrate and MediaInfo is buggy. For the French track the situation is a bit more complicated:

General
Unique ID : 181109783328528269767092120785351483635 (0x884079B9EF4E2234A69025BC420A20F3)
Complete name : I:\The Great Wall\STREAM\00245_0 (1).mka
Format : Matroska
Format version : Version 4 / Version 2
File size : 2.81 MiB
Duration : 20 s 384 ms
Overall bit rate mode : Constant
Overall bit rate : 1 155 kb/s
Encoded date : UTC 2017-10-03 16:49:14
Writing application : mkvmerge v16.0.0 ('Protest') 64-bit
Writing library : libebml v1.3.5 + libmatroska v1.4.8

Audio
ID : 1
Format : E-AC-3
Format/Info : Enhanced Audio Coding 3
Format profile : E-AC-3+Atmos / E-AC-3
Format settings, Endianness : Big
Codec ID : A_EAC3
Duration : 20 s 384 ms
Bit rate mode : Constant
Bit rate : 500 b/s
Channel(s) : 11 objects / 6 channels
Channel positions : 11 objects / Front: L C R, Side: L R, LFE
Sampling rate : 48.0 kHz
Frame rate : 187.500 FPS (256 SPF)
Compression mode : Lossy
Stream size : 2.80 MiB (100%)
Service kind : Complete Main
Default : Yes
Forced : No

Besides being wrong about the bitrate, MediaInfo 0.7.99 claims that somehow this track contains an Atmos extension. Don't know if this is true. (It claims that there are 11 objects in this track when one examines the original m2ts file.) MediaInfo 0.7.92.1 does not detect an Atmos extension in the French track (and not in the German track). I don't know which version of MediaInfo claims that these tracks are 576 kb/s.
By the way, if I use ffmpeg to copy these streams, the same applies.
If one extracts the streams, MediaInfo claims they 13h 2m long. I don't know if the bitrate is miscalculated resulting in the wrong duration or if it's the other way around. Probably the former.

*: mkvmerge assigns id 0 to the video track, id 1 to the TrueHD Atmos stream, 2 to the core embedded in the TrueHD Atmos stream, 3 to the French and 4 to the German track.

[hubblec4]

My Pioneer Reciever shows the Dolby Atmos sign when I play the BD with TMT5.
The file size of the extracted audio(with eac3to) is almost the same what mkvmerge output(only the audio in an mka).
So I guess all is ok with this streams, only the detection is not correct(channels, bitrate).

[madshi]

eac3to v3.32 released

http://madshi.net/eac3to.zip

* added support for UHD Blu-Ray playlists and clip infos
* added support for h265/HEVC bitstream parsing
* added support for m2ts h265/HEVC splitting
* added support for MKV h265/HEVC splitting
* added support for h265/HEVC framerate modifications
* added detection for Dolby Vision Enhancement Layer

[Thunderbolt8]

nice, thanks for an update!

can you comment on how accurate HEVC UHD BD support & remuxing is? Im referring to that you said you dont have/didnt use to have exact UHD specifications to implement this.

is all metadata like HDR and dolby vision remuxed as well?

[madshi]

Well, I can't *re*mux HEVC, just *de*mux, because I'm still using the Haali DirectShow filter for MKV muxing, which doesn't support HEVC. But other than that I think everything should work just fine. But of course I can't guarantee that there are no bugs, so some testing might make sense.

[Gizzmo]

Hi,
thanks for the update. Tryed it and now the UHD BD structure is recognized. But the videostream with h265/HEVC is not recognized
as videostream. It is shown as audiostream.

Maybe only a UI issue, with Clown_BD ? Will do the demux now.

[Thunderbolt8]

has the format and structure of .sup files changed with UHD BDs in any way compared to regular BDs? or did everything stay the same?

[XadoX]

Is there a way to get the eac3to console output in PowerShell in multiple lines? For me eac3to outputs everthing only into the first line.

[madshi]

thanks for the update. Tryed it and now the UHD BD structure is recognized. But the videostream with h265/HEVC is not recognized as videostream. It is shown as audiostream.

Maybe only a UI issue, with Clown_BD ? Will do the demux now.
This is very likely to be an issue with Clown_BD, not with eac3to.

has the format and structure of .sup files changed with UHD BDs in any way compared to regular BDs? or did everything stay the same?
I don't know, haven't tested that yet, why don't you give it a try and report back? ;)

Is there a way to get the eac3to console output in PowerShell in multiple lines? For me eac3to outputs everthing only into the first line.
Weird. It works for me with PowerShell in Windows 10. I wouldn't really comment using PowerShell for eac3to, though. A simple command line window is probably the better choice. E.g. PowerShell already stumbles if you do something like "eac3to 1)", it doesn't like the ")".

[XadoX]

...Weird. It works for me with PowerShell in Windows 10. I wouldn't really comment using PowerShell for eac3to, though. A simple command line window is probably the better choice. E.g. PowerShell already stumbles if you do something like "eac3to 1)", it doesn't like the ")".

Hmm... maybe I have to change some setting in PowerShell.
Yeah I am still using CMD, but since Microsoft removed the feature "Start CMD here" with right click in Win10 I am using PowerShell often for quick demuxing.

[madshi]

It's easy to get the "Start CMD here" back, a quick google search will tell you how.

[XadoX]

I know the reg value. But somehow it feels wrong to use things that are not official implemented :)
It's just a matter of time CMD dies completely.

[LigH]

Is there a way to get the eac3to console output in PowerShell in multiple lines? For me eac3to outputs everthing only into the first line.

The reason for this issue is probably that eac3to prepends every output text line with a large number of backspace characters.

https://frupic.frubar.net/shots/36082.png

I have no clue why Madshi believes this to be necessary.

[hubblec4]

Thanks for the new version, madshi.
Nice to see that few first features for UHD-Bluray are come.

Have you new (hidden)switches implemented?

[d3rd3vil]

So how can I use eac3to to just show all the information about a video stream like the Dolby Vision layer and stuff?

[hubblec4]

has the format and structure of .sup files changed with UHD BDs in any way compared to regular BDs? or did everything stay the same?

I have extracted UHD-BD sup files with the new eac3to and all seems to work. BDSup2Sub read the sups and extract also forced sups.

[LigH]

Originally, eac3to was a tool to process audio streams. If you want to know "all details" about video streams, you will probably need different tools.

[Megalith]

I just realized I haven't updated eac3to since 3.27 (the version that comes with Clown_BD). One of the most significant fixes appears to be "TrueHD Atmos streams could not be demuxed or decoded," but I have been able to extract these just fine for remuxing. Is that weird?

[tebasuna51]

Thanks madshi.

In my first test all seems ok.

Can you do anything with EAC3 tracks?
At last to preserve the name 'eac3'to :)

[madshi]

Can you do anything with EAC3 tracks?
What do you mean? E-AC3 tracks should be fully supported.

[tebasuna51]

Not all:

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

https://forum.doom9.org/showthread.php?p=1802092#post1802092 and next

[LigH]

Blu-ray EAC3 (as core AC3 extension only) is still a different case, I believe... ffmpeg can't handle it yet, either:

Dependent substream decoding is not implemented.

[tebasuna51]

To sumarize the problem here.

I found two kind of EAC3 streams detected by eac3to without core (2 English) or with core (3 Spanish) like is show here (sample (https://www.sendspace.com/file/dzj4qm)):

eac3to Eac30.m2ts
------------------------------------------------------------------------------
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)

Over this m2ts we can do, adding or not the -core, some times [ignored]:

eac3to Eac30.m2ts 2: output.eac3 [-core] -> OK eac3 output
eac3to Eac30.m2ts 2: output.ac3 [-core] -> OK recoded to ac3

All ok over track 2

eac3to Eac30.m2ts 3: output.eac3 -> ERROR
eac3to Eac30.m2ts 3: output.eac3 -core -> OK AC3.eac3 output
eac3to Eac30.m2ts 3: output.ac3 -> ERROR
eac3to Eac30.m2ts 3: output.ac3 -core -> useless, recoded core

Over track 3, can't extract or recode (maybe because: Dependent substream decoding is not implemented) the full eac3.
Can extract a perfect AC3 core (when demand eac3 extensión) or recode the core (useless)

Now the same m2ts converted to mkv with MkvToolNix is show by eac3to like:

eac3to Eac30.mkv
------------------------------------------------------------------------------
MKV, 1 video track, 2 audio tracks, 0:01:23, 24p /1.001
1: h264/AVC, English, 1080p24 /1.001 (16:9)
2: EAC3, English, 5.1 channels, 48kHz
3: EAC3, Spanish, 5.1 channels, 48kHz
Bitstream parsing for tracks 2 and 3 failed. <WARNING>
Demuxing these tracks may still produce correct results - or not. <WARNING>

And now:

eac3to Eac30.mkv 2: output.eac3 [-core] -> OK eac3 output
eac3to Eac30.mkv 2: output.ac3 [-core] -> ERROR

eac3to Eac30.mkv 3: output.eac3 [-core] -> OK eac3 output
eac3to Eac30.mkv 3: output.ac3 [-core] -> ERROR

Can extract both tracks but never decode/recode

EDIT:
Decoding track 3 with ffmpeg show always (from m2ts or mkv) the message: Dependent substream decoding is not implemented.
Then MkvToolNix don't extract the AC3 core, preserve the substream to convert 5.1 -> 7.1 even when eac3to show only 5.1 on mkv.
Remuxed the mkv to m2ts with tsMuxeR the track 3 is showed by eac3to like original Eac30.m2ts

Without free tools to decode EAC3 7.1 (always decode the core) the recommended option, by now, is extract the core from m2ts
eac3to Eac30.m2ts 3: output.eac3 -core
and rename to .ac3

[mini-moose]

thanks Madshi for the great tool!

I think there are a few things that may/could use some updating:

1) E-AC3 audios with embedded Core (mentioned before)
2) libFLAC (very old version used, maybe worth updating)
3) libDcaDec (I think it's part of ffmpeg now?)

[SeeMoreDigital]

To sumarize the problem here.

I found two kind of EAC3 streams detected by eac3to without core (2 English) or with core (3 Spanish) like is show here (sample (https://www.sendspace.com/file/dzj4qm)):
Hi @tebasuna51,

It would seem that most if not all our current tools are unable to correctly identify Dolby Digital Plus audio streams :(

Have you noticed that when you run your 'Eac30.m2ts' sample file through TSmuxerGUI that it reports the first audio track as having 17 channels. And when you run the same file through MediaInfo it's unable to detect the AC3 cores at all. Or identify that the second audio track has been encoded with 7.1 channels.

By contrast, my Oppo UDP-203 is able to identify the first audio track as being a 5.1 channel Dolby Digital Plus stream and second audio track as being a 7.1 channel Dolby Digital Plus stream. But neither of my LG 'smart' TV's (which support Dolby Digital Plus) are able to detect either of the audio streams at all (ie: total silence). But my Panasonic 'smart' TV can detect the first audio stream.


Cheers

[zeropc]

madshi, can you please bring back the feature to show the applied delay/ amount x delay could not be fixed in the log?

this was actually quite helpful when syncing/editing audio.

thanks :)

[Snowknight26]

It's still there.

[zeropc]

It's still there.

nope

eac3to v3.32
command line: "C:\Users\zeropc\Desktop\eac3to 3.32\eac3to.exe" "H:\xmen atmos.mkv" 2: "H:\x\xa1.thd" +18ms
------------------------------------------------------------------------------
MKV, 1 video track, 3 audio tracks, 29 subtitle tracks, 2:23:58, 24p /1.001
1: h265/HEVC, 3840x1600 24p /1.001 (12:5), 10 bits
2: TrueHD (Atmos), English, 7.1 channels, 48kHz
3: AC3, English, 5.1 channels, 640kbps, 48kHz
4: AAC, English, 2.0 channels, 48kHz, 9ms
"Commentary with director Bryan Singer and writer Simon Kinberg"
5: Subtitle (PGS), English, "Forced"
6: Subtitle (PGS), English, "SDH"
7: Subtitle (PGS), Spanish
8: Subtitle (PGS), French
9: Subtitle (PGS), French
10: Subtitle (PGS), Spanish
11: Subtitle (PGS), Danish
12: Subtitle (PGS), Dutch
13: Subtitle (PGS), Finnish
14: Subtitle (PGS), German
15: Subtitle (PGS), Italian
16: Subtitle (PGS), Norwegian
17: Subtitle (PGS), Swedish
18: Subtitle (PGS), Chinese
19: Subtitle (PGS), Chinese
20: Subtitle (PGS), Japanese
21: Subtitle (PGS), English, "Commentary"
22: Subtitle (PGS), Spanish, "Commentary"
23: Subtitle (PGS), French, "Commentary"
24: Subtitle (PGS), French, "Commentary"
25: Subtitle (PGS), Spanish, "Commentary"
26: Subtitle (PGS), Danish, "Commentary"
27: Subtitle (PGS), Dutch, "Commentary"
28: Subtitle (PGS), Finnish, "Commentary"
29: Subtitle (PGS), German, "Commentary"
30: Subtitle (PGS), Italian, "Commentary"
31: Subtitle (PGS), Japanese, "Commentary"
32: Subtitle (PGS), Norwegian, "Commentary"
33: Subtitle (PGS), Swedish, "Commentary"
[a02] Extracting audio track number 2...
[a02] Creating file "H:\x\xa1.thd"...
Video track 1 contains 207088 frames.
eac3to processing took 16 minutes, 6 seconds.
Done.

[hubblec4]

Hi madshi

Is it possible that eac3to scan the sup tracks and find forced sup's before I demux the sup track.

While eac3to scans the mpls (all tracks are listed) and for the sup tracks, you could add an info "forced sup's detected".
M2TS, 1 video track, 3 audio tracks, 10 subtitle tracks, 0:43:05, 24p /1.001
1: Chapters, 8 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
4: DTS Master Audio, German, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
5: DTS Master Audio, Spanish, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
6: Subtitle (PGS), English
7: Subtitle (PGS), English
8: Subtitle (PGS), Danish
9: Subtitle (PGS), Finnish
10: Subtitle (PGS), German, forced detected
11: Subtitle (PGS), Hindi
12: Subtitle (PGS), Norwegian
13: Subtitle (PGS), Spanish
14: Subtitle (PGS), Swedish
15: Subtitle (PGS), Turkish

Really great would be if you could add a new "virtual" forced-sup track to the track list. Demux the forced-sup brings only the forced sups.

M2TS, 1 video track, 3 audio tracks, 10 subtitle tracks, 0:43:05, 24p /1.001
1: Chapters, 8 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
4: DTS Master Audio, German, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
5: DTS Master Audio, Spanish, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
6: Subtitle (PGS), English
7: Subtitle (PGS), English
8: Subtitle (PGS), Danish
9: Subtitle (PGS), Finnish
10: Subtitle (PGS), German
11: Subtitle (PGS), Hindi
12: Subtitle (PGS), Norwegian
13: Subtitle (PGS), Spanish
14: Subtitle (PGS), Swedish
15: Subtitle (PGS), Turkish
16: Subtitle (PGS), German, forced

LG,
hubble

[tebasuna51]

2) libFLAC (very old version used, maybe worth updating)
You can replace libFLAC.dll with libFLAC_dynamic.dll (renamed) from
http://www.rarewares.org/files/lossless/flac_dll-1.3.2-git-20170314-icl.zip

3) libDcaDec (I think it's part of ffmpeg now?)
Yes, that was requested already.

It would seem that most if not all our current tools are unable to correctly identify Dolby Digital Plus audio streams :(

Have you noticed that when you run your 'Eac30.m2ts' sample file through TSmuxerGUI that it reports the first audio track as having 17 channels.
Yep.

Just by curiosity I check some tsMuxeR versions and the first audio track is:
tsMuxeR 1.8.8: track not recognized
tsMuxeR 2.6.9: channels 94
tsMuxeR 2.6.11 or 12: channels 91

What is your version than show 17 channels?.

madshi, can you please bring back the feature to show the applied delay/ amount x delay could not be fixed in the log?

this was actually quite helpful when syncing/editing audio.

I don't see differences between log in 3.31 and 3.32 with this command.

BTW is know than eac3to can't delay thd tracks.
Only decoding that delay can be applied.

Is it possible that eac3to scan the sup tracks and find forced sup's before I demux the sup track.

The first scan must be quick, a depth scan becomes slow.
When you demux these tracks the log show the forced captions found.

For me is enough, there are other tools to manage sup's.

[SeeMoreDigital]

Just by curiosity I check some tsMuxeR versions and the first audio track is:
tsMuxeR 1.8.8: track not recognized
tsMuxeR 2.6.9: channels 94
tsMuxeR 2.6.11 or 12: channels 91

What is your version than show 17 channels?.Hi @tebasuna51,

I'm using tsMuxeR GUI 2.6.12. But I've just noticed that if you drop the same 'Eac30.m2ts' file into the application, then remove it, and re-add it, the application reports different information...


Cheers (oh bugger... I've run out of beer :scared:)

[Q-the-STORM]

Now the same m2ts converted to mkv with MkvToolNix is show by eac3to like:

eac3to Eac30.mkv
------------------------------------------------------------------------------
MKV, 1 video track, 2 audio tracks, 0:01:23, 24p /1.001
1: h264/AVC, English, 1080p24 /1.001 (16:9)
2: EAC3, English, 5.1 channels, 48kHz
3: EAC3, Spanish, 5.1 channels, 48kHz
Bitstream parsing for tracks 2 and 3 failed. <WARNING>
Demuxing these tracks may still produce correct results - or not. <WARNING>

And now:

eac3to Eac30.mkv 2: output.eac3 [-core] -> OK eac3 output
eac3to Eac30.mkv 2: output.ac3 [-core] -> ERROR

eac3to Eac30.mkv 3: output.eac3 [-core] -> OK eac3 output
eac3to Eac30.mkv 3: output.ac3 [-core] -> ERROR

Can extract both tracks but never decode/recode
btw decode/recode should work after you demuxed eac3 and renamed to ac3....

So:
eac3to Eac30.mkv 2: output.eac3
then
rename to output.ac3
then
eac3to output.ac3 output2.wav -whatever


At least it works with my files that have eac3 tracks and no core. I always need to demux and rename, then decoding and recoding work.

That seems like a eac3to bug to me.

[hubblec4]

The first scan must be quick, a depth scan becomes slow.
When you demux these tracks the log show the forced captions found.

For me is enough, there are other tools to manage sup's.

I know but all other tools can't read sup's from the m2ts file. You have to extract them first.

I hope madshi has also the Bluray-specs and can do what makemkv does. There is not long scan to identify the forced sup's.

[Megalith]

Anyone know of a GUI that is still regularly updated? It seems like most of them have been abandoned.

[tebasuna51]

So:
eac3to Eac30.mkv 2: output.eac3
then
rename to output.ac3
then
eac3to output.ac3 output2.wav -whatever

Yep with track 2, with track 3 don't work.
And you don't need rename, eac3to identify the stream by the header not by extension, work fine also:

eac3to Eac30.mkv 2: output2.eac3
then
eac3to output2.eac3 output2.wav

With track 3:

eac3to Eac30.mkv 3: output3.eac3
then
eac3to output3.eac3 output3.wav -->
E-AC3, 7.1 channels, 0:01:23, 768kbps, 48kHz, dialnorm: -27dB
(core: AC3, 5.1 channels, 0:01:23, 448kbps, 48kHz, dialnorm: -27dB)
Removing AC3 dialog normalization...
Extracting E-AC3 core...
Decoding with libav/ffmpeg...
[libav] frame CRC mismatch <WARNING>
[libav] get_buffer() failed <WARNING>
The libav decoder reported error -22 while decoding. <ERROR>
Aborted at file position 262144. <ERROR>

You need:

ffmpeg -drc_scale 0 -i output3.eac3 -acodec pcm_s24le output3.wav
...
[eac3 @ 00000000006bf5e0] Dependent substream decoding is not implemented.
Update your FFmpeg version to the newest one from Git.
If the problem still occurs, it means that your file has a feature which has not been implemented.
...

Or directly from the mkv:

ffmpeg -drc_scale 0 -i Eac30.mkv -map 0:2 -acodec pcm_s24le output3.wav

with the same WARNING

[tebasuna51]

Anyone know of a GUI that is still regularly updated? It seems like most of them have been abandoned.

HdBrStreamExtractor is updated in MeGUI -> Tools -> HD Streams Extractor (https://forum.doom9.org/showthread.php?p=1821541#post1821541)

MeGUI v2771 [HD Streams Extractor] added support for HEVC video tracks (requires eac3to 3.32+)
[Muxer] added support for the h265 file extension

Or UsEac3to v1.2.7 (https://forum.doom9.org/showthread.php?t=145574)

All other GUI's are abandoned.

[mini-moose]

You can replace libFLAC.dll with libFLAC_dynamic.dll (renamed) from
[url]http://www.rarewares.org/files/lossless/flac_dll-1.3.2-git-2

Thanks. I know I can do that but be nice if it's updated in eac3to package too. Also, wasn't there some bug 1.3.2 and 1.3.2-1 was released? I don't see it in the flac website but it's on megui.

[stax76]

HdBrStreamExtractor is updated in MeGUI -> Tools -> HD Streams Extractor (https://forum.doom9.org/showthread.php?p=1821541#post1821541)



Or UsEac3to v1.2.7 (https://forum.doom9.org/showthread.php?t=145574)

All other GUI's are abandoned.

Last major feature megui implemented is x264 released 2004, it lacks anything major since then:

High DPI
VapourSynth
x265 full implementation
AV1
NVEnc
QSVEnc

[Atak_Snajpera]

I would like to suggest small cosmetic addition to eac3to. It would be cool if extension .265 was also supported during demuxing. Currently only .h265 is recognized.

[stax76]

There is also hvc due to most file extensions being 3 characters in Windows.

264 h264 avc
265 h265 hvc hevc

[tebasuna51]

Last major feature megui implemented is x264 released 2004...

??

Here we are talking about eac3to GUI's.

[tebasuna51]

The .h265 extracted with eac3to v3.32 from this sample (https://drive.google.com/open?id=0BzDk6Axq8AyJb1dFbEhMZUVCcFk) don't play fine.

Extracted with MkvExtract is ok.

[SeeMoreDigital]

The .h265 extracted with eac3to v3.32 from this sample (https://drive.google.com/open?id=0BzDk6Axq8AyJb1dFbEhMZUVCcFk) don't play fine.Just-so-you-know...

I downloaded your sample, transferred it to an USB pen-drive and can confirm that it plays fine in my LG television.


Cheers

[Megalith]

HdBrStreamExtractor is updated in MeGUI -> Tools -> HD Streams Extractor (https://forum.doom9.org/showthread.php?p=1821541#post1821541)



Or UsEac3to v1.2.7 (https://forum.doom9.org/showthread.php?t=145574)

All other GUI's are abandoned.

Thank you, MeGUI seems to be working great. It looks like I can finally move away from Clown_BD.

I am still confused about how eac3to handles Atmos tracks, though. Is an Atmos track the same thing as a TrueHD track, or is eac3to converting Atmos to a TrueHD track for compatibility reasons? Dolby's naming convention isn't helping me understand it any better.

[Megalith]

Well, looks like eac3to and MeGUI has failed, too. This is what an UHD HEVC file looks like when muxed to MKV. Only MKVToolNix appears to do it correctly. (Actually, maybe the problem lies with MKVToolNix in not handling separate/extracted files correctly...)

https://i.imgur.com/WvvaJHp.jpg

[tebasuna51]

I downloaded your sample, transferred it to an USB pen-drive and can confirm that it plays fine in my LG television.

The uploaded sample work fine, but if you extract the .h265 with eac3to and play it, even re-remuxed to mkv, fails (like the Megalith image).

[tebasuna51]

I am still confused about how eac3to handles Atmos tracks, though. Is an Atmos track the same thing as a TrueHD track, or is eac3to converting Atmos to a TrueHD track for compatibility reasons?

A Atmos track is a TrueHD track with extra metadata than only a receiver with Atmos support (and ceiling speakers) can understand.

The metadata remain in thd extracted by eac3to/mkvextract.
Standard receivers ignore that metadata.

[tebasuna51]

Another sample (https://mega.nz/#!CZQimT4I!VCjqq6boCs8O5yWB0LAp34g5X9Nev33V30Dhv29WIuM) with problems.

Remark the different size of extracted .h265

15/10/2017 22:03 177.693.906 Arrival UHD Sample.mkv_ 0.h265 (mkvextract)
15/10/2017 21:59 157.722.039 Arrival UHD Sample.mkv_1eng.h265 (eac3to)

[VAMET]

Dear madshi

I have used eac3to to identify my .mkv files and I have got such results:

MKV, 2 video tracks, 1 audio track, 2:15:58, 24p /1.001
1: h265/HEVC, English, 2160p24 /1.001 (16:9), 10 bits
2: h265/HEVC, English, 1080p24 /1.001 (16:9), 10 bits - Dolby Vision Enhancement Layer
3: DTS Master Audio, English, 7.1 channels, 24 bits, 48kHz (core: DTS, 5.1 channels, 1509kbps, 48kHz)

Source disk has got DTS:X audio track, why it's not identified as DTS:X, but DTS Master Audio? The next .mkv file, which I show results below, has got in source disk Dolby Atmos and in the .mkv file it's Dolby Atmos too, which is correctly identified.

MKV, 2 video tracks, 1 audio track, 2:34:39, 24p /1.001
1: h265/HEVC, English, 2160p24 /1.001 (16:9), 10 bits
2: h265/HEVC, English, 1080p24 /1.001 (16:9), 10 bits - Dolby Vision Enhancement Layer
3: TrueHD (Atmos), English, 7.1 channels, 48kHz, 32ms

My second question, why there is not core embedded in track 3 TrueHD (Atmos) like in first .mkv file for DTS Master Audio (core: DTS, 5.1 channels, 1509kbps, 48kHz)?

Thank you in advance for your support.

Sincerely

[tebasuna51]

why it's not identified as DTS:X, but DTS Master Audio?
Because eac3to can't detect DTS:X, maybe can be a feature request (not important for me, because can be extracted preserving DTS:X info, and can be decoded like standard DTS-MA)

why there is not core embedded in track 3 TrueHD (Atmos)
Because TrueHD don't have any 'core', like DTS-HD have.
The AC3 track interleaved with TrueHD track is only possible in m2ts container, in mkv must be a different track.

[tebasuna51]

Posts about E-AC3+Atmos moved to new thread E-AC3+Atmos (https://forum.doom9.org/showthread.php?t=174998)

Because eac3to can't recognize the tracks like Atmos.
BTW eac3to can decode them like standard 5.1.

[LigH]

When eac3to was "fixed" not to crash when Atmos metadata are contained, the fix only meant to properly ignore them.

[Music Fan]

But I believe you can't remove the Atmos part and keep only the True HD part, thus it's not ignored but treated as is.

[nevcairiel]

"Removing" would not be "ignoring". It just ignores if its present or not, and copies the entire TrueHD as-is. Removing it would require modifiying the entire TrueHD stream, since its not a separate sub-stream or anything like that, but actually integrated into the TrueHD stream itself.

[pcroland]

Hi! What's the difference between using and not using the -down16 option? I tried both way, but the output is still has 16bit bit-depth but the md5sum's are different: https://vgy.me/Tflfue.png

[LigH]

If you use the -down16 option, the precision of your input WAV is first reduced from 32 bit float down to 16 bit integer, before encoding it to AC3.

Omitting it, more or less the full input precision is encoded to AC3.

[pcroland]

And what do you suggest? Should I use it or omit it?

[LigH]

Why would you want to possibly reduce the precision, thus quality, in the middle of a conversion? If you omit it, the precision might be better than just 16 bits.

[pcroland]

Alright, thanks for the help :D

[hubblec4]

-logpids

For most Blurays match the PID list to the track info list,
but for this Bluray with 2 video streams and DTS Express not.
PID: 256/4113 ($ea), "VC-1", VC-1
PID: 256/4352 ($86), DTS Master Audio
PID: 256/4353 ($85), DTS Hi-Res
PID: 256/4354 ($86), DTS Master Audio
PID: 256/4355 ($81), "AC-3", AC3
PID: 256/4356 ($81), "AC-3", AC3
PID: 256/4357 ($81), "AC-3", AC3
PID: 256/4358 ($81), "AC-3", AC3
PID: 256/4608 ($90), Subtitle (PGS)
PID: 256/4609 ($90), Subtitle (PGS)
PID: 256/4610 ($90), Subtitle (PGS)
PID: 256/4611 ($90), Subtitle (PGS)
PID: 256/4612 ($90), Subtitle (PGS)
PID: 256/4613 ($90), Subtitle (PGS)
PID: 256/4614 ($90), Subtitle (PGS)
PID: 256/4615 ($90), Subtitle (PGS)
PID: 256/4616 ($90), Subtitle (PGS)
PID: 256/4617 ($90), Subtitle (PGS)
PID: 256/4618 ($90), Subtitle (PGS)
PID: 256/4619 ($90), Subtitle (PGS)
PID: 256/4620 ($90), Subtitle (PGS)
PID: 256/4621 ($90), Subtitle (PGS)
PID: 256/4622 ($90), Subtitle (PGS)
PID: 256/4623 ($90), Subtitle (PGS)
PID: 256/4624 ($90), Subtitle (PGS)
PID: 256/4625 ($90), Subtitle (PGS)
PID: 256/4626 ($90), Subtitle (PGS)
PID: 256/6656 ($a2), DTS Express
PID: 256/6657 ($a2), DTS Express
PID: 256/6912 ($ea), "VC-1", VC-1

M2TS, 2 video tracks, 9 audio tracks, 19 subtitle tracks, 2:36:07, 24p /1.001
1: Chapters, 80 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: VC-1, 480p24 /1.001 (3:2)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS-ES, 5.1 channels, 1509kbps, 48kHz)
5: DTS Hi-Res, French, 5.1 channels, 3018kbps, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
6: DTS Master Audio, German, 7.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
7: AC3, English, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
8: AC3 Surround, English, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
9: AC3 Headphone, English, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
10: AC3 Headphone, German, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
11: DTS Express, English, 2.0 channels, 192kbps, 48kHz
12: DTS Express, English, 2.0 channels, 192kbps, 48kHz
13: Subtitle (PGS), English
14: Subtitle (PGS), French
15: Subtitle (PGS), German
16: Subtitle (PGS), Turkish
17: Subtitle (PGS), English
18: Subtitle (PGS), French
19: Subtitle (PGS), German
20: Subtitle (PGS), English
21: Subtitle (PGS), French
22: Subtitle (PGS), German
23: Subtitle (PGS), English
24: Subtitle (PGS), French
25: Subtitle (PGS), English
26: Subtitle (PGS), English
27: Subtitle (PGS), French
28: Subtitle (PGS), German
29: Subtitle (PGS), English
30: Subtitle (PGS), French
31: Subtitle (PGS), German


The change of the stream order breaks simple parsing.

@madshi
Could you add the eac3to stream number at the end of the PID info line for each PID?

[pstn]

Could anyone explain what is going on with this pls?

1) 00003.mpls, 00009.m2ts+00004.m2ts, 1:51:42
- Chapters, 12 chapters
- h264/AVC, 1080i50 (16:9)
- DTS Master Audio, Norwegian, multi-channel, 48kHz
- DTS Master Audio, French, multi-channel, 48kHz

since the 00009.m2ts file is a 10frame (1/2 second) black screen with no audio track, followed by the movie, should I be doing the demuxing directly on the 00004.m2ts (the actual movie) file instead?

thanks

[LigH]

I would assume that this is an "authoring trick" prefix (e.g. to catch a "double PREV button", as I remember from DVD authoring). So, yes, try to work with the 00004.m2ts only, but check well if audio is in sync in your result. (No warranties, though.)

[tebasuna51]

Yes, you must use the 00004.m2ts (I do not expect problems with audio sync).

If you try extract tracks from the mpls eac3to crash because no audio in first m2ts.
The mpls is only usefull to extract the chapters, but you need correct the times with the duration of first m2ts.

[hubblec4]

Could anyone explain what is going on with this pls?

1) 00003.mpls, 00009.m2ts+00004.m2ts, 1:51:42
- Chapters, 12 chapters
- h264/AVC, 1080i50 (16:9)
- DTS Master Audio, Norwegian, multi-channel, 48kHz
- DTS Master Audio, French, multi-channel, 48kHz

since the 00009.m2ts file is a 10frame (1/2 second) black screen with no audio track, followed by the movie, should I be doing the demuxing directly on the 00004.m2ts (the actual movie) file instead?

thanks

I don't know what your aim is.
But you could try my chapterEditor.
Such disturbing m2ts can be ignored/removed and the chapter times will be recalculated automatically.

[mini-moose]

But you could try my chapterEditor.
Such disturbing m2ts can be ignored/removed and the chapter times will be recalculated automatically.

Interesting, didn't know that's a feature in CE!

Please see pm hubblec4.

[pcroland]

D:\_encode\s09
λ eac3to.exe 00.eac3 00_hd.eac3 -6006ms
E-AC3, 2.0 channels, 0:23:43, 224kbps, 48kHz, dialnorm: -24dB
Removing E-AC3 dialog normalization...
Applying (E-)AC3 delay...
A remaining delay of +10ms could not be fixed.
Creating file "00_hd.eac3"...
eac3to processing took 1 second.

Sampling rate : 48.0 kHz
Frame rate : 187.500 FPS (256 spf)

With 187.5 fps the length of one frame is 5.33ms, not 32ms, isn't it? So the "remaining delay" that "cannot be fixed" should be less than 3ms (2.66ms) in the worst case with E-AC3 (that has 256spf).

[kron]

Hi, friends,

is it a problem? I used the version 31 and everything was ok, but now using v32 shows this:

eac3to v3.32
command line: "C:\Program Files\eac3to\eac3to.exe" "C:\Users\Desktop\Audio HD\Overdrive.2017.BluRay.1080p.DTS-HD.MA.5.1.AVC.REMUX-FraMeSToR_track2_eng.dts" "C:\Users\Desktop\Audio HD\Overdrive.2017.BluRay.1080p.DTS-HD.MA.5.1.AVC.REMUX-FraMeSToR_track2_eng.dts_.flac" -progressnumbers -log="C:\Program Files\eac3to\UsEac3To.log"
------------------------------------------------------------------------------
DTS Master Audio, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
Decoding with libDcaDec DTS Decoder...
libDcaDec reported the warning "XLL output not lossless". <WARNING>
Encoding FLAC with libFlac...
Creating file "C:\Users\Desktop\Audio HD\Overdrive.2017.BluRay.1080p.DTS-HD.MA.5.1.AVC.REMUX-FraMeSToR_track2_eng.dts_.flac"...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 10 minutes, 51 seconds.
Done.

Thanks for helping.

[LigH]

Forum attachments take a while to get approved. Could you please use an image hoster, or simply quote full CLI parameters and the error message from the console?

As uncomplicated image hoster, I would recommend e.g. https://frupic.frubar.net; and for screenshots from a console window, I would prefer PNG over JPG.

[tebasuna51]

It's not a problem.

That was repported before with previous eac3to versions.
Sometimes libdcadec show that WARNING, I suspect is when a DTS frame don't need extra info to convert in lossless (like silent frames).

[kron]

Forum attachments take a while to get approved. Could you please use an image hoster, or simply quote full CLI parameters and the error message from the console?

As uncomplicated image hoster, I would recommend e.g. https://frupic.frubar.net; and for screenshots from a console window, I would prefer PNG over JPG.

Thanks for answering, I'll remember this.

It's not a problem.

That was repported before with previous eac3to versions.
Sometimes libdcadec show that WARNING, I suspect is when a DTS frame don't need extra info to convert in lossless (like silent frames).

Reading some posts back, I've seen it was corrected in v31, do you think it will be also done in this 32?

BTW, do you friends still use Arcsoft to convert DTS to flac or have just forget it?

Sorry for so many questions, but I believe in the good will here.

Thanks for the attention.

[LigH]

If there is no "regression" (accidental loss), then all the fixes introduced in v31 should persist in v32.

The Arcsoft decoder is commercial (and of course not shipped together with eac3to), and the free libavcodec decoder is just as good (can be bit-exact as well where possible, or at least with negligible difference), so I believe there is little reason today to insist in Arcsoft. I believe it also had some issues providing different feature sets per version... but I never used it.

[mtamimi]

Guys, any update on that issue with some decrypted UHD discs?!

Thnx

[tebasuna51]

The Arcsoft decoder is commercial (and of course not shipped together with eac3to), and the free libavcodec decoder is just as good (can be bit-exact as well where possible, or at least with negligible difference)

You are right, sometimes there are a little diference (last bit +- 1, some rounding error) between ArcSoft and libdcadec decoders.

But in my tests always the decoded with libdcadec are bitexact with the source wav, and the decoded with ArcSoft is the unexact.

Now ArcSoft is only necessary to decode DTS-Express, not supported in last libdcadec.dll. But you can use ffmpeg than now include a more recent libdcadec.

@kron
Use the 'Save Log' button in UsEac3to and attach the log file here.
More easy, and save space in forum, instead a image capture sometimes incomplete.

[Boulder]

Not directly eac3to-related but what kind of commandline should be used for ffmpeg when piping from eac3to in case of DTS-Express to WAV?

[sneaker_ger]

Piping from eac3to implies eac3to has to decode DTS-Express. Which it can't. You need to open the file directly with ffmpeg.

[Boulder]

Can't you pipe the extracted DTS file? Otherwise it's two passes (extract core, decode file with ffmpeg) at least if the playlist has multiple m2ts files.

[tebasuna51]

Yep, you need two passes with multiple m2ts

[kron]

Hi, guys, sorry for the mistakes, let's see if I did right this time.

eac3to v3.32
command line: "C:\Program Files\eac3to\eac3to.exe" "C:\Users\Desktop\Audio HD\Overdrive.2017.BluRay.1080p.DTS-HD.MA.5.1.AVC.REMUX-FraMeSToR_track2_eng.dts" "C:\Users\Desktop\Audio HD\Overdrive.2017.BluRay.1080p.DTS-HD.MA.5.1.AVC.REMUX-FraMeSToR_track2_eng.dts_.flac" -progressnumbers -log="C:\Program Files\eac3to\UsEac3To.log"
------------------------------------------------------------------------------
DTS Master Audio, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
Decoding with libDcaDec DTS Decoder...
libDcaDec reported the warning "XLL output not lossless". <WARNING>
Encoding FLAC with libFlac...
Creating file "C:\Users\Desktop\Audio HD\Overdrive.2017.BluRay.1080p.DTS-HD.MA.5.1.AVC.REMUX-FraMeSToR_track2_eng.dts_.flac"...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 10 minutes, 51 seconds.
Done.

Thanks for the patience.

[mtamimi]

I need your help regarding this a BD that I have...

1. Listing the playlist works...But it reports 2 similar playlists, the 2nd one is the correct one since the stream 9031 is the beginning of the movie...
eac3to.exe E:\

1) 00108.mpls, 1:29:23
[14993+4761+6837+15438+15239+17347+25303+268+23180+16207+86+15862+1470+6953+15999+12333+23096+24749+21465+25620+13396+29818+20351+8413+12583+20631+21951+25683+21820+23783+10226+6888+29843+12750+13212+0+14298+20771+9415+21118+23276+8535+14235+18830+138+14053+17728+30132+14167+14332+18422+31252+3096+26217+21108+18946+17707+7113+13953+22220+9031].m2ts
- Chapters, 10 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48kHz
- AC3, English, stereo, 48kHz

2) 01002.mpls, 1:29:23
[9031+22220+13953+7113+17707+18946+21108+26217+3096+31252+18422+14332+14167+30132+17728+14053+138+18830+14235+8535+23276+21118+9415+20771+14298+0+13212+12750+29843+6888+10226+23783+21820+25683+21951+20631+12583+8413+20351+29818+13396+25620+21465+24749+23096+12333+15999+6953+1470+15862+86+16207+23180+268+25303+17347+15239+15438+6837+4761+14993].m2ts
- Chapters, 10 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48kHz
- AC3, English, stereo, 48kHz

2. Selecting the 2nd playlist reports this
eac3to.exe E:\ 2)

M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 1:29:19, 24p /1.001
1: Chapters, 10 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
4: AC3, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
5: Subtitle (PGS), English

3. Encoding the audio to DD 5.1 @ 640 fails with error...Can you guys help?!
eac3to.exe E:\ 2) 3:Audio.ac3 -640

M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 1:29:19, 24p /1.001
1: Chapters, 10 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
4: AC3, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
5: Subtitle (PGS), English
a03 Extracting audio track number 3...
a03 Decoding with libDcaDec DTS Decoder...
a03 libDcaDec reported the warning "XLL output not lossless".
a03 Remapping channels...
a03 Encoding AC3 <640kbps> with libAften...
a03 Creating file "Audio.ac3"...
a03 0:06:26 The pes parser received incomplete data.
a03 0:06:26 Skipping dirty PES bytes...
a03 0:06:26 Skipping dirty PES bytes...
a03 0:06:26 Skipping dirty PES bytes...
a03 0:06:26 Skipping dirty PES bytes...
a03 This track is not clean.
a03 libDcaDec reported the error "Invalid bitstream format".
Aborted at file position 1655990272.


P.S. Handbrake handles this disc w/o issues at all. It can encodes both the audio and video w/o issues...

:thanks:

[tebasuna51]

I never see that, 2 mpls with the same m2ts just in reverse order.

The problem seems located at time 0:06:26
You can load each m2ts in [9031+22220+13953+7113+17707+18946...] until reach that point.
Maybe is a corrupt m2ts or without the same audio stream than crash eac3to.

I don't know for what Handbrake work, maybe skip the problems without crash.
Are audio/video in sync after the point 0:06:26?

[hubblec4]

I never see that, 2 mpls with the same m2ts just in reverse order.


A more strange Bluray for such behaviour is: BASTILLE_DAY.

There are more then 500 playlists with the same m2ts files in different orders.

@mtamimi
Looks like the m2ts files are corrupt.

[Boulder]

A more strange Bluray for such behaviour is: BASTILLE_DAY.

There are more then 500 playlists with the same m2ts files in different orders.

They're not strange, that's a form of copy protection.

[mtamimi]

I never see that, 2 mpls with the same m2ts just in reverse order.

The problem seems located at time 0:06:26
You can load each m2ts in [9031+22220+13953+7113+17707+18946...] until reach that point.
Maybe is a corrupt m2ts or without the same audio stream than crash eac3to.

I don't know for what Handbrake work, maybe skip the problems without crash.
Are audio/video in sync after the point 0:06:26?

Handbrake reports 2 playlists just like eac3to, however Handbrake also reports only 4 chapters for the 1st playlist, and 10 for the 2nd one. Hence the 2nd one is the correct one. I did encode the disc in Handbrake and the audio is in sync, no issues at all...

I was told that eac3to encodes audio better than Handbrake, so I thought I will use eac3to instead of Handbrake to encode the audio...

Any suggestions?

[mtamimi]

@mtamimi
Looks like the m2ts files are corrupt.

So how come Handbrake successfully processed the whole disc?!

[kron]

Hey, guys, is there someway to be sure the audio converted (using eac3to) from DTS HDMA to flac is really lossless?

I mean a program to show the result of such conversion?

:thanks:

[LigH]

Any compressed audio format (DTS-HD as well) is first decoded to uncompressed PCM samples. And this uncompressed audio is then compressed to FLAC.

The encoding to FLAC is lossless, certainly. The name already tells you: Free Lossless Audio Codec.

[tebasuna51]

So how come Handbrake successfully processed the whole disc?!

Some soft can skip a error and continue.
eac3to abort with a error.

Maybe is a unique DTS frame corrupt and is only 11 ms lossed.
And the async is unnoticeable.

You can try extract only the dts:

eac3to.exe E:\ 2) 3: Audio.dts

without decode maybe eac3to do the job ok.
And after encode to AC3 with ffmpeg (preferred over libaften):

ffmpeg -i Audio.dts -acodec ac3 -ab 640k -center_mixlev 0.707 Audio.ac3

[mtamimi]

Some soft can skip a error and continue.
eac3to abort with a error.

Maybe is a unique DTS frame corrupt and is only 11 ms lossed.
And the async is unnoticeable.

You can try extract only the dts:

eac3to.exe E:\ 2) 3: Audio.dts

without decode maybe eac3to do the job ok.
And after encode to AC3 with ffmpeg (preferred over libaften):

ffmpeg -i Audio.dts -acodec ac3 -ab 640k -center_mixlev 0.707 Audio.ac3

eac3to.exe E:\ 2) 3:Audio.DTSMA

a03 0:27:28 Skipping dirty PES bytes...
a03 0:27:28 Skipping dirty PES bytes...
a03 0:27:28 Skipping dirty PES bytes...
a03 0:27:28 Skipping dirty PES bytes...
a03 0:27:28 Skipping dirty PES bytes...
a03 0:27:28 Skipping dirty PES bytes...
a03 0:27:28 Skipping dirty PES bytes...
a03 0:27:28 Skipping dirty PES bytes...
a03 0:27:28 Skipping dirty PES bytes...
a03 0:27:28 Skipping dirty PES bytes...
a03 0:27:28 Skipping dirty PES bytes...


I kept getting hundreds of messages similar to the above, so I halted eac3to...What do you think?!

[tebasuna51]

I kept getting hundreds of messages similar to the above, so I halted eac3to...What do you think?!

Then is bug in eac3to or corrupt data in source.

In order to identify the problem you can test the first m2ts with, for instance:

eac3to.exe E:\BDMV\STREAM\22220.m2ts 2: Audio.dts

and upload a small m2ts than show the problem to http://eac3to.bugs.madshi.net/

[pstn]

Does anyone know of a way to demux the whole playlist instead of just the first title in the playlist ? (for blurays of TV series)

[tebasuna51]

eac3to show the mpls ordered by duration (maybe is not by the chapters order) and you can select the desired playlist.

In the sample before https://forum.doom9.org/showthread.php?p=1823609#post1823609

eac3to E:\ 2)

open the second playlist.

[pstn]

eac3to show the mpls ordered by duration (maybe is not by the chapters order) and you can select the desired playlist.

In the sample before https://forum.doom9.org/showthread.php?p=1823609#post1823609

eac3to E:\ 2)

open the second playlist.

sorry bud, I think I used the wrong terms, what I meant was how do I demux all .mpls at once from a disc? instead of going

"eac3to C:/ejfslflsifsihf 1) -demux"
"eac3to C:/ejfslflsifsihf 2) -demux"
"eac3to C:/ejfslflsifsihf 3) -demux"
"eac3to C:/ejfslflsifsihf 4) -demux"

[Music Fan]

You can put all these commands in a bat, it will execute them one after the other.

[Revgen]

Feature Request:

Detect DTS:X audio streams.

[XMEN3]

Another sample (https://mega.nz/#!CZQimT4I!VCjqq6boCs8O5yWB0LAp34g5X9Nev33V30Dhv29WIuM) with problems.

Remark the different size of extracted .h265

15/10/2017 22:03 177.693.906 Arrival UHD Sample.mkv_ 0.h265 (mkvextract)
15/10/2017 21:59 157.722.039 Arrival UHD Sample.mkv_1eng.h265 (eac3to)
found problem also with last UHD pirates and man of steel ectracted without problem but video size is 2gb less and have artefacts

[pstn]

You can put all these commands in a bat, it will execute them one after the other.

thank you

[manuelin]

Other samples of h265 videos that eac3to extracts its video track with artifacts:
Baby Driver (https://mega.nz/#!CIwGUACQ!4N5yUhUWCdigglWZzYPHksd-GoOiq2Gt7RBFMcjPHNw)
Pirates of the Caribbean: Dead Men Tell No Tales (https://mega.nz/#!qE4XEZxZ!78-a2oM88Kr6-eMXSpdSnFE9jA0PsxZZZd8HWr9xsPY)
Independence Day (https://mega.nz/#!qdwmVJQA!9D0ZzVFNFZVj3gJjelAWsBPlCJcz-5MM9A6Zsihw68k)

I have also seen that this happens with The Fifth Element and Mad Max: Fury Road.

[madshi]

Thanks for the samples, everyone. Could you please confirm that this test build fixes the HEVC/h265 issues?

http://madshi.net/eac3toHevcTest.rar

[Thunderbolt8]

Thanks for the samples, everyone. Could you please confirm that this test build fixes the HEVC/h265 issues?

http://madshi.net/eac3toHevcTest.rarwere you able to find enough documentation about HEVC specs?

[manuelin]

Thanks for the samples, everyone. Could you please confirm that this test build fixes the HEVC/h265 issues?

http://madshi.net/eac3toHevcTest.rar

Thank you very much madshi, I have made a demux of the UHD disk of Baby Driver and Mad Max and now everything works perfect, there are no more artifacts.

Sorry for my bad english and thank you very much again!

[madshi]

Glad to hear that! I suppose you could also remux/demux with mkvtoolnix, so I'm not sure how useful the functionality in eac3to really is. But eac3to can change HEVC bitstream framerate on the fly (e.g. 23.976 <-> 24.000), not sure if other tools can do that.

Anyway, I'll probably wait until next weekend, to give others a chance to double check the test build, too, then I'll release a new official build.

[Atak_Snajpera]

Works here as well.

Ps. Madshi do not forget to add .264/.265 extensions as valid when demuxing video stream please :)

[mtamimi]

regarding eac3to capability of listing the playlists, is there a chance that it misses some of the extras on a standard BD or UHD?

Thnx

[hubblec4]

Glad to hear that! I suppose you could also remux/demux with mkvtoolnix, so I'm not sure how useful the functionality in eac3to really is. But eac3to can change HEVC bitstream framerate on the fly (e.g. 23.976 <-> 24.000), not sure if other tools can do that.

Your demux function is more than important!!!!

Its the only way to get proper streams from seamless branching Blurays, and for Audios to remove the "dialnorm".

[madshi]

regarding eac3to capability of listing the playlists, is there a chance that it misses some of the extras on a standard BD or UHD?
IIRC, there's currently a limit set at 15 minutes. Any playlists shorter than 15 minutes are not shown.

Your demux function is more than important!!!!

Its the only way to get proper streams from seamless branching Blurays, and for Audios to remove the "dialnorm".
Yes, of course. I was just wondering how useful the demuxing of HEVC video streams was. But it shouldn't harm (when working correctly), in any case, and maybe the framerate rewriting capability could be useful.

[hubblec4]

I was just wondering how useful the demuxing of HEVC video streams was. But it shouldn't harm (when working correctly), in any case, and maybe the framerate rewriting capability could be useful.

For the moment I have not seen a seamless branching UHD-Bluray, but it will be come and then we need the HEVC-demux function.

[Funcha]

Hi all. So just to make sure I'm getting all my information straight, if I use eac3to version 3.31 or 3.32 and convert a DTS HD-Master Audio file with dcadec I may or may not get the "XXL output not lossless" message but the same extensions in the master audio file will still properly get decoded to wav just like with Arcsoft?

I used to use Arcsoft decoder exclusively, but upon a new system build I'm having a bizarre error where 24-bit files are being decoded as a bit longer than the original resulting in a slow, muddled sounding file out of sync with the original. Am hoping dcadec can be my fix for this issue but definitely don't want to lose the lossless parts of the files, if possible.

[tebasuna51]

Anyway, I'll probably wait until next weekend, to give others a chance to double check the test build, too, then I'll release a new official build.
Thanks madshi.
The .h265 files extracted with the new eac3to version seems play fine now.

But I don't know the h265 format to confirm than is bug free.
About the size of extracted streams:

Samples/Size eac3to3.32 eac3to3.33B mkvextract greater
------- ----------- ----------- ----------- --------
Sample1 157.722.039 177.704.162 177.693.906 1,000058
Sample2 223.059.063 247.715.510 247.705.342 1,000041

Now the filesizes are a little greater than extracted by mkvextract but I don't know if it can be a problem.

[tebasuna51]

@Funcha
https://forum.doom9.org/showthread.php?p=1823550#post1823550

[madshi]

Thanks madshi.
The .h265 files extracted with the new eac3to version seems play fine now.

But I don't know the h265 format to confirm than is bug free.
About the size of extracted streams:

Samples/Size eac3to3.32 eac3to3.33B mkvextract greater
------- ----------- ----------- ----------- --------
Sample1 157.722.039 177.704.162 177.693.906 1,000058
Sample2 223.059.063 247.715.510 247.705.342 1,000041

Now the filesizes are a little greater than extracted by mkvextract but I don't know if it can be a problem.
eac3to's h264/h265 bitstream parser requires there to be AUDs (access unit delimiters) between each frame. (UHD) Blu-Rays do have AUDs in the video bitstream. However, mkvtoolnix removes the AUDs when muxing the bitstream. When demuxing again, mkvextract doesn't add them back in. eac3to adds AUDs when demuxing MKVs, which is why eac3to's demux size is slightly bigger. That's nothing to worry about. It goes as far as eac3to not being able to parse the video bitstream extracted by mkvextract at all because the AUDs are missing, which makes the video unparsable to eac3to, because without AUDs eac3to doesn't know where one video frame starts and ends exactly. eac3to can parse the bitstream within MKV files without AUDs because the MKV container information defines where one video frame starts and ends.

A small size change could also come from rewriting the fps value. One sample I saw had 480000/20020 stored which eac3to replaces with 24000/1001. Such a replacement can in theory change the size of the video bitstream because the numbers are stored in a weird bitstream format.

[Music Fan]

Are you talking about decrypted or non crypted UHD discs ?

@ tebasuna51 : you often write than instead of that and nearly always forget the word it before is, no offense ;)

[madshi]

Are you talking about decrypted or non crypted UHD discs ?
What is the difference?

[Music Fan]

If the original disc is not crypted, you don't have to decrypt it.

[madshi]

Ok, but what does that have to do with eac3to? eac3to does not decrypt.

[Music Fan]

I'm simply astonished that you talk about UHD demuxing, I didn't know that these discs could be copied.

[madshi]

You may want to google "DeUHD".

[Music Fan]

Thanks, this tool doesn't seem evoked in the decrypting part of the forum ;
https://forum.doom9.org/forumdisplay.php?s=&f=9&page=1&pp=25&sort=lastpost&order=desc&daysprune=75

[mtamimi]

IIRC, there's currently a limit set at 15 minutes. Any playlists shorter than 15 minutes are not shown.


Please, is it possible to allow us to set that value?! A lot of extras are actually less than 15 minutes. :thanks:

[hubblec4]

Please, is it possible to allow us to set that value?! A lot of extras are actually less than 15 minutes. :thanks:

This limitation is useful!

You can load each mpls separate with eac3to or
you can use my chapterEditor BD2mkv which shown more/all mpls(depend on the settings).

[Music Fan]

And also tsMuxer.

[mtamimi]

This limitation is useful!

You can load each mpls separate with eac3to or
you can use my chapterEditor BD2mkv which shown more/all mpls(depend on the settings).

Can I ask you why you think this limitation is useful?

Loading mpls is not a good idea since most of the time we are talking about multiple files w/o knowing where to start from. Just scanning for titles like 1 minute long titles will help us to demux those tracks also...

Regarding your tools, I appreciate what you do, I don't know much about them and how reliable they are compared to eac3to and MakeMKV, but I think using one app -i.e. eac3to- to do everything that I want for UHD BD/BD is much more convenient...

That should not stop me from wishing you all the best for your hard work, it is really good to see what you are doing...

[mtamimi]

And also tsMuxer.
Sorry was this a suggestion for the extra playlists?

If that's the case, does it support UHD? Is it still active? It looks dead to me...

[Music Fan]

Sorry was this a suggestion for the extra playlists?
Yes.

If that's the case, does it support UHD?
I don't know, I never tried, I was thinking to Blu-ray's mpls actually.

[mtamimi]

Yes.
I don't know, I never tried, I was thinking to Blu-ray's mpls actually.

MakeMKV handles regular Blu-Rays w/o issues, and is able to extract all extra features, and is active. I am a paying customer. I don't need tsMuxer for that especially since it was last updated back in 2014, however the fact that it still does work for old Blu-Rays is great.

eac3to is totally different, very powerful audio conversion utility that is also capable of demuxing FHD and UHD discs...It just misses extracting all extra features from UHD and FHD discs...

:thanks:

[Bandits]

Please, is it possible to allow us to set that value?! A lot of extras are actually less than 15 minutes. :thanks:

I asked about this a long time ago. I would find this useful to grab an occasional small clip on a disc without changing programs.

The normal 15 minute restriction is great to limit playlists but having a choice to show everything would be amazing.

A switch option maybe? -allplaylists

[LigH]

Or "-minplaytime", to customize the lower threshold.

[hubblec4]

Can I ask you why you think this limitation is useful?

Some Blurays have so many extra playlists, which are all ignored.
OK, the time should/could be changeable.


That should not stop me from wishing you all the best for your hard work, it is really good to see what you are doing...
Thanks a lot.

[madshi]

eac3to v3.33 released

http://madshi.net/eac3to.zip

* fixed a couple of HEVC bitstream bugs
* added new parameter "-minPlayListRuntime=0:15:00"
* added new parameter "-maxPlayListRuntime=4:00:00"
* added support for ".264" and ".265" file extensions
Thanks to mtamimi for "sponsoring" the new parameters.

[mtamimi]

eac3to v3.33 released

http://madshi.net/eac3to.zip

* fixed a couple of HEVC bitstream bugs
* added new parameter "-minPlayListRuntime=0:15:00"
* added new parameter "-maxPlayListRuntime=4:00:00"
* added support for ".264" and ".265" file extensions
Thanks to mtamimi for "sponsoring" the new parameters.

You are awesome, thank you very much :thanks: :thanks: :thanks:

[mtamimi]

I am getting this error message for both UHD and regular Blu-Ray discs
Source file "4k-test2.ts" not found.
I am not using the new parameters...

[act.anon]

eac3to v3.33 released

http://madshi.net/eac3to.zip

* fixed a couple of HEVC bitstream bugs
* added new parameter "-minPlayListRuntime=0:15:00"
* added new parameter "-maxPlayListRuntime=4:00:00"
* added support for ".264" and ".265" file extensions
Thanks to mtamimi for "sponsoring" the new parameters.

Receiving the error:
"Source file "4k-test2.ts" not found."

[Megalith]

>HEVC bitstream bugs

Have these been elaborated on?

[madshi]

I am getting this error message for both UHD and regular Blu-Ray discs

I am not using the new parameters...
Sorry, that was me being stupid. I've updated the zip, please re-download.

>HEVC bitstream bugs

Have these been elaborated on?
One bug was reported earlier in this thread (graphical corruption when demuxing certain files). The other one is this:

http://bugs.madshi.net/view.php?id=512

[Thunderbolt8]

eac3to's h264/h265 bitstream parser requires there to be AUDs (access unit delimiters) between each frame. (UHD) Blu-Rays do have AUDs in the video bitstream. However, mkvtoolnix removes the AUDs when muxing the bitstream. When demuxing again, mkvextract doesn't add them back in. eac3to adds AUDs when demuxing MKVs, which is why eac3to's demux size is slightly bigger. That's nothing to worry about. It goes as far as eac3to not being able to parse the video bitstream extracted by mkvextract at all because the AUDs are missing, which makes the video unparsable to eac3to, because without AUDs eac3to doesn't know where one video frame starts and ends exactly. eac3to can parse the bitstream within MKV files without AUDs because the MKV container information defines where one video frame starts and ends.whats the real world advantage/disadvantage of keeping or removing the AUDs? when using a PC or using stand alone hardware? in which situations?

[hubblec4]

eac3to v3.33 released

http://madshi.net/eac3to.zip


* added new parameter "-minPlayListRuntime=0:15:00"
* added new parameter "-maxPlayListRuntime=4:00:00"


Many many thanks for your continuous work on eac3to.

The time format is strictly limited to h:mm:ss? or is hh:mm:ss also working?

[madshi]

whats the real world advantage/disadvantage of keeping or removing the AUDs? when using a PC or using stand alone hardware? in which situations?
The same always applied to h264/AVC. Did you ever worry about it? If not, why worry about it now?

The time format is strictly limited to h:mm:ss? or is hh:mm:ss also working?
hh:mm:ss should also work.

[madshi]

eac3to v3.34 released

http://madshi.net/eac3to.zip

* fixed: "min/maxPlayListRuntime" worked for listing, but not processing

[mtamimi]

Tested and working as expected :)

:thanks:

[AYColumbia]

Thank you madshi. Any plans to update the help to include the newer parameters? Thanks again.

[hubblec4]

Thanks for your info and the new version!!

[hubblec4]

Mmh,
I tested the new version3.34 and I get no mpls list, with eac3to3.33 works fine.

-minPlayListRuntime=0:10:00 -maxPlayListRuntime=4:00:00

[madshi]

Mmh,
I tested the new version3.34 and I get no mpls list, with eac3to3.33 works fine.

-minPlayListRuntime=0:10:00 -maxPlayListRuntime=4:00:00
Argh. There are so many ways to use eac3to that it's hard to test them all properly. I'll fix this next week. Anyway, you can use it like "eac3to . -minPlayListRuntime=0:10:00 -maxPlayListRuntime=4:00:00" for now to workaround the problem. Or alternatively just don't use "maxPlayListRuntime" at all, then there's no problem.

[jpsdr]

Do you intend by any chance to update some " old DLL" (ffmpeg, flac, ...) to newer versions ?
For exemple, dcadec is now implemented into ffmpeg, so all improvements of it are not anymore in the standalone version.

[wiggaz]

I still get a corrupt video (Passengers) with the latest version (3.34). Am I missing something?
Thanks in advance.

[madshi]

I still get a corrupt video (Passengers) with the latest version (3.34). Am I missing something?
Can you post a small video extract with which I could reproduce the problem? Thanks!

[tormento]

Sometimes I get Invalid bitstream format error with DTS to AC3 conversion and I have to use ffmpeg.

I want to supply you an example (https://ufile.io/ifxhm), so probably a fix will be out.

eac3to.exe eng.dts eng.ac3
DTS Master Audio, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
Decoding with libDcaDec DTS Decoder...
Remapping channels...
Encoding AC3 <640kbps> with libAften...
Creating file "eng.ac3"...
libDcaDec reported the error "Invalid bitstream format".
Aborted at file position 372244480.

[wiggaz]

Can you post a small video extract with which I could reproduce the problem? Thanks!
First of all, thanks for your answer.
Since I thought maybe it was the source corrupted, I tried muxing the m2ts directly and it was flawless, so it couldn't be the disc.
And that's why there are two sample: one muxing the .h265 extract with eac3to [corrupted], and the second muxing directly the m2ts (only the video of course) [flawless].
They are 120s each.

http://www.mediafire.com/file/ts3fsbjgzmrwwzc/eac3to_sample-001.mkv
http://www.mediafire.com/file/6x3tlidcloo1b30/mkvtoolnix_sample-001.mkv


eac3to is update to 3.34 with Useac3to at 1.28 and mkvtoolnix is update to 18.0.0.
Thanks in advance again.

[madshi]

Thanks. But what I really need is to be able to reproduce the problem on my PC, so I can double check if the problem is really fixed. So I need an extract of the m2ts file.

[wiggaz]

Thanks. But what I really need is to be able to reproduce the problem on my PC, so I can double check if the problem is really fixed. So I need an extract of the m2ts file.

The second file is the m2ts file muxed into mkv, with all the tracks disabled except of the video.
Or should I give you the mux with all the tracks, divided into a small piece (120sec like the others) with mkvtoolnix?

[madshi]

I don't need the mux, I need the original m2ts file (a small part of it)!

[sneaker_ger]

@wiggaz
You can use DGSplit (http://rationalqm.us/dgsplit/dgsplit12.zip) to cut e.g. 50 MB from the beginning of the m2ts.

[madshi]

That should work, but only if the corruption is right at the start of the movie?

[wiggaz]

Here's, splitted with DGSplit as suggested (thx sneaker_ger): http://www.mediafire.com/file/24om27uq5bukho3/00001_0.m2ts

And this is an example: http://thumbs2.imagebam.com/27/45/00/0beea0667232263.jpg (http://www.imagebam.com/image/0beea0667232263)

[madshi]

Thanks! And the problem also occurs with this small m2ts file, correct? (Don't have time to test right now.)

[sneaker_ger]

@wiggaz
I can't reproduce the problem with the "00001_0.m2ts". Only with "eac3to_sample-001.mkv". Please check again that you really are on eac3to 3.34 and not 3.32.

[wiggaz]

allright. I'm a moron.
I had two eac3to folders. One updated, and another one from which I extracted the movie, who wasn't.
Now I have only one folder, updated, and with it I extract the movie flawlessly.

Sorry I have wasted your time.
At least everything is fine now. Thanks anyway

[tebasuna51]

Sometimes I get Invalid bitstream format error with DTS to AC3 conversion and I have to use ffmpeg.

I want to supply you an example (https://ufile.io/ifxhm), so probably a fix will be out.

Your example seems corrupted.

Using Arcsoft show the same message:

The ArcSoft DTS Decoder reported an error while decoding. <ERROR>
Aborted at file position 372244480. <ERROR>

Using ffmpeg also show a error (but continue decoding):

[dca @ 000000000050fb40] Invalid bit allocation index
Error while decoding stream #0:0: Invalid data found when processing input

Don't exist a fix for that.

[pstn]

Does anyone know if it is still necessary to do eac3to in.dtshd out.dtshd -21ms after converting PCM > DTS-HD with DTS Master Suite?

[tebasuna51]

Like all encoders DTS Master Suite can add a initial delay to the encoded stream, but the output have a special global header than show some global info than include the delay used.

Whith my tool LeeAudBi4 (https://www.sendspace.com/file/vvdv2p) can read that info, for instance:

File ........: D:\tmp\DTSENC.dtshd
Size ........: 3330600 bytes

---------------------------------------------- Header Info
ChunkID .....: DTSHDHDR (Length: 16)
SubchunkID ..: CORESSMD (Length: 12)
Core SampleR : 48000
Core Bitrate : 1509
Core ChMask : 2059 (L,R,C,LFE,Lss,Rss)
Core FrameLen: 2012
SubchunkID ..: EXTSS_MD (Length: 8)
Ext. Avg. Bit: 388
SubchunkID ..: AUPR-HDR (Length: 24)
Samplerate...: 48000
Num. frames..: 1315
Samples frame: 512
Num samp. ori: 671919
Channel Mask.: 2123 (L,R,C,LFE,Lss,Rss,Lsr,Rsr)
Samples Delay: 1024 [1]
SubchunkID ..: STRMDATA (Length: 3324188)
Offset data .: 140 (Extrachunks at end of file: 6272 bytes) [2]
------------------------------------------ End Header Info
Delay .......: 21.333 ms., (2 frames) [1]
Duration ....: 14.027 sec., (0h. 0m. 14.027s.)
Avg. Bitrate : 1896 Kb/s.
------------------------------------------------- End Info

eac3to remove the global header DTSHDHDR and also the final Extrachunks [2], but don't apply the delay [1].
Then yes you can remove it with the parameter -21ms

[pstn]

Like all encoders DTS Master Suite can add a initial delay to the encoded stream, but the output have a special global header than show some global info than include the delay used.

Whith my tool LeeAudBi4 (https://www.sendspace.com/file/vvdv2p) can read that info, for instance:

eac3to remove the global header DTSHDHDR and also the final Extrachunks [2], but don't apply the delay [1].
Then yes you can remove it with the parameter -21ms

That's great thanks,

I just tried it on a DTS HD file made with DTS Master Suite,

this is what I got :

File ........: C:\Users\pstn\Desktop\track2_eng.dts
Size ........: 691429400 bytes

----------------------------------------- First Frame Info
CRC present .................: 0 (Not)
Number of PCM Sample Blocks .: 15 ( 512 samples/frame)
Primary Frame Byte Size .....: 2011 ( 2012 bytes/frame)
Audio Channel Arrangement ...: 9 (5 C + L + R + SL + SR)
Core Audio Samp. Frequency ..: 13 (48 kHz)
Transmission Bit Rate .......: 24 (1536 Kb/s)
Embedded Down Mix Enabled ...: 0 (Not)
Embedded Dynamic Range Flag .: 0 (Not)
Embedded Time Stamp Flag ....: 0 (Not)
Auxiliary Data Flag .........: 0 (Not)
Mastered in HDCD format .....: 0 (Not)
Extension Audio Descr. Flag .: 0 (Channel Extension XCh)
Extended Coding Flag ........: 0 (Not)
Audio Sync Word Insert. Flag : 1 (Sub-sub-frame)
Low Frequency Effects Flag ..: 2 (Present, interpolation factor 64)
Predictor History Flag Switch: 1 (Yes)
Multirate Interpolator Switch: 0 (Non-perfect Reconstruction)
Encoder Software Revision ...: 7 (Current)
Copy History ................: 1 (Definition deliberately omitted)
Source PCM Resolution .......: 0 (16 bits)
Front Sum/Difference Flag ...: 0 (Not)
Surrounds Sum/Difference Flag: 0 (Not)
Dialog Normalization Param. .: - 0 dB
--------------------------------------------- Revised Info
Final bytes after core: 716
Total Frames ......: 236770
Duration ..........: 2525.547 seconds. ( 0 h. 42 m. 5.547 s.)
Master A. min./max.: 68 / 2620 (HD-MA subframe bytes)
------------------------------------------------- End Info

For some reason it doesn't show me the delay, but it was made with DTS Master Suite, so it should have one I think right? I used "extended info" in your app.

[tebasuna51]

I just tried it on a DTS HD file made with DTS Master Suite:

File ........: C:\Users\pstn\Desktop\track2_eng.dts

You need use a .dtshd just encoded with DTS Master Suite, after mixed in a container the global header DTSHDHDR disapear.

EDIT:
MkvMerge and tsMuxer remove the global header (and end Extrachunks) of a .dtshd just created with DTS Master Suite, but don't apply the delay.
You can't know if you need apply the -21ms or not.

[pstn]

You need use a .dtshd just encoded with DTS Master Suite, after mixed in a container the global header DTSHDHDR disapear.

EDIT:
MkvMerge and tsMuxer remove the global header (and end Extrachunks) of a .dtshd just created with DTS Master Suite, but don't apply the delay.
You can't know if you need apply the -21ms or not.

I see, thank you

[tormento]

Your example seems corrupted.
As FFMPEG seems more resilient, a flag to ignore corrupted frames would be ok.

[hubblec4]

Hi madshi

eac3to don't find the correct mpls for the Bluray Spider Man Homecoming (https://forum.videohelp.com/attachments/43873-1512076591/Spider%20Man%20-%20Homecomig%20BD.7z)


mpls 00420 with 16 chapters is missing.

eac3to v3.33
command line: "D:\eac3to.exe" "K:\" -log="E:\Spider Man - Homecoming\title.txt"
------------------------------------------------------------------------------
1) 00421.mpls, 00001.m2ts, 2:13:28
- Chapters, 83 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48kHz
- DTS Master Audio, German, multi-channel, 48kHz
- AC3, Turkish, multi-channel, 48kHz

2) 00422.mpls, 0:32:00
[334+342+343+344+345+346+347+348+349+350+351+352+353+354+355+356+357+335+338].m2ts
- Chapters, 17 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48kHz

3) 00353.mpls, 0:16:17
[342+343+344+345+346+347+348+349+350+351+352].m2ts
- Chapters, 10 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48kHz

[mtamimi]

I have a problem with a movie, the movie plays well w/o issues...However when I tried to convert the audio track I've had some issues...

eac3to v3.34
command line: eac3to.exe L:\Movie.REMUX.BluRay.AVC.DD.2.0.mkv -progressnumbers -log=log.1.txt
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 1:38:49, 24p /1.001
1: h264/AVC, English, 1080p24 /1.001 (16:9)
2: AC3, English, 2.0 channels, 192kbps, 48kHz
"Stereo"

As you see, it is AC3 2.0 and its 1:38:49...This is correct...Now, extracting...

eac3to v3.34
command line: eac3to.exe L:\Movie.REMUX.BluRay.AVC.DD.2.0.mkv 2:Audio.ac3 -progressnumbers -log=log.2.txt
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 1:38:49, 24p /1.001
1: h264/AVC, English, 1080p24 /1.001 (16:9)
2: AC3, English, 2.0 channels, 192kbps, 48kHz
"Stereo"
[a02] Extracting audio track number 2...
[a02] Creating file "Audio.ac3"...
Video track 1 contains 284321 frames.
eac3to processing took 3 minutes, 48 seconds.
Done.


Now, check this
eac3to v3.34
command line: eac3to.exe Audio.ac3 -progressnumbers -log=log.3.txt
------------------------------------------------------------------------------
AC3, 2.0 channels, 3:50:24, 192kbps, 48kHz

it reports the file as 3:50:24. If I converted the file to anything, I will have a 3:50:24 long audio track, and it sounds as if it was slowed down...

Can you guys investiate this?

The extracted audio track is here
https://mega.nz/#!9uYFmQJb!Mk3i3QY0GAVVoPLb1U6md3GuzSGvHEQGnNjA8GFv8Sw

[tebasuna51]

Can you guys investiate this?

The first 8 seconds are 2.0 the rest is:

AC3, 5.1 channels, 1:38:41, 448kbps, 48kHz

[XadoX]

@madshi Thx for the Update now I can use PowerShell without probs.

[mtamimi]

The first 8 seconds are 2.0 the rest is:

AC3, 5.1 channels, 1:38:41, 448kbps, 48kHz

Great, how did you figure this out? And what is the proper/precise way to trim the audio track ?

thnx :thanks: :thanks: :thanks:


*edit*
I tried to use your utility LaaAudBi4, Avira reported a threat and blocked it...
http://i67.tinypic.com/2lcsjyo.png

[LigH]

Heuristic warnings are usually false alarms, report them to your AV company.

Regarding automatic splitting of AC3 related to a changing channel layout, I am not sure what to do these days; I remember that ProjectX was able to detect channel layout changes, but only in SD TS with MPEG-2 video, it can't handle MKV. HeadAC3he was able to extract time ranges from AC3 files frame-wise, but you would have to know the exact timecodes.

[mtamimi]

Heuristic warnings are usually false alarms, report them to your AV company.

Regarding automatic splitting of AC3 related to a changing channel layout, I am not sure what to do these days; I remember that ProjectX was able to detect channel layout changes, but only in SD TS with MPEG-2 video, it can't handle MKV. HeadAC3he was able to extract time ranges from AC3 files frame-wise, but you would have to know the exact timecodes.

He was able to detect when the change started, so perhaps he uses a utility that will help in splitting the 2.0 track from the 5.1...?

[sneaker_ger]

Too lazy to test now but maybe you can use delay in this simple case. Something like:
eac3to input.ac3 temp.ac3 -8000ms
eac3to temp.ac3 output.ac3 +8000ms

Or with edit function. I'm sure tebasuna51 will post a proper solution(*) if this doesn't work. (If there is sound then maybe recode the first 8 seconds to 5.1)


(* if you want to regard this as a problem to begin with)

[tebasuna51]

He was able to detect when the change started, so perhaps he uses a utility that will help in splitting the 2.0 track from the 5.1...?
Try SplitAc3 from https://forum.doom9.org/showthread.php?p=1447695#post1447695

EDIT: The sneaker_ger solution work fine because the first 8 sec are silence.

[mtamimi]

Try SplitAc3 from https://forum.doom9.org/showthread.php?p=1447695#post1447695

EDIT: The sneaker_ger solution work fine because the first 8 sec are silence.

:thanks: a lot :)

*edit*
Anyone aware of a utility that helps me to trim the 1st 8000ms of the h264 video. If eac3to is capable of doing that, would someone show me an example?!

[sneaker_ger]

mkvtoolnix can trim video but it is problematic because of the keyframe limitation. Might need better tool like SolveigMM Video Splitter, VideoReDo or similar. Easiest to just leave the 8s.

[LigH]

Well, the optimal tool to split audio streams is one that doesn't care about video frames because it handles the audio only.

[sneaker_ger]

He wants to cut the 8s away altogether (video and audio).

[mtamimi]

He wants to cut the 8s away altogether (video and audio).

I did add +8000ms to the trimmed 5.1 auido track, and remuxed with the original untouched video and everything is in sync w/o issues. This is also good since I don't need to modify the chapters also...

I am only curious if I decided to trim the video by -8000ms instead of adding +8000ms of silence to the audio track...how can I do that easily?

:thanks:

[LigH]

But Dolby Digital AC-3 has 32 ms frames (at 48 kHz, with 1536 samples). Fortunately, 8000 ms are an integer multiple of its frame duration. But what if not? Will the duration be rounded up or down to the next audio frame border? It may be (generally) more reliable to split audio streams at the frame boundary between different channel layouts, no matter which duration they have written in seconds and milliseconds.

[sneaker_ger]

Depends on the software. With mkvtoolnix you can just do "--split parts:00:08-" and it will cut all tracks accordingly and also set delays in container to make precision <=1 ms if cuts aren't exactly on frame boundaries (which - like you said - seldomly align). It's not really a problem anyone should lose sleep about.

[tebasuna51]

Cut all tracks of a container without recode (VideoReDo or similar can recode only boundaries) at a exact point need:

- The time point is a exact multiple of audio framelength. Here, with AC3 48 Khz, 8000 ms are 250 x 32 ms, then cut the first 250 frames.

- The time point is a exact multiple of video framelength. If video have 25 fps you can cut the first 8000 / 25 = 320 frames, but you need than the 321ª frame was a Keyframe.
With other fps or without a Keyframe at exact point you can't cut exactly without recode.

