eac3to - audio conversion tool

[Kurtnoise]

Anyway, using -log has a serious drawback. I have no way to distinguish the error message (if any) from the other messages. There is no leading string such as "ERROR:" to identify the line containing the error in the log.
This is not true...as soon as an error occurs, you have the <ERROR> string that appears.

[r0lZ]

OK, I think I have a working solution.
In fact, if you use the -log option, the log file is written to disc only at the end of the process (or when the process is aborted due to an error). But STDOUT still contains the normal messages (including the progress numbers when --progressnumbers is specified, as well as all that nasty BS and Spaces, that can be stripped out). Therefore, it is still possible to display the messages and update a progress bar while eac3to is working. Only the error messages are missing. But if the process returns a non-zero return code, you can then open the log file that has been written to disc, and search for the lines ending with "<ERROR>". They are the lines that are theoretically printed to stdout, but that are unavailable when STDOUT is redirected. It's a somewhat convoluted method, but it works.
Thanks again to everybody.

[EDIT] Kurtnoise has posted his message when I was writing mine and I was editing my previous message. Yes, <ERROR> is printed in the log at the end of the lines containing the error messages. I haven't noticed that when I did my first tests.

[stax76]

This and hoping for a fix is all we can do, I could read from a simple colored test app so it's possible.

[frenshprince]

Hello,

Coule you tell mi if its possible to use eac3to to convert a bluray playlist (mpls) with multiple m2ts to one m2ts ?

Of course I could use tsmuxer to do that, but I don't know why, it doen't work well' and the length of the m2ts is doubled.

Thanks

[Music Fan]

Coule you tell mi if its possible to use eac3to to convert a bluray playlist (mpls) with multiple m2ts to one m2ts ?
Previous page ;
http://forum.doom9.org/showthread.php?p=1704811#post1704811

[tebasuna51]

eac3to can extract the video to one mkv (or one video stream .h264/.vc1), can extract/recode the audio streams and extract subs.

But can't mux the streams to m2ts.

[frenshprince]

That's what I thought.

Thank you for your help.

[stinman]

Wil this program ever be able to decode TrueHD with Atmos or at least still decode TrueHD and disregard the atmos data like Lav Filters do? Or is there any way to get eac3to use the Lav Filters kind of like it will use the ArcSoft Decoders? I tried to put the Lav Filters in the folder with eac3to 3.27 and the HDDVD Bluray Stream Extracter to see if it would work. The only way to to extract a TrueHD atmos stream now is to run a MKV thru MakeMKV and out put the TrueHD atmos to flac, then the flac can be converted to wavs. I can't code like you guys can, if I could I would try and make eac3to use the new lav filters.:confused: ;)

[Music Fan]

That's what I thought.
You can use eac3to anyway to demux streams then use TSMuxer to remux in ts (or m2ts).

[tebasuna51]

The only way to to extract a TrueHD atmos stream now is to run a MKV thru MakeMKV...

You can use also ffmpeg, search in this thread. Or better in ffmpeg syntax (http://forum.doom9.org/showthread.php?p=1695623#post1695623) thread.

The problem was already reported to madshi.

[dade49]

The only way to to extract a TrueHD atmos stream now is to run a MKV thru MakeMKV and out put the TrueHD atmos to flac, then the flac can be converted to wavs.
This is much simpler. Extract the TrueHD Atmos stream to a multi-channel wav via ffmpeg, and then run that wav through eac3to to dts or whatever.
ffmpeg -i 00000.m2ts -map 0:1 extract.wav
eac3to extract.wav extract.dts

[tebasuna51]

Post about ffmpeg moved to ffmpeg syntax (http://forum.doom9.org/showthread.php?p=1707195#post1707195) thread.

Please don't use this thread for ffmpeg discussions.

[Arm3nian]

I'm experimenting with the dolby headphone wrapper plugin that's available with foobar, so I'm remapping channels with eac3to to see the effects. My source file is the song from the end of the first hobbit movie (7.1 channel dts-hd ma). I downmix to 5.1 and then analyze the channels with the foobar oscilloscope. The default output of eac3to is L,R,C,LFE,SL,SR, and is verified by the waveforms. The loudest is the center, followed by the left and right, then the two surrounds, and the LFE is near non existent, but is there.

http://imgur.com/DeIVxyd

I did the same thing but remapped the channels to L,R,SL,SR,LFE,C. As can be seen in the oscilloscope, the center is still the loudest (6th waveform), followed by the left and right (first and second waveforms), but the LFE somehow has more activity now and is louder (5th waveform). The SL and SR are closely related in activity and amplitude to each other, just like before remapping, but have lower activity and amplitude than the LFE... and also have lower amplitude than before the remap.

http://imgur.com/mTc9GgA

What's going on? Does eac3to apply different mix levels while disregarding which channel is what and going just based on sequential order, or is foobar doing something.

[tebasuna51]

...I did the same thing but remapped the channels to L,R,SL,SR,LFE,C.

Please put the log or at least the command line used to do this.

Does eac3to apply different mix levels while disregarding which channel is what and going just based on sequential order,

Of course, the channel order define which channel is what.

or is foobar doing something.

I don't know what you do with foobar.

[stax76]

Hi everybody,

I've improved StaxRip's eac3to support quite much lately, maybe a few people here have some interest to look at the latest release:

http://forum.doom9.org/showthread.php?p=1707791#post1707791

http://s17.postimg.org/vls7dqkaz/Unbenannt4.png (http://postimg.org/image/vls7dqkaz/)

http://s7.postimg.org/f62v5hs07/Unbenannt.jpg (http://postimg.org/image/f62v5hs07/)

[Arm3nian]

Please put the log or at least the command line used to do this.
Of course, the channel order define which channel is what.

14612
So if the channels are not in the default order when remapped then they get mixed at wrong levels?

[tebasuna51]

eac3to input71.dt output.wav -down6
finish with:
FL' = FL (0)
FR' = FR (1)
FC' = FC (2)
LF' = LF (3)
SL' = BL + SL (4 + 6)
SR' = BR + SR (5 + 7)

But with remap: eac3to input71.dt output.wav -down6 -0,1,4,5,3,2
finish with:
FL' = FL (0)
FR' = FR (1)
FC' = BL (4)
LF' = BR (5)
SL' = LF + SL (3 + 6)
SR' = FC + SR (2 + 7)

Seems first remap the first 6 channel and after mix the Side channels .

Maybe you need 2 pass, first to only -down6 and after remap with -0,1,4,5,3,2

[Arm3nian]

eac3to input71.dt output.wav -down6
finish with:
FL' = FL (0)
FR' = FR (1)
FC' = FC (2)
LF' = LF (3)
SL' = BL + SL (4 + 6)
SR' = BR + SR (5 + 7)

But with remap: eac3to input71.dt output.wav -down6 -0,1,4,5,3,2
finish with:
FL' = FL (0)
FR' = FR (1)
FC' = BL (4)
LF' = BR (5)
SL' = LF + SL (3 + 6)
SR' = FC + SR (2 + 7)

Seems first remap the first 6 channel and after mix the Side channels .

Maybe you need 2 pass, first to only -down6 and after remap with -0,1,4,5,3,2

Yeah two passes works as expected. Madshi can change it to were the order of the commands matter; as of right now it always remaps first. I personally don't care though, just experimenting.

[kevmitch]

This seems to be a semi-popular topic, but I don't think my specifