eac3to - audio conversion tool

[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
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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

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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.