[LigH]

But the problem of mtamimi here is not cutting at an arbitraty timecode. It is cutting at the audio frame where the channel layout changes. And it is knowing the position of that audio frame, not in milliseconds, but in audio frame number. And you already suggested SplitAc3 (https://forum.doom9.org/showthread.php?p=1447695#post1447695) for this job. All is fine. :)

[mtamimi]

Cut all tracks of a container without recode (VideoReDo or similar can recode only boundaries) at a exact point need:

- The time point is a exact multiple of audio framelength. Here, with AC3 48 Khz, 8000 ms are 250 x 32 ms, then cut the first 250 frames.

- The time point is a exact multiple of video framelength. If video have 25 fps you can cut the first 8000 / 25 = 320 frames, but you need than the 321ª frame was a Keyframe.
With other fps or without a Keyframe at exact point you can't cut exactly without recode.

"VideoReDo or similar can recode only boundaries" << Does this make a difference?

"you can't cut exactly without recode." :(

[tebasuna51]

"VideoReDo or similar can recode only boundaries" << Does this make a difference?

Recode always lose quality, but it's the best choice if you need that cut.

If you need more info please open a thread in video subforum, is off topic here.

[mtamimi]

Recode always lose quality, but it's the best choice if you need that cut.

Does not worth it. I prefer the 8s silence... :thanks:

[ashlar42]

Is there any up to date GUI available?

[pstn]

Does anyone know if there are any risks or downside to demuxing from a share (LAN) or a mounted ISO?

[LigH]

It's not much different to copying files. Mostly sequential reading from one file, and mostly sequential writing into a few other files. I would try to avoid both reading and writing over LAN due to low speed.

[Megalith]

Can someone clarify whether or not "Removing AC3 dialog normalization" affects TrueHD tracks? Or does it only affect the AC3 "core" track?

[Ripman]

I was trying to convert a .ts file to mkv. I got a message about the audio track not being "clean". See below. What does that mean? Thanks and happy new year.


eac3to v3.34
command line: eac3to war_requiem.ts 1: v.h264 2: a.flac
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 1:30:00, 50p
1: h264/AVC, 720p50 (16:9)
2: AC3, German, 2.0 channels, 448kbps, 48kHz, dialnorm: -23dB, 9ms
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Decoding with libav/ffmpeg...
[a02] Applying RAW/PCM delay...
[a02] Reducing depth from 64 to 24 bits...
[a02] Encoding FLAC with libFlac...
[v01] Creating file "v.h264"...
[a02] This track is not clean. <WARNING>
[a02] Creating file "a.flac"...
[a02] Audio has a gap of 32ms at playtime 0:00:02. <WARNING>
[a02] Starting 2nd pass...
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Decoding with libav/ffmpeg...
[a02] Applying RAW/PCM delay...
[a02] Reducing depth from 64 to 24 bits...
[a02] Encoding FLAC with libFlac...
[a02] Realizing RAW/PCM gaps...
[a02] Creating file "a.flac"...
Video track 1 contains 270021 frames.
eac3to processing took 1 hour, 18 minutes.
Done.


[edit] the output flac from the command above was 270min - was supposed to be 90 min. Must be a bad audio stream in that .Ts file. I’ll post the mediainfo.

[edit2]mediainfo -f

General
Count : 327
Count of stream of this kind : 1
Kind of stream : General
Kind of stream : General
Stream identifier : 0
ID : 1
ID : 1 (0x1)
Count of video streams : 1
Count of audio streams : 1
Video_Format_List : AVC
Video_Format_WithHint_List : AVC
Codecs Video : AVC
Audio_Format_List : AC-3
Audio_Format_WithHint_List : AC-3
Audio codecs : AC3
Audio_Language_List : German
Complete name : C:\0\War_Requiem.ts
Folder name : C:\0
File name : War_Requiem
File extension : ts
Format : MPEG-TS
Format : MPEG-TS
Format/Extensions usually used : ts m2t m2s m4t m4s tmf ts tp trp ty
Commercial name : MPEG-TS
Internet media type : video/MP2T
Codec : MPEG-TS
Codec : MPEG-TS
Codec/Extensions usually used : ts m2t m2s m4t m4s tmf ts tp trp ty
File size : 6321633292
File size : 5.89 GiB
File size : 6 GiB
File size : 5.9 GiB
File size : 5.89 GiB
File size : 5.887 GiB
Duration : 5400380.000000
Duration : 1 h 30 min
Duration : 1 h 30 min 0 s 380 ms
Duration : 1 h 30 min
Duration : 01:30:00.380
Duration : 01:30:00:23
Duration : 01:30:00.380 (01:30:00:23)
Overall bit rate mode : VBR
Overall bit rate mode : Variable
Overall bit rate : 9364703
Overall bit rate : 9 365 kb/s
Maximum Overall bit rate : 35500000
Maximum Overall bit rate : 35.5 Mb/s
Frame rate : 50.000
Frame rate : 50.000 FPS
Frame count : 270023
Stream size : 315367443
Stream size : 301 MiB (5%)
Stream size : 301 MiB
Stream size : 301 MiB
Stream size : 301 MiB
Stream size : 300.8 MiB
Stream size : 301 MiB (5%)
Proportion of this stream : 0.04989
File creation date : UTC 2017-12-31 22:57:08.222
File creation date (local) : 2017-12-31 17:57:08.222
File last modification date : UTC 2015-10-16 14:42:01.150
File last modification date (local) : 2015-10-16 09:42:01.150
OverallBitRate_Precision_Min : 9364702
OverallBitRate_Precision_Max : 9364704

Video
Count : 338
Count of stream of this kind : 1
Kind of stream : Video
Kind of stream : Video
Stream identifier : 0
StreamOrder : 0-0
ID : 4113
ID : 4113 (0x1011)
Menu ID : 1
Menu ID : 1 (0x1)
Format : AVC
Format/Info : Advanced Video Codec
Format/Url : http://developers.videolan.org/x264.html
Commercial name : AVC
Format profile : High@L4
Format settings : CABAC / 4 Ref Frames
Format settings, CABAC : Yes
Format settings, CABAC : Yes
Format settings, ReFrames : 4
Format settings, ReFrames : 4 frames
Format settings, GOP : M=4, N=32
Internet media type : video/H264
Codec ID : 27
Codec : AVC
Codec : AVC
Codec/Family : AVC
Codec/Info : Advanced Video Codec
Codec/Url : http://developers.videolan.org/x264.html
Codec profile : High@L4
Codec settings : CABAC / 4 Ref Frames
Codec settings, CABAC : Yes
Codec_Settings_RefFrames : 4
Duration : 5400460
Duration : 1 h 30 min
Duration : 1 h 30 min 0 s 460 ms
Duration : 1 h 30 min
Duration : 01:30:00.460
Duration : 01:30:00:23
Duration : 01:30:00.460 (01:30:00:23)
Bit rate mode : VBR
Bit rate mode : Variable
Bit rate : 8449420
Bit rate : 8 449 kb/s
Maximum bit rate : 40000000
Maximum bit rate : 40.0 Mb/s
Width : 1280
Width : 1 280 pixels
Height : 720
Height : 720 pixels
Sampled_Width : 1280
Sampled_Height : 720
Pixel aspect ratio : 1.000
Display aspect ratio : 1.778
Display aspect ratio : 16:9
Frame rate : 50.000
Frame rate : 50.000 FPS
Frame count : 270023
Resolution : 8
Resolution : 8 bits
Colorimetry : 4:2:0
Color space : YUV
Chroma subsampling : 4:2:0
Chroma subsampling : 4:2:0
Bit depth : 8
Bit depth : 8 bits
Scan type : Progressive
Scan type : Progressive
Interlacement : PPF
Interlacement : Progressive
Bits/(Pixel*Frame) : 0.183
Delay : 600000.000
Delay : 10 min 0 s
Delay : 10 min 0 s 0 ms
Delay : 10 min 0 s
Delay : 00:10:00.000
Delay, origin : Container
Delay, origin : Container
Stream size : 5703844345
Stream size : 5.31 GiB (90%)
Stream size : 5 GiB
Stream size : 5.3 GiB
Stream size : 5.31 GiB
Stream size : 5.312 GiB
Stream size : 5.31 GiB (90%)
Proportion of this stream : 0.90227
Buffer size : 30000000

Audio
Count : 306
Count of stream of this kind : 1
Kind of stream : Audio
Kind of stream : Audio
Stream identifier : 0
StreamOrder : 0-1
ID : 4352
ID : 4352 (0x1100)
Menu ID : 1
Menu ID : 1 (0x1)
Format : AC-3
Format/Info : Audio Coding 3
Commercial name : AC-3
Format settings, Endianness : Big
Codec ID : 129
Codec : AC3
Codec : AC3
Duration : 5400384
Duration : 1 h 30 min
Duration : 1 h 30 min 0 s 384 ms
Duration : 1 h 30 min
Duration : 01:30:00.384
Duration : 01:30:43:29
Duration : 01:30:00.384 (01:30:43:29)
Bit rate mode : CBR
Bit rate mode : Constant
Bit rate : 448000
Bit rate : 448 kb/s
Channel(s) : 2
Channel(s) : 2 channels
Channel positions : Front: L R
Channel positions : 2/0/0
ChannelLayout : L R
Samples per frame : 1536
Sampling rate : 48000
Sampling rate : 48.0 kHz
Samples count : 259218432
Frame rate : 31.250
Frame rate : 31.250 FPS (1536 spf)
Frame count : 168762
Resolution : 16
Resolution : 16 bits
Bit depth : 16
Bit depth : 16 bits
Compression mode : Lossy
Compression mode : Lossy
Delay : 600009.000
Delay : 10 min 0 s
Delay : 10 min 0 s 9 ms
Delay : 10 min 0 s
Delay : 00:10:00.009
Delay, origin : Container
Delay, origin : Container
Delay relative to video : 9
Delay relative to video : 9 ms
Delay relative to video : 9 ms
Delay relative to video : 9 ms
Delay relative to video : 00:00:00.009
Video0 delay : 9
Video0 delay : 9 ms
Video0 delay : 9 ms
Video0 delay : 9 ms
Video0 delay : 00:00:00.009
Stream size : 302421504
Stream size : 288 MiB (5%)
Stream size : 288 MiB
Stream size : 288 MiB
Stream size : 288 MiB
Stream size : 288.4 MiB
Stream size : 288 MiB (5%)
Proportion of this stream : 0.04784
Language : de
Language : German
Language : German
Language : de
Language : deu
Language : de
Service kind : CM
Service kind : Complete Main
bsid : 6
dialnorm : -23
dialnorm : -23 dB
compr : -0.28
compr : -0.28 dB
dsurmod : 1
dsurmod : Not Dolby Surround encoded
acmod : 2
lfeon : 0
dialnorm_Average : -23
dialnorm_Average : -23 dB
dialnorm_Minimum : -27
dialnorm_Minimum : -27 dB
dialnorm_Maximum : -23
dialnorm_Maximum : -23 dB
dialnorm_Count : 1885
compr_Average : -0.13
compr_Average : -0.13 dB
compr_Minimum : -0.28
compr_Minimum : -0.28 dB
compr_Maximum : 2.36
compr_Maximum : 2.36 dB
compr_Count : 1885
dynrng_Average : 0.11
dynrng_Average : 0.11 dB
dynrng_Minimum : 0.00
dynrng_Minimum : 0.00 dB
dynrng_Maximum : 2.15
dynrng_Maximum : 2.15 dB
dynrng_Count : 1885
format_identifier : AC-3

[tebasuna51]

...the output flac from the command above was 270min - was supposed to be 90 min.

Your AC3 seems have some frames 2.0, but most of them are 5.1.

Extract and split the AC3 track.
http://forum.doom9.org/showthread.php?p=1447695#post1447695

[Ripman]

Your AC3 seems have some frames 2.0, but most of them are 5.1.

Extract and split the AC3 track.
http://forum.doom9.org/showthread.php?p=1447695#post1447695

Thanks for that tip. Which line in the mediainfo report caught your attention? I tried the SplitAc3 program. Here is the log:


FileSize : 302421504 bytes
---------- First valid Header
Time eq. : 5400384 ms.
SamplCod : 0 (0:48, 1:44.1, 2:32 KHz.)
BitRate : 448 Kb/s
ChanMode : 2 (1:1/0, 2:2/0, 3:3/0, 4:2/1, 5:3/1, 6:2/2, 7:3/2)
FrameSize: 1792 bytes
---------- Process ( 25.000000 fps is used for Trim)
Time: 0 ms. Written: 69 frames 2.0 ( 2208 ms.)
Time: 2208 ms. Written: 2 frames 5.1 ( 64 ms.) Trim(55, 56)
Time: 2272 ms. Skipped: 1792 bytes (Maybe: 32 ms.)
Time: 2304 ms. Written: 168665 frames 5.1 ( 5397280 ms.) Trim(58, 134989)
Time: 5399584 ms. Written: 6 frames 2.0 ( 192 ms.)
Time: 5399776 ms. Written: 9 frames 5.1 ( 288 ms.) Trim(134994, 135000)
Time: 5400064 ms. Written: 1 frames 2.0 ( 32 ms.)
Time: 5400096 ms. Written: 3 frames 5.1 ( 96 ms.) Trim(135002, 135003)
Time: 5400192 ms. Skipped: 1792 bytes (Maybe: 32 ms.)
Time: 5400224 ms. Written: 5 frames 2.0 ( 160 ms.)
---------- End of File
Total time: 5400384 ms. at EOF
T. written: 168679 frames 5.1.
T. written: 81 frames 2.0.


[edit] looking at the splitac3 log, it seems like my tv signal got a bit corrupted — a little at the start, and a little at the end. It seems like splitac3 put 2ch samples in one output file, and 5.1ch samples in another file.

I could go back to the source .ts file, trim 2.304 secs from the start, preserve the next 5397280 ms (01:29:57.28), discard everything after that, and then extract the audio. Or, I could "zero-out" the first 2 frames (64ms) from the SplitAc3 5.1ch output, and everything after (64ms + 5397280 ms). Does this sound about right? How do I account for delays in the original audio stream? Thanks again.

[tebasuna51]

Which line in the mediainfo report caught your attention?
Nothing in MediaInfo.
Is a typical problem, the output length is 3 times the expected because 5.1 have 3 times samples than 2.0

I could go back to the source .ts file, trim 2.304 secs from the start, preserve the next 5397280 ms (01:29:57.28), discard everything after that, and then extract the audio. Or, I could "zero-out" the first 2 frames (64ms) from the SplitAc3 5.1ch output, and everything after (64ms + 5397280 ms). Does this sound about right? How do I account for delays in the original audio stream? Thanks again.

If you don't need the video I think is better cut the ts file, maybe commercials.

But if the video is OK you can use the 5.1 extracted by SplitAc3 with:

eac3to split51.ac3 one.ac3 -64ms
(the first 2 frames)
eac3to one.ac3 two.ac3 +2304ms
(audio/video delay)

And, maybe, you can delete the last 12 frames with DelayCut, only if sound bad.

[Ripman]

Nothing in MediaInfo.
Is a typical problem, the output length is 3 times the expected because 5.1 have 3 times samples than 2.0
>clever fellow


If you don't need the video I think is better cut the ts file, maybe commercials.

But if the video is OK you can use the 5.1 extracted by SplitAc3 with:

eac3to split51.ac3 one.ac3 -64ms
(the first 2 frames)
eac3to one.ac3 two.ac3 +2304ms
(audio/video delay)

And, maybe, you can delete the last 12 frames with DelayCut, only if sound bad.

Thanks for the tip. Video and audio recovered. SplitAc3 worked nicely - if'd be cool to have an option with that simple program to leave samples in contiguous numbered files.

[Ripman]

Is there any plans to integrate eac3to with mkvextract/mkvmerge? I know it works with haali, but I wondered about the others too. Almost positive I read about this within the thread last year.

[nautilus7]

ffmpeg mkv muxing support would be easier.

[Midzuki]

I agree, Haali is dead, and Nero is dead as well, it's time to drop them.

[stax76]

mp4 muxing for avc/hevc would also be useful because LSMASHVideoSource/LibavSMASHSource can open it without indexing. (something I miss from the mkv format)

[SeeMoreDigital]

mp4 muxing for avc/hevc would also be useful because LSMASHVideoSource/LibavSMASHSource can open it without indexing. (something I miss from the mkv format)Agreed ;)

[Ripman]

ffmpeg mkv muxing support would be easier.

I like ffmpeg too. But I ended up at mkvmerge years ago, and the developer is pretty accessible via doom9.

Are there specific ffmpeg mkv capabilities that you use or prefer? Any capabilities that exist in ffmpeg but not in mkvmerge? Thanks.

[stax76]

I like ffmpeg too. But I ended up at mkvmerge years ago, and the developer is pretty accessible via doom9.

Are there specific ffmpeg mkv capabilities that you use or prefer? Any capabilities that exist in ffmpeg but not in mkvmerge? Thanks.

I think the discussion was meant about muxing libraries which mkvtoolnix probably don't expose, the ffmpeg library for muxing/demuxing is called libavformat I believe.

[sneaker_ger]

At what point does it become easier to add eac3to features to ffmpeg than vice versa?

[nautilus7]

Who said that?

If you're referring to me, I said that it would be easier to add mkv muxing with ffmpeg (libavformat), as eac3to already uses ffmpeg for decoding.

[sneaker_ger]

No one. Just wondering...

[LigH]

Adding eac3to features to other tools will not be very easy as long as eac3to stays closed-source. That's no complaint, just my thoughts...

[AYColumbia]

@madshi,
I noticed that the haali splitter URL is outdated in the -test info and is now located at the link below if you want to update it.

https://haali.su/mkv/

[LigH]

The whole Haali Media Splitter software as such is outdated. There are fortunately not many usage cases left where parts of it can still be useful...

[Bandits]

When demuxing a UHD disc I got an audio file with DELAY -5ms in the filename.

- Does this mean there was a delay placed on the file so it would line up correctly when remuxed?
- Does this mean I need to place a delay on the file when remuxing?
- Is there a way to stop eac3to.exe from changing the output filename when a delay is encountered?

A different disc has an issue with the video.

First pass:
1) 00000.mpls, 00001.m2ts, 2:13:12
- Chapters, 16 chapters
- h265/HEVC, 2160p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48kHz
- AC3, Spanish, multi-channel, 48kHz
- AC3, French, multi-channel, 48kHz
- AC3, English, stereo, 48kHz
- DTS, English, stereo, 48kHz

Second pass:
M2TS, 1 video track, 5 audio tracks, 5 subtitle tracks, 2:13:12, 11.987p
1: Chapters, 16 chapters
2: h265/HEVC, unknown parameters
3: DTS Master Audio, English, 7.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
4: AC3, Spanish, 5.1 channels, 640kbps, 48kHz
5: AC3, French, 5.1 channels, 640kbps, 48kHz
6: AC3 Surround, English, 2.0 channels, 224kbps, 48kHz
7: DTS, English, 2.0 channels, 768kbps, 48kHz
8: Subtitle (PGS), English
9: Subtitle (PGS), French
10: Subtitle (PGS), English
11: Subtitle (PGS), French
12: Subtitle (PGS), Spanish

For some reason it cannot understand the video track on the second pass and will result in a corrupt file if demuxed. I get same result pointing straight at the M2TS also. Using version 3.34.

[tebasuna51]

- eac3to can't apply delay only in TrueHD tracks, and add the string "DELAY -5ms" (I don't know how change this) to remember you add the delay when remux.

With other tracks, not TrueHD, eac3to cut audio frames or add silent audio frames. then can't be always exact, and show the remaining delay in log without modify the track name.

BTW delays smaller than a video frame duration (~40 ms) can be safely ignored.

- The problem with the video track with parameters detected in mpls but not in m2ts seems:

a) If the m2ts play fine maybe it is not supported by eac3to. Cut a sample and upload it to http://eac3to.bugs.madshi.net

b) Or maybe it is corrupt.

[Bandits]

Loading the mpls or the m2ts gives same result for the video file.

I will send a sample. What tool can cut the m2ts correctly?

[Music Fan]

What tool can cut the m2ts correctly?
Probably ffmpeg (in command line), and maybe TsMuxer.

[MeteorRain]

I tend to hard cut m2ts using a binary file cutter -- it preserves the file structure.
You can just cut, for example, 200 megabytes from the beginning, reproduce against it and then upload.

[hubblec4]

What tool can cut the m2ts correctly?


DGSplit (https://forum.videohelp.com/attachments/44370-1515634594/DGSplit.7z) is for me the best binary cut tool.

[rco133]

Hi.

This is a general eac3to question, but also has to do with audioconversion.

I am using eac3to 3.34 and am trying to demux a disk with several audiotracks like this:

8: E-AC3, Czech, 7.1 channels, 768kbps, 48kHz, dialnorm: -27dB
(core: AC3, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB)

When I do "eac3to 1) -demux" it says:

a08 Applying (E-)AC3 delay failed.

Seems like eac3to can't demux audio streams like that. Maybe because it is 7.1?

I would then like to demux all other tracks than the e-ac3 tracks. Is there anyway to tell eac3to to ignore some tracks when using -demux? There is 20 subtitles or so on the disk, so making a manual command line including all tracks but the e-ac3 tracks is quite cumbersome.

When I have to name the tracks manually like 15: English.sup I also Loose all the information that eac3to usually uses for the filename. Number of subs in track, and if there is any forced subs in the track.

Any help will be much appreciated.

Thanks in advance.

rco133

[tebasuna51]

Yes, is a bug already reported here: http://bugs.madshi.net/view.php?id=450

More info: https://forum.doom9.org/showthread.php?p=1821743#post1821743

I don't know how help you.

[AlvoErrado2]

Is it possible to change the "framerate" for eac3 audio without converting to ac3 using this tool??

[sneaker_ger]

No, changing speed requires re-encoding and eac3to can only encode to the following formats:
(1) RAW, (L)PCM
(2) WAV (PCM only), W64, RF64, AGM
(3) WAVs (multiple mono WAV files, PCM only)
(4) AC3
(5) DTS
(6) AAC
(7) FLAC

You can output to WAV and encode that to eac3. But be aware there are different eac3 "flavors": https://forum.doom9.org/showthread.php?t=174418

[LigH]

Video is a discrete medium, frames appear at specific times and stay for a specific while, therefore you can change quite easily their display duration; but audio is a rather continuous medium, changing the playing time (with or without preserving the pitch doesn't matter much) is always a substantial change of the whole content, usually requires decoding, resampling, and new encoding. There may be a quite simple AC3 encoder in eac3to, but E-AC3 is not supported, AFAIK.

[r0lZ]

MkvMerge can shrink or stretch the audio streams by a specific factor without re-encoding. I don't know how it does it, and if the quality of the audio is less good after the operation. I suppose that it changes just the sampling rate (and therefore also the pitch), but I'm not sure.

[sneaker_ger]

MkvMerge simply stretches/compresses the timecodes. In the end you either have all frames slightly overlapping with each other or tiny gaps between all of them. Players may react badly to this so I don't recommend it. Better to change video fps in case re-encoding is not desired.

[tebasuna51]

MkvMerge simply stretches/compresses the timecodes. In the end you either have all frames slightly overlapping with each other or tiny gaps between all of them. Players may react badly to this so I don't recommend it. Better to change video fps in case re-encoding is not desired.

100% agree

[AlvoErrado2]

Thank's all for your answers!!

I understood what I need to do, thank you one more time.:)

[Eseninzhiv]

can extract Dolby Atmos master file (.atmos) with eac3to from an audio file Dolby Atmos/TrueHD Audio (BluRay)?

[LigH]

AFAIK, eac3to can only safely skip Atmos metadata.

[tebasuna51]

Last 5 post moved to Create Dolby Atmos (https://forum.doom9.org/showthread.php?t=175239) thread.

[Sakura-chan]

Sorry I can't resist: maybe we should have asked Kurt Gödel while he was still around. It might be lossless but we will never be able to prove it since the source is unknown and will be forever. ;)

I start to wonder how lossless my rips I thought they were, actually are.

Gives the same message for "Jurassic World" as well. Since I'm paranoid now, I let decode the same DTS-HD MA track twice to FLAC in one run (dcaDec & Arcsoft) and compare if the FLAC matches.

The FLACs match despite the XLL-warning. Really lossless? Well ...
Sorry if I'm beating a dead horse here...

But I'd want to be 101% doubt free. Is DTS-HD MA decoding (for all channels and bit depths) *really* lossless?

I've seen the tired "libDcaDec reported the warning "XLL output not lossless"." since the start of times for every single DTS-HD MA track I converted. Looking for old posts it all points to it indeed being lossless.

And another thing. On the libDcaDec page it says
This program is deprecated!

This decoder has been fully integrated into FFmpeg master branch and further development will continue there. Using FFmpeg for DTS decoding is now recommended.
Why is eac3to still using it instead ffmpeg? On the other hand if libDcaDec is indeed bit perfect it's kinda a moot point.

[LigH]

The state of libDcaDec as a separate project with separate source repository may be deprecated. But its development continues inside the ffmpeg project. So if eac3to reports to use libDcaDec, it does not explain explicitly that this may mean "libDcaDec as developed further as a part of ffmpeg".

[sneaker_ger]

eac3to still uses the deprecated, separate library. (Similar with Aften AC3 encoding.)

[SquallMX]

Is there a way to remove DialogNorm on EAC3 7.1 tracks? I keep getting this error when trying:

M2TS, 1 video track, 6 audio tracks, 11 subtitle tracks, 2:10:31, 24p /1.001
1: Chapters, 27 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
4: AC3, English, 2.0 channels, 320kbps, 48kHz, -14ms
5: AC3, French, 5.1 channels, 640kbps, 48kHz, -14ms
6: E-AC3, Spanish, 7.1 channels, 1024kbps, 48kHz, dialnorm: -27dB
(core: AC3, 5.1 channels, 576kbps, 48kHz, dialnorm: -27dB)
7: AC3, Portuguese, 5.1 channels, 640kbps, 48kHz, -14ms
8: AC3, English, 2.0 channels, 192kbps, 48kHz, -14ms
9: Subtitle (PGS), English
10: Subtitle (PGS), French
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), Portuguese
13: Subtitle (PGS), English
14: Subtitle (PGS), French
15: Subtitle (PGS), Spanish
16: Subtitle (PGS), Portuguese
17: Subtitle (PGS), French
18: Subtitle (PGS), Spanish
19: Subtitle (PGS), Portuguese
[a06] Extracting audio track number 6...
[a06] Removing AC3 dialog normalization...
[a06] Applying (E-)AC3 delay failed. <ERROR>
Aborted at file position 1048576. <ERROR>

[LigH]

Removing DN succeeded.

Applying delay failed.

[tebasuna51]

Hello SquallMX!

eac3to don't support that E-AC3 tracks, bug already reported: http://bugs.madshi.net/view.php?id=450
Pending to fix.

[hubblec4]

Hello SquallMX!

You could use the "-core" switch, this will work.

[SquallMX]

Hello SquallMX!

eac3to don't support that E-AC3 tracks, bug already reported: http://bugs.madshi.net/view.php?id=450
Pending to fix.

Thanks for the info!

Hello SquallMX!

You could use the "-core" switch, this will work.

Yes, is a workaround that works fine, unfortunately you lose the 7.1 data.

[LigH]

It may still be possible not to try to bring the audio stream in sync by changing it, but by multiplexing it (demultiplexed without any changes) into the copy with the same delay.

[`Orum]

To make a longer post (https://forum.doom9.org/showthread.php?p=1835193#post1835193) brief, there seem to be issues with the Haali muxer eac3to uses for matroska muxing. Is there any chance that this could be replaced with support for more recent muxers using libebml/libmatroska?

Alternatively, could users at least get a warning when using mkv output, so we know to expect problems?

[Thunderbolt8]

does eac3to have difficulties with UHD BDs which have gaps due to multiple .m2ts files? I heard makemkv does handle them correctly.

[Asmodian]

I have never had any issues in that regard, eac3to has always worked well with UHD BDs for me. I point it to the playlists, like I do for regular blurays.

[Thunderbolt8]

I have never had any issues in that regard, eac3to has always worked well with UHD BDs for me. I point it to the playlists, like I do for regular blurays.

refering to those UHD BDs with seamless branching titles

[hubblec4]

Hi madshi

Only eac3to is able to remove dialog normalization(DN) from streams(ac3).
I have many mkv files muxed from VOB files with ac3 streams which include the DN.

Feature request:
Is it possible that eac3to remove the DN from the ac3 streams without demuxing-remuxing?

[tormento]

I am not using eac3to anymore as the ac3 encoding output is really outdate and somehow problematic. My work path now is demux audio tracks and feed them to FFMpeg. Is there any plan to use it as encoding/decoding library or is it a long way to go?

[Richard1485]

You could pipe to ffmpeg and continue using eac3to.

[tormento]

You could pipe to ffmpeg and continue using eac3to.
Decoding is buggish too. How can I use ffmpeg for both decoding and encoding?

[Richard1485]

I wasn't aware that decoding was buggy. In what way?

[mtamimi]

I tried to change an audio file from 29.970 to 25.000 using the following command

eac3to source29970.mp2 output25000.mp2 -29.970 -changeTo25.000

but the output file is identical to the source, nothing changed...am I doing it wrong?

Thnx

[nevcairiel]

Audio doesn't really have a frame rate as such, its a video property. What exactly do you want to happen?

[mtamimi]

Audio doesn't really have a frame rate as such, its a video property. What exactly do you want to happen?

So speedingup and down is just for video? Can I use the audio from 29.970 video in a 25.000 video?

[Richard1485]

eac3to source29970.mp2 output25000.mp2 -speedup

[nevcairiel]

So speedingup and down is just for video? Can I use the audio from 29.970 video in a 25.000 video?

It depends how those videos are made. Are they exactly the same length? Then yes, the audio doesn't need to be changed.

Often 29.97 to 25 would be IVTC'ed to 23.976 and then speed-up from there, so durations may differ slightly, but you would have to do that speedup step, and not a 29.97 -> 25 slowdown.

[thecoreyburton]

Sometimes when dealing with AC3 tracks, I get the message "Reducing depth from 64 to 24 bits". The source audio in these cases is always reported by MediaInfo to be 16-bit. Is it practical to use "-down16" in these cases, or is it reducing (or increasing, rather?) the depth to 24 bit for a specific reason?

[sneaker_ger]

You can ignore the MediaInfo information. AC3 tracks do not have any inherent bitdepth (TrueHD is different). The AC3 decoder decodes it at 64 bit. Then eac3to automatically dithers that down to 24 bit because humans cannot hear a difference anyways and 64 bit files would be huge (or the target format doesn't support 64 bit in the first place). You can reduce the bitdepth further to 16 bit if you want and will probably not hear any difference. But this only makes sense if you are compressing to a format that does have a set bitdepth (e.g. FLAC, wav), not if you target AC3, AAC, MP3, ...

[thecoreyburton]

Perfect! That not only resolves my question, but also also a few other nagging thoughts I had as well.

[wolflop]

Can someone tell me how I can demux a Dolby Digital Plus 7.1 into individual wavs?

thanks

wolflop

[Music Fan]

This should work (add s after wav) to get 24 bit ;
eac3to input.ac3 output.wavs

Or add -down16 if you need 16 bit ;
eac3to input.ac3 output.wavs -down16

[wolflop]

This does not work with DD. with other tracks that goes (I use the eac3togui) with DD aborts the program with error message

[LigH]

And why do you believe that we do not want to know the exact error message? Or even better, a whole console output log, if the GUI doesn't hide that from the user.

[Music Fan]

Did you get an error message ?

edit : LigH was faster :D

[tebasuna51]

Can someone tell me how I can demux a Dolby Digital Plus 7.1 into individual wavs?

EAC3 7.1 can't be decode to 7.1 channels with free soft.

ffmpeg can decode it to 5.1:

ffmpeg -i input.eac3 decoded.w64

and after to individual 6 wavs (maybe ffmpeg can do this directly):

eac3to decoded.w64 output.wavs

[BonumMonstrum]

Hello everyone!

I'm using eac3to quite often, but by now i couldnt figure out how to ONLY demux subtitles.... maybe you guys can help me with that...? Demuxing the whole files only for the subtitles takes soo much time and I would love to reduce this ...


Thanks a lot in advance for your help!!!

[LigH]

Well, the main purpose of eac3to is processing audio streams. To demultiplex subtitles only, you may find tools better specialized in this task. It would help to know your source format (DVD Video, Blu-ray?) to suggest accordingly.

[richardpl]

EAC3 7.1 can't be decode to 7.1 channels with free soft.

ffmpeg can decode it to 5.1:

ffmpeg -i input.eac3 decoded.w64

and after to individual 6 wavs (maybe ffmpeg can do this directly):

eac3to decoded.w64 output.wavs

Have you found actual EAC3 7.1 which are not test streams?

Maybe if you report such streams, there is higher chance of it being supported in FFmpeg.

[r0lZ]

how to ONLY demux subtitles....
You can demux the subtitles exactly like any other stream.

Example: demux English subtitles from a BD playlist.

First, issue this command to inspect the content of the playlist (assuming angle 1):

path\to\eac3to.exe V:\BDMV\PLAYLIST\00200.mpls 1)

You will see something like this:

M2TS, 2 video tracks, 6 audio tracks, 4 subtitle tracks, 0:07:13, 24p /1.001
1: Chapters, 3 chapters
2: h264/AVC (left eye), 1080p24 /1.001 (16:9)
3: h264/MVC (right eye), 1080p24 /1.001 (16:9)
4: DTS Master Audio, English, 7.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
5: DTS Hi-Res, English, 5.1 channels, 2046kbps, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
6: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz, -24ms
7: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz, -24ms
8: AC3, French, 5.1 channels, 640kbps, 48kHz, -24ms
9: AC3, Spanish, 5.1 channels, 640kbps, 48kHz, -24ms
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), French
13: Subtitle (PGS), Spanish

Now, demux the subtitle streams 10 and 11:

path\to\eac3to.exe V:\BDMV\PLAYLIST\00200.mpls 1) 10: english1.sup 11: english2.sup

Works perfectly for a BD. I suppose that the same method works with any input file.

[Boulder]

Well, the main purpose of eac3to is processing audio streams. To demultiplex subtitles only, you may find tools better specialized in this task. It would help to know your source format (DVD Video, Blu-ray?) to suggest accordingly.

And as the subtitles are muxed into the corresponding files (m2ts, VOB, mkv etc.), eac3to needs to process the whole stream anyway so extracting only the subs is not much faster than extracting audio and subs.

[BonumMonstrum]

You can demux the subtitles exactly like any other stream.

Example: demux English subtitles from a BD playlist.

First, issue this command to inspect the content of the playlist:

path\to\eac3to.exe V:\BDMV\PLAYLIST\00200.mpls 1

You will see something like this:

M2TS, 2 video tracks, 6 audio tracks, 4 subtitle tracks, 0:07:13, 24p /1.001
1: Chapters, 3 chapters
2: h264/AVC (left eye), 1080p24 /1.001 (16:9)
3: h264/MVC (right eye), 1080p24 /1.001 (16:9)
4: DTS Master Audio, English, 7.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
5: DTS Hi-Res, English, 5.1 channels, 2046kbps, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
6: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz, -24ms
7: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz, -24ms
8: AC3, French, 5.1 channels, 640kbps, 48kHz, -24ms
9: AC3, Spanish, 5.1 channels, 640kbps, 48kHz, -24ms
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), French
13: Subtitle (PGS), Spanish
Track 4 is used for destination file "1".
This audio conversion is not supported.
Now, demux the subtitle streams 10 and 11:

path\to\eac3to.exe V:\BDMV\PLAYLIST\00200.mpls 10: english1.sup 11: english2.sup

Works perfectly for a BD. I suppose that the same method works with any input file.

Thanks a lot, it works, but not perfectly for my goal: My problem was, that I couldn't get the subtitles from some sources due to some audio streams that wasn't supported - and I always only wanted the subtitles. BUT I also would like to know whether there are forced subtitles or not. Normally eac3to creates a file like

"00125 - 7 - Subtitle (PGS), English, 1080 captions" or "1080 captions, 3 Forced captions"

So the automatic naming of eac3to for files would be nice now :D

At least I can get the subtitles now, thanks a lot for your help!!!!

[nevcairiel]

Have you found actual EAC3 7.1 which are not test streams?

Maybe if you report such streams, there is higher chance of it being supported in FFmpeg.

There is several Blu-ray discs which use this channel setup. If you need more samples, let me know.

[FrasierCrane]

Hi!

Whenever I try to demux a BluRay (m2ts files) containing an E-AC3 (7.1) audio track, eac3to fails. It can not even extract the core and it just aborts. See the log:

eac3to v3.34
command line: "F:\Clown_BD\eac3to\eac3to.exe" "V:" 1) 6: "T:\Clown_BD\Audio_6_German.AC3" -progressnumbers -LOG="T:\Clown_BD\LOGS\eac3to_PASS3_LOG.LOG"
------------------------------------------------------------------------------
M2TS, 1 video track, 6 audio tracks, 7 subtitle tracks, 1:45:00, 72p /1.001
1: Chapters, 28 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 channels, 24 bits, 48kHz, dialnorm: -3dB
(core: DTS, 5.1 channels, 1509kbps, 48kHz, dialnorm: -3dB)
4: DTS Hi-Res, English, 5.1 channels, 2046kbps, 48kHz, dialnorm: -3dB
(core: DTS, 5.1 channels, 1509kbps, 48kHz, dialnorm: -3dB)
5: AC3, English, 2.0 channels, 320kbps, 48kHz, dialnorm: -28dB
6: E-AC3, German, 7.1 channels, 896kbps, 48kHz, dialnorm: -27dB
(core: AC3, 5.1 channels, 512kbps, 48kHz, dialnorm: -27dB)
7: E-AC3, Italian, 7.1 channels, 896kbps, 48kHz, dialnorm: -27dB
(core: AC3, 5.1 channels, 512kbps, 48kHz, dialnorm: -27dB)
8: AC3, English, 2.0 channels, 320kbps, 48kHz, dialnorm: -27dB
9: Subtitle (PGS), English
10: Subtitle (PGS), German
11: Subtitle (PGS), Italian
12: Subtitle (PGS), English
13: Subtitle (PGS), German
14: Subtitle (PGS), Italian
15: Subtitle (PGS), English
[v02] The video bitstream framerate field doesn't seem to match the timestamps. <WARNING>
[a06] Extracting audio track number 6...
[a06] Removing AC3 dialog normalization...
[a06] Extracting E-AC3 core...
[a06] Decoding with libav/ffmpeg...
[a06] The libav decoder reported error -22 while decoding. <ERROR>
Aborted at file position 1048576. <ERROR>


What is the problem and how do I fix it? When demuxxing the other tracks, they are mostly damaged as well (especially the subtitle tracks) - this only happens when there is such an E-AC3 track present in the m2ts file.

Cheers!

[tebasuna51]

The issue is already reported (http://bugs.madshi.net/view.php?id=450) to madshi.

The eac3 7.1 tracks reported by eac3to like:

6: E-AC3, German, 7.1 channels, XXXkbps, 48kHz, dialnorm: -27dB
(core: AC3, 5.1 channels, YYYkbps, 48kHz, dialnorm: -27dB)

Fails when try to extract to EAC3 with:

eac3to "INPUT" 1) 6: output.eac3
...
[a06] Extracting audio track number 6...
[a06] Removing AC3 dialog normalization...
[a06] Applying (E-)AC3 delay failed. <ERROR>
Aborted at file position ... <ERROR>

Also fails when try to recode to AC3 (seems than can't extract the core):

eac3to "INPUT" 1) 6: output.ac3
...
[a06] Extracting audio track number 6...
[a06] Removing AC3 dialog normalization...
[a06] Extracting E-AC3 core...
[a06] Decoding with libav/ffmpeg...
[a06] The libav decoder reported error -22 while decoding. <ERROR>
Aborted at file position ... <ERROR>

The unique workaround than work is recode the core to AC3 with a explicit -core parameter:

eac3to "INPUT" 1) 6: output.ac3 -core
...
[a06] Extracting audio track number 6...
[a06] Extracting E-AC3 core...
[a06] Removing AC3 dialog normalization...
[a06] Decoding with libav/ffmpeg...
[a06] Remapping channels...
[a06] Encoding AC3 <640kbps> with libAften...
[a06] Creating file "output.ac3"...
Video track 1 contains X frames.
eac3to processing took Z seconds.
Done.

[LigH]

So it is one of the usual "notorious" Blu-ray E-AC3 streams with 5.1 AC3 core + 2 differential E-AC3 channels as substream?

[r0lZ]

BUT I also would like to know whether there are forced subtitles or not. Normally eac3to creates a file like

"00125 - 7 - Subtitle (PGS), English, 1080 captions" or "1080 captions, 3 Forced captions"

So the automatic naming of eac3to for files would be nice now :D
I'm not sure I understand. eac3to (and all M2TS demuxers) cannot know if there are forced subtitles without analysing the content of the M2TS file, because that information is not stored in a header or in the MPLS file. So, you have to launch the analysis anyway, then inspect the output and manually give proper file names for the demuxed tracks. I suppose that a good GUI should be able to automate that, but with a GUI, you will have to select manually the streams you want to demux. So, unless you write yourself a tool to do exactly what you want, there will always be things to do manually.

BTW, I forgot to specify that if your M2TS is multi-angle, you have to specify the angle to demux with this (assuming angle 1):
path\to\eac3to.exe V:\BDMV\PLAYLIST\00200.mpls 1) 10: english1.sup 11: english2.sup
(I have edited my previous post accordingly.)

[mbcd]

This is right, if you have a single forced subtitle at the end of the stream / film, you can not recognize it at beginning of the stream, so you have parse the whole stream to find it. If you dont have some addition file which contains a listing of all streams, you have to decode the complete stream.

[Smithy]

So it is one of the usual "notorious" Blu-ray E-AC3 streams with 5.1 AC3 core + 2 differential E-AC3 channels as substream?

The NEW DD+ @ BD have 4 (EAC3) Surround Ch's that replace the 2 Surrounds of the 5.1 AC3 Core Track when 7.1 DD+ is playing !

[SeeMoreDigital]

After a stalled start with HD-DVD there's no doubt that Dolby Digital Plus is back and expanding it's feature-set, especially as more streaming services embrace object-based (Atmos) audio.

With Atmos in mind... Have any of you guys downloaded Dolby's 7.1.4 and 9.1.6 (DD+ with Atmos) samples, that were muxed into the .mp4 container?

[nevcairiel]

The NEW DD+ @ BD have 4 (EAC3) Surround Ch's that replace the 2 Surrounds of the 5.1 AC3 Core Track when 7.1 DD+ is playing !

This is not exactly "new", this setup has been allowed on Blu-ray since forever, it was just rarely used. Still is somewhat rare, but it occurs sometimes.

All that said, FFmpeg should be able to handle this very soon. The patch is basically ready for inclusion. It should come to FFmpeg based players and tools then eventually.

[tebasuna51]

...FFmpeg should be able to handle this very soon.

Nice to know.

BTW, I suggest a parameter to extract/decode only the 5.1 part for users of 5.1 audio equipment (like me) without need downmix.

[richardpl]

Yes, there is already eac3_core bitstream filter in git master.

[LigH]

Only loosely related: After more than a year, the ffmpeg trac ticket #3595 (https://trac.ffmpeg.org/ticket/3595#comment:8) is active again ... its title got renamed. :o

[richardpl]

And everybody donated single penny.

[LigH]

If I knew it would certainly accelerate something, I would even donate more.

[pstn]

does anyone know which command to use to demux a 3d bluray but isolate the 2d part?

[Boulder]

does anyone know which command to use to demux a 3d bluray but isolate the 2d part?

Isn't the video for both eyes as its own track? IIRC the left eye stream is the "main" one to demux.

[LigH]

This is true when the one eye's content is encoded as AVC video stream, and the other eye's content as MVC substream.

[nevcairiel]

That is the case for 3DBDs, always. Although left/right could be swapped and the AVC stream actually be the right eye, but there is always one main AVC stream you can just take without the MVC stream to get 2D.

[pstn]

That is the case for 3DBDs, always. Although left/right could be swapped and the AVC stream actually be the right eye, but there is always one main AVC stream you can just take without the MVC stream to get 2D.

yes that's what I'm talking about, is there any specific command?

thanks

[r0lZ]

To grab the 2D movie, you have to demux the MPLS, and select the AVC video stream and the audio and subtitle streams you are interested in, and drop the MVC. If you want to demux directly the M2TS file(s), you have to locate the one that contains the AVC stream. It's easy with a good player. The M2TS files with the MVC stream cannot be played directly.

The command to demux is exactly identical to the command to demux a 2D BD, just omit the MVC stream. But personally, I would use tsMuxeR instead of eac3to to demux a BD. It has a good GUI, easy to use.

[pstn]

To grab the 2D movie, you have to demux the MPLS, and select the AVC video stream and the audio and subtitle streams you are interested in, and drop the MVC. If you want to demux directly the M2TS file(s), you have to locate the one that contains the AVC stream. It's easy with a good player. The M2TS files with the MVC stream cannot be played directly.

The command to demux is exactly identical to the command to demux a 2D BD, just omit the MVC stream. But personally, I would use tsMuxeR instead of eac3to to demux a BD. It has a good GUI, easy to use.


thanks buddy

[Megalith]

Anybody have issues with the audio being completely out of sync on Blu-ray titles that feature both a theatrical and extended cut? It seems as if the audio tracks between the two releases are getting mixed up...

Edit: Okay, so I looked at the log, and it isn't pretty:

[a04] Audio has a gap of 31ms at playtime 0:16:23. <WARNING>
[a04] Audio has a gap of 31ms at playtime 0:19:56. <WARNING>
[a04] Audio has a gap of 31ms at playtime 0:32:59. <WARNING>
[a04] Audio has a gap of 31ms at playtime 0:34:25. <WARNING>
[a04] Audio has a gap of 31ms at playtime 0:54:09. <WARNING>
[a04] Audio has a gap of 31ms at playtime 0:58:47. <WARNING>
[a04] Audio has a gap of 31ms at playtime 1:09:57. <WARNING>
[a04] Audio has a gap of 31ms at playtime 1:11:45. <WARNING>
[a04] Audio has a gap of 31ms at playtime 1:21:48. <WARNING>
[a04] Audio has a gap of 31ms at playtime 1:27:02. <WARNING>
[a04] Audio has a gap of 31ms at playtime 1:42:03. <WARNING>
[a04] Audio has a gap of 31ms at playtime 1:52:59. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:05:13. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:06:03. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:10:04. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:15:49. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:23:46. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:25:37. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:31:14. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:34:07. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:44:08. <WARNING>
[a10] Audio has a gap of 20ms at playtime 0:16:23. <WARNING>
[a10] Audio has a gap of 10ms at playtime 0:19:56. <WARNING>
[a10] Audio has a gap of 10ms at playtime 0:32:59. <WARNING>
[a10] Audio has a gap of 31ms at playtime 0:34:25. <WARNING>
[a10] Audio has a gap of 20ms at playtime 0:54:09. <WARNING>
[a10] Audio has a gap of 20ms at playtime 0:58:47. <WARNING>
[a10] Audio has a gap of 10ms at playtime 1:09:57. <WARNING>
[a10] Audio has a gap of 20ms at playtime 1:11:45. <WARNING>
[a10] Audio has a gap of 31ms at playtime 1:21:48. <WARNING>
[a10] Audio has a gap of 31ms at playtime 1:27:02. <WARNING>
[a10] Audio has a gap of 10ms at playtime 1:42:03. <WARNING>
[a10] Audio has a gap of 10ms at playtime 1:52:59. <WARNING>
[a10] Audio has a gap of 20ms at playtime 2:05:13. <WARNING>
[a10] Audio has a gap of 31ms at playtime 2:06:03. <WARNING>
[a10] Audio has a gap of 31ms at playtime 2:10:04. <WARNING>
[a10] Audio has a gap of 31ms at playtime 2:15:49. <WARNING>
[a10] Audio has a gap of 20ms at playtime 2:23:46. <WARNING>
[a10] Audio has a gap of 31ms at playtime 2:25:37. <WARNING>
[a10] Audio has a gap of 31ms at playtime 2:31:14. <WARNING>
[a10] Audio has a gap of 31ms at playtime 2:34:07. <WARNING>
[a10] Audio has a gap of 10ms at playtime 2:44:08. <WARNING>
[a11] Audio has a gap of 52ms at playtime 0:16:23. <WARNING>
[a11] Audio has a gap of 10ms at playtime 0:19:56. <WARNING>
[a11] Audio has a gap of 42ms at playtime 0:32:59. <WARNING>
[a11] Audio has a gap of 31ms at playtime 0:34:25. <WARNING>
[a11] Audio has a gap of 52ms at playtime 0:54:09. <WARNING>
[a11] Audio has a gap of 20ms at playtime 0:58:47. <WARNING>
[a11] Audio has a gap of 42ms at playtime 1:09:57. <WARNING>
[a11] Audio has a gap of 52ms at playtime 1:11:45. <WARNING>
[a11] Audio has a gap of 31ms at playtime 1:21:48. <WARNING>
[a11] Audio has a gap of 31ms at playtime 1:27:02. <WARNING>
[a11] Audio has a gap of 42ms at playtime 1:42:03. <WARNING>
[a11] Audio has a gap of 42ms at playtime 1:52:59. <WARNING>
[a11] Audio has a gap of 20ms at playtime 2:05:13. <WARNING>
[a11] Audio has a gap of 31ms at playtime 2:06:03. <WARNING>
[a11] Audio has a gap of 31ms at playtime 2:10:04. <WARNING>
[a11] Audio has a gap of 31ms at playtime 2:15:49. <WARNING>
[a11] Audio has a gap of 52ms at playtime 2:23:46. <WARNING>
[a11] Audio has a gap of 31ms at playtime 2:25:37. <WARNING>
[a11] Audio has a gap of 31ms at playtime 2:31:14. <WARNING>
[a11] Audio has a gap of 31ms at playtime 2:34:07. <WARNING>
[a11] Audio has a gap of 42ms at playtime 2:44:08. <WARNING>

Does this mean that the disc/release is bad? I understand that eac3to tries to fix these errors, but apparently it isn't working.

[tebasuna51]

@Megalith

Please put the rest of the full log.

eac3to must recover the gaps, but maybe there are other problem.

[Megalith]

M2TS, 2 video tracks, 8 audio tracks, 20 subtitle tracks, 2:50:56, 11.988p
1: Chapters, 28 chapters
2: h265/HEVC, 2160p24 /1.001 (16:9), 10 bits
3: h265/HEVC, 1080p24 /1.001 (16:9), 10 bits - Dolby Vision Enhancement Layer
4: DTS Master Audio, English, 7.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
5: DTS, English, 2.0 channels, 255kbps, 48kHz
6: DTS, French, 5.1 channels, 768kbps, 48kHz
7: DTS, Russian, 5.1 channels, 768kbps, 48kHz
8: DTS, Russian, 5.1 channels, 768kbps, 48kHz
9: DTS, Czech, 5.1 channels, 768kbps, 48kHz
10: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
11: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
12: Subtitle (PGS), English
13: Subtitle (PGS), French
14: Subtitle (PGS), Dutch
15: Subtitle (PGS), Danish
16: Subtitle (PGS), Finnish
17: Subtitle (PGS), Norwegian
18: Subtitle (PGS), Swedish
19: Subtitle (PGS), Czech
20: Subtitle (PGS), Modern Greek
21: Subtitle (PGS), English
22: Subtitle (PGS), French
23: Subtitle (PGS), Russian
24: Subtitle (PGS), Modern Greek
25: Subtitle (PGS), English
26: Subtitle (PGS), French
27: Subtitle (PGS), Russian
28: Subtitle (PGS), Modern Greek
29: Subtitle (PGS), French
30: Subtitle (PGS), Russian
31: Subtitle (PGS), Czech
Creating file "G:\00 Mux\Gladiator\Chapters_1.txt"...
[v02] Extracting video track number 2...
[a10] Extracting audio track number 10...
[s21] Extracting subtitle track number 21...
[a10] Removing AC3 dialog normalization...
[a11] Extracting audio track number 11...
[s12] Extracting subtitle track number 12...
[s25] Extracting subtitle track number 25...
[a04] Extracting audio track number 4...
[a11] Removing AC3 dialog normalization...
[v02] Creating file "G:\00 Mux\Gladiator\Audio_2_Undetermined.h265"...
[a04] Creating file "G:\00 Mux\Gladiator\Audio_4_English.DTS"...
[a10] Creating file "G:\00 Mux\Gladiator\Audio_10_English.AC3"...
[a11] Creating file "G:\00 Mux\Gladiator\Audio_11_English.AC3"...
[s21] Creating file "G:\00 Mux\Gladiator\Subtitles_21_English.sup"...
[s25] Creating file "G:\00 Mux\Gladiator\Subtitles_25_English.sup"...
[s12] Creating file "G:\00 Mux\Gladiator\Subtitles_12_English.sup"...
[a11] Skipping identical AC3 frames (seamless branching)...
[a04] Audio has a gap of 31ms at playtime 0:16:23. <WARNING>
[a04] Audio has a gap of 31ms at playtime 0:19:56. <WARNING>
[a04] Audio has a gap of 31ms at playtime 0:32:59. <WARNING>
[a04] Audio has a gap of 31ms at playtime 0:34:25. <WARNING>
[a04] Audio has a gap of 31ms at playtime 0:54:09. <WARNING>
[a04] Audio has a gap of 31ms at playtime 0:58:47. <WARNING>
[a04] Audio has a gap of 31ms at playtime 1:09:57. <WARNING>
[a04] Audio has a gap of 31ms at playtime 1:11:45. <WARNING>
[a04] Audio has a gap of 31ms at playtime 1:21:48. <WARNING>
[a04] Audio has a gap of 31ms at playtime 1:27:02. <WARNING>
[a04] Audio has a gap of 31ms at playtime 1:42:03. <WARNING>
[a04] Audio has a gap of 31ms at playtime 1:52:59. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:05:13. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:06:03. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:10:04. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:15:49. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:23:46. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:25:37. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:31:14. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:34:07. <WARNING>
[a04] Audio has a gap of 31ms at playtime 2:44:08. <WARNING>
[a10] Audio has a gap of 20ms at playtime 0:16:23. <WARNING>
[a10] Audio has a gap of 10ms at playtime 0:19:56. <WARNING>
[a10] Audio has a gap of 10ms at playtime 0:32:59. <WARNING>
[a10] Audio has a gap of 31ms at playtime 0:34:25. <WARNING>
[a10] Audio has a gap of 20ms at playtime 0:54:09. <WARNING>
[a10] Audio has a gap of 20ms at playtime 0:58:47. <WARNING>
[a10] Audio has a gap of 10ms at playtime 1:09:57. <WARNING>
[a10] Audio has a gap of 20ms at playtime 1:11:45. <WARNING>
[a10] Audio has a gap of 31ms at playtime 1:21:48. <WARNING>
[a10] Audio has a gap of 31ms at playtime 1:27:02. <WARNING>
[a10] Audio has a gap of 10ms at playtime 1:42:03. <WARNING>
[a10] Audio has a gap of 10ms at playtime 1:52:59. <WARNING>
[a10] Audio has a gap of 20ms at playtime 2:05:13. <WARNING>
[a10] Audio has a gap of 31ms at playtime 2:06:03. <WARNING>
[a10] Audio has a gap of 31ms at playtime 2:10:04. <WARNING>
[a10] Audio has a gap of 31ms at playtime 2:15:49. <WARNING>
[a10] Audio has a gap of 20ms at playtime 2:23:46. <WARNING>
[a10] Audio has a gap of 31ms at playtime 2:25:37. <WARNING>
[a10] Audio has a gap of 31ms at playtime 2:31:14. <WARNING>
[a10] Audio has a gap of 31ms at playtime 2:34:07. <WARNING>
[a10] Audio has a gap of 10ms at playtime 2:44:08. <WARNING>
[a11] Audio has a gap of 52ms at playtime 0:16:23. <WARNING>
[a11] Audio has a gap of 10ms at playtime 0:19:56. <WARNING>
[a11] Audio has a gap of 42ms at playtime 0:32:59. <WARNING>
[a11] Audio has a gap of 31ms at playtime 0:34:25. <WARNING>
[a11] Audio has a gap of 52ms at playtime 0:54:09. <WARNING>
[a11] Audio has a gap of 20ms at playtime 0:58:47. <WARNING>
[a11] Audio has a gap of 42ms at playtime 1:09:57. <WARNING>
[a11] Audio has a gap of 52ms at playtime 1:11:45. <WARNING>
[a11] Audio has a gap of 31ms at playtime 1:21:48. <WARNING>
[a11] Audio has a gap of 31ms at playtime 1:27:02. <WARNING>
[a11] Audio has a gap of 42ms at playtime 1:42:03. <WARNING>
[a11] Audio has a gap of 42ms at playtime 1:52:59. <WARNING>
[a11] Audio has a gap of 20ms at playtime 2:05:13. <WARNING>
[a11] Audio has a gap of 31ms at playtime 2:06:03. <WARNING>
[a11] Audio has a gap of 31ms at playtime 2:10:04. <WARNING>
[a11] Audio has a gap of 31ms at playtime 2:15:49. <WARNING>
[a11] Audio has a gap of 52ms at playtime 2:23:46. <WARNING>
[a11] Audio has a gap of 31ms at playtime 2:25:37. <WARNING>
[a11] Audio has a gap of 31ms at playtime 2:31:14. <WARNING>
[a11] Audio has a gap of 31ms at playtime 2:34:07. <WARNING>
[a11] Audio has a gap of 42ms at playtime 2:44:08. <WARNING>
[a04] Starting 2nd pass...
[a04] Realizing DTS gaps...
[a04] Creating file "G:\00 Mux\Gladiator\Audio_4_English.DTS"...
[a10] Starting 2nd pass...
[a10] Realizing (E-)AC3 gaps...
[a10] Creating file "G:\00 Mux\Gladiator\Audio_10_English.AC3"...
[a11] Starting 2nd pass...
[a11] Realizing (E-)AC3 gaps...
[a11] Creating file "G:\00 Mux\Gladiator\Audio_11_English.AC3"...
Video track 2 contains 245891 frames.
Video track 3 contains 245891 frames.
Subtitle track 12 contains 1453 captions.
Subtitle track 21 contains 2463 captions.
Subtitle track 25 contains 2842 captions.
eac3to processing took 35 minutes, 4 seconds.
Done.

[Megalith]

It turns out that the sync issue isn't there when I use MakeMKV instead of Clown_BD and MKVMerge. Apparently, the problem stems from demuxing the playlist file first instead of muxing directly to MKV. I have no idea why that is.

[tebasuna51]

I don't know if your eac3to input is a .mpls or a .m2ts.

Also I don't know the eac3to version, Clown_BD is a old GUI for eac3to, you need latest eac3to version to manage h265/HEVC BD's.
You can see how manage Clown_BD the video:
[v02] Creating file "G:\00 Mux\Gladiator\Audio_2_Undetermined.h265"...

BTW, if MakeMKV work fine problem solved.

[Megalith]

Yes, but I am still curious about how I would get a proper file using eac3to, as it is my preferred tool.

The file is an .m2ts, and I am using the latest version of eac3to. I have also ruled out mkvmerge being the issue.

I guess the solution is to force eac3to not to adjust the audio, but I don't know if that is possible.

[tebasuna51]

Only one m2ts directly from BD with theatrical and extended cut?

BTW you can stop the second pass with the parameter -no2ndpass, and see if work without adjust the audio.

[hubblec4]

Hi Megalith

Muxing Multi-Edition BDs with mkvmerge don't work really well. The issue are the identical frames (https://gitlab.com/mbunkus/mkvtoolnix/issues/2051) and maybe the audio-gaps also.

For such BDs I use eac3to to demux the audios. Like tebasuna51 wrote:
You could test a demux with the parameter "-no2ndpass".

My chapterEditor is special designed to preserve both(or more) editions of the movie in one mkv and eac3to is used to fix such issues.

[Megalith]

The BD features both a theatrical and extended cut, but I am only attempting to make an MKV of the extended version. I will try the -no2ndpass parameter, assuming I can figure out how to work eac3to from the command line. (I have only ever used a GUI for eac3to.)

I made a mistake when I said I was working with an .m2ts file earlier; the input is actually an .mpls playlist file. Interestingly, I was able to drag this directly into mkvmerge to create a problem-free file, which would imply that the -no2ndpass parameter should work.

[hubblec4]

Yes mkvmerge can handle mpls playlist files and audio almost always in synch but your BD has identical frames and that shift the chapter points a little bit (not always but often) without separate demuxing the audios.

[hubblec4]

Hi madshi

I know you are very busy, but could you fix this E-AC3 7.1 channel issue (http://bugs.madshi.net/view.php?id=450) please?

[rhaz]

Does anyone have any examples how to use -edit -loop/silence?

I can't figure out how this works and there is no examples on google.

All it says on wiki is:
-edit=0:00:00.000,0ms - any audio format loops or removes audio data at the specified runtime
-silence/-loop - forces usage of silence (or looping) for audio edits

I need to add 950ms of loop/silence at -edit=0:57:58.000,0ms

[tebasuna51]

From the changelog.txt:

* for gaps, edits & repairs > 1000ms eac3to now inserts silence by default
* for gaps, edits & repairs < 1000ms eac3to now loops audio by default
* option "-silence" forces eac3to to insert silence instead of looping audio
* option "-loop" forces eac3to to loop audio instead of inserting silence

If you want add 950 ms of silence:

-edit=0:57:58.000,950ms -silence

If you want repeat 950 ms, like is the default (<1000ms) is enough:

-edit=0:57:58.000,950ms

[mini-moose]

Anybody have issues with the audio being completely out of sync on Blu-ray titles that feature both a theatrical and extended cut? It seems as if the audio tracks between the two releases are getting mixed up...
You are having trouble with a UHD disc not a normal blu-ray.
It's been discussed here earlier on that eac3to can't create properly syncd audios from seamlessly branched UHD discs. I think it someone suggested that has to do with the ffmpeg lib it's using, and that using nero7 switch makes it better (but you probably need to have that nero installed).

Did you have more success with -no2ndpass ?

[iSeries]

Hi, any known issues updating libflac.dll to the latest version?

[Music Fan]

Hi,
I encoded a 24 bit wav into ac3 256 kbps with eac3to and discovered that a short silence (27 ms) was added at the end :o
That's problematic because I have to append another sample after and I hear a gap.
I see the short silence in an audio editor (after converting back to wav).
Is there a solution with eac3to or should I use another encoder ?
Thanks.

[sneaker_ger]

Like many other codecs AC3 hs fixed-length frames and encoder delay. One AC3 frame is 1536 samples long so at 48 kHz you have a frame-length of 32 ms. That will be the same with every AC3 encoder. Either work in WAV format incl. the append operation and encode after or at least try to find a better (e.g. silent) spot where you can append better.

[Music Fan]

Mmh, thanks for the explanation but that's what I didn't want to read :o
It's strange because I cut an ac3 file in 2 parts (with mkvmerge), I modified the first (a few seconds), re-encoded it in ac3 and the silence appeared at the end.
Normally, mkvmerge can't cut in the middle of an AC3 frame, thus the new ac3 file should be as the first (a multiple of 1536 samples).
I will verify if I can add silence at the beginning (which would not be problematic in this case) to get a size being a multiple of 1536 samples.

[sneaker_ger]

With AC3 we usually have 5ms encoder delay. It would fit your 27ms because 32ms-5ms = 27ms. So maybe remove 5ms in the beginning.

eac3to input.wav output.ac3 -5ms

[Music Fan]

It's better, but there is still a 1ms silence at the end (and the same with -6ms). I will try it and see if I hear it.

[Music Fan]

I don't hear anymore the gap with a 1ms silence.
But curiously, I had also tried to create a wav being a multiple of 1536 samples and eac3to still added a silence (a few ms) at the end.

[Megalith]

By default, does eac3to convert .wav files to .w64 files? All of my demuxing programs seem to do this, but I've never run into an LPCM file on a Blu-ray that actually exceeded 4GB.

[LigH]

No, you have to use .w64 as output format extension so eac3to knows you want it.

And I easily exceeded 4 GB decoding 5.1 audio to a 6 channel WAV from a DVD Video AC3 track; the other alternative eac3to offers is .wavs which creates mono WAV files per audio channel.

[Yoshi]

Sorry if I'm beating a dead horse here...

Well if you already give the honor to quote 3 year old messages of mine which I have to re-read in order to remember what it was about again, then let's beat that poor little horse together as unfortunately, it seems that it still deserves it. ;)

But I'd want to be 101% doubt free. Is DTS-HD MA decoding (for all channels and bit depths) *really* lossless?

That I still wonder as well. Considering madshi's remark which had followed my posting at the time, I doubt it though:

eac3to still cannot fully remove dialnorm from DTS-HD tracks, though, because doing so would require to rewrite the whole HD frame structure, including CRCs etc, which is very complicated.

If I should be wrong about this, everyone shall feel free to correct me but to my understanding, getting rid of dialog normalization is a fundamental requirement to have a technically lossless conversion so I am somewhat confused that this issue hasn't been brought to higher attention here as it would mean that the conversion of DTS-HD MA to let's say FLAC still isn't really lossless.

Of course it is most probably rather an academic issue as most (slightly) flawed conversions are audibly still transparent but then the question has to be valid why to care about lossless formats in the first place and not just stick to the cores (at least I wouldn't trust myself to stand a blind test in this life).

The only practical reason for me to even bother keeping the lossless stuff (taking psychological voodoo ambitions aside) is that interestingly (and almost ironically), many FLAC results turn out to be in fact smaller than their DTS core counterparts, at least when dithered down to 16 bit which technically isn't lossless anymore, either but on the other hand, any lower 8 bit from the studio master most probably contain nothing but noise there anyway.

In practice, even the best converters achieve around 20 bit performance at best, not to speak about the ears where 16 bit probably are already more than enough.


@madshi

In regard to that, just a thought about eac3to's dither function:

As I have noticed with a few series, sometimes their channels never reach at least 6dB, but let's say peak around -8 dBFS. Now if I convert the 24 bit DTS-HD MA source to 16 bit applying dither, since the peak of course remains the same, from my understanding I will effectively end up using only about 15 bits SNR wise (~ 6dB / bit).

Now most probably that doesn't matter at all as even only 13~14 bit sound damn good and are very quiet but I wonder if technically it wouldn't be the better approach to apply normalization during that dithering stage in order to make use of the full 16 bit. After all, isn't that exactly what all the mastering-in-24-bit-fuss despite having a 16 bit end-format is all about?

[nevcairiel]

If I should be wrong about this, everyone shall feel free to correct me but to my understanding, getting rid of dialog normalization is a fundamental requirement to have a technically lossless conversion so I am somewhat confused that this issue hasn't been brought to higher attention here as it would mean that the conversion of DTS-HD MA to let's say FLAC still isn't really lossless.


If you transcode audio to FLAC with eac3to using the new dcadec decoder, then it'll ignore dialnorm. What it cannot do is remove dialnorm from an encoded DTS-HD bitstream without decoding it.

Although one could argue that ignoring dialnorm is actually not a correct way to decode DTS-HD and the end-result would differ from what any commercial DTS-HD decoder would produce, and is therefor incorrect?

[Yoshi]

Many thanks for your remarks, nevcairiel.

I just realized that I apparently haven't distinguished between the flag removal and the ignorance during decoding before then, good point.

However, that also raises some confusion because from what I've understood that dialog normalization is considered to be only critical during re-encoding processes because there it's irreversible. On the other hand, I don't see the unconditional necessity to remove the flag from the undecoded sources as then it will be decoded at some point in the future anyway to probably high bitdepth PCM within some AVR and then, any dialog normalization is just a loudness change which shouldn't affect the audio quality at all.

So eac3to states

"- can remove dialog normalization from AC3, E-AC3, DTS and TrueHD tracks"

Maybe it should also note something like this then in order to prevent confusion:

"- can ignore dialog normalization from AC3, E-AC3, DTS, DTS-HD MA, and TrueHD tracks during decoding"

[tebasuna51]

"- can remove dialog normalization from AC3, E-AC3, DTS and TrueHD tracks"

Thats remain true.

"- can ignore dialog normalization from AC3, E-AC3, DTS, DTS-HD MA, and TrueHD tracks during decoding"

- When decode any Dialog Normalization in source is ignored unless you add the parameter -keepDialnorm

[Richard1485]

Is there is any reason why decoding DTS-HD MA or Dolby HD on a seamless-branching UHD BD to WAVs with eac3to would be problematic?

[Yoshi]

@tebasuna51

Thanks for your clarification. However, could you elaborate on having the dialog normalization removed without decoding it to PCM? From what I've learned now, as long as I use eac3to, it won't matter since the flag will be ignored anyway, right? However, unless I want to feed other programs or decoders with those encoded files, what is the essential purpose then of having it removed in the first place?

Isn't the approach to have the dialog normalization embedded just in case for playback purposes and ignoring it selectively whenever decoding it to PCM?

In other words - if I got this correctly, the only thing eac3to still cannot do is removing the dialog normalization info from a DTS-HD source while otherwise keeping it "as is", however decoding it will give us the PCM with dialog normalization ignored. What would be the purpose of having an altered DTS-HD file then with adjusted frames, CRC, etc. then?

[tebasuna51]

The Dialog Normalization is a field in AC3 and DTS header (not used with other formats) than can force the decoders to attenuate the volume to obtain the same level between tracks...(in a ideal world).

Some decoders ca be forced to ignore the Dialog Normalization, but not others.

In AC3 header the field store a value between 0 and 31 (-31 dB), if is 27 (many times) the volume is attenuate 4 dB, if is 31 -> 0 dB (no attenuation).
Remove the Dialog Normalization is store 31 in this field to avoid attenuation. After that all decoders produce the same output (the source without attenuation).

In DTS header the field value is directly the value to attenuate, but is rarely used, between dozens of samples I only found one to force -4 dB, the rest are always 0.
I never see problems with DTS volume.

The problem is only when follow Dolby recomendations explained here: How To Properly Encode Dolby Digital Audio (AC3) (https://forum.doom9.org/showthread.php?t=56020) because we obtain low volume compared with CD-Audio, TV commercials or other audio encoded with other formats (MP3, AAC, FLAC, ...).
The free AC3 encoders by default put 31 in Dialog Normalization to be comparable with other sources.

[Yoshi]

Many thanks again for your further explanation. However, in summary, even if the normalization isn't removed or ignored during conversion, the worst which will happen is the loss of a few dB of dynamic range / SNR depending on the setting of course. However, I'd rate that issue to be less severe than any dynamic range compression. Do you agree?

Something unrelated I just encountered and hopefully hasn't been discussed here already at length (searched the thread but couldn't find anything):

Given a 24 bit DTS-HD MA source which actually only contains 16 bit audio, eac3to gives me different results if using the parameter -down16 or not - which doesn't make any sense because if the actual bit depth is in fact 16 bit, then there is no need to apply dither just because I told so originally (as a user, at first I can't tell whether the 24 bit file really contains that amount of information or not). The resulting file without the -down16 option is smaller so probably eac3to unnecessarily and stubbornly applies dither here although it shouldn't.

Do I really have to let eac3to convert all 24 bit files now without the -down16 option just to see whether they are really 24 bit or only 16? :(

[nevcairiel]

eac3to will only know if a file is actually only 16-bit after it finished processing the entire file, and would ordinarily then do a 2nd pass to remove the empty 8-bits.
If you force -down16, it'll process the 24-bit audio as if its really 24-bit audio and always apply dithering - because it doesn't know if all future data is only 16-bit.

Here is a quote from madshi from a few years ago explaining the same thing (fyi, it took me 2 minutes to find it :p)

If your source claims to be 24bit, and if you use -down16, then eac3to will apply dithering - even if it's only really 16bit. The reason for that is that eac3to doesn't know which bitdepth the source has until processing is complete. You surely don't want eac3to to scan the whole file first, everytime, before starting processing, do you?

[Yoshi]

Damn it, apparently I used the wrong keywords then because I also already assumed that madshi had mentioned that back then. Thank you for having done my task then. There was also a discussion in terms of MakeMKV which probably till this day doesn't strip unused bits like eac3to does.

However, I think it wouldn't hurt to have such an option for eac3to indeed to check the whole file beforehand if necessary and then decide what to do. I mean what speaks against it - right now, I have to do it manually which is even more bothersome. However, I greatly assume that only decoding the first few minutes or short random pieces of a movie's soundtrack should do was well - if that leads to 16 bit only of real content, it should be quite unlikely that this becomes true 24 bit audio (if there is such a thing at all) all of the sudden.

Of course one could argue that if someone is already willing to sacrifice quality on the paper by dithering down the audio to 16 bit, then 16 bit sources again dithered to 16 bit isn't exactly the end of days either. ;)

Update: anyone else running into this issue, as a not exactly pretty but usable workaround, one can use the special device name "nul" as the filename of a second decoding instance. This way, one ends up knowing whether the original was only 16 bit or not while at least not using any hard disk I/O resources as the file is never created but processed by eac3to nonetheless. Of course, I think there would be better ways to do this including having a switch and also by maybe doing a quick random check of the source to save time.

[rockydon]

Thanks @madshi I have reached this page via Google,I would like to ask,first of all I am a newbie to this world and I am learning things with help of Google and forums.

I want to ask can eac3to run on Linux vps,I have iso file and I want to extract few tracks out of it,i am just learning new things,can you help me,I have mounted iso now I am looking for tutorial for track extraction.

I need audio track to be extracted from it.

I have ssh access but no setup desktop.

I am sorry again if something is missing from my side,thanks and I am waiting for reply.

I don't need remux or want to make Any thing,my sole intention is to extract audio track as it is.

[Yoshi]

@rockydon: not sure if this helps in your particular VPS environment but eac3to worked flawlessly for me via WINE under Ubuntu, Mint and Fedora at least.

[rockydon]

@yoshi thanks for your reply,I have been through many Google articles and yes you are absolutely right that it works under wine,I am not confident about running wine,I am using ubuntu 14,should I go and install wine and are their any effects of wine on other parts of Ubuntu,will wine completely change my Ubuntu environment,I am already running couple of things on it.will wine mess with my Ubuntu environment.I am a newbie and i am scared to mess up.

One more thing I want to bring into notice that I am using headless server,I don't have virtual desktop,I can only access with ssh.can I still use wine

[r0lZ]

Wine is totally independent from the system and it should not produce any conflict. (A small subset of the Windows OS is included with wine, but it is kept within the wine installation directory and is never mixed with the Linux OS.)

Not sure for ssh, but IMO it should work, and it doesn't hurt to try.

[rockydon]

Wine is totally independent from the system and it should not produce any conflict. (A small subset of the Windows OS is included with wine, but it is kept within the wine installation directory and is never mixed with the Linux OS.)

Not sure for ssh, but IMO it should work, and it doesn't hurt to try.

Its really difficult to understand how to operate eac3to in wine i found no exact guide to run it.

i have installed it in home/USER/ where there is .wine/drive_c onwards...........

any one to help me to get how to use eac3to.

didnt understand where shall i place eac3to in "program files" or "programdata" my sole intention to use eac3to is give folder input and extract tracks from BDMV,i will not be doing any converion or encoding simply i didnt find any guide which ahve ffmpeg folder input to extract tracks becasue my BDMV folder has multiple m2ts file and its not single big file which has main content,its split over few m2ts files,also eac3to can take *****.MLPS input so i want to run eac3to to just extract few tracks,if at all some one can guide me.i tried waste one whole day but had no success with it also i have only SSH access.

I have gone through this wiki page https://en.m.wikibooks.org/wiki/Eac3to/How_to_Use but nowhere it says how to setup eac3to and how to setup in wine so it can be used,one more thing my vps doesn't allow X server display setup,it's giving me error,but i wont mind just extracting tracks because I have BDInfo for my Bluray tracks.

From above second paragraph I just mean I have installed wine on my Ubuntu vps but I have done Any special set up if required after that i have tried winecfg that's it,but no X server display is available.

I have my Bluray content on mounted cdrom.
Also I tried bringing folders to /home/USER/.wine/drive_c/ but still unable to execute eac3to.

[rockydon]

Ok I read article that log.text can be turn off but is there a way I can create log.txt of Bluray metadata I find with help of eac3to????????

Because eac3to will create log only if there is some conversion or file extraction.

https://forum.doom9.org/showthread.php?p=1511031#post1511031

[Groucho2004]

Ok I read article that log.text can be turn off but is there a way I can create log.txt of Bluray metadata I find with help of eac3to????????

Because eac3to will create log only if there is some conversion or file extraction.

https://forum.doom9.org/showthread.php?p=1511031#post1511031
eac3to "playlist" > info.txt
or
eac3to "BDMV directory" > info.txt

Alternatively, use BDInfo (https://www.videohelp.com/software/BDInfo).

[rockydon]

Yeah got my issue solved I needed VNC to see display.

[rockydon]

while extracting audio in wine enviornment

https://i.imgur.com/ypFTHo6.png

how can i solve this

My purpose is to see playlist which i can do only with help of eac3to in ubuntu and extract ac3 track

[tebasuna51]

Please put the command line.
Do you try to write a file in a mounted CD-ROM?

The problem seems not related with eac3to but with your wine-linux.
I think than ALSA lib is not from eac3to.

[heerschop]

I have a DTS-MA 7.1 audio track from a bluray movie that I want to convert to AC3 5.1. Eac3to report that the DTS-MA 7.1 track has a strange setup.
With the down6 option, eac3to seems to create a correct 5.1 AC3 file.
Is MeGUI also able to create a correct AC3 from this DTS-MA strange setup audio track ? (see image for config options used in MeGui)


https://forum.doom9.org/attachment.php?attachmentid=16131&stc=1&d=1510955589

[MrVideo]

I have a DTS-MA 7.1 audio track from a bluray movie that I want to convert to AC3 5.1.
Why would you want to convert to AC3? Just extract the DTS core and avoid adding recoding errors.

[tebasuna51]

@heerschop
The MeGUI downmix 7.1 -> 5.1 works the same for standard or strange setup. No problem.

Even the MeGUI downmix (or BeHappy) works better than eac3to or ffmpeg downmix because, thanks to AviSynth plugin AudioLimiter (only 32 bits), mix only the 4 surround channels of 7.1 to the 2 surround channels of 5.1 leaving the other channels untouched preserving the full volume.

@MrVideo
There are two reasons to recode: save space and compatibility with some players (not all suport DTS)

If the source is lossless, like DTS MA is, the encode have only errors of the AC3 encoder like the 'core' DTS have also.
The quality difference between the 'core' DTS 1509 Kb/s and the AC3 640 Kb/s can only be listen with expensive audio equipment and 'gold' ears.

[MrVideo]

There are two reasons to recode: save space and compatibility with some players (not all suport DTS)
The saving space part was a given. The DTS not being playable by all players would seem kinda strange after all these years of DTS existing. If the player doesn't, it is time to upgrade or replace with one that does. :D

VLC is free and plays one helluv a lot of stuff. :cool:

[SeeMoreDigital]

If you disregard the slight difference in quality....

Rather than use the lossless 7.1 DTS-HD MA stream it's often easier to use the lossy 5.1 DTS core to create a 5.1 Dolby Digital encode ;)

[heerschop]

@heerschop
The MeGUI downmix 7.1 -> 5.1 works the same for standard or strange setup. No problem.


Thanks for your explanation , that's all I wanted to know.

Greetz

[Richard1485]

Even the MeGUI downmix (or BeHappy) works better than eac3to or ffmpeg downmix because, thanks to AviSynth plugin AudioLimiter (only 32 bits), mix only the 4 surround channels of 7.1 to the 2 surround channels of 5.1 leaving the other channels untouched preserving the full volume.

I didn't know that! What source do you select to do this? Or do you have to decode 7.1 to W64 in eac3to first and then use BeHappy?

[tebasuna51]

LWLibavAudioSource (or FFAudioSource) can decode DTS HD or TrueHD 7.1 lossless (even inside a container), you don't need decode it previously.
But of course you can do it with eac3to.

[MrVideo]

LWLibavAudioSource (or FFAudioSource) can decode DTS HD or TrueHD 7.1 losless (even inside a container), you don't need decode it previously.
Oops, it must have been lossy, as an "s" is missing. :eek:

[Richard1485]

^^ Thanks, tebasuna!

[shpankey]

This has probably been mentioned, but on the first post w/ the links, the link to eac3to µGUI is sending to unknown webpages (seems to vary on each try).

[tebasuna51]

Yes, the domain www homecinema-hd com don't work.
I don't know where find µGUI, but I can't recommend use it.

Also Clown BD link (forum slysoft com/showthread.php?t=25818) don't work.
Replaced with https://www.videohelp.com/software/Clown-BD v0.81 (May 31, 2013)

[73ChargerFan]

thanks

[Megalith]

Can anyone think of a reason why eac3to would process a 23.976 FPS video stream as a 27.730 FPS one? I think it is the reason why the DTS track in one of my MKVs is completely out of sync. I did not encounter the same error when I used MakeMKV instead.

[tebasuna51]

eac3to don't change the fps of video streams unless you specify a new one. For instance:

eac3to INPUT 1: output.h264 -changeTo23.976

and only fix values like -changeTo23.976, 24.000, 25.000, ... are available.
You never can change to 27.730 with eac3to.

Put a MediaInfo of your source.

[Megalith]

Yes, "process" was probably a poor choice of words. But this is what I noticed when I looked at the log for "Deep Rising":

1) 00004.mpls, 00008.m2ts+00004.m2ts, 1:46:22
- Chapters, 8 chapters
- h264/AVC, 1080p24 /1.001 (16:9)

M2TS, 1 video track, 3 audio tracks, 1 subtitle track, 1:46:22, 27.730p
1: Chapters, 8 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, 5.1 channels, 16 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
4: DTS Master Audio, 2.0 channels, 16 bits, 48kHz
(core: DTS, 2.0 channels, 1509kbps, 48kHz)
5: AC3, 2.0 channels, 192kbps, 48kHz, 32ms
6: Subtitle (PGS)
Creating file "H:\Mux\Chapters_1.txt"...
[a05] Extracting audio track number 5...
[v02] Extracting video track number 2...
[s06] Extracting subtitle track number 6...
[a04] Extracting audio track number 4...
[a03] Extracting audio track number 3...
[a05] Applying (E-)AC3 delay...
[v02] Creating file "H:\Mux\Video_2.h264"...
[a03] Creating file "H:\Mux\Audio_3_Undetermined.DTS"...
[a04] Creating file "H:\Mux\Audio_4_Undetermined.DTS"...
[a05] Creating file "H:\Mux\Audio_5_Undetermined.AC3"...
[s06] Creating file "H:\Mux\Subtitles_6_Undetermined.sup"...
Video track 2 contains 153012 frames.
Subtitle track 6 contains 1063 captions.
eac3to processing took 15 minutes, 2 seconds.
Done.

MediaInfo is showing that the video stream in the relevant playlist (.0004 mpls) is 23.976.

Video #1
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, RefFrames : 3 frames
Format settings, GOP : M=3, N=12
Codec ID : 27
Duration : 11 s 11 ms
Bit rate mode : Variable
Bit rate : 492 kb/s
Maximum bit rate : 32.3 Mb/s
Width : 1 920 pixels
Height : 1 080 pixels
Display aspect ratio : 16:9
Frame rate : 23.976 (24000/1001) FPS
Standard : NTSC
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.010
Stream size : 662 KiB
Color range : Limited
Color primaries : BT.709
Transfer characteristics : BT.709
Matrix coefficients : BT.709
format_identifier : HDMV
Source : 00008.m2ts

Video #2
ID : 4113 (0x1011)
Menu ID : 1 (0x1)
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L4.1
Format settings : CABAC / 4 Ref Frames
Format settings, CABAC : Yes
Format settings, RefFrames : 4 frames
Codec ID : 27
Duration : 1 h 46 min
Bit rate mode : Variable
Maximum bit rate : 35.0 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
format_identifier : HDMV
Source : 00004.m2ts

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 : 1 h 46 min
Bit rate mode : Variable
Channel(s) : 6 channels
Channel layout : C L R Ls Rs LFE
Sampling rate : 48.0 kHz
Frame rate : 93.750 FPS (512 SPF)
Bit depth : 16 bits
Compression mode : Lossless
Language : English
Source : 00004.m2ts

Audio #2
ID : 4353 (0x1101)
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 : 1 h 46 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 : 16 bits
Compression mode : Lossless
Language : English
Source : 00004.m2ts

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 : 1 h 46 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)
Bit depth : 16 bits
Compression mode : Lossy
Stream size : 146 MiB
Language : English
Service kind : Complete Main
bsid : 8
dialnorm : -31
dialnorm : -31 dB
compr : -0.28
compr : -0.28 dB
dsurmod : 1
dsurmod : Not Dolby Surround encoded
acmod : 2
lfeon : 0
dialnorm_Average : -31
dialnorm_Average : -31 dB
dialnorm_Minimum : -31
dialnorm_Minimum : -31 dB
dialnorm_Maximum : -31
dialnorm_Maximum : -31 dB
dialnorm_Count : 777
format_identifier : AC-3
Source : 00004.m2ts

[tebasuna51]

And what is the fps of extracted video?:

[v02] Creating file "H:\Mux\Video_2.h264"...

Is 27.730p? Or 2: h264/AVC, 1080p 24 /1.001 (16:9)?

I think than the 27.730p is only a wrong info but the video is not changed.

Please use MediaInfo with the two 00008.m2ts and 00004.m2ts in Advanced Mode, to see durations in seconds, not only minutes.
Maybe the audio tracks are different.

[madshi]

The 27.730p comes from trying to interpret the m2ts packet time stamps. *Usually* this matches the video fps, but sometimes not. You can safely ignore this information. It's probably eac3to misunderstanding the timestamps somehow, or the timestamps containing unusually high jitter or something...

[Megalith]

I've figured out why there's an incorrect audio delay. The problem is that the DTS track begins playing at the very beginning of the file (00008.m2ts, which is supposed to be a silent 11-second title card), even though it isn't supposed to start until the actual movie (00004.m2ts).

I'm not sure where the issue stems from, since the 00004.mpls file itself is fine. Could it actually be a problem with mkvmerge?

[tebasuna51]

When the first m2ts don't have the same audio streams, eac3to most the times crash.

If not you can forget the first 11-second credit and work only with the 00004.m2ts, or delay all streams not present in 00008.m2ts.

eac3to "H:\Mux\Audio_3_Undetermined.DTS" "H:\Mux\Audio_3_Delayed.DTS" +11100ms
...

[Anakunda]

Hello there, I'm having question to BR demuxing.
If the media contains two or more surround audio tracks being DTS-HD Master Audio and TrueHD, which one is expected the have lossless layer more fidelity?
And is eac3to always using the lossless layer when extracting to PCM/FLAC/WAV?
Last question, how do I view which codec is eac3to using to decode DTS(HD) and TrueHD, can I configure it?

[SeeMoreDigital]

If the media contains two or more surround audio tracks being DTS-HD Master Audio and TrueHD, which one is expected the have lossless layer more fidelity?As both are lossless audio formats, they have the capability of sounding exactly the same.

[mrh335]

I am using EAC3TO 3.34 with a GUI and using Staxrip 1.9 with version 3.34.

I am using a bluray ISO file which I have ripped and want to encode to a smaller MKV file.

If I use either software, I am able to choose the playlist I want to perform work on and then I select only the video portion, no audio and no subtitle (have tried both ways) and the process stops after a few seconds.

There is no log file generated. The size or playlist or bluray file do not matter.

This same process works on one 2015 Macbook Pro and one 2010 custom desktop, but not on newer 2017 Macbook Pro and 2017 iMac. All these computers are running Windows 10 natively via bootcamp.

If I change the output from MKV to h264, the process completes, but not vice versa.

Copy of status window is below and last line is the last status print out at 1% with no further progress even after hours.

Can anyone advise how to extract a bluray playlist to MKV given this issue? I have tried every version of EAC3TO over the last 1 year with the same results.


M2TS, 1 video track, 6 audio tracks, 10 subtitle tracks, 1:48:06, 24p /1.001
1: Chapters, 28 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 (strange setup) channels, 24 bits, 48kHz
(core: DTS-ES, 5.1 channels, 1509kbps, 48kHz)
4: AC3, English, 5.1 channels, 448kbps, 48kHz
5: AC3, Spanish, 5.1 channels, 448kbps, 48kHz
6: AC3, French, 5.1 channels, 448kbps, 48kHz
7: AC3, English, 2.0 channels, 224kbps, 48kHz
8: AC3, English, 2.0 channels, 224kbps, 48kHz
9: Subtitle (PGS), English
10: Subtitle (PGS), Spanish
11: Subtitle (PGS), French
12: Subtitle (PGS), English
13: Subtitle (PGS), Spanish
14: Subtitle (PGS), French
15: Subtitle (PGS), English
16: Subtitle (PGS), Spanish
17: Subtitle (PGS), French
18: Subtitle (PGS), English
Creating file "C:\Users\markh\Desktop\New folder\dp_chapters.txt"...
v02 Extracting video track number 2...
v02 Muxing video to Matroska...

[Boulder]

Why not use MKVToolnix GUI to do the job if there's no need to process audio but use it as it is? Eac3to is really not a muxing tool.

[mrh335]

Why not use MKVToolnix GUI to do the job if there's no need to process audio but use it as it is? Eac3to is really not a muxing tool.

I want to use Staxrip to compress using h265 and either convert the audio or transfer it over depending on what I want to use the smaller file to do.

The trial without the audio was to focus on the fact that I can't create a MKV regardless of the options I choose and it is linked to issues with EAC3TO whether being called through Staxrip or as a stand along program.

[LigH]

Can anyone advise how to extract a bluray playlist to MKV...?

Usually with the eternal beta version of makeMKV. The purpose of eac3to is mainly to process audio only, not everything.

[justnobody]

hi all
actually I have a question, "auto"-audio delay question, it's a bit complicated, I've uploaded samples for you guys
situation is source file already has delay in audio

example one
a video from iTunes having -42ms in audio (almost every iTunes video has delay in audio, audio format is AAC, iTunes loves AAC), 23.976fps

I cut a small piece out of it (mkvtoolnix), delay is now 5ms

and we demux this file with eac3to and put them (all tracks) together again with mkvtoolnix, delay is gone, 0ms now

since these delay are small, I can't heard difference between final 0ms or the 10ms or the original -42ms, seems all are fine

example two
this file freaks me out, video is 29.970 fps, a DVD and its encode, both are 29.970 fps, I will just leave the DVD alone, and I extract the audio track from VOB and mux it with the encode video (mkvtoolnix), result showing -100ms delay but it's in sync

I cut a piece out of this encode (sample showing 2ms delay), demux in eac3to, sometimes h.264 file becomes 25fps, if that happens I need to mark the video file with "*.h264 -29.970" (when I demux the whole film, framerate becomes 25fps in this case, but demux this sample it automatically makes the video 29.970)

and I put the demux video and audio together with mkvtoolnix, out of sync, the final result (the output from mkvtoolnix) is obviously more than 2ms.. even more than 100ms, totally not working if we demux-and-mux a file like this (of course the final file is showing 0ms now) (when the h.264 file becomes 25fps I also tried to label it as 29.970fps in mkvtoolnix during muxing, the result is also out of sync)


my question: with example one, should we demux and mux again to eliminate the delay, makes it 0ms? or just keep it there? exp one is a file from iTunes that coming with their own syncing audio since the start, in exp two, the audio is extracted from VOB and muxed with encode, once muxed (with mkvtoolnix) it's perfectly sync but showing -100ms delay

example two, when I demux-and-mux (both samples are in sync) the whole film, same thing happens, whole film out of sync, so when we come across a file like this (a DVDRip that has audio delay) we better leave it alone and not trying to demux it? That's my guess... Sorry for the long words


samples
http://www.mediafire.com/file/v550n7ugev3qug0

ps, if syncing is a big problem then in movie production how do they make it smooth?

[sneaker_ger]

example one
a video from iTunes having -42ms in audio (almost every iTunes video has delay in audio, audio format is AAC, iTunes loves AAC), 23.976fps
Maybe it is a delay from video re-ordering (bframes)? 1000/(24000/1001) ~= 41.7ms ~= 42ms

Or could be because of AAC encoder delay (though I expect 44ms, then. ~42.66.. ms is length of 2 AAC frames) . The most common lossy audio formats have this.
https://developer.apple.com/library/archive/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html

Now whether or not this is really stored correctly in your original file ... no idea. ~40ms is hard to tell confidently.

I cut a small piece out of it (mkvtoolnix), delay is now 5ms
This is misleading because video and audio frames have different lengths and they won't 100% align at the cut point. Mkvmerge stores delay via container to make up for this.

and we demux this file with eac3to and put them (all tracks) together again with mkvtoolnix, delay is gone, 0ms now
If you demux to "raw" (ADTS) AAC there is no way to store the delay infos that mkv or mp4 container can store. It is lost. You will introduce a slight delay (here: -5ms). MediaInfo doesn't know this. It is not an AI that analyzes e.g. speech to mouth movement of video but only looks at the delay values stored in the container (mkv/mp4).

[arrgh]

hi
in some UHD discs eac3to detects an "Dolby Vision Enhancement Layer" which it Shows as a separate track... it obviously can also be extracted...
is it really a separate track? if yes, can it be remuxed? if yes, how (e.g. in a *.mkv)?
thanks

[Dulus_No]

example one
a video from iTunes having -42ms in audio (almost every iTunes video has delay in audio, audio format is AAC, iTunes loves AAC), 23.976fps

Maybe it is a delay from video re-ordering (2 bframes?)? 2 * 1000/(24000/1001) ~= 41.7ms = 42ms

Delay shows up because DRM removing tools (except Requiem) also drop edit/elst atoms from source m4v file. And after MKVToolNix fixed the issue #1889 (https://gitlab.com/mbunkus/mkvtoolnix/issues/1889) mkv files with iTunes source have that delay.

[sneaker_ger]

Assuming there's also aac encoder delay they almost (44ms - 42ms = 2ms) cancel each other out completely if the edit lists of both tracks are lost, right?

[mkver]

@arrgh: The only way to store Dolby Vision data in Matroska is if it is embedded in the video stream and is copied with the video stream into the Matroska file by the muxer. (I don't know how Dolby Vision is stored in the video stream if it is stored in the video stream so I don't know if the Dolby data looks like invalid/garbage data and is therefore likely filtered out by muxers.) Even if the data ends up in the file, no indication of this would be stored at the container level, there is no header element saying "This track contains Dolby Vision data". I don't think that this can be meaningfully extracted as a separate track (if it is shown as a separate track, then it is likely not stored in the video stream, so it has its own PID and one could "extract the track by using a PID filter).
@Justnobody:
1. Your samples are not very informative. You should have given us files before you remuxed them so that we can analyze its sync. (This wouldn't work with the iTunes file because if Dulus_No
2. That the "delay" value changes when you extract a part of a file is (as sneaker_ger already said) a result of audio and video frames having different durations. Here is an easy example: You have an AC-3 track (duration of a each audio frame: 32ms) and a PAL 25p video (duration of each frame 40ms). If both audio and video start at 0:00 (and don't have any gaps) and there is a keyframe at 1s and you cut everything before 1s away, then your new file will have a delay of 24ms: The first video frame that will be kept was at 1s and is now at 0:00, the first audio frame that will be kept was at 1024ms and is now at 24ms.
This effect also implies that if you want to upload remuxed files and not the original files for us to analyze, you should upload the beginning of the file, not something in the midle (checking with a subtitle for the movie this scene is at 28:26 (if the subtitles and your movie have the same fps)).
3. But one can nevertheless say something meaningful about your second sample. NTSC DVDs were designed to work with old analog 6/1.001 i TVs and so they had to output 60/1.001 i, but this doesn't mean that they had to use this format internally. One can store 24/1.001p material as progressive on a NTSC DVD by using repeat-field-flags. The typical way to do this is by using a 2:3 pulldown: One frame will be the basis for two fields and therefore shown 2/(60/1.001)s (about 33ms) long, the next frame will be the basis for three fields and therefore shown 3/(60/1.001) s long. Of course, the natural framerate for such material is 24/1.001fps and that's how modern equipment will play it.
Your second sample "beginning sample Ibitsu [720x480][H264][AAC] (1) (1).mkv" before extraction and remuxing is actually 24/1.001p although the default duration in the header says 30/1.001fps. Here is my guess to what happened: You remuxed this file and the remuxing tool just used the frame rate info the bitstream signalled (one should keep in mind that this is not wrong -- the "default duration" is not intended to be the reciprocal of the frame rate) and wrote 30/1.001fps into the Matroska header. But lateron you used a tool to reencode it to H.264 that is smart in the sense that it noticed that this is just a cfr movie that has been subjected to a 2:3 pulldown and that therefore created 24/1.001p output pictures, but unfortunately it still wrote 30/1.001 fps in the Matroska header and it gave the bitstream a timebase of 60/1.001 (but didn't set the fixed_frame_rate_flag). When mkvmerge remuxd the video elemetary stream, the only timing information the bitstream contained suggested 30/1.001fps and that is how mkvmerge remuxed it. The result is totally out of sync and there is no video for the last few seconds of audio. The asynchronity in this case has nothing to do ith an initial audio delay, instead it is a result of mismatched framerates and as such gets gradually worse.
4. Actually, the VOB/Program Stream demultiplexer and the MPEG-2 module of mkvmerge have some bugs, but the developer of mkvmerge considers MPEG-2 obsolete and doesn't want to spend time on this. One should not really use mkvmerge to remux DVDs, use MakeMKV instead (it can also parse the ifo files so that one gets the correct DVD color palette and the chapters).

[sneaker_ger]

@arrgh: The only way to store Dolby Vision data in Matroska is if it is embedded in the video stream and is copied with the video stream into the Matroska file by the muxer. (I don't know how Dolby Vision is stored in the video stream if it is stored in the video stream so I don't know if the Dolby data looks like invalid/garbage data and is therefore likely filtered out by muxers.) Even if the data ends up in the file, no indication of this would be stored at the container level, there is no header element saying "This track contains Dolby Vision data". I don't think that this can be meaningfully extracted as a separate track (if it is shown as a separate track, then it is likely not stored in the video stream, so it has its own PID and one could "extract the track by using a PID filter).
There are 2 ways to store Dolby Vision. One as SEI messages in the base video stream. On Blu-Ray it seems they use a second "track" instead. As you say there isn't really any way to "properly" copy that into mkv yet.

[mkver]

Assuming there's also aac encoder delay they almost (44ms - 42ms = 2ms) cancel each other out completely if the edit lists of both tracks are lost, right?
You are making presumptions about the aac encoder delay that needn't be true. Not every aac encoder has a delay of 2 frames. The best bet would probably to test Apple's AAC encoder. Or maybe Dulus_No knows a bit more about this?
[Edit]: This (https://developer.apple.com/library/archive/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html) and this (https://forum.doom9.org/showthread.php?p=1563714#post1563714) both suggest 2112 samples which are exactly 44ms. You knew that, didn't you?
[Edit2]: Yes, you did. Of course you did.

[sneaker_ger]

This is the link I posted earlier: https://developer.apple.com/library/archive/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html
Apple's AAC encoder uses 2112 samples of delay. At 48 kHz that's exactly 44ms. If you use QAAC it confirms this.

(Of course we still can't be 100% sure that's what they use for their streaming but it's the best guess until someone has a properly decrypted sample.)

[sneaker_ger]

and we demux this file with eac3to and put them (all tracks) together again with mkvtoolnix, delay is gone, 0ms nowIf you demux to "raw" (ADTS) AAC there is no way to store the delay infos that mkv or mp4 container can store. It is lost. You will introduce a slight delay (here: -5ms). MediaInfo doesn't know this. It is not an AI that analyzes e.g. speech to mouth movement of video but only looks at the delay values stored in the container (mkv/mp4).
@justnobody
One thing I forgot to add:
When you extract certain audio formats with eac3to it will delete/add audio frames to preserve the delay value of the source container as best as possible (within length of 1 audio frame). The eac3to log should tell you if that happens. Not sure if that works with AAC as eac3to's AAC handling seems to be very limited.

[Dulus_No]

You are making presumptions about the aac encoder delay that needn't be true. Not every aac encoder has a delay of 2 frames. The best bet would probably to test Apple's AAC encoder. Or maybe Dulus_No knows a bit more about this?
[Edit]: This (https://developer.apple.com/library/archive/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html) and this (https://forum.doom9.org/showthread.php?p=1563714#post1563714) both suggest 2112 samples which are exactly 44ms. You knew that, didn't you?
[Edit2]: Yes, you did. Of course you did.
The same delay shows up not only on AAC tracks but also on AC3 (video has a +42ms / 1 frame delay). mp4dump of original m4v with drm (https://pastebin.com/hRqADJcu) and decrypted with TunesKit m4v (https://pastebin.com/zH6GJAh5).

[sneaker_ger]

Ok, so only one edts/elst with "media time" != 0.

[arrgh]

There are 2 ways to store Dolby Vision. One as SEI messages in the base video stream. On Blu-Ray it seems they use a second "track" instead. As you say there isn't really any way to "properly" copy that into mkv yet.

yes, I see a second Video track...

https://www.bilder-upload.eu/bild-d12e51-1539893458.jpg.html

so, that means it is relly a second independant track and not something like a "forced pgs" which sometimes comes as an individual track, sometimes enbedded in a general track where some frames are marked as forced...
or, in other words, the contend of this track is not also included in the first video stream; to maintain the complete data one would need to mux the second track into a container?
there are also no other containers (instead of mkv) which could be used for this and enable rederers like madVR to use the data?

[sneaker_ger]

I don't know any free software that supports Dolby Vision. (This includes madVR.)

[killerpot]

eac3to wav to ac3 bug in v3.34 and all other version i try 3.27,3,29,3.30,3.31,3.34 and others all bug

only old one 3.24 not bug!!
bug 3.34 and other give this error: proof here https://puu.sh/BTB58/7697ab92ae.mp4 (3,34 for this example)

aborded at file position 262144 !!! spend 6 hour try fix that nothing work i try all king of thing

different folder location just in case not work and other thing i try PLZ HELP ME SOMEONE!

why eac3to 3.24 work???
eac3to wav to ac3 only work! with v3.24
check the proof here https://puu.sh/BTAUz/8be7a5e60d.mp4

FILE HERE ON GOOGLE DRIVE (FOR TESTING) vff.wav
https://drive.google.com/open?id=1aBXKnjbAeFohrAWsSm_kJdA4CA_G74Wa

some one can test it and help me plz! i want to know what is the probleme annoy me so much thx!

in 3.34 the only time it's work it's making this line "eac3to vff.wav vff.ac3 -resampleTo48000 -down16

-448" WHY EAC3TO SHOULD WORK WITH 44,1 KHZ RIGHT? it's work in 3.24!

[tebasuna51]

@killerpot

I can reproduce the bug but the Aften encoder included in eac3to is deprecated, use ffmpeg instead without problems:

ffmpeg.exe -i vff.wav -acodec ac3 -center_mixlev 0.707 -ab 448k vff.ac3

[killerpot]

but the problem i hate ffmpeg if you try to encode ac3 in audacity (ffmpeg audacity version) your ac3

will be a shit for shure.and the command are more complicated in ffmpeg.question #1 what this does
-center_mixlev 0.707 (its and 5,1 wav to ac3 i wanna encode)

question #2 no other freeware are better then aften to encode ac3.but a question did ffmpeg use aften in

their thing to encode ac3 ?

question #3 i think in the past eac3to tell he encode with aften my ac3 and now that tell libaften do

you know if it's the case ?

question#4 if it's the case what the difference between aften and libaften ?

[tebasuna51]

1) By default ffmpeg put in ac3 header a value of -center_mixlev 0.595, the correct default is 0.707 to avoid low dialog volume if downmixed to stereo.

2) The ac3 encoder in ffmpeg is a improved aften encoder (https://forum.doom9.org/showthread.php?p=1794293#post1794293) (by the same developer Justin Ruggles)

3) eac3to always use libaften.dll to encode ac3, but you can use pipe to encode with aften.exe (https://github.com/wieslawsoltes/AftenWindowsBuilds/releases). Or better use Aften directly:

aften vff.wav vff.ac3

(defaults: -b 448 for 5.1, -cmix 0 [= -3.0 dB = 0.707], -readtoeof 1 only needed if wav > 4 GB, -pad 0 to avoid 5.3 ms of delay)

4) Don't exist differences between aften and libaften, here seems a eac3to bug when call libaften with a 44.1KHz source.

If the command are complicated for you use my UsEac3to GUI and the A/V Recode option to use ffmpeg for ac3 recode directly.

[Masutin]

To convert to mono, what more options are required to avoid the error 'No channel mapping to AAC defined'?stdout.wav -mono | QAAC

[tebasuna51]

stdout.wav -mono -simple | QAAC

Work with a simple wav header, with a WAVE_FORMAT_EXTENSIBLE header (default) the channel mask can be other than 4 (mono center channel).

Remember: the undocumented switch "-mono" extract the center channel from a multichannel input or left channel from stereo input, don't make a downmix from input channels.

[Lyle_JP]

1) By default ffmpeg put in ac3 header a value of -center_mixlev 0.595, the correct default is 0.707 to avoid low dialog volume if downmixed to stereo.

I always use -dialnorm -27 on my ffmpeg ac3 encodings. Does that do the same thing, or should I also add -center_mixlev 0.707?

[LigH]

The "Dialog Normalization" and the "Center Mix Level" are independent parameters.

[tebasuna51]

I always use -dialnorm -27

Always?
Not all sources need that parameter.
My recommendation is let the ffmpeg default -31

or should I also add -center_mixlev 0.707?

Like LigH say is not the same.

Here I recommend use 0.707, default used by Dolby, instead the ffmpeg default 0.595.

[Richard1485]

Here I recommend use 0.707, default used by Dolby, instead the ffmpeg default 0.595.

Does the appropriate value for -center_mixlev vary from source to source (in the way that -dialnorm does) or is it always a matter of preference?

[tebasuna51]

Of course if there are options is because it can vary.

I only recommend defaults without analyze the source.
For instance when recode soundtracks from others formats.

[Richard1485]

Of course if there are options is because it can vary.

Yes, or because there are preferences.

I only recommend defaults without analyze the source.

Understood. Thanks, tebasuna! :)

[GCRaistlin]

madshi, can you please take a look at this bug: http://bugs.madshi.net/view.php?id=377

[asarian]

Same here:

eac3to.exe valerian.m2ts 2: c:\video\temp.pcm -down6
MKV, 1 video track, 2 audio tracks, 3 subtitle tracks, 2:17:25, 24p /1.001
1: h265/HEVC, 2160p24 /1.001 (16:9), 10 bits
2: TrueHD (Atmos), English, 7.1 channels, 48kHz, 2419ms
"TrueHD Atmos 7.1"
3: AC3 EX, English, 5.1 channels, 640kbps, 48kHz, 2419ms
"AC-3 5.1-EX"
4: Subtitle (PGS), English, "English (Forced)"
5: Subtitle (PGS), English, "English (SDH)"
6: Subtitle (PGS), German, "German"
a02 Extracting audio track number 2...
a02 Decoding with libav/ffmpeg...
a02 Applying RAW/PCM delay...
a02 Mixing surround channels...
a02 Swapping endian...
a02 Remapping channels...
a02 Creating file "c:\video\temp.pcm"...
----[truehd @ 026336c0] Lossless check failed - expected 00, calculated c1.
[truehd @ 026336c0] Lossless check failed - expected 00, calculated 46.
[truehd @ 026336c0] Lossless check failed - expected 00, calculated 34.
--[truehd @ 026336c0] Lossless check failed - expected 00, calculated 24.
-[truehd @ 026336c0] Lossless check failed - expected 00, calculated 5b.
[truehd @ 026336c0] Lossless check failed - expected 00, calculated 81.
[truehd @ 026336c0] Lossless check failed - expected 00, calculated 83.
------------[truehd @ 026336c0] Lossless check failed - expected 00, calculated e7.
-[truehd @ 026336c0] Lossless check failed - expected 00, calculated 3b.
--[truehd @ 026336c0] Lossless check failed - expected 00, calculated e2.
-[truehd @ 026336c0] Lossless check failed - expected 00, calculated 96.
[truehd @ 026336c0] Lossless check failed - expected 00, calculated 78.

This is with a brand new Blu-ray (Valerian and the City of a Thousand Planets).


Thing is, I'd love to use ffmpeg directly, but I can't extract the Atmos stream with tsMuxer first (as it doesn't recognize it).

[mkver]

Thing is, I'd love to use ffmpeg directly, but I can't extract the Atmos stream with tsMuxer first (as it doesn't recognize it).
But you can use ffmpeg directly. A Bluray contains a slightly modified version of a transport stream that ffmpeg can handle. And if you build ffmpeg with libbluray, it can even parse playlists.

[asarian]

But you can use ffmpeg directly. A Bluray contains a slightly modified version of a transport stream that ffmpeg can handle. And if you build ffmpeg with libbluray, it can even parse playlists.

Thx.

Hmm, turns out ffmpeg (4.1) yielded the same errors. I can't say I'm hearing any distortion, but still.

[mkver]

This means that either something is wrong with your source or that something is wrong with ffmpeg's decoder. The best place to continue would probably be ffmpeg's bugtracker (https://trac.ffmpeg.org/).
What happens if you use the new truehd_core bitstream filter (designed to drop atmos extensions; usage: ffmpeg -i <inputfile> -vn -sn -c:a copy -bsf:a truehd_core -f <outputfile>)?
If the errors always happen at the same place, you could upload just the few seconds of the track for an issue. (Btw: Does this Bluray use seamess branching and is the error at a branchpoint?)

[asarian]

This means that either something is wrong with your source or that something is wrong with ffmpeg's decoder. The best place to continue would probably be ffmpeg's bugtracker (https://trac.ffmpeg.org/).
What happens if you use the new truehd_core bitstream filter (designed to drop atmos extensions; usage: ffmpeg -i <inputfile> -vn -sn -c:a copy -bsf:a truehd_core -f <outputfile>)?


Thx for the tip! To what can I output the TrueHD stream, though? FFmpeg won't let me output it to .wav (probably because wav is 6 channels only, iirc). I will need a stream I can downmix with eac3to (-down6) again.

[mkver]

The reason that ffmpeg won't let you mux the truehd stream into wav is because wav is a container for uncompressed PCM data, not for truehd data. You can e.g. mux it into Matroska. Just use a file with a "mka" extension and ffmpeg will know that you want to mux into Matroska.

Also keep in mind that if you intend to open a ticket in ffmpeg's bugtracker, you need to provide the truehd track with the atmos extension (or at least a snippet of it that allows to reproduce the issue).

[tebasuna51]

Same here

Same to what?

eac3to.exe valerian.m2ts 2: c:\video\temp.pcm -down6
MKV, 1 video track, 2 audio tracks, 3 subtitle tracks, 2:17:25, 24p /1.001
1: h265/HEVC, 2160p24 /1.001 (16:9), 10 bits
2: TrueHD (Atmos), English, 7.1 channels, 48kHz, 2419ms
"TrueHD Atmos 7.1"
3: AC3 EX, English, 5.1 channels, 640kbps, 48kHz, 2419ms
"AC-3 5.1-EX"
...
[truehd @ 026336c0] Lossless check failed - expected 00, calculated 78.

Is a m2ts or a mkv?

Please don't use .pcm, use .w64 or .wav instead.

Some messages can be ignored.

, but I can't extract the Atmos stream with tsMuxer first (as it doesn't recognize it).

tsMuxeR is a muxer, to extract use eac3to or ffmpeg or mkvextract if it is a mkv.

FFmpeg won't let me output it to .wav (probably because wav is 6 channels only, iirc). I will need a stream I can downmix with eac3to (-down6) again.

wav can have more than 6 channels, but can't be greater than 4 GB (limit for strict wav format) use .w64

If: eac3to valerian.m2ts 2: c:\video\temp.w64

and: ffmpeg -i valerian.m2ts -vn -acodec pcm_s24le c:\video\temp.w64

produce the same messages and output you can ignore them or assume is a corrupt input.

After that you can downmix the w64 (recommended with MeGUI or BeHappy).

[asarian]

Is a m2ts or a mkv?

Good catch. Too used to .m2ts files. :) This test actually was on the mkverge .mkv version of the .m2ts (video and audio only), on my media server. You got me thinking, though, that maybe it got corrupted somehow. I just loaded up the disc again, but it makes no difference: about the same amount of errors.

Please don't use .pcm, use .w64 or .wav instead.

What's wrong with .pcm?! I always convert every 7.1 source to .pcm (with -down6). .pcm is supposed to be lossless, right?

If: eac3to valerian.m2ts 2: c:\video\temp.w64

and: ffmpeg -i valerian.m2ts -vn -acodec pcm_s24le c:\video\temp.w64

produce the same messages and output you can ignore them or assume is a corrupt input.

After that you can downmix the w64 (recommended with MeGUI or BeHappy).

Hmm, that's sad: the latest ffmpeg also produces the same errors. So, maybe I should file a small bug report with them; but, in all honesty, I can't hear any errors at all; so maybe I'll just forget about it.

[LigH]

Raw PCM has no header. No information which sample rate, which bit depth, which number format, how many channels, which channel layout ... no details, only a byte stream of sample data. You will have to tell a reading application every attribute, or it won't be able to interpret the byte stream.

WAV has a header but is limited to 32 bit size fields (max. 4 GB); W64 expands this limit.

FLAC is losslessly compressed.

[asarian]

Raw PCM has no header. No information which sample rate, which bit depth, which number format, how many channels, which channel layout ... no details, only a byte stream of sample data. You will have to tell a reading application every attribute, or it won't be able to interpret the byte stream.

WAV has a header but is limited to 32 bit size fields (max. 4 GB); W64 expands this limit.

FLAC is losslessly compressed.

Thanks. Thing is, .mkv is just an intermediate stage for me: eventually everything gets muxed to an .m2ts container again. So, it kinda needs to stay LPCM (for which I then use the good old Pcm2Tsmu, to get tsMuxer to accept it).

[tebasuna51]

...then use the good old Pcm2Tsmu, to get tsMuxer to accept it.

Thanks, I write Pcm2Tsmu for that, but last tsMuxeR versions accept already .w64 like pcm input greater 4 GB.

Pcm2Tsmu is old and obsolete now.

[asarian]

Thanks, I write Pcm2Tsmu for that, but last tsMuxeR versions accept already .w64 like pcm input greater 4 GB.

Pcm2Tsmu is old and obsolete now.

Hehe, didn't even realize you wrote Pcm2Tsmu. :) Thanks for that! It's been an instrumental tool!

And yeah, even a 5.1 LPCM stream is nearly 7G. :p So, good tip on using .w64 format.

[Doom195051]

I have a bunch of MKVs with FLAC audio (varying from FLAC 1.0, 2.0 and 7.1) that I want to convert to DTS-HD MA.

I thought i could attempt to do this with eac3to alongside the arcsoft, surcode or DTS-HD Master Suite software. I'm running Windows 10 FYI.

Has anyone had good experience with latest windows trying to get this done? I run eac3to (using the UsEac3to GUI) and select the functionality to convert Track #2: FLAC to DTS from within eac3to.

I always end up getting the error: "Pressing the Surcode "Encode" button didn't seem to work..." I tried making the file names 1 character long and on a very short file path (i.e. F:/Movies/a.mkv) but still get the same error, on either 1.0.21 or 1.0.29 Surcode version.

I thought...well how about I just convert the FLAC to wavs (I tried it on a mono FLAC giving me just one center .wav file) then run it through Surcode...well Surcode converts that center mono file into a five channel DTS track (which I can successfully mux with the video but I feel like this is wrong...).

So lastly, I wanted to try with the actual DTS-HD Master Suite software, which seems like the most viable. I tried converting the single center channel .wav file but the encode just seems frozen in my queue as pending...


So any idea what would be the best way to tackle my goal of converting FLAC 1.0/2.0/7.1 from an MKV into DTS-HD MA 1.0/2.0/7.1 (respectively) in an MKV?

I saw this which seemed to indicate people don't reccommend DTS-HD Master Suite...
https://forum.doom9.org/showthread.php?t=158611

Also Arcsoft 6 doesn't even work on Window 10 anymore. I used this work around from another post that, when I run eac3to test, it detects Arcsoft dtsdecoder dll to be working (idk if it actually is though...).
https://forum.doom9.org/showthread.php?t=154070

[tebasuna51]

1) ArcSoft is a decoder, not a encoder. And is not necesary anymore because the free libdcadec.dll can do the decode of DTS-MA.

2) Surcode is a DTS encoder than can run with eac3to but can't produce DTS-MA, only standard DTS 768 or 1536 Kb/s. Forget it to recode FLAC.

3) If you want DTS-MA you need DTS-HD Master Suite and you only need eac3to to decode the FLAC's to monowav's:

eac3to input.flac output.wavs

Or in UsEac3to selecting 'wavs' in Output format
After that you can use DTS-HD Master Suite with the monowavs input:

Stream type DTS-HD Master Audio
Channel Layout: from 1.0 to 7.1

[Masutin]

Can I add something to "stdout.wav | LAME" to make it show progress? Otherwise any option of LAME for verbosity is ignored and no progress shows. E.g. with QAAC it shows fine.

[asarian]

Ran into this problem: HEVC multi-part demuxing? (https://forum.doom9.org/showthread.php?t=176040) Any chance HEVC demuxing will become a reality with eac3to?

Thanks.

[Atak_Snajpera]

Can you just join them with simple
copy /b 800.m2ts+801.m2ts+802.m2ts+803.m2ts+804.m2ts combined.m2ts

[asarian]

Can you just join them with simple
copy /b 800.m2ts+801.m2ts+802.m2ts+803.m2ts+804.m2ts combined.m2ts

Never mind my stupidity. :o Turns out you CAN demux the HEVC stream, just not to .mkv yet.

[JimmyBarnes]

Yes, is a workaround that works fine, unfortunately you lose the 7.1 data.

Your workaround was to add -core to -demux when demuxing E-AC3 streams which caused eac3to to abort extraction due to "Applying (E-)AC3 delay failed" error.

The audio streams I wanted were not the problem E-AC3 ones, so I extracted just that stream => X.thd+ac3.

Then I ran eac3to [path] 1) -demux -core and let it extract all the streams. The X.thd+ac3 produced was binary identical to the one above.

As you suggested, I would have imagined using -core would lose something, but it doesn't seem to have here, do you have any explanation?

[tebasuna51]

...As you suggested, I would have imagined using -core would lose something, but it doesn't seem to have here, do you have any explanation?

The -core eac3to parameter is intended to extract the core part (limited to 5.1) of a single audio track, before only with DTS-HD but now also EAC3 7.1.

A thd+ac3 is not a single audio track but 2 independent tracks interleaved.
To extract each track you can use:

eac3to X.thd+ac3 X.thd X.ac3

The -core parameter is ignored here.

[Masutin]

Can I remap a 4.1 audio to bring the second voice channel up front when playing in stereo? Software players can mix voice to both channels but my TV can't.

[tebasuna51]

Remap? Maybe you need downmix.

Try -downstereo

Put the log to know the input layer and formats.

[Masutin]

Correction: the audio is 3.1 - Front & Side L R. I decoded part of it and found the L voice channel (must be channel 0) silent. I did come across a 2.0 mix of this audio, it had voice in one channel. Downmixing as it is probably won't do. How do I go about mixing or remapping this? Other than decoding to WAVs and replacing the silent channel. Mono? I didn't keep the log and the source AC3 but if you think it can help I'll get it.

[tebasuna51]

3.1 with FL channel silent?
Is strange, without a sample I don't know the problem.

BTW eac3to have a way to remap channels, for instance:

eac3to input output -1,2,0,3

that send the input FL channel (0) to FC channel (3) in output

[zveroboy]

madshi

If I use VLC 228 or lower, then there are no errors in the recorded files.
But if I use VLC 306 or higher, then there are a lot of errors in the recorded files ([v01] Video overlaps for 7 frames at playtime 0:01:30. <WARNING>).

Could you explain the nature of these errors?
Are these realy errors in the records or is this an incorrect work of the eac3to?

Here are two videos recorded in 228 and 306

228 https://yadi.sk/i/TATk2aBmExgHcw

306 https://yadi.sk/i/ELXZ03Jk3hU-vQ

[Masutin]

Tebasuna, gracias! Do I need anything more to successfully apply -1,2,0,3? I get "Command line parameter "1,2,0,3" is unknown."

[tebasuna51]

@Masutin

Sorry, seems the remap parameter need at least 6 channel even if source have only 4:

eac3to input output -1,2,0,3,4,5

tested over a 3.1 input.

[Masutin]

Hola, Tebasuna! If I can trouble you again. I did what I should have and decoded the original 3.1 into C L R BC. C (ch. 2?) is the only voice channel. To get voice in both fronts, can you suggest how to mix or remap this? Sorry for misleading, the wrong details were from the encode.

[tebasuna51]

If you decode a source like:
eac3to v3.34
command line: eac3to 4a310.ac3 4a310.ac3_.wavs
----------------------------------------------
AC3, 3/1 channels, 0:00:20, 224kbps, 48kHz
Decoding with libav/ffmpeg...
Reducing depth from 64 to 24 bits...
Writing WAVs...
Creating file "D:\tmp\4a310.ac3_.L.wav"...
Creating file "D:\tmp\4a310.ac3_.R.wav"...
Creating file "D:\tmp\4a310.ac3_.C.wav"...
Creating file "D:\tmp\4a310.ac3_.BC.wav"...
eac3to processing took 1 second.
Done.

and the ..._.C.wav is the only voice channel I recommend you use:

eac3to input output.ac3 -downStereo -normalize

Or other output format (AAC recommended for stereo)

[wonkey_monkey]

A warning, as people still seem to be posting directly to threads and may not be aware:

It appears that the forum may have been hacked. There is a suspicious "test" announcement, apparently from tebasuna51 (but probably not), parts of the forum are not working, and there appear to be some malicious javascript files.

I'm not speaking in any official capacity here, but I would recommend, at the very least, NOT entering your password anywhere on Doom9 for the time being.

Please refer to this post: https://forum.doom9.org/showthread.php?goto=newpost&t=176128

tebasuna51, as you seem to have posted recently to this thread in particular, can you enlighten us?

[asarian]

Just a quick question. :) When extracting DTS-MA audio from an m2ts stream, giving a file a .dts extension will yield the same result as putting .dtshd, right? Whereas '.dts -core' would just get the old DTS core.

The 2 resultant streams look the same to me in tsMuxer (but you never know).

[tebasuna51]

Yes, a .dts can be a standard CBR DTS or a VBR DTS-HD (MasterAudio or HighResolution).

[Masutin]

Tebasuna, thanks again! Converting 3.1 (voice channel C) to 2.0 works! Voice in both channels! One last question on this. When recoding as 3.1, can there be a way to make QAAC preserve the order as L C R BC? A few remapping sequences I've tried produce only FL FR SL SR. Probably remapping won't help here.

[asarian]

Yes, a .dts can be a standard CBR DTS or a VBR DTS-HD (MasterAudio or HighResolution).

Thanks! :) (Just wanted to make sure it wasn't some header tricking me or something)

P.S. Glad to see you post again! Puts the matter of your allegedly compromised account at rest (see above).

[tebasuna51]

Tebasuna, thanks again! Converting 3.1 (voice channel C) to 2.0 works! Voice in both channels! One last question on this. When recoding as 3.1, can there be a way to make QAAC preserve the order as L C R BC? A few remapping sequences I've tried produce only FL FR SL SR. Probably remapping won't help here.

You dont need remap, all must be automatic.
A 3.1 with WAV order FL,FR,FC,BC can be converted to AAC with qaac to:

LC 48000Hz 4.0 (C L R Cs) -- 128,144,160,192,224,256,288,320,384,448,512,576,640 (bitrates)

qaac -v 128 --adts -o outLC128.aac inFL_FR_FC_BC.wav
work fine

but not to HE than only support:

HE 48000Hz 4.0 (L R Ls Rs) -- 64,80,96,112,128,160

qaac -v 128 --he --adts -o outHE128.aac inFL_FR_FC_BC.wav
ERROR: Channel layout not supported

Run: qaac --formats

to know the formats allowed

[asarian]

Hmm, eac3to marks a video at 1080i50. That's not true interlaced, right? (So, no need to give it 'InputType=0' in QTGMC, right?) And when I export it to .mkv, it just says 25fps.

[nevcairiel]

Why would that not be true interlaced? Its used all over europe in interlaced broadcasts.

[asarian]

Why would that not be true interlaced? Its used all over europe in interlaced broadcasts.

Because there are different kinds of 1080i (https://en.wikipedia.org/wiki/1080i). Pay particular attention to this sentence: "However, when 1080p material is captured at 25 or 30 frames/second, it is converted to 1080i at 50 or 60 fields/second, respectively, for processing or broadcasting." The blu-ray I am talking about appears to be 'pseudo-interlaced' like that (aka, no different top- and bottom field).

And since eac3to also just exports it to a 25fps stream, I am strengthened in my belief it's just 'fake' interlaced.

[nevcairiel]

50 fields interlaced makes 25 frames per second, thats how all PAL interlaced streams look. There is no real way to know without visual inspection of the footage. 1080i50 could be interlaced, or it could be fake-interlaced. The number alone tells you nothing.
Also, you gave no context that its even from a Blu-ray. But even then, those can also contain true interlaced streams.

[sneaker_ger]

Because there are different kinds of 1080i (https://en.wikipedia.org/wiki/1080i). Pay particular attention to this sentence: "However, when 1080p material is captured at 25 or 30 frames/second, it is converted to 1080i at 50 or 60 fields/second, respectively, for processing or broadcasting." The blu-ray I am talking about appears to be 'pseudo-interlaced' like that (aka, no different top- and bottom field).

And since eac3to also just exports it to a 25fps stream, I am strengthened in my belief it's just 'fake' interlaced.
How would we know when you don't provide any sample.

Also I'm not sure what you expect from running QTGMC "without 'InputType=0'" (which is the same as running it with that parameter as that's the default value) on progressive content.

[asarian]

Well, here's a sample (https://1drv.ms/u/s!AhSxhQ9g_mrMlgupngrxwSkSBV2n) :)

[asarian]

How would we know when you don't provide any sample.

Also I'm not sure what you expect from running QTGMC "without 'InputType=0'" (which is the same as running it with that parameter as that's the default value) on progressive content.

For every progressive material, I normally use 'InputType=1'. Only interlaced gets InputType=0.

[sneaker_ger]

Hmm, eac3to marks a video at 1080i50.
Well, here's a sample (https://1drv.ms/u/s!AhSxhQ9g_mrMjwZVIy9FW7JrOtCa) :)
That sample is 1080p60.

[asarian]

That sample is 1080p60.

Yikes, I linked the wrong sample. :) Here is the real one: proper sample (https://1drv.ms/u/s!AhSxhQ9g_mrMlgupngrxwSkSBV2n)

[sneaker_ger]

Progressive 1080p25 encoded as interlaced.

[asarian]

Progressive 1080p25 encoded as interlaced.

:thanks: That's what I thought. :)

[mkver]

Progressive 1080p25 encoded as interlaced.
That's not the whole story. It's MBAFF with pic_struct SEIs declaring every coded frame to be progressive; furthermore, bottom_field_pic_order_in_frame_present_flag is set to zero (meaning that the top and bottom fields of the coded frames have the same pic order count and even in the absence of a pic_struct SEI the coded frames should be considered progressive*). But it also has ct_type equal to 1 which means that the original source material is interlaced; furthermore the specs contain the clause that "Two consecutive fields in output order shall have different values of clockTimestamp when the value of ct_type for either field is 1 (interlaced)." This means that this stream is simply out-of-spec!

The reason that this particular sample is treated as interlaced is that FFmpeg uses the ct-type to override the pic_struct (https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/h264_slice.c#L1178) (notice that ct_type isn't simply the value read from the bitstream (https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/h264_sei.c#L93)). So changing the ct_type is one way of fixing this.

Notice that ffmpeg has a bug because it always flags MBAFF as interlaced in the absence of SEI (see here (https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/h264_slice.c#L1181)) regardless of the pic order count of the fields involved. This means that simply deleting the SEI would not change the behaviour of FFmpeg based players unless FFmpeg is fixed, too (and the player updated).

*: From the semantics of pic_struct:
"NOTE 6 – When pic_struct_present_flag is equal to 0, then in many cases default values may be inferred. In the absence of other indications of the intended display type of a picture, the decoder should infer the value of pic_struct as follows:
– If field_pic_flag is equal to 1, pic_struct should be inferred to be equal to (1 + bottom_field_flag).
– Otherwise, if TopFieldOrderCnt is equal to BottomFieldOrderCnt, pic_struct should be inferred to be equal to 0 [progressive frame].
– Otherwise, if TopFieldOrderCnt is less than BottomFieldOrderCnt, pic_struct should be inferred to be equal to 3 [TFF].
– Otherwise (field_pic_flag is equal to 0 and TopFieldOrderCnt is greater than BottomFieldOrderCnt), pic_struct should be
inferred to be equal to 4 [BFF]."

[sneaker_ger]

I always wonder what the guys writing MPEG specs are smoking.

[asarian]

@mkver, wow, that's a complicated story! :)

This means that this stream is simply out-of-spec!

That might explain why tsMuxer totally tripped on it (badly broken and jittery output). I was able to extract the main stream correctly with eac3to.

But it also has ct_type equal to 1 which means that the original source material is interlaced;

There's some visual evidence for that too; darn if I can find one now, but earlier, I saw a few frames with your typical interlaced 'stripes' artifacts in them (as if badly deinterlaced). It's this way on the original .m2ts blu-ray too (so it's not an eac3to extract thing).

[Xor]

Please simply help, i formatted and reinstall last version of eac3to 334 (also the surcode suite)

Test result

H:\eac3to334>EAC3TO -TEST
eac3to (v3.34) is up to date
Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed
http://www.nero.com/eng/store-blu-ray.html
CAUTION: You need Nero 7. Nero 8 won't work with eac3to.
ArcSoft DTS Decoder doesn't seem to be installed
http://www.arcsoft.com/products/totalmediatheatre
Sonic Audio Decoder (3.34.0.0) doesn't seem to be installed
Haali Matroska Muxer doesn't seem to be installed
http://haali.net/mkv
Nero AAC Encoder (1.5.4.0) is installed
Surcode DTS Encoder (1.0.29.0) is installed




I tried to convert a DTS track to AC3, for DEcoding using other tool (Decoding with libDcaDec DTS Decoder...)

H:\eac3to334>eac3to Test.dts Test.ac3 -640
DTS, 5.1 channels, 1:53:27, 1509kbps, 48kHz
Decoding with libDcaDec DTS Decoder...
Remapping channels...
Encoding AC3 <640kbps> with libAften...
Creating file "Test.ac3"...
eac3to processing took 0 minute, 41 seconds.
Done.


Surcode DTS Encoder (1.0.29.0) include DECODER or not ?

Thank's

[LigH]

No, the SurCode DTS encoder is an encoder only. But libavcodec in eac3to has a free DTS decoder, no need for additional software.

[asarian]

Another quick extension question. :) I extracted a TrueHD stream to a .thd file. So far so good. Is there a way I can get tsMuxer to accept it, though?

[tebasuna51]

tsMuxeR only accept tracks .thd+ac3

Extract the full TrueHD track like thd+ac3 or convert the .thd stream to .thd+ac3 with eac3to.

[asarian]

tsMuxeR only accept tracks .thd+ac3

Extract the full TrueHD track like thd+ac3 or convert the .thd stream to .thd+ac3 with eac3to.

Oops! Guess that's what I did wrong: I assumed (like with DTS-MA and .dts), that .thd would include the .ac3 sub-core. :thanks:

[Xor]

No, the SurCode DTS encoder is an encoder only. But libavcodec in eac3to has a free DTS decoder, no need for additional software.

d:\eac3to334>eac3to t1.dts t1_768.dts -768
DTS, 5.1 channels, 2:34:09, 1509kbps, 48kHz
Decoding with libDcaDec DTS Decoder...
Writing WAVs...
Creating file "t1_768.L.wav"...
Creating file "t1_768.R.wav"...
Creating file "t1_768.LFE.wav"...
Creating file "t1_768.C.wav"...
Creating file "t1_768.SR.wav"...
Creating file "t1_768.SL.wav"...
Encoding DTS <768kbps> with Surcode...
Surcode DTS Encoder doesn't seem to be installed.

Please show me an alternative encoder (to Minnetonka Surcode DTS) can support ENCODE DTSHD (@ 1509kbps) ????

[filler56789]

d:\eac3to334>eac3to t1.dts t1_768.dts -768
DTS, 5.1 channels, 2:34:09, 1509kbps, 48kHz
Decoding with libDcaDec DTS Decoder...
Writing WAVs...
Creating file "t1_768.L.wav"...
Creating file "t1_768.R.wav"...
Creating file "t1_768.LFE.wav"...
Creating file "t1_768.C.wav"...
Creating file "t1_768.SR.wav"...
Creating file "t1_768.SL.wav"...
Encoding DTS <768kbps> with Surcode...
Surcode DTS Encoder doesn't seem to be installed.

Please show me an alternative encoder (to Minnetonka Surcode DTS) can support ENCODE DTSHD (@ 1509kbps) ????

ffdcaenc-2 doesn't support DTSHD, only ordinary DCA, cannot be used by eac3to, but it is free.

The Master Audio Suite supports all types of DTS, including pure lossless DTSHD, but it's not free and cannot be used by eac3to.

[LigH]

I tried to convert a DTS track to AC3

I didn't expect you want to encode to DTS instead of AC3; eac3to has a (DCA Core) DTS decoder with libav (libdcadec), and an AC3 encoder with libav (ffmpeg AC3). For this task, my remark "no need for additional software" is valid. I did not mean to extend it to encoding to DTS, that's a different topic.

Lossy DTS (DCA Core) with high bitrate, compatible to DVD Video discs (1509.75 kbps), is not the lossless "DTS-HD", compatible to Blu-ray discs.

[tebasuna51]

Only 2 points:

ffdcaenc-2 doesn't support DTSHD, only ordinary DCA, cannot be used by eac3to
It can be used with 'pipe':

eac3to INPUT stdout.wav | ffdcaenc -i - -o output.dts -l -b 1509.75

eac3to has a (DCA Core) DTS decoder with libav (libdcadec), and an AC3 encoder with libav (ffmpeg AC3).
eac3to can't use libav (ffmpeg), it use libAften.dll (obsolete). But also can use ffmpeg with 'pipe':

eac3to INPUT stdout.w64 | ffmpeg -i - -c:a ac3 -b:a 640k -center_mixlev 0.707 output.ac3

[Xor]

ffdcaenc-2 doesn't support DTSHD, only ordinary DCA, cannot be used by eac3to, but it is free.

The Master Audio Suite supports all types of DTS, including pure lossless DTSHD, but it's not free and cannot be used by eac3to.

I have installed "Suite Surcode 1.0.23", work fine on ENcoding DTS.

But this version support DEcoding DTSHD ?

If Yes, how to force eac3to to use surcode for deconding dtshd (not use libDcaDec free ) ????

http://thumbs2.imagebam.com/97/d4/c5/20cc041195028104.jpg (http://www.imagebam.com/image/20cc041195028104)

Thank's

[nevcairiel]

But this version support DEcoding DTSHD ?

If Yes, how to force eac3to to use surcode for deconding dtshd (not use libDcaDec free ) ????

http://thumbs2.imagebam.com/97/d4/c5/20cc041195028104.jpg (http://www.imagebam.com/image/20cc041195028104)

Thank's

There is no reason to even entertain such a thought, because libdcadec supports full and flawless DTS-HD decoding. You don't need any external software for DTS-HD decoding.

[tebasuna51]

Like nevcairiel say there is no reason to eac3to use other decoder than libdcadec, btw eac3to can use other decoders than defaults like is explained in the first post of this thread.

To decode DTS-HD can't use Surcode but yes ArcSoft or Sonic (if installed), to override default decoders can be used parameters like:
-arcsoft
-sonic
-nero

[Bandits]

Request explanation of the following audio output.

Audio_3_English DELAY -17ms.THD+AC3

Does the "DELAY -17ms" mean:

A delay of -17ms on the audio needs to be applied when remuxing?
A delay of -17ms was applied to the audio output and nothing has to be done when remuxing?

If the delay was applied to the audio output, is there a way to turn off the creation of the filename with "DELAY -17ms" in it? Even when I specify a filename for the output eac3to still changes it when it wants a DELAY in the filename.

[tebasuna51]

Does the "DELAY -17ms" mean:

A delay of -17ms on the audio needs to be applied when remuxing

eac3to can't apply delay's to thd streams.

[Overdrive80]

Hi, I am converting audio between NTSC and FILM and I noticed that the final result, in duration, is distinctive to the expected one.

For example, starting with an audio of the following characteristics:

Duration: 24 min 37 sec 684 ms
Origin: NTSC
Samplerate: 96000 Hz
Bitdepth: 32 float

The conversion factor gives me a samplerate change of 95904.096.

When truncating this amount, in audition change the sampling rate to 95904 and the result is an audio with the following duration: 24 min 39 sec 163 ms

If I use the amount 95905, I get: 24 min 39 sec 147 ms which is approximated to that obtained with eac3to being 146 ms.

I can understand that we are using a multiplo of 5, for reasons that I do not understand, since I have made a conversion table, and this is the only conversion along with PAL to FILM, which disagree with the operations made from another editor. In the latter case, the conversion factor is 92067,932, which must be rounded to 92068 to obtain the expected value. This rounding I can understand but the previous one seems to me that rounding to 95905 is too much.

https://i.postimg.cc/j5CKTRBt/Captura-de-pantalla-2019-04-18-19-09-17.png

[tebasuna51]

Only can confirm your data:

The FILM -> NTSC conversion, with %_.wav -24.000 -changeTo23.976, over 1477.684 seconds wav output 1479.147 seconds when must be 1479.162
There are a error of 15 ms, all other conversions work fine.

You can use -r8brain parameter to obtain a exact output instead use the default ssrc routines.

[rco133]

Hi.

I have just installed a new PC, and am quite puzzled by how eac3to is reacting.

For some reason eac3to is now 2-3 times slower at demuxing than it is on my old system.

Demuxing a bluray on old system takes for example 9 minutes.

Demuxing the same bluray on the new system takes 20-30 minutes.

I am using the same command line on both systems, and have tried to use both HDD and SSD on both systems. I have not been able to find anything that makes the new system demux as fast as the new one though.

I have simply copied the eac3to folder from the old system to the new one. so there shouldn't be any changes in the installation.

I have also tried to do the demux through MeGUI, but I think it also uses eac3to, and is just as slow.

Old system is an 8 core i7 6900k and new one is 16 core i9 9960x.

Any ideas as to what might cause this slowdown?

Thanks in advance.

rco133

[Overdrive80]

Ok, thanks Tebasuna51 for the workaround.

[Overdrive80]

Hi again, I am trying use this line command in batch file: "-24.000 -changeTo23.976 -down32 -normalize -resampleTo48000 -r8brain", but process stop (blocked or paused) when *.pass1 file reach 1 GB.

Why this limitation?

Source file:

Origin: NTSC
Samplerate: 96000 Hz
Bitdepth: 32 float
Size: 3GB


Batch file:

@echo off
Title NTSC to FILM
COLOR A8

set "eac3to=C:\Portables\MeGUI\tools\eac3to\eac3to.exe" rem Put you directory for eac3to

md %~dp0Completed

for %%@ in (*.wav) do (

"%eac3to%" "%%@" "%%~dpn@_FILM.wav" -24.000 -changeTo23.976 -down32 -normalize -resampleTo48000 -r8brain -log=NUL
move "%%@" "%~dp0Completed" /Y
)

pause&exit

[Sparktank]

Aside from diagnostics working with blurays,etc.
Is there any advantage to using eac3to over FFMPEG for transcoding BluRay audio (THD/DTSHDMA/etc) to something else?

I notice that FFMPEG won't do stats about gaps like eac3to does.
But, FFMPEG is more friendly for updates.

[tebasuna51]

Hi again, I am trying use this line command in batch file: "-24.000 -changeTo23.976 -down32 -normalize -resampleTo48000 -r8brain", but process stop (blocked or paused) when *.pass1 file reach 1 GB.

I don't know what is the problem.

The process work without -normalize?

Maybe is not necesary at all or can be do in a second separate pass, now you force to do the resample for -changeTo23.976 and the -resampleTo48000 two times, in pass1 and in pass2.

[Bandits]

Hi.

I have just installed a new PC, and am quite puzzled by how eac3to is reacting.

For some reason eac3to is now 2-3 times slower at demuxing than it is on my old system.

Demuxing a bluray on old system takes for example 9 minutes.

Demuxing the same bluray on the new system takes 20-30 minutes.

I am using the same command line on both systems, and have tried to use both HDD and SSD on both systems. I have not been able to find anything that makes the new system demux as fast as the new one though.

I have simply copied the eac3to folder from the old system to the new one. so there shouldn't be any changes in the installation.

I have also tried to do the demux through MeGUI, but I think it also uses eac3to, and is just as slow.

Old system is an 8 core i7 6900k and new one is 16 core i9 9960x.

Any ideas as to what might cause this slowdown?

Thanks in advance.

rco133

Are you demuxing straight from the Blu-Ray? If you are, are you using the same Blu-Ray drive?

Riplock is a drive function that can slow your reading of video files. Ensure your drive does not use Riplock.

[tebasuna51]

@Sparktank
- Decode audio from THD/DTSHDMA/etc can be do with ffmpeg, more actualized than eac3to decoder dll's.

- Also ffmpeg have the best free AC3 encoder or the output can be piped to other free encoders like qaac.

- But I don't know how ffmpeg can manage audio from multiple m2ts sources like can be present in BD's.
Like you say eac3to can manage gaps/overlaps in the best know way.
By the moment I extract the audio tracks from BD's with eac3to (gui's UsEac3to or MeGUI) and recode after to desired format with ffmpeg or qaac (gui's UsEac3to, MeGUI or BeHappy).

[Overdrive80]

I don't know what is the problem.

The process work without -normalize?

Maybe is not necesary at all or can be do in a second separate pass, now you force to do the resample for -changeTo23.976 and the -resampleTo48000 two times, in pass1 and in pass2.

Tested separating process and same result. New script:

@echo off
Title NTSC to FILM
COLOR A8

IF NOT EXIST "%~dp0Completados" (md "%~dp0Completados")

set "eac3to=C:\Users\Isra-PC\Desktop\eac3to\eac3to.exe" rem Put you directory for eac3to

for %%@ in (*.wav) do (
::First pass, conversion
"%eac3to%" "%%@" "%temp%\temp.wav" -24.000 -changeTo23.976 -resampleTo48000 -down32 -r8brain -log=NUL

::Two pass, normalization
"%eac3to%" "%temp%\temp.wav" "%%~dpn@_FILM.wav" -normalize -down32 -log=NUL

::Remove temps files
del "%temp%\temp.wav"

::Move originals to completed
move "%%@" "%~dp0Completados"
)

pause&exit

Will try ffmpeg:

"%ffmpeg%" -hide_banner -nostdin -y -channel_layout stereo -i "%input%" -filter:a "atempo=0.9990009" -c:a pcm_f32le -ar 48000 -vn -sn "%temp%\temp.wav"

[an3k]

If anyone wants to get SurCode DTS Encoder working on Windows 10 this is how it goes:

1) Get the "SurCode DVD-DTS Professional" Setup, you can buy it at https://www.minnetonkaaudioshop.com/epages/MASIShop.sf/seccaf11703b2/?ObjectPath=/Shops/MASIShop/Products/3001-00106-000
2) Get "UniExtract" from https://www.legroom.net/software/uniextract (you want the "UniExtract Binary Archive" which is the "portable" version)
3) Start UniExtract.exe and select the Data1.cab from the "SurCode DVD-DTS Professional" zip. - It will extract the contents into a directory called "data1".
4) Create a directory called "Minnetonka Audio Software" inside "C:\Program Files (x86)"
5) Copy or Move directory "data1" into "C:\Program Files (x86)\Minnetonka Audio Software"
6) Rename directory "C:\Program Files (x86)\Minnetonka Audio Software\data1" to "C:\Program Files (x86)\Minnetonka Audio Software\SurCode DVD-DTS"
7) Create a new text document on your Desktop or somewhere else and rename the file "install.reg". Make sure it is not called "install.reg.txt" (show file-extensions). It should have a green-ish icon in front of the filename.
8) Copy&Paste the text from the quote-block below into that "install.reg" file.
9) Double-click the "install.reg" file and let Windows import the Registry settings.
10) run eac3to.exe -test and you should have SurCode support.
11) Activate your copy of "SurCode DVD-DTS Professional".

Windows Registry Editor Version 5.00

[HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion\App Paths\surcodedvd.exe]
"Path"="C:\\Program Files (x86)\\Minnetonka Audio Software\\SurCode DVD-DTS"
@="C:\\Program Files (x86)\\Minnetonka Audio Software\\SurCode DVD-DTS\\surcodedvd.exe"

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Minnetonka Audio Software]

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Minnetonka Audio Software\SurCode DVD-DTS]
"Home"="C:\\Program Files (x86)\\Minnetonka Audio Software\\SurCode DVD-DTS"

[HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Minnetonka Audio Software\SurCode DVD-DTS\1.0.10]


Q: Why?
A: Because the Installer used by Minnetonka for the SurCode DVD-DTS Encoder is so old it doesn't run on Windows 10 (even though it just copies files from A to B and writes some Registry values, nothing fancy).

Q: Is this piracy?
A: No. We just eliminate that crappy Installer. Activation of the product is done by the surcodedvd.exe itself.

Q: But I can use the encoder already without activating it. So this is piracy?
A: Congratulations, you have found the trial period :)

[Overdrive80]

Thanks for you tip, for complete it:

@echo off
Title Install SurCode
Color E0

goto check_Permissions

:check_Permissions
echo Administrative permissions required. Detecting permissions...

net session >nul 2>&1
if %errorLevel% == 0 (
goto main
) else (
ECHO.
ECHO. Failure: Current permissions inadequate. & ECHO.
ECHO. Press key for trying again.
pause>nul&exit
)


:main
ECHO The process has begin...

::Set paths to your files.
set "uniExtract=%~dp0uniextract161_noinst\UniExtract.exe"
set "dataCab=%~dp0Surcode DVD Pro DTS Encoder v1.0.29\Data1.cab"

::Extract the content
IF EXIST "%~dp0SurCode DVD-DTS" (
DEL "%~dp0SurCode DVD-DTS" /Q /F
)

"%UniExtract%" "%dataCab%" "%~dp0SurCode DVD-DTS"

::Create folder system
IF NOT EXIST "%ProgramFiles(x86)%\Minnetonka Audio Software" md "%ProgramFiles(x86)%\Minnetonka Audio Software"

::Move folder extracted to system
robocopy "%~dp0SurCode DVD-DTS" "%ProgramFiles(x86)%\Minnetonka Audio Software\SurCode DVD-DTS" /E /njh /njs /ndl /nc /ns >nul

::Install key registry
reg add "HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion\App Paths\surcodedvd.exe" /ve /t REG_SZ /d "C:\Program Files (x86)\Minnetonka Audio Software\SurCode DVD-DTS\surcodedvd.exe" /f >nul
reg add "HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion\App Paths\surcodedvd.exe" /v "Path" /t REG_SZ /d "C:\Program Files (x86)\Minnetonka Audio Software\SurCode DVD-DTS" /f >nul
reg add "HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Minnetonka Audio Software\SurCode DVD-DTS" /v "Home" /t REG_SZ /d "C:\\Program Files (x86)\\Minnetonka Audio Software\\SurCode DVD-DTS" /f >nul
reg add "HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Minnetonka Audio Software\SurCode DVD-DTS\1.0.10" /ve /t REG_SZ /f >nul

cls&ECHO The process has ended.
pause>nul&exit

[rco133]

Are you demuxing straight from the Blu-Ray? If you are, are you using the same Blu-Ray drive?

Riplock is a drive function that can slow your reading of video files. Ensure your drive does not use Riplock.

No, I am demuxing from SSD drive.

What I have found is that if I use megui and only select the tracks that I want, instead of using -demux directly with eac3to it goes much faster. Pretty much 4-5 times faster than if I extract all tracks.

If I use tsmuxer to extract all tracks, it takes half the time that it takes with eac3to.

I have no explanation, other then it must be something about how eac3to writes stuff to disk, which gives a big negative impact on Windows 10 compared to Windows 7. Other wise I have no explanation for what I am seing.

Anyway, I will just use megui and select the few tracks that I need, which will make the process much shorter, eventhough it also uses eac3to.

rco133

[an3k]

Anyway, I will just use megui and select the few tracks that I need, which will make the process much shorter, eventhough it also uses eac3to.

rco133

When MeGUI is using eac3to to demux the audio tracks you can open the task manager, go to "Processes" (the first tab), right-click onto the column header (Name, Status, CPU, etc.) and enable "command line". Go to the eac3to process launched by MeGUI and check the actual command line eac3to is invoked with. Maybe this gives a hint.

[Wildfire]

Perhaps it's just me, but I seem to be unable to find a way to automate converting DTS-HD audio tracks from MKV files (or better said, replacing the DTS-HD tracks with the core DTS tracks).

Currently, I'm using gMKVExtractGUI to extract the DTS-HD audio track from each file, then I extract the core DTS track using eac3to and finally I use MKVToolnix to replace the DTS-HD track with the core DTS track.

For dozens of files, that's a lot of work. Surely there's a way to automate the process, but for the moment I'm at a loss how to automate the process. I'm sure using eac3to and MKVToolnix there must be some way to script the entire process...

[sneaker_ger]

mkvmerge has a --reduce-to-core option.
https://mkvtoolnix.download/doc/mkvmerge.html

or ffmpeg dca_core bitstream filter
https://ffmpeg.org/ffmpeg-bitstream-filters.html#dca_005fcore

[Wildfire]

mkvmerge has a --reduce-to-core option.
https://mkvtoolnix.download/doc/mkvmerge.html

Thanks for the tip, now I'll have to figure out how to automate the process. Unfortunately, it's not as easy as "mkvmerge --reduce-to-core *.mkv *-DTS.mkv" or something like that :P

[Vicio]

Hi Guys, I'm having a question regarding audio conversion 7.1 (DTS MA-HD / TrueHD) to 5.1 AC3.
I was reading this tutorial: https://bodoicuho.ucoz.ru/ and I noticed that the conversion process for AC3 audios is somewhat laborious, needing to convert it to .w64 with eac3to and then using Sound Forge to get the RMS value from the central band and finally convert it for Dolby Digital.

However this method seems to me a bit old and searching here I came across this other method (which seems to me simpler and more current) of @tebasuna51: https://forum.doom9.org/showthread.php?t=175044

https://forum.doom9.org/showpost.php?p=1850153&postcount=14689

Where it recommends using the Megui for this purpose, the doubt is even with the Megui still need to convert the audio to .w64 and do the Sound Forge step to get the RMS value?

Below I'll be leaving two samples of two audios 7.1.
https://send.firefox.com/download/bff23123430354f6/#311orhBTI8w70DP2CMUTCA

I hope I have managed to express myself and apologize for my English, I am using a translator.

[tebasuna51]

Where it recommends using the Megui for this purpose, the doubt is even with the Megui still need to convert the audio to .w64 and do the Sound Forge step to get the RMS value?

MeGUI, or other free software, can't encode to Dolby Digital.
The best AC3 free encoder (ffmpeg) used with MeGUI or BeHappy is not a Dolby Digital encoder.

The recommended Dialog Normalization and Dynamic Range Compression with this encoder is the default (-31 dB and None), then is not necesary know the RMS value of center channel in Sound Forge (or with other method).

Take in mind than you are recoding a previous track, not encoding material recorded by you. Many encoder options in commercial encoders (filters, phase shift, etc.) are now unnecesary.

[Vicio]

MeGUI, or other free software, can't encode to Dolby Digital.
The best AC3 free encoder (ffmpeg) used with MeGUI or BeHappy is not a Dolby Digital encoder.
Yes, I'm aware of this, Megui is just a GUI, I just wanted to ask the question straight to the point without dwelling on technical terms. :)

The recommended Dialog Normalization and Dynamic Range Compression with this encoder is the default (-31 dB and None), then is not necesary know the RMS value of center channel in Sound Forge (or with other method).
Okay, that's what I wanted to hear. :goodpost:

Take in mind than you are recoding a previous track, not encoding material recorded by you. Many encoder options in commercial encoders (filters, phase shift, etc.) are now unnecesary.
Understood, noted here.
And thank you very much for the help, I read more half of the pages of this topic and I observed that you are a helpful person.
Greetings from Brazil.

[zeropc]

Guys, can eac3to no longer slowdown videos in h265?
I get the following error message

C:\Users\zeropc>"\\Mac\Home\Desktop\eac3to more gui\eac3to\eac3to.exe" "X:\evil2mkv"
1: "\\Mac\Home\Desktop\dead2\hdr.h264" 2: "\\Mac\Home\Desktop\dead2\dv.h264" -slowdown
MKV, 2 video tracks, 2 audio tracks, 1 subtitle track, 1:24:32, 24p
1: h265/HEVC, 2160p24 (16:9), 10 bits
2: h265/HEVC, 1080p24 (16:9), 10 bits - Dolby Vision Enhancement Layer
3: DTS Master Audio, English, 1.0 channels, 16 bits, 48kHz
(core: DTS, 1.0 channels, 768kbps, 48kHz)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
5: Subtitle (PGS), English
v01 The video framerate is correct, but rather unusual.
v02 The video framerate is correct, but rather unusual.
This video conversion is not supported.

[tebasuna51]

eac3to can't edit h265 (slowdown), but you want also recode to h264, for that the message is:
"This video conversion is not supported."

[zeropc]

eac3to can't edit h265 (slowdown), but you want also recode to h264, for that the message is:
"This video conversion is not supported."

AHHH!!!

Would be sweet if madshi can add this option for h265 :)

[SeeMoreDigital]

eac3to can't edit h265 (slowdown), but you want also recode to h264, for that the message is:
"This video conversion is not supported."Out of interest... Has anybody created a tool that can change some of the attributes of h265 elementary streams?

[nevcairiel]

Out of interest... Has anybody created a tool that can change some of the attributes of h265 elementary streams?

Depending on what partciular properties you are talking about, the h265_metadata bitstream filter in ffmpeg can change a bunch of them.

[sneaker_ger]

* hevc_metadata (https://ffmpeg.org/ffmpeg-bitstream-filters.html#hevc_005fmetadata)

[SeeMoreDigital]

Marvellous...

Thank-you Mr sneaker or is it Mr ger ;)

[nevcairiel]

How misleading, the file was called h265_metadata_bsf.c :D

[an3k]

I noticed a bug with eac3to v3.34 and Surcode DTS Encoder. When you convert a multichannel AC3 or WAV (for FLAC, etc.) to DTS it works fine. If you however want to convert a singlechannel (1.0 aka. mono) to DTS eac3to - or better Surcode throws an error. The very same file eac3to doesn't handle correctly is perfectly working if used directly inside the Surcode DTS Encoder software.

The error is:
C:\Program Files (x86)\MeGUI\tools\eac3to>eac3to.exe "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.wav" "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.dts"
WAV, 1.0 channels, 2:32:13, 24 bits, 1152kbps, 48kHz
Reading WAV...
Writing WAVs...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.C.wav"...
The original audio track has a constant bit depth of 24 bits.
Encoding DTS <768kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.29.0.
Surcode says/asks: "Invalid Wave File D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.dts.".
Surcode says/asks: "At least one valid source file must be specified to encode.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

Somehow eac3to pushes the destination filename to Surcode as the source filename.

Another bug is that if your source file is named eg. "foobar.C.wav" and you want to convert it to AC3 or DTS eac3to wants to convert that file to "foobar.C.wav" to make it compatible with the format AC3 or DTS expects. However, it throws an error because it just overwrote the source audio file with the temporary one :)

[tebasuna51]

@an3k
Encode a monowav to standard DTS 768 Kb/s with Surcode is usseless, you obtain a file with the same size and less quality than the PCM 16 bits (WAV).

eac3to don't need manage that conversion.

I can't reproduce the name problem encoding to AC3.

[an3k]

@an3k
Encode a monowav to standard DTS 768 Kb/s with Surcode is usseless, you obtain a file with the same size and less quality than the PCM 16 bits (WAV).

eac3to don't need manage that conversion.

I can't reproduce the name problem encoding to AC3.

The sense may be non-existend. I agree with you on that point but non-the-less it is a bug in eac3to.

Using the same file directly in Surcode works (even though the resulting DTS file is 5.1 channels). Because of that I went and created a DTS-HD MA using the DTS Master Audio Suite which created a mono file.

Yeah, if the target is AC3 it works without issues.

[an3k]

I noticed another weird behavior and don't know if eac3to is doing something wrong or me not understanding it.

eac3to v3.34
command line: eac3to.exe "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.ac3" "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.wavs"
------------------------------------------------------------------------------
AC3, 5.1 channels, 2:32:13, 448kbps, 48kHz
Decoding with libav/ffmpeg...
Reducing depth from 64 to 24 bits...
Writing WAVs...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.L.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.SL.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.C.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.SR.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.LFE.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.R.wav"...
Clipping detected, a 2nd pass will be necessary. <WARNING>
Starting 2nd pass...
Decoding with libav/ffmpeg...
Reducing depth from 64 to 24 bits...
Writing WAVs...
Applying -0.55dB gain...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.L.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.LFE.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.R.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.C.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.SL.wav"...
Creating file "D:\Encode\Movie (2008)\title_t03_track2_[ger]_DELAY 0ms.SR.wav"...
eac3to processing took 5 minutes, 17 seconds.
Done.

eac3to says Reducing depth from 64 to 24 bits... but MediaInfo says the ac3 is 16 bits. The second audio stream is TrueHD (MediaInfo says it's 16 bits too) and if I transcode that into individual wavs eac3to correctly use 16 bits.

[sneaker_ger]

AC3 and other lossy codecs don't have any inherent bitdepth despite what MediaInfo might suggest. The decoder can decode at 64 bits but it would be to wasteful to store 64 bit PCM/WAV so eac3to reduces the depth.

[an3k]

Oh, I see. Thank you for the information.

What about Dolby TrueHD? MediaInfo says 24 bits but eac3to says 16 bits and is reducing from 24 to 16 bits.

I want to keep as much quality as possible and the size of the intermediate wav files isn't a thing I have to worry about since I'm going to encode to a different format.

[sneaker_ger]

Lossless codecs do have "inherent" bitdepths. In your case it was encoded as 24 bits. But eac3to detects that throughout the complete file all of the respective 8 minor bits are 0s. So it decides to encode to 16 bit instead. This has absolutely no influence on quality.

[MeteorRain]

Is there any way to stop eac3to from adding DELAY to output file name?

I'm writing script to batch extract eac3 from files and convert them to wav for further processing.

eac3to x.mkv x.wav tells me This audio conversion is not supported..

eac3to x.mkv x.eac3 then eac3to x.eac3 x.wav works.
However eac3to will add DELAY XYZ ms to the file name so it's kinda hard to script that into a makefile.
Since this delay is usually less than 1 frame, I really don't care if it's applied correctly.

[sneaker_ger]

I would recommend using ffmpeg. It seems better at reading from mkv and also has a more feature-complete eac3 decoder IIRC.
ffmpeg -i "x.mkv" -map 0:a:0 "x.wav"

It can correct delays, too. Hope this is correct:
ffmpeg -i "x.mkv" -map 0:a:0 -af aresample=first_pts=0 "x.wav"

[MeteorRain]

Thanks. I'm aware of ffmpeg but still prefer eac3to if possible.

[thecoreyburton]

I often use eac3to to demux BD streams. In the past I've noted a common delay of just above 21ms (there's a remaining delay of less than 1ms) when dealing with DTSHD track transcoding (eg. DTSHD > FLAC). At the time, I read up on this a fair bit and did some of my own testing, eventually concluding that a simple "-21ms" parameter would be ideal when dealing with DTSHD in this way.

There's plenty of discussion about this, but a lot of it seems related to specific issues and I can't find anything conclusive about whether or not this is the right way to approach this. I'm not sure if there is a "correct" way to do this (for instance a trim based on samples / frames to make it align perfectly or some internal function that does this), or if I should be touch this value at all. I believe it's an encoder delay (and not a decoder delay, but I could be wrong) and so it stands to reason that if this is true, perhaps some BDs have been authored with compensation for this delay in mind.

I was wondering if the way I'm handling it is the correct procedure, or is there a better way?

[tebasuna51]

It is know than encoders add initial silence samples to initialize some encoders buffers, for instance if you recode to AC3 you obtain a delay of 5 ms, until 38 ms with some AAC encoders.

But with some encoders that delay is stored in metadata and any decoder must restore the audio without delay. For instance AAC in mp4 container (.m4a) must play without delay.
I can't reproduce the problem recoding DTSHD to FLAC, for me the obtained .flac is decoded without delay.

[zeropc]

Are we going to see a new version that supports HEVC? Last version is almost a year old

[sneaker_ger]

eac3to v3.32 "added support for h265/HEVC bitstream parsing". So your feature request seems too unspecific. Either way eac3to development seems pretty much stalled. Not much you can do except make a proper feature request on madshi's tracker and keep fingers crossed he finds time/interest to work on it again ...

[ryu34]

Hey I'm trying to use a sony x700 uhd bd player to play back my 4k uhd remuxes from a usb flash drive. It doesn't support true-hd playback from mkv containers, it can only do true-hd from m2ts, however dts-hd ma works fine from either. I thought i could over come this by just using the m2ts straight from the rip but when i playback that file i lose the ability to seek or skip chapters, anyone have any suggestions barring getting a new playback device? Thanks

[SeeMoreDigital]

Hey I'm trying to use a sony x700 uhd bd player to play back my 4k uhd remuxes from a usb flash drive. It doesn't support true-hd playback from mkv containers, it can only do true-hd from m2ts, however dts-hd ma works fine from either. I thought i could over come this by just using the m2ts straight from the rip but when i playback that file i lose the ability to seek or skip chapters, anyone have any suggestions barring getting a new playback device? ThanksSadly... With regard to av file/container support, all hardware playback devices have their limitations. And just-so-you-know, the Panasonic UHD players are just as limited as the Sony's in this regard!

[ryu34]

OK new question what kind of tools do i need to use with eac3to to demux 4k UHD dics rips and remux them into my own custom m2ts , edit i tried ts muxer and the audio desyncs from the video when using a true-hd track demuxed from the disc rip using eac3to.

[tebasuna51]

I don't know any free tool to manage 4k UHD m2ts.

tsMuxeR can't do it, read this (https://forum.doom9.org/showthread.php?p=1880216#post1880216), now the source code is released and maybe it can work in the future.

[stax76]

Can somebody with a trustworthy reputation provide updated libraries?

https://github.com/staxrip/staxrip/issues/32

[Sparktank]

Can somebody with a trustworthy reputation provide updated libraries?

https://github.com/staxrip/staxrip/issues/32

you can use Rarewares for libflac update (remember to use only the x86 build for eac3to).
www.rarewares.org/lossless.php

They use latest library.
According to hydrogenaud.io forums, the only real differences for FLAC 1.3.2 and 1.3.3 is mainly for behind the curtain updates.
Nothing useful for the end-user and not worth re-compressing your existing library that has used 1.3.2 in the past.

[stax76]

you can use Rarewares for libflac update (remember to use only the x86 build for eac3to).
www.rarewares.org/lossless.php

They use latest library.
According to hydrogenaud.io forums, the only real differences for FLAC 1.3.2 and 1.3.3 is mainly for behind the curtain updates.
Nothing useful for the end-user and not worth re-compressing your existing library that has used 1.3.2 in the past.

Thanks, I found everything I needed.

[samf88]

Hi all... not sure if this doubles as a help thread but hoping someone can spot where I'm going wrong with eac3to. I searched and found others have had this error, but help threads have fizzled out before anything was resolved.

All I want to do is demux *all* video/audio/subs/chapters tracks from a BD playlist on my hard drive to a separate directory (obviously I don't want to demux within the BD structure). I don't want to convert or encode anything, but it seems to think that's what I want to do. Where am I going wrong?

C:\>"D:\eac3to.exe" "F:\BD\MOVIE" 2) -demux "F:\eac3toDemux"
M2TS, 1 video track, 1 audio track, 6 subtitle tracks, 0:21:02, 24p /1.001
1: MPEG2, 1080p24 /1.001 (16:9)
2: AC3, English, 2.0 channels, 192kbps, 48kHz
3: Subtitle (PGS), English
4: Subtitle (PGS), Chinese
5: Subtitle (PGS), Dutch
6: Subtitle (PGS), French
7: Subtitle (PGS), Italian
8: Subtitle (PGS), Spanish
Track 2 is used for destination file "".
This audio conversion is not supported.

I understand I can manually select each track and give it a filename, but I was hoping to do that stuff further on in my workflow as this isn't a one off job.

All of the eac3to help pages seem to imply the -demux flag is for demuxing *everything* so I'm not sure where I'm going wrong. Thanks in advance!

[tebasuna51]

If tou want specify a separate directory try:

"D:\eac3to.exe" "F:\BD\MOVIE" 2) "F:\eac3toDemux\title.*"

[stax76]

Has anybody experienced:

The program can't start because hs is missing from your computer.

It first occurred after updating the DTS library. So far only one user reported it.

https://forum.doom9.org/showthread.php?p=1885544#post1885544

[tebasuna51]

Nope. I never see that message with eac3to 3.34 (W 10).

What is the eac3to version included in Staxrip 2.0.2.0 than work?

[stax76]

Nope. I never see that message with eac3to 3.34 (W 10).

What is the eac3to version included in Staxrip 2.0.2.0 than work?

Sorry about the late reply, the versions were probably the same, the difference was an updated DTS library I had compiled. The problem was solved meanwhile, it had something to do with a VC++ runtime dependency.

[tebasuna51]

... the difference was an updated DTS library I had compiled. The problem was solved meanwhile, it had something to do with a VC++ runtime dependency.

Please let us know if the updated library work with eac3to better than old one.

[kws53]

Hi all... not sure if this doubles as a help thread but hoping someone can spot where I'm going wrong with eac3to. I searched and found others have had this error, but help threads have fizzled out before anything was resolved.

All I want to do is demux *all* video/audio/subs/chapters tracks from a BD playlist on my hard drive to a separate directory (obviously I don't want to demux within the BD structure). I don't want to convert or encode anything, but it seems to think that's what I want to do. Where am I going wrong?

C:\>"D:\eac3to.exe" "F:\BD\MOVIE" 2) -demux "F:\eac3toDemux"
M2TS, 1 video track, 1 audio track, 6 subtitle tracks, 0:21:02, 24p /1.001
1: MPEG2, 1080p24 /1.001 (16:9)
2: AC3, English, 2.0 channels, 192kbps, 48kHz
3: Subtitle (PGS), English
4: Subtitle (PGS), Chinese
5: Subtitle (PGS), Dutch
6: Subtitle (PGS), French
7: Subtitle (PGS), Italian
8: Subtitle (PGS), Spanish
Track 2 is used for destination file "".
This audio conversion is not supported.

I understand I can manually select each track and give it a filename, but I was hoping to do that stuff further on in my workflow as this isn't a one off job.

All of the eac3to help pages seem to imply the -demux flag is for demuxing *everything* so I'm not sure where I'm going wrong. Thanks in advance!
I do this all the time.
1) Create a directory for the new files
2) Right click on the directory listing (at top) in Windows File Explorer, and select "Copy as Text"
3) In your CLI (Command Prompt in Windows), navigate to the correct DRIVE for your copied directory listing (typically type Drive letter, then colon), type "CD", then Right click and paste the directory text. Hit Enter.
4) Then in that directory, put in the location of your eac3to executable and the optical (or MOUNTED hard drive) that your ISO file is on.
5) For example: create a directory "Martian_1080" on Drive D. In Explorer, right click and copy this directory as text. Then in Command Prompt, type "D:", then ENTER. Then "CD ", then right click, paste. Then ENTER. You are now in the directory you wish to save your files.

Here is what it would look like: D:\Martian_1080>F:\video\eac3to a: 1) -demux (Actually omit the "1) -demux" portion initially to get a listing of all of the MPLS streams. Then select the appropriate stream 1), 2), etc.)

This will demux EVERYTHING on the ISO on drive a: and put it into D:\Martian_1080\

[Harrenkyym]

Hola, tebasuna, ¿qué tal? Escribiré en inglés, for everyone, but I miss so much mundodivx... :(

I have a question, as always, I keep building my own video library. When I want to sync the spanish audio with microhd movie, I get for example: 2: stdout.wav +6240ms -25.000 -changeTo23.976 | qaac -V 99 --ignorelength --adts --no-delay -o %_2eng.aac - At this point everything is OK, both movies have the same fps and I have calculated the difference between them, but what happens when everything is set and i discover that the audio in the spanish movie comes with a delay of 600ms already? How do I have to proceed? Thank you.

[tebasuna51]

Hello Harrenkyym! I don't know you with this nick in dead spanish forum mundodivx.

I can't explain 600ms but maybe less.

eac3to always apply the delay before the time expansion (25 -> 23.976), no matter if you use diferent parameter order:
+6240ms -25.000 -changeTo23.976
is the same than
-25.000 -changeTo23.976 +6240ms

Then the delay is expanded also to 6240 * (25*1,001/24) = 6506 ms
266 bigger than 6240

If you want a final delay of 6240 ms you must use:
+5984ms -25.000 -changeTo23.976

[SquallMX]

Seamless branching for new 4K titles seems to be broken. Audio goes out of sync, subtitles are also affected.

[l00t]

Seamless branching for new 4K titles seems to be broken. Audio goes out of sync, subtitles are also affected.

Not just for new titles, but for almost every seamless branched UHD discs. For these it is recommended to use DGDecNV or MakeMKV.

[SquallMX]

Not just for new titles, but for almost every seamless branched UHD discs. For these it is recommended to use DGDecNV or MakeMKV.

Yeah, I using MakeMKV as a workaround, unfortunately ac3to also fails when transcoding TrueHD to WAV/FLAC if the MKV was created from a Seamless Branched disc using MakeMKV.

[Boulder]

I think you should create an issue at http://bugs.madshi.net/main_page.php, madshi has fixed critical bugs even if the application is not actively developed.

[redbtn]

I think you should create an issue at http://bugs.madshi.net/main_page.php, madshi has fixed critical bugs even if the application is not actively developed.
It's already there over 1.5 years
http://bugs.madshi.net/view.php?id=542

[Boulder]

It's already there over 1.5 years
http://bugs.madshi.net/view.php?id=542

Uh, doesn't look too good then :(

I wonder if ffmpeg can handle such streams joined by MakeMKV?

[redbtn]

Uh, doesn't look too good then :(
I wonder if ffmpeg can handle such streams joined by MakeMKV?I've heard that DGDecNV can demux this correctly, but I don't have it to check.

Yeah, I using MakeMKV as a workaround, unfortunately ac3to also fails when transcoding TrueHD to WAV/FLAC if the MKV was created from a Seamless Branched disc using MakeMKV.
It fails with error or output flac doesn't sync with video?

[videoh]

I've heard that DGDecNV can demux this correctly, but I don't have it to check. The latest version can indeed correctly demux all elementary streams, subtitles, and chapters in sync with the video. It's been tested on several disks, such as the UHD version of Cars 2 (which has boatloads of M2TS files in the playlist). If any other streams are not handled correctly please let us know and they'll be addressed straight away. Here is an interesting thread on the matter:

http://rationalqm.us/board/viewtopic.php?f=8&t=780

wherein Bullwinkle advised how to do this sync correction correctly. Possibly madshi is interested in his HW offerings these days.

[Boulder]

The latest version can indeed correctly demux all streams in sync with the video. I have tested it on several disks, such as Cars 2. If any other streams are not handled correctly please let me know and I will address them straight away. Here is an interesting thread on the matter:

http://rationalqm.us/board/viewtopic.php?f=8&t=780

Hehe, must have been the best 15 bucks I ever spent that I put in the DGDecNV license :D

[videoh]

DGDecNV wants to thank you for all your great contributions over the years, Boulder, and for your contributions to the entire video community!

[videoh]

We could make a stand-alone demuxer that doesn't require an nVidia card.

[l00t]

Hehe, must have been the best 15 bucks I ever spent that I put in the DGDecNV license :D

It's definitely worth buying ;)

We could make a stand-alone demuxer that doesn't require an nVidia card.

That would be like Christmas :)

[videoh]

Ordering my Santa outfit. :)

We can make it freeware, why not? DGDemux sounds good. Take a bluray/uhd playlist and deliver all elementary streams, subtitles, and chapters in sync.

MakeMKV uses container timestamps to fix things up AFAIK, for ES demuxing you are in trouble.

[SeeMoreDigital]

Ordering my Santa outfit. :)

We can make it freeware, why not? DGDemux sounds good. Take a bluray/uhd playlist and deliver all elementary streams, subtitles, and chapters in sync.Amazing :)

[SquallMX]

I've heard that DGDecNV can demux this correctly, but I don't have it to check.


It fails with error or output flac doesn't sync with video?

It fails with an 'Lossless Check Failed - Expected XX, Calculated YY' error, usually around a seamless branching scene.

[Boulder]

I think those errors are not an issue, they've always been there as far as I remember.

[redbtn]

It fails with an 'Lossless Check Failed - Expected XX, Calculated YY' error, usually around a seamless branching scene.It happens often when I convert DTS XXL to FLAC. But with TrueHD I didn't notice it. And of course any not lossless format gives this messages.

[SquallMX]

I think those errors are not an issue, they've always been there as far as I remember.

I think MakeMKV is not fixing the truehd audio properly.

TrueHD audio from seamless branching title created by DGIndexNV to FLAC conversion works OK
TrueHD audio from seamless branching title created by eac3to to FLAC conversion works OK but is out of sync
TrueHD audio from seamless branching title created by MakeMKV to FLAC conversion fails

Looks like DGIndexNV is the best tool for the job,
who would have thought that, best 15 dollars I've ever spent.

[videoh]

I think MakeMKV is not fixing the truehd audio properly.

TrueHD audio from seamless branching title created by DGIndexNV to FLAC conversion works OK
TrueHD audio from seamless branching title created by eac3to to FLAC conversion works OK but is out of sync
TrueHD audio from seamless branching title created by MakeMKV to FLAC conversion fails
Are you using the TrueHD stream or its embedded AC3?

Looks like DGIndexNV is the best tool for the job,
who would have thought that, best 15 dollars I've ever spent. Looks like I got lucky, because I don't even do gap processing for TrueHD (although I do for the embedded AC3). It appears to not be necessary for TrueHD, and perhaps the problem for eac3to is that it does this processing when not needed. The eac3to algorithm does not look good to me (deleting duplicated audio frames AFAIK). Can you try with that -no2ndpass thingie (assuming you are using the TrueHD and not its embedded AC3)?

[videoh]

Amazing :) He he, now I have to deliver it. No way I can let you down. :D

Created the project today and started coding... Whee!

[hubblec4]

Hi videoh

That are great news.
Could you keep in mind your project should run on Linux and Windows and we need all what eac3to can do plus a bit more.
In my case I have often seamless branching BDs and eac3to has some bugs for e-ac3 streams.
Skip identical frames works almost for all cases but not 100%.

[von Suppé]

He he, now I have to deliver it. No way I can let you down. :D

Created the project today and started coding... Whee!

Thanks in advance, great!

Would next week be possible? I've decided to spend my end-of-year holidays early and I'm eager to unpack my X-mas presents. :D

[jpsdr]

Be realistic.
As good as he is, you can't expect videoh to produce all the eac3to features within the snap of a fingers.
It will be allready very great if there is a simple demuxer wich demux properly streams where eac3to fails.

[von Suppé]

I was just joking. My apologies.

[videoh]

I took it as a joke. :)

[videoh]

Just successfully did a first full demux from Cars 2 UHD using DGDemux. I have to clean up some stuff, remove the licensing, and make a manual. Maybe by Thanksgiving, or even next week! ;)

I will start a new thread for further status reports, as it doesn't belong here.

EDIT: New thread is here:

https://forum.doom9.org/showthread.php?p=1889434#post1889434

[MrVideo]

Found a problem with eac3to v3.34. It freezes while attempting to analyze a full ABC network feed mux. It gets to 96% and halts.

A sample tsreader capture of the mux is here (http:/vidiot.com/video/ABC-H265-mux-test.zip) (53,879,018 bytes).

MediaInfo reads it just fine.

[LigH]

MediaInfo does not analyze the whole file, only the file parts where it assumes to find the headers with the attributes describing the content (usually in the beginning of a file, sometimes in the end as well). It will most probably skip parts with possible broadcast data errors.

[tebasuna51]

Maybe there are some problem at end of mux (96%), MediaInfo never read all the stream, only the headers.

ffmpeg info say:
ts streams:

41: hevc (Rext), yuv422p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], 59.94 fps
36(aaa,aab): mp2, 48000 Hz, stereo, s16p, 384 kb/s
38(aac,aad): mp2, 48000 Hz, stereo, s16p, 384 kb/s
37(aae,aaf): mp2, 48000 Hz, stereo, s16p, 384 kb/s
35(aag,aah): mp2, 48000 Hz, stereo, s16p, 384 kb/s
29(eng): mp2, 48000 Hz, stereo, s16p, 256 kb/s
15: hevc (Rext), yuv422p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], 59.94 fps
22(aaa,aab): mp2, 48000 Hz, stereo, s16p, 384 kb/s
24(aac,aad): mp2, 48000 Hz, stereo, s16p, 384 kb/s
20(aae,aaf): mp2, 48000 Hz, stereo, s16p, 384 kb/s
23(aag,aah): mp2, 48000 Hz, stereo, s16p, 384 kb/s
12(eng): mp2, 48000 Hz, stereo, s16p, 256 kb/s
1: hevc (Rext), yuv422p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], 59.94 fps,
30(aaa,aab): mp2, 48000 Hz, stereo, s16p, 384 kb/s
32(aac,aad): mp2, 48000 Hz, stereo, s16p, 384 kb/s
31(aae,aaf): mp2, 48000 Hz, stereo, s16p, 384 kb/s
33(aag,aah): mp2, 48000 Hz, stereo, s16p, 384 kb/s
39(eng): mp2, 48000 Hz, stereo, s16p, 256 kb/s
40: hevc (Rext), yuv422p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], 59.94 fps
17(aaa,aab): mp2, 48000 Hz, stereo, s16p, 384 kb/s
21(aac,aad): mp2, 48000 Hz, stereo, s16p, 384 kb/s
16(aae,aaf): mp2, 48000 Hz, stereo, s16p, 384 kb/s
18(aag,aah): mp2, 48000 Hz, stereo, s16p, 384 kb/s
8(eng): mp2, 48000 Hz, stereo, s16p, 256 kb/s
19: hevc (Rext), yuv422p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], 59.94 fps
25(aaa,aab): mp2, 48000 Hz, stereo, s16p, 384 kb/s
28(aac,aad): mp2, 48000 Hz, stereo, s16p, 384 kb/s
26(aae,aaf): mp2, 48000 Hz, stereo, s16p, 384 kb/s
27(aag,aah): mp2, 48000 Hz, stereo, s16p, 384 kb/s
34(eng): mp2, 48000 Hz, stereo, s16p, 256 kb/s
42: hevc (Rext), yuv422p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], 59.94 fps
3(aaa,aab): mp2, 48000 Hz, stereo, s16p, 384 kb/s
5(aac,aad): mp2, 48000 Hz, stereo, s16p, 384 kb/s
4(aae,aaf): mp2, 48000 Hz, stereo, s16p, 384 kb/s
6(aag,aah): mp2, 48000 Hz, stereo, s16p, 384 kb/s
10(eng): mp2, 48000 Hz, stereo, s16p, 256 kb/s

And show many errors and warnings when try to extract some track.
Try yourself:
ffmpeg -i "ABC-H265-mux-test.ts" -map 0:42 -vcodec copy "ABC-H265-mux-test.ts_42.h265"

[MrVideo]

I tried opening the file with tsmuxer (2.6.15) and it complained about pieces of it and ignored a couple of video streams and audio streams. So, I did another capture of 500 MB and tsmuxer didn't like it either. So I read the new file into VideoReDoPro and extracted each of the 6 programs (1 video stream and 5 audio streams). Not one error was reported. I remuxed the 6 TS files into a M2TS file via tsmuxer and eac3to liked it.

I have no idea what is wrong with a complete mux capture that causes eac3to and tsmuxer to not like it. If I just capture one of the program streams, eac3to doesn't have an issue with it.

Thanks for the pointers.

As a side note, I normally do not capture the ABC network feed. I'm only doing it lately because of the Jeopardy! special programming. And because ABC switched over from H.264 to H.265, I was also testing out my ZShell script extract a single stream and extract three of the audio streams to convert the MPA to DD5.1 AC3. When eac3to hung I thought that the new 3.34 version had issues. Turns out that the ABC network feed has something in it that eac3to and tsmuxer do not like, when captured as a whole.

[SeeMoreDigital]

Hi @MrVideo,

It's been a few years since I've played around with transport streams that contain multiple TV channels. But I seem to remember that Mosu's MKVToolNix did quite a good job at identifying all the separate streams...

[MrVideo]

I know what all of the streams are. I don't use eac3to to identify the streams, but to extract one of the program's video and audio streams in order to create 6 separate wave files from the MPA stereo audio streams so that I can use ffmpeg to create an AC3 5.1 file that I'll then tsmuxer into a new TS file. I have a Zshell script that runs all of the pieces. If eac3to can't find all of the streams, it means that I have to use VideoReDo to pull what I want into a new stream that will then be used to create the AC3 file. That said, it means that I should only capture the program that I need, which is what I am doing as I write this.

There is always a way around an issue.

[Bandits]

I have a UHD disc with different audio streams with different delays. The THD+AC3 (Atmos) stream has a -5ms delay and I was told earlier in this thread that eac3to cannot apply a delay to those streams. This results in the file name being appended with a "DELAY -5ms". The other audio streams show a -17ms delay but the file names are not appended. Looking at the log, it shows the -17ms streams have had a adjustment applied to the stream by eac3to. It also shows that +15ms of the delay could not be applied to the same streams.

Is the timing of these audio streams correct when eac3to corrects them even when showing it couldn't correct it completely?

eac3to v3.34
command line: "C:\eac3to.exe" "C:\XXXXX" 1) 1: "C:\Chapters.TXT" 2: "C:\Video_2_HEVC.h265" 3: "C:\Audio_3_English.THD+AC3" 4: "C:\Audio_4_French.AC3" 5: "C:\Audio_5_Italian.AC3" 6: "C:\Audio_6_Japanese.AC3" 7: "C:\Audio_7_Portuguese.AC3" 8: "C:\Audio_8_Spanish.AC3" 9: "C:\Audio_9_Thai.AC3" 10: "C:\Audio_10_English.AC3" 11: "C:\Subtitle_11_English.sup" 12: "C:\Subtitle_12_English.sup" 13: "C:\Subtitle_13_English.sup" 14: "C:\Subtitle_14_English.sup" 15: "C:\Subtitle_15_Chinese.sup" 16: "C:\Subtitle_16_Chinese.sup" 17: "C:\Subtitle_17_French.sup" 18: "C:\Subtitle_18_Italian.sup" 19: "C:\Subtitle_19_Japanese.sup" 20: "C:\Subtitle_20_Korean.sup" 21: "C:\Subtitle_21_Portuguese.sup" 22: "C:\Subtitle_22_Spanish.sup" 23: "C:\Subtitle_23_Thai.sup" 24: "C:\Subtitle_24_French.sup" 25: "C:\Subtitle_25_Japanese.sup" 26: "C:\Subtitle_26_English.sup" 27: "C:\Subtitle_27_Italian.sup" 28: "C:\Subtitle_28_Japanese.sup" 29: "C:\Subtitle_29_Korean.sup" 30: "C:\Subtitle_30_Spanish.sup" -progressnumbers -minPlayListRuntime=0:01:00 -log="C:\log.txt"
------------------------------------------------------------------------------
M2TS, 1 video track, 8 audio tracks, 20 subtitle tracks, 2:18:01, 11.988p
1: Chapters, 16 chapters
2: h265/HEVC, 2160p24 /1.001 (16:9), 10 bits
3: TrueHD/AC3 (Atmos), English, 7.1 channels, 48kHz, -5ms
(embedded: AC3, 5.1 channels, 640kbps, 48kHz)
4: AC3 Surround, French, 2.0 channels, 192kbps, 48kHz, -17ms
5: AC3, Italian, 5.1 channels, 640kbps, 48kHz, -17ms
6: AC3, Japanese, 5.1 channels, 640kbps, 48kHz, -17ms
7: AC3, Portuguese, 5.1 channels, 640kbps, 48kHz, -17ms
8: AC3, Spanish, 5.1 channels, 384kbps, 48kHz, -17ms
9: AC3, Thai, 5.1 channels, 384kbps, 48kHz, -17ms
10: AC3 Surround, English, 2.0 channels, 192kbps, 48kHz, -17ms
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), English
14: Subtitle (PGS), English
15: Subtitle (PGS), Chinese
16: Subtitle (PGS), Chinese
17: Subtitle (PGS), French
18: Subtitle (PGS), Italian
19: Subtitle (PGS), Japanese
20: Subtitle (PGS), Korean
21: Subtitle (PGS), Portuguese
22: Subtitle (PGS), Spanish
23: Subtitle (PGS), Thai
24: Subtitle (PGS), French
25: Subtitle (PGS), Japanese
26: Subtitle (PGS), English
27: Subtitle (PGS), Italian
28: Subtitle (PGS), Japanese
29: Subtitle (PGS), Korean
30: Subtitle (PGS), Spanish
Creating file "C:\Chapters.TXT"...
[a06] Extracting audio track number 6...
[v02] Extracting video track number 2...
[s30] Extracting subtitle track number 30...
[s22] Extracting subtitle track number 22...
[s20] Extracting subtitle track number 20...
[s27] Extracting subtitle track number 27...
[s21] Extracting subtitle track number 21...
[a09] Extracting audio track number 9...
[a09] Applying (E-)AC3 delay...
[a09] A remaining delay of +15ms could not be fixed.
[s29] Extracting subtitle track number 29...
[s19] Extracting subtitle track number 19...
[a04] Extracting audio track number 4...
[a04] Applying (E-)AC3 delay...
[a04] A remaining delay of +15ms could not be fixed.
[s17] Extracting subtitle track number 17...
[a03] Extracting audio track number 3...
[a10] Extracting audio track number 10...
[a10] Applying (E-)AC3 delay...
[a10] A remaining delay of +15ms could not be fixed.
[a07] Extracting audio track number 7...
[a07] Applying (E-)AC3 delay...
[a07] A remaining delay of +15ms could not be fixed.
[s18] Extracting subtitle track number 18...
[s16] Extracting subtitle track number 16...
[s12] Extracting subtitle track number 12...
[a08] Extracting audio track number 8...
[a08] Applying (E-)AC3 delay...
[a08] A remaining delay of +15ms could not be fixed.
[s23] Extracting subtitle track number 23...
[s28] Extracting subtitle track number 28...
[a05] Extracting audio track number 5...
[a05] Applying (E-)AC3 delay...
[a05] A remaining delay of +15ms could not be fixed.
[a06] Applying (E-)AC3 delay...
[a06] A remaining delay of +15ms could not be fixed.
[s14] Extracting subtitle track number 14...
[s24] Extracting subtitle track number 24...
[s25] Extracting subtitle track number 25...
[s11] Extracting subtitle track number 11...
[s13] Extracting subtitle track number 13...
[s26] Extracting subtitle track number 26...
[s15] Extracting subtitle track number 15...
[v02] Creating file "C:\Video_2_HEVC.h265"...
[a03] Creating file "C:\Audio_3_English.THD+AC3"...
[a06] Creating file "C:\Audio_6_Japanese.AC3"...
[a05] Creating file "C:\Audio_5_Italian.AC3"...
[a07] Creating file "C:\Audio_7_Portuguese.AC3"...
[a08] Creating file "C:\Audio_8_Spanish.AC3"...
[a09] Creating file "C:\Audio_9_Thai.AC3"...
[a04] Creating file "C:\Audio_4_French.AC3"...
[a10] Creating file "C:\Audio_10_English.AC3"...
[s26] Creating file "C:\Subtitle_26_English.sup"...
[s27] Creating file "C:\Subtitle_27_Italian.sup"...
[s30] Creating file "C:\Subtitle_30_Spanish.sup"...
[s28] Creating file "C:\Subtitle_28_Japanese.sup"...
[s29] Creating file "C:\Subtitle_29_Korean.sup"...
[s13] Creating file "C:\Subtitle_13_English.sup"...
[s11] Creating file "C:\Subtitle_11_English.sup"...
[s15] Creating file "C:\Subtitle_15_Chinese.sup"...
[s12] Creating file "C:\Subtitle_12_English.sup"...
[s14] Creating file "C:\Subtitle_14_English.sup"...
[s16] Creating file "C:\Subtitle_16_Chinese.sup"...
[s18] Creating file "C:\Subtitle_18_Italian.sup"...
[s23] Creating file "C:\Subtitle_23_Thai.sup"...
[s21] Creating file "C:\Subtitle_21_Portuguese.sup"...
[s22] Creating file "C:\Subtitle_22_Spanish.sup"...
[s17] Creating file "C:\Subtitle_17_French.sup"...
[s20] Creating file "C:\Subtitle_20_Korean.sup"...
[s19] Creating file "C:\Subtitle_19_Japanese.sup"...
[s24] Creating file "C:\Subtitle_24_French.sup"...
[s25] Creating file "C:\Subtitle_25_Japanese.sup"...
Video track 2 contains 198533 frames.
Subtitle track 11 contains 1946 captions.
Subtitle track 12 contains 1832 captions.
Subtitle track 13 contains 1946 captions.
Subtitle track 14 contains 1854 captions.
Subtitle track 15 contains 2101 captions.
Subtitle track 16 contains 2145 captions.
Subtitle track 17 contains 1250 normal and 5 forced captions.
Subtitle track 18 contains 1720 normal and 6 forced captions.
Subtitle track 19 contains 1874 captions.
Subtitle track 20 contains 1863 captions.
Subtitle track 21 contains 1870 normal and 6 forced captions.
Subtitle track 22 contains 1988 normal and 5 forced captions.
Subtitle track 23 contains 1832 normal and 9 forced captions.
Subtitle track 24 contains 4 forced captions.
Subtitle track 25 contains 4 captions.
Subtitle track 26 contains 2055 captions.
Subtitle track 27 contains 1713 captions.
Subtitle track 28 contains 346 captions.
Subtitle track 29 contains 1983 captions.
Subtitle track 30 contains 1754 captions.
eac3to processing took 13 minutes, 49 seconds.
Done.

[tebasuna51]

The only way to correct delay, without recode, is add/delete full audio frames to extracted streams.

- eac3to can't do it to TrueHD streams.
- Over AC3 streams can add/delete frames of 32 ms, then over -17 ms delete 1 frame and the output need a + 15 ms of furter delay to be applied in the next container.
- BTW delays lower than a video frame duration, here 41.7 ms, can be ignored safely because can't be detected.
- If you insist to extract a stream with 0 ms of delay you need recode the AC3 (not recommended) with something like:

8: stdout.w64 -17ms | ffmpeg -i - -c:a ac3 -b:a 640k -center_mixlev 0.707 %_8spa.ac3

[Bandits]

I have a UHD disc that outputs the following:

eac3to v3.34
command line: "C:\eac3to.exe" "C:\XXXXX" 1) -log="C:\log.txt"
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 5 subtitle tracks, 2:13:12, 11.987p
1: Chapters, 16 chapters
2: h265/HEVC, unknown parameters
3: DTS Master Audio, English, 7.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
4: AC3, Spanish, 5.1 channels, 640kbps, 48kHz
5: AC3, French, 5.1 channels, 640kbps, 48kHz
6: AC3 Surround, English, 2.0 channels, 224kbps, 48kHz
7: DTS, English, 2.0 channels, 768kbps, 48kHz
8: Subtitle (PGS), English
9: Subtitle (PGS), French
10: Subtitle (PGS), English
11: Subtitle (PGS), French
12: Subtitle (PGS), Spanish

The video track is not correctly identified.

Using bdinfo.exe on the disc shows:

2160p / 23.976 fps / 16:9

Using the new tsMuxer.exe on the disc shows:

Profile: Main 10@5.1 Resolution 3840:2160p Frame rate 23.976

I don't know if tsMuxer.exe is just faking the detection or can see something that eac3to.exe and bdinfo.exe can't. If I use txMuxer.exe to demux/remux the output is still detected the same way as "unknown parameters".

Is there something that eac3to.exe is missing or is the video track just not made right? This is the first time I have seen this message.

[tebasuna51]

Last eac3to version is from 2017, the new tsMuxeR, updated at present, maybe can do a better job.

The last version of BDInfo is from 2018.

[Yoshi]

I'm not sure whether this issue has already been covered here before as on a pre-search, the closest I found was only this post (https://forum.doom9.org/showpost.php?p=1804163&postcount=14210).

With several movies now which provide a 7.1 mix such as "Dredd" or "Ford v Ferrari" (and no matter whether Dolby TrueHD or DTS-HA MA), I encountered clipping when downmixing to 5.1. It goes undetected and eac3to doesn't issue a second pass. However, if I inject any volume change, like -0.1dB, then the audio all of the sudded isn't only treated as a 64 bit source (probably due to the internal workings of the libav/ffmpeg decoder's gain control), but also clipping is correctly recognized, triggering a second pass with an often more severe volume reduction.

To me, this doesn't make any sense as the clipping should be recognized without any manual gain intervention by the user.

Either I missed something here or this is a real bug in eac3to.

[Richard1485]

With several movies now which provide a 7.1 mix such as "Dredd" or "Ford v Ferrari" (and no matter whether Dolby TrueHD or DTS-HA MA), I encountered clipping when downmixing to 5.1.

Yes, I've come across such clipping as well, so I usually check the output in Audacity. You could decode the 7.1 to w64 with eac3to and then use BeHappy to downmix to 5.1. It uses AviSynth, like this:


LoadPlugin("nicaudio.dll")
LoadPlugin("AudioLimiter.dll")
a=RaWavSource("file.w64", 1)
\.ConvertAudioToFloat()

flr = Getchannel(a, 1, 2, 3, 4)
blr = Getchannel(a, 5, 6)
slr = Getchannel(a, 7, 8)
sur = MixAudio(blr, slr, 1.0, 1.0).SoftClipperFromAudX(0.0)
return mergechannels(flr,sur)

[Teknon]

Hey guys,

sorry I'm new here since I've literally never had a problem worth asking about on a forum with eac3to or clownbd or whatever. This time I have to ask, though, since I don't know what to do.

I set up a new PC with a fresh Windows 10 installation last week and since then eac3to won't read blu-rays anymore. I still remux blu-rays with clownbd but after the 1st pass it doesn't show me any playlists to choose from (see attached image).

When I try to read out the blu-rays within eac3to directly and not with clownbd it just says "blu-ray/hddvd structure not found", that's why I figured it's an issue with eac3to and not clownbd.
I also tried out Blu-ray ripper, an alternative to clownbd, but it doesn't work either.

I've already reinstalled everything several times, tried out different versions for the programs I use but nothing has worked so far. Anyone got any tips?

Thanks and best regards,
Teknon

[LigH]

You will have had a resident ripping software installed on your previous PC, because eac3to never was a ripper, it is not able to decrypt. Usually you are expected to copy the decrypted content to your harddisk before using eac3to.

[tebasuna51]

I still remux blu-rays with clownbd but after the 1st pass it doesn't show me any playlists to choose from (see attached image).

I delete your empty image because don't help.

LigH is right, you need a BD decripter to work after with eac3to.

You can also try with MakeMKV (https://www.makemkv.com/)

[Teknon]

Sorry, but I don't really understand (not an expert on this stuff). On my previous PC I remuxed blu-rays/blu-ray isos with clownbd using eac3to and tsmuxer and at the end it gave out a blu-ray folder structure with only one m2ts file (the one that was the movie).

I do have all of these programs installed, they even come with clownbd itself. Although, I did install them separately just to try if that works but no matter what I do, eac3to doesn't find the blu-ray structure, neither within clownbd nor by itself. Which is why I'm thinking that it's an issue with eac3to, not clownbd or tsmuxer.

[Flanker-B]

why is eac3to ignoring e-ac3 part of a e-ac3 7.1 file and converting only ac3 5.1 core part? I tried to convert a e-ac3 french track from a french movie "Le chant du loup" aka "The Wolf's Call", and it always ignores eac3 part and only converts ac3 core!

[SeeMoreDigital]

why is eac3to ignoring e-ac3 part of a e-ac3 7.1 file and converting only ac3 5.1 core part? I tried to convert a e-ac3 french track from a french movie "Le chant du loup" aka "The Wolf's Call", and it always ignores eac3 part and only converts ac3 core!What audio format are you converting to?

[tebasuna51]

@Teknon
From Clown_BD v0.81 info:
You need to have the UDF2.5 driver installed running under XP, and of course AnyDVD-HD (https://www.videohelp.com/software/AnyDVD) when dealing directly from the disks

AnyDVD-HD, to decript BD disks, is not free, for that I recommend MakeMKV free (while beta) to decript BD disks.

[Teknon]

@tebasuna51
Thanks for your response. MakeMKV seems to work for now, at least it reads my blu-ray ISOs and I can make an MKV from it

Is there no way to get clownbd to run like it used to on my old pc so I can get a blu-ray structure at the end? I never needed to install AnyDVD-HD or anything other than eac3to, tsMuxer and clownbd (at least to my knowledge) for it to properly read and remux blu-ray ISOs, it just seems really weird to me that it doesn't work anymore, and I liked having demuxing and remuxing into a blu-ray structure all in one program.

Thanks for your help!

[videoh]

MakeMKV can rip to a bluray structure.

[sneaker_ger]

why is eac3to ignoring e-ac3 part of a e-ac3 7.1 file and converting only ac3 5.1 core part? I tried to convert a e-ac3 french track from a french movie "Le chant du loup" aka "The Wolf's Call", and it always ignores eac3 part and only converts ac3 core!
eac3to's libraries are very outdated. You should be able to fully decode 7.1 e-ac3 using ffmpeg.

[tormento]

You should be able to fully decode 7.1 e-ac3 using ffmpeg.
Approved. (cit Davie504 :p)

I switched to ffmpeg months ago and I never had problems.

[Megalith]

Anyone running into issues with recent UHD releases, like Midway and Knives Out?

[foobar2000]

how to select right channel while doing -mono? thanks

[DMD]

Good morning
I wonder if there is any possibility of demuxing the audio stream from Eac3 (8ch) to Waws (8ch)
Currently I always get error.
Thank you

https://i.postimg.cc/jqB3p4NC/Screenshot-003.png

https://i.postimg.cc/3wZZWgdj/Screenshot-002.png

[tebasuna51]

@DMD

eac3to have some limits decoding eac3 streams, but if you have UsEac3to and a recent ffmpeg.exe version (https://ffmpeg.zeranoe.com/builds/) in the same folder you can use the 'A/V Recode' button.

1) Load the same file and 'A/V Recode'

2) Select track (first track for ffmpeg is '0' video, then '1' for first audio track)

3) Select decode to w64 (we expect a big file)

4) 'Run' (or 'Enqueue')

5) Load the .w64 output and now 'Output' to .wavs

[tebasuna51]

how to select right channel while doing -mono? thanks

You can't do it, with stereo input the output is the left channel (with multichannel the output is the center channel).

If you need only the right channel use the output to .wavs and discard the left channel.

[foobar2000]

You can't do it, with stereo input the output is the left channel (with multichannel the output is the center channel).

If you need only the right channel use the output to .wavs and discard the left channel.

thanks, i never noticed .wav and .wavs.......

[chompy]

1. "normal" (not "strange setup") DTS-HD MA 7.1 has the following channel layout:

http://thumbnails31.imagebam.com/9780/43bb3497794528.jpg (http://www.imagebam.com/image/43bb3497794528)

which in the way how Microsoft (and thus eac3to) named the channels is the same as:

http://thumbnails30.imagebam.com/9780/160e7497795034.jpg (http://www.imagebam.com/image/160e7497795034)

or DTS channel names mapped to Microsoft channel names are as follows:

DTS channel name <---> Microsoft channel name

L <---> L
R <---> R
C <---> C
LFE <---> LFE
Lsr <---> BL
Rsr <---> BR
Lss <---> SL
Rss <---> SR


Hello,

Is this DTS <---> Microsoft channel map correct?

I thought that if demuxing a DTS-HD 7.1 track and then encode it again in DTS-HD Master Suite I should use SL for LSS and SR for RSS (surround channels) and BL for LSR and BR for RSR (back channels). And similary with a DTS-HD 7.1 strange setup I thought SL is LS, SR is RS, BL is LSR and BR is RSR.

According to the Microsoft channel layout image it should be as I've said, but then in the text table DTS channel name <---> Microsoft channel name BL and BR seem to be exchanged with SL and SR.

Thanks

[tebasuna51]

...I should use SL for LSS and SR for RSS (surround channels) and BL for LSR and BR for RSR (back channels).
OK.

And similary with a DTS-HD 7.1 strange setup I thought SL is LS, SR is RS, BL is LSR and BR is RSR.
By your text seems you talk about Other Setup (see image 2), then is OK, but I never see a DTS like this.
The Strange Setup, than I see more often (see the image 1), table maybe can be:

Left <---> ? FrontLeft of Center
Right <---> ? FrontRight of Center
Center <---> FrontCenter
LFE <---> LFE
Leftwide <---> ? FrontLeft
Rightwide <---> ? FrontRight
Leftsusrround <---> don't exist, in 5.1 can be BackLeft (old) or SideLeft (now)
Rightsurround <---> don't exist, in 5.1 can be BackRight (old) or SideRight (now)

But like nobody have the speakers in the correct position, the table used is like the standard 7.1:

Left <---> FrontLeft
Right <---> FrontRight
Center <---> FrontCenter
LFE <---> LFE
Leftwide <---> SideLeft
Rightwide <---> SideRight
Leftsusrround <---> BackLeft
Rightsurround <---> BackRight

... in the text table DTS channel name <---> Microsoft channel name BL and BR seem to be exchanged with SL and SR.
The text table for Standard Setup (see image 3) is OK for me:

Leftsurroundrear <---> BackLeft
Leftsurroundside <---> SideLeft

[chompy]

The Strange Setup, than I see more often (see the image 1), table maybe can be:

Left <---> ? FrontLeft of Center
Right <---> ? FrontRight of Center
Center <---> FrontCenter
LFE <---> LFE
Leftwide <---> ? FrontLeft
Rightwide <---> ? FrontRight
Leftsusrround <---> don't exist, in 5.1 can be BackLeft (old) or SideLeft (now)
Rightsurround <---> don't exist, in 5.1 can be BackRight (old) or SideRight (now)


Thanks, I thought that the strange setup was this:

https://images2.imagebam.com/cf/f4/a0/e1ab321338032211.png

[tebasuna51]

Thanks, I thought that the strange setup was this:

I never see a DTS like this (in commercial movies of course, not created by you), you can upload a sample?

[chompy]

I never see a DTS like this (in commercial movies of course, not created by you), you can upload a sample?

In fact I don't know how to test if the strange setup is your image 1 or your image 2, but as in the DTS-HD Encoder Suite version I have apart from the standard setup, the only other 7.1 config available is the one shown in your image 2, I thought that was the strange setup.

Here you have a sample of the original track I'm using: https://www.upload.ee/files/11326828/sample.dtshd.html

EDIT: I've seen that there's an option to show all 7.1 layouts and then the one in your image 1 (among others) appear... Is then an easy way to know the config used in the original DTS-HD 7.1 track to use it again in the new edited track?

[DMD]

@DMD

eac3to have some limits decoding eac3 streams, but if you have UsEac3to and a recent ffmpeg.exe version (https://ffmpeg.zeranoe.com/builds/) in the same folder you can use the 'A/V Recode' button.

1) Load the same file and 'A/V Recode'

2) Select track (first track for ffmpeg is '0' video, then '1' for first audio track)

3) Select decode to w64 (we expect a big file)

4) 'Run' (or 'Enqueue')

5) Load the .w64 output and now 'Output' to .wavs

Thank you very much for the information.
the file "ffmpeg.exe" which download Shared or Static?
Thank you

[tebasuna51]

the file "ffmpeg.exe" which download Shared or Static?

The Static, only one file without need *.dll's

[tebasuna51]

...Is then an easy way to know the config used in the original DTS-HD 7.1 track to use it again in the new edited track?

Of course, with your sample (image 2) :
eac3to.exe sample.dtshd -logdts

+ DTS-Core
- frameSize 2012
- DTS-ES +
- channelNo 5
- lfe 1
- channelDescr 5.1
- samplingRate 48000
- bitDepth 24
- bitrate 1509000
- samplesPerFrame 512
- copyHistory 1
+ DTS-HD
- fullSize 116
- headerSize 32
- refClockCode 1/48000
- frameDurationCode 1
- activeMasks [1], [[1]]
+ Asset [0]
- fullSize 84
- headerSize 14
- corePackets Core
- extSubStrPackets XLL
- bitResolution 24
- maxSampleRate 48000
- totalNumChannels 8
- activeSpeakers C L R Ls Rs LFE Lsr Rsr ($4f)
DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48kHz
(core: DTS-ES, 5.1 channels, 1509kbps, 48kHz)

Other samples (image 1 and 3):
- activeSpeakers C L R Ls Rs LFE Lw Rw ($40f)
- activeSpeakers C L R LFE Lsr Rsr Lss Rss ($84b)

[chompy]

Of course, with your sample (image 2) :
eac3to.exe sample.dtshd -logdts

Great, lots of thanks again!!

[DMD]

The Static, only one file without need *.dll's
Thank you

[DMD]

Good morning
I apologize for my question, as far as the EAC3 Encoder is concerned, bitrate values higher than 640 Kb / s are not expected?
Thank you

https://i.postimg.cc/y6bY7tBK/Screenshot-001.png

[filler56789]

Good morning
I apologize for my question, as far as the EAC3 Encoder is concerned, bitrate values higher than 640 Kb / s are not expected?


Surely that's a limitation (design flaw) in the GUI
(btw, could you please name it?).

Besides, just like .dca (DTS "coherent acoustics"), the .eac3 compression has no predefined set of "allowed bitrates".

[DMD]

Surely that's a limitation (design flaw) in the GUI
(btw, could you please name it?).

Besides, just like .dca (DTS "coherent acoustics"), the .eac3 compression has no predefined set of "allowed bitrates".

I tried to force the settings, manually entering the value of 1509 kb/s.
The procedure started regularly with the set bit-rate value.
The only limitation that the maximum number of channels that can be managed is 6, the initial test track is 8ch (7.1).
I don't know how it is possible to manage 8 ch tracks.

https://i.postimg.cc/FFJ0Ggw2/Senza-titulo-1.png

https://i.postimg.cc/ZKSjdM0f/Senza-titulo-2.png

[SeeMoreDigital]

I had a go this morning trying to generate an 8 channel EAC3 encode using an 8 channel PCM source. I also obtained a 6 channel EAC3 elementary stream.

[SeeMoreDigital]

As a bit of an addition...

I also created a 6 channel EAC3 encode at 1536 Kbps using one of my DTS audio CD's as a source. And I have to say that the playback results are very good indeed.

[tebasuna51]

Surely that's a limitation (design flaw) in the GUI
(btw, could you please name it?).

The GUI is UsEac3to, and yes is a GUI limit, I can't understand for what we can need more than 640 Kb/s for lossy 5.1

[DMD]

If it were possible to manage 8 channels, it would be fantastic.

[LigH]

The reason why ffmpeg won't encode E-AC3 with >6 channels (https://trac.ffmpeg.org/ticket/3595#comment:13) is not really obvious to me. And I am obviously wrong with my guessed reason... :o

[nevcairiel]

The reason why ffmpeg won't encode E-AC3 with >6 channels (https://trac.ffmpeg.org/ticket/3595#comment:13) is not really obvious to me. And I am obviously wrong with my guessed reason... :o

Noone has spent time making it, and noone seems interested in doing so unless someone is willing to pay for it?

Thats as logical a reason as for anything. People usually work on one of two things: Stuff they have a personal interest for, or stuff other people pay them for.

[tebasuna51]

In my opinion 7.1 don't offer a better 2D surround than 5.1, I think is a commercial placebo, then for me is ok the ffmpeg behaviour.

Other question is Atmos 3D surround.

[justnobody]

ffmpeg will not support eac3 7.1 encode, just forget it and go


my question: is it possible to cut the end of the file using eac3to? like cut 10s, we can cut the beginning with simple -10000ms but..

anyway for cutting like this ffmpeg is pretty handy, eac3to doesn't come with this function i guess

[r0lZ]

is it possible to cut the end of the file using eac3to?
Not sure for eac3to, but I use mkvtoolnix (https://mkvtoolnix.download/) to losslessly cut any audio file. Just mux it to MKA, and use the Split After specific timestamps feature (in the Output tab) to split it where you want. Then, you can extract the part you need from the MKA segment with gMKVExtractGUI (https://sourceforge.net/projects/gmkvextractgui/).

[stax76]

Not sure for eac3to, but I use mkvtoolnix to losslessly cut any audio file.

Except the FLAC format. :)

[sneaker_ger]

my question: is it possible to cut the end of the file using eac3to? like cut 10s, we can cut the beginning with simple -10000ms but..
I think only if you specify the length with -edit. E.g. if the file is 1h30m long and you want to cut the last 10 seconds:
eac3to "input.eac3" "output.eac3" "-edit=1:29:50,-10000ms"
(Or maybe make the "-10000" to e.g. "-11000ms" to have some reserve for rounding/alignments.)

[tebasuna51]

If you know the movie length, for instance 1:50:40.000 you can cut the 10 last seconds with:

-edit=1:50:30.000,-10000ms

EDIT: to slow to make the test

[richardpl]

ffmpeg will not support eac3 7.1 encode, just forget it and go

It will, just pay.

[filler56789]

It will, just pay.

People will pay IF:

1) the price of the new encoder is lower than the price of the "official"/certified encoders

&&

2) the price of the new encoder is declared publicly;
I mean, no "send me an email and I'll tell you how much it will cost",
alright :)

[videoh]

Except the FLAC format. :) FLAC is easy to cut with sox.exe. Use the trim effect.

[orion44]

Can "-ac 2" switch in ffmpeg be used to downmix 7.1 DTS audio to stereo WAV?

Or is that switch just used for 5.1 AC3 to stereo downmixing?

[Sparktank]

Can "-ac 2" switch in ffmpeg be used to downmix 7.1 DTS audio to stereo WAV?

Yes, but I don't know which switch it is to normalize if clipping goes beyond 0dBFS.

I'd use eac3to for that:
eac3to source.dts stereo.wav -downStereo -normalize

It should normalize if clipping beyond 0dBFS is detected, by default. So you don't really need to add that switch to eac3to.
Recommended to use stereo if you have plain stereo playback environment (tv,etc).
-downProLogic if you have a surround sound environment that can decode Dolby Pro Logic II.

At least with eac3to, you get a log that gives you useful information.
Regarding updates, FFMPEG and eac3to should be be equal in updates. Not a whole lot has happened with DTS regarding last updates.

[Bigmango]

The GUI is UsEac3to, and yes is a GUI limit, I can't understand for what we can need more than 640 Kb/s for lossy 5.1

For all the devices that can't play TrueHD or DTSMA. (i.ex. people streaming from their plex or emby server to their TVs. Even more so, look at the high end 10'000+$ Samsung TVs, they can't even play DTS, but they do play ac3 and eac3).

In the above example, streaming to the TV, with the audio out via HDMI-ARC to a receiver with the TrueHD/DTSMA re-ecoded to eac3 1664k (which is the highest bitrate for the eac3 bluray spec according to the Dolby Media Encoder) makes it as transparent as the lossless audio (depending on your ears you might need less, but better be safe, especially with audiophile headphones, etc... :) ).

That being said, the new E-ARC now passes TrueHD and DTSMA through, so the eac3 encoding will become less relevant in the near future.

[Bigmango]

In my opinion 7.1 don't offer a better 2D surround than 5.1, I think is a commercial placebo, then for me is ok the ffmpeg behaviour.


Using a 7.1 source, the 5.1 at 1536k eac3 produced by ffmpeg is poor compared to the 7.1 produced by the Dolby Media Encoder. (I have tested with both).

There is a big soundstage difference.

The difference is night and day with some good audiophile reference headphones with a big soundstage (take a Sennheiser HD 650 for example...).

[Bigmango]

ffmpeg will not support eac3 7.1 encode, just forget it and go


According to the ffmpeg tracker, the developper who implemented the 5.1 eac3 said that he will work on adding the 7.1, but that was a couple years ago and nothing has happened since...

[Bigmango]

I also created a 6 channel EAC3 encode at 1536 Kbps using one of my DTS audio CD's as a source. And I have to say that the playback results are very good indeed.

FYI, these are the maximum bitrates eac3 can go up to with the Dolby Media Encoder:

DD+ for bluray: 1664k. (this works fine over HDMI-ARC).
DD+ standard: 3024k (I have not tested this over ARC, but according to the information I could find online, ARC supports ~1Mbs (EARC 38Mbs); so this would be the reason for the DD+ bluray spec.).

[SeeMoreDigital]

Using a 7.1 source, the 5.1 at 1536k eac3 produced by ffmpeg is poor compared to the 7.1 produced by the Dolby Media Encoder. (I have tested with both).

There is a big soundstage difference.

The difference is night and day with some good audiophile reference headphones with a big soundstage (take a Sennheiser HD 650 for example...).Can you upload both of your test sample encodes for us please?

Have you compared both encoders using the same 5.1 PCM source encoded at the same bit-rate?

[Sonic3R]

Hey, there’s a source code for eac3to? I wish, if possible, to adapt it for cross platform means to be used in Linux/Mac too...

[jpsdr]

As far as i know, the source code is not avaible, and it's in delphi.

[tebasuna51]

@Bigmango
I accept, but not share your opinions (headphones for 7.1, better than 5.1?).
Of course commercial encoders can be better than free ones.

BTW this is a eac3to thread than can't encode to eac3 byself.
The eac3 bitrate limitation is in the UsEac3to GUI than use ffmpeg to encode.
That GUI can be modified, I supply the autoit source code, and recompiled for free (https://www.autoitscript.com/site/)

You only need change the line 1895:
GUICtrlSetData($bit_2, "640|576|512|448|384|320|256|224|192|160|128|112|E640|E576|E512|E448|E384|E320|E256|E224|E192|E160|E128|E112|E96|E80|E64", "640")

And include the bitrates desired:
GUICtrlSetData($bit_2, "640|576|512|448|384|320|256|224|192|160|128|112|E6144|...|E768|E640|E576|E512|E448|E384|E320|E256|E224|E192|E160|E128|E112|E96|E80|E64", "640")

[tebasuna51]

As far as i know, the source code is not avaible, and it's in delphi.

You are right, madshi never want public the source code.

[Bigmango]

Can you upload both of your test sample encodes for us please?

I don't have the time now for some demuxing/cutting, but I'll see if I can get this for you a little later.

In the mean time, I will give you all the info I can below:

Have you compared both encoders using the same 5.1 PCM source encoded at the same bit-rate?

Yes the difference is there, not only with 7.1, but also with 5.1 sources.

In comparison, the ffmpeg DD+ sounds more "flat" / bland.

The Dolby Media Encoder audio has more "presence". On some good reference headphones like the Sennheiser HD 650 which have an amazing soundstage, the surround is perfectly positioned and moves around in an amazing way.

With the Dolby Encoder audio, the movies just "move" more. Immersion is much better. The result is that the movies are better (I often feel like I am rediscovering my movies after replacing the 5.1 ffmpeg DD+ @ 1536k with the Dolby Encoder version @ 1664k).

The difference is not huge, but it is there. It's sometimes subtle (but even subtle when compared on a short clip, if you watch a movie for a longer time the experience/immersion is a lot better thanks to these improvements) and perhaps to understand the subtlety level I am talking about, you could compare it to something like the difference between a 384k ac3 and lossless (or DTS 1509). (But here I am not talking about audio quality, but "presence". More Flat vs more Alive).

I suspect this better/stronger presence of the surround effects to be caused by the way the Dolby Encoder is applying the metadata, and using an AV receiver that knows how to decode this metadata.

These are the settings the Dolby Media Encoder uses by default (which I use):

Encoder settings:

Dialog level: -31
Data rate : 1280k for 7.1 , 768k for 5.1 (I set it @ 1664k for both)
Bitstream mode: "Complete main" (Other options are: "Music & Effects", "Visually Impaired", "Hearing Impaired", "Dialog", "Commentary", "Emergency", "Voice over").

Preprocessing:

Line mode profile: Film standard (Other options are : "Film light", "Music standard", Music light", "Speech", "None").
RF mode profile: Film Standard (Other options: all the same as Line mode profile)

Channel bandwidth low pass filter: ON
LFE low pass filter: ON
DC high pass filter: ON
Surround channel 90-Degree phase shift: ON
Surround channel 3db attenuation: OFF

(All of these settings and what they do is explained in the Dolby Media Encoder PDFs, which you can easily find with a google search).

As you can see, my guess is that these settings (which are the defaults) is where the magic is happening.

[Bigmango]

Hey, there’s a source code for eac3to? I wish, if possible, to adapt it for cross platform means to be used in Linux/Mac too...

It works perfectly on Linux with wine (I use "wine UsEac3to.exe" all the time, works perfect).

But yes, a native port would be great.

[kdantas]

It works perfectly on Linux with wine (I use "wine UsEac3to.exe" all the time, works perfect).

But yes, a native port would be great.

Hi @Bigmango, nice to know. Can you indicate a tutorial explaining, if possible, how to install and run eac3to with UsEac3to on Ubuntu Server 16.04 Xenian Xerus? I would like to remux my BluRays from my dedicated server. I also know that I will need to run mkvtoolnix to join the demuxed files. Any information is welcome. And sorry if this is a noob question, but my knowledge of linux is almost zero.

[cypher007]

trying to convert a THD stream. ive demuxed the mkv using eac3to then tried to extract the wavs but it complains about Lossless check failed and eventually errors with:

The libav decoder reported error -1094995529 while decoding. <ERROR>
Aborted at file position 1268252672. <ERROR>

[tebasuna51]

Try decode to w64 with ffmpeg, if it also abort maybe you have a corrupt track.

[cypher007]

Try decode to w64 with ffmpeg, if it also abort maybe you have a corrupt track.

same error. but if i use ripbot264 it will create a W64.

i did wonder if i could use this to then extract wavs from. when i tried it worked but when i checked the wavs they were 16bit not 24bit as i expected.

[tebasuna51]

For what do you expect 24 bits, what is the source?

[cypher007]

its a truehd stream so assumed 24bit. though my knowledge of bit depth in truehd is limited.

why is it ripbot can demux the truehd to w64? and not eac3to.

[tebasuna51]

Many TrueHD are only 16 bitdepth.

I think ripbot uses AviSynth decoders, maybe are more permissives for errors and skip some bad frames, check if the length is correct.

[cypher007]

Many TrueHD are only 16 bitdepth.

I think ripbot uses AviSynth decoders, maybe are more permissives for errors and skip some bad frames, check if the length is correct.

will do. thanks very much for the help.

[StormMeows]

Hi guys, hopefully someone can help. I'm trying to do something that I thought would be relatively easy but cannot figure it out.

Here is my goal first of all: I am trying to figure out what subtitles are "forced" and also which title is the main title of the movie on blu-ray files, folders, and ISOs so that I can use MKVtoolnix to convert the movie to .MKV and then I can check the box for "default track" when there is a forced track. A movie example would be something like "Avatar" where there is foreign language spoken but only in certain parts of the movies. If I use something like MakeMKV and do the backup blu-ray disc to .MKV then I can select all of the sub title tracks and save it, but when I play the file in something like Kodi or Plex, then I will have to manually select a subtitle track and know which movies have forced subtitles. There are movies like 10 Things I Hate About You that have a French forced subtitle track that I would have no clue it existed unless I knew the movie.

I was advised that I could use eac3to and figure out which title on the disc is the "main title" and if it has a forced subtitle track within 30 seconds or so. Then I could use something like BDSupforsub and get the correct subtitle, then remux the movie using the MKVtoolnix.

So here's my problem. I downloaded the eac3to and have it in a folder on my desktop (it has the .exe file in here and some .dll files) then I went into the System C drive to locate CMD.exe . I copy and pasted the CMD.exe into the folder for eac3to.

Then I am trying to type in this: C:\Users\robma\Desktop\Remux\eac3to>D:\Pulp Fiction.1994\ and I get "DNS server not authoritative for zone."

What am I doing wrong? I just want to figure out which movies have forced subtitles so I can select the subtitle track in MKVtoolnix as the "default" track to blu-rays and 4K .MKV files so when I play them in Plex and Kodi I won't have to worry about finding out which sub track is the correct one, etc.

Thanks for any help from a newb.

[tebasuna51]

I copy and pasted the CMD.exe into the folder for eac3to.
For what? It's not needed.

Then I am trying to type in this: C:\Users\robma\Desktop\Remux\eac3to>D:\Pulp Fiction.1994\

For what the >?

C:\Users\robma\Desktop\Remux\eac3to "D:\Pulp Fiction.1994"

[Edited because r0lZ next post]

[r0lZ]

For what the >?

C:\Users\robma\Desktop\Remux\eac3to D:\Pulp Fiction.1994\
Or:
C:\Users\robma\Desktop\Remux\eac3to "D:\Pulp Fiction.1994"
The quotes are necessary due to the space between pulp and fiction.

[tebasuna51]

Yep, I don't see the space.

[StormMeows]

For what? It's not needed.



For what the >?

C:\Users\robma\Desktop\Remux\eac3to "D:\Pulp Fiction.1994"

[Edited because r0lZ next post]

Or:
C:\Users\robma\Desktop\Remux\eac3to "D:\Pulp Fiction.1994"
The quotes are necessary due to the space between pulp and fiction.

Hey guys I have tried both of these and it just gives me this error: Desktop\Remux\eac3to is not recognized as an internal or external command, operable program or batch file. Any thoughts? Thanks for the help!

For what? It's not needed.



For what the >?

C:\Users\robma\Desktop\Remux\eac3to "D:\Pulp Fiction.1994"

[Edited because r0lZ next post]

I put them in the same folder so I do not have to type out the first part every time. I can just start with the "D:" portion, but as mentioned in my last post, I cannot get the eac3to to even work. I really want to figure out what blu-rays have forced subtitles and this is the only way I know that I can do this. Hope you can help. thanks

[StormMeows]

edited my other post.

[r0lZ]

Hey guys I have tried both of these and it just gives me this error: Desktop\Remux\eac3to is not recognized as an internal or external command, operable program or batch file. Any thoughts? Thanks for the help!
Desktop\Remux\eac3to is not the complete path. You have to type:
C:\Users\robma\Desktop\Remux\eac3to
Or, easier, just open the command prompt window and the folder containing eac3to. Then, drag eac3to.exe to the command prompt window. The full path of the command will be inserted automatically. Then, all you have to do is to type the path of the folder to process, like "D:\Pulp Fiction.1994". You can also type just the beginning of the path (like D:\p) and then use the TAB key to complete automatically the path name.

cmd.exe must NOT be copied, and it is not necessary to type its full path. It's a system command, available from everywhere. If you really want to open it from a specific folder, right-click the background of the folder and select New -> Shortcut. In the dialog, type cmd and hit OK twice. Now, right-click the new shortcut and select Properties. Remove whatever path is in the "Start in:" field. Hit OK. Now, when you will double-click that shortcut, the command prompt window will be opened, with the current directory already defined.

[dade49]

Any chance Madshi would be willing to look at the following to properly fix seamless branching of MLP/Atmos tracks? Been using eac3to forever, but there does appear to be issues with TrueHD seamless branching. Thanks.

https://github.com/domyd/mlp

[tormento]

Is this project dead?

I see FFMPEG doing leaps and circles around eac3to. What a pity.

[richardpl]

Is eac3to open source at all?

[tebasuna51]

Nope, madshi don't free the source code.

[tormento]

Nope, madshi don't free the source code.
Isn't possible to write a patch or a pipeline to allow ffmpeg shared dll use instead of almost ancient eac3to included ones?

I mean: they are from March 2015.

[tebasuna51]

The avcodec/avutil from ffmpg aren't compatibles with eac3to (https://forum.doom9.org/showthread.php?p=1078513#post1078513) in 2007.

I sugest to madshi make them compatibles (https://forum.doom9.org/showthread.php?p=1781207#post1781207) in 2016 but without answer

[tormento]

The avcodec/avutil from ffmpg aren't compatibles with eac3to (https://forum.doom9.org/showthread.php?p=1078513#post1078513) in 2007.
That's the purpose of my message.

With all the brilliant code writers here, could be possible to compile some "fake" old ffmpeg dlls to redirect to the new ones?

[LigH]

Probably not that easy, because application interfaces only change when there is an important reason. Usually because the previous interface lacked of a feature a new format needs. If someone would try to wrap it into an older interface, such newer features would be missing, and that may break using several formats completely.

[Sonic3R]

It works perfectly on Linux with wine (I use "wine UsEac3to.exe" all the time, works perfect).

But yes, a native port would be great.

Wine won't work on Ubuntu Server until you install desktop version.

Very bad that madshi don't want to share code, a lot of people want to improve it ... it's his choice

[richardpl]

I really wonder why such software is used at all.

[LigH]

"Such software" as eac3to? ... Probably because of some unique features no other software provided for a long while. Are there now equal alternatives? Great. That was not always the case.

[kdantas]

Hi, is there any way (chenge dll, edit registry, etc) to correct the version information of Haali Matrosk Muxer? I already installed the latest version found on the site... MatroskaSplitter23062013, as well as, I installed the latest version of the K-Lite Mega Codec Pack ... v15.5.7. I know it's just a date error, but I would like to see the correct information. Thanks.

Haali Matroska Muxer (2013-04-14) is installed
There's a new version (2013-06-23) available
http://haali.net/mkv

[kdantas]

Hi, is there any way to get a progress bar for eac3to running on the Windows command line, like this one below?

https://i.imgur.com/1hrIKcc.gif

[LigH]

Only if madshi would code that. And the console output of eac3to is already a little weird (if you redirect the help output into a text file, its formatting is most peculiar). In case of output redirection, one may not wish the output to contain a frequently updated progress message.

[tormento]

Trying to extract eac3 track from Solo – A Star Wars story. The mpls combines different m2ts as there are 3 different language scrolling screen at the beginning.

eac3to v3.34
command line: "D:\Eseguibili_x86\Media\eac3to\eac3to.exe" "F:\Raw\Solo — A Star wars story" 2) 4:"E:\in\2_14 Solo — A Star wars story\2_4_audio.flac" -progressnumbers
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 8 subtitle tracks, 2:14:47, 24p /1.001
1: Chapters, 50 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
4: E-AC3, Italian, 7.1 channels, 896kbps, 48kHz, dialnorm: -27dB
(core: AC3, 5.1 channels, 512kbps, 48kHz, dialnorm: -27dB)
5: DTS, Spanish, 5.1 channels, 1509kbps, 48kHz
6: Subtitle (PGS), English
7: Subtitle (PGS), Italian
8: Subtitle (PGS), Spanish
9: Subtitle (PGS), Portuguese
10: Subtitle (PGS), Modern Greek
11: Subtitle (PGS), English
12: Subtitle (PGS), Italian
13: Subtitle (PGS), Spanish
[a04] Extracting audio track number 4...
[a04] Removing AC3 dialog normalization...
[a04] Extracting E-AC3 core...
[a04] Decoding with libav/ffmpeg...
[a04] The libav decoder reported error -22 while decoding. <ERROR>
Aborted at file position 1048576. <ERROR>

Never tried to extract tracks from mpls with ffmpeg. Need some help about command line. :o

[Boulder]

You are not demuxing but decoding and encoding. To get around the issue, you could try demuxing first to ac3 and then transcode that to FLAC with a recent ffmpeg build.

[tebasuna51]

Is know than eac3to can't decode eac3 tracks like this.
You can extract it with:

eac3to "D:\Eseguibili_x86\Media\eac3to\eac3to.exe" "F:\Raw\Solo — A Star wars story" 2) 4:"E:\in\2_14 Solo — A Star wars story\2_4_audio.eac3" -keepDialnorm

(when eac3to try to change Dialog Normalization crash)

After that my recomendation is preserve the track as is, recode to flac doesn't make sense for me.
A lossy track when is decoded the output is 32 float (even 64 float) and make a flac with this...

[Thunderbolt8]

has anyone tried if this https://github.com/domyd/mlp

is it more accurate than demuxing THD tracks with eac3to (or any other tool)? could this tool be incorporated here as well like other components?

[tebasuna51]

Here?

Seems madshi don't want make new versions.

I will try to include it to the UsEac3to GUI.

I can't get binary with stream selection, maybe in next release.

[planet1]

Seems madshi don't want make new versions.

Ohh, so there is not much of a point in posting any feature requests here or at http://eac3to.bugs.madshi.net then?

[Thunderbolt8]

I hope there still is and that madshi is currently just prioritizing madvr

[stax76]

madvr is closed source and under-maintained as well I believe, meaning hardly progress for both in the last 2-3 years, or maybe I'm just not well-informed.

[tebasuna51]

has anyone tried if this https://github.com/domyd/mlp

is it more accurate than demuxing THD tracks with eac3to (or any other tool)? could this tool be incorporated here as well like other components?

Included the use of the version than can select the TrueHD stream (https://github.com/domyd/mlp/actions/runs/120022192) in UsEac3To (https://forum.doom9.org/showthread.php?p=1920050#post1920050)

[pottzman]

Hi, is there any way (chenge dll, edit registry, etc) to correct the version information of Haali Matrosk Muxer? I already installed the latest version found on the site... MatroskaSplitter23062013, as well as, I installed the latest version of the K-Lite Mega Codec Pack ... v15.5.7. I know it's just a date error, but I would like to see the correct information. Thanks.

Haali Matroska Muxer (2013-04-14) is installed
There's a new version (2013-06-23) available
http://haali.net/mkv

Yes, download a program called "attribute changer" https://www.petges.lu/download/ and use it on "splitter.ax" and "splitter64.ax" in the haali media splitter directory to change the creation and modified dates to 2013-06-23. eac3to will not complain anymore.

[Skyrocket]

Can anyone please help me. Been trying for a few days to learn how to fix it with no luck.
I am wanting to convert my audio with eac3to to aac. When trying it comes up with the Nero error. I have Nero 7 installed on my computer but dont think its linked with the eac3to. What can or should i do to convert to aac?

[tebasuna51]

Nero 7 is used to decode AAC, to encode to AAC you need NeroAacEnc.exe from here (https://www.videohelp.com/software/Nero-AAC-Codec) in the same folder than eac3to

[Csimbi]

Hi there,
is there any plan to add Atmos to wav conversion?
(with metadata support, not just TrueHD conversion)
Thanks!

[SeeMoreDigital]

Hi there,
is there any plan to add Atmos to wav conversion?
(with metadata support, not just TrueHD conversion)
Thanks!Are you're hoping to encode the 'spatially-coded sub-stream meta-data' for the height/object channels into separate mono PCM streams?

[stax76]

I would like to request Windows 10 long path support if this app isn't dead. It works via prefix workaround so probably only misses the manifest info.

https://github.com/staxrip/staxrip/wiki/Windows-10-long-path-support

[pottzman]

Let me know if you get it to work.
I am using arcsoft, nero, sonic and surcode with eac3to, all without installing them completely on my system.


If someone knows how to get the sonic decoder working qithout installing it could they post it here. I can get nero, arcsoft and surcode all working without installing them but not the sonic decoder

[tebasuna51]

The sonic or arcsoft decoders are not needed at all with eac3to v3.34 (and nero only for aac sources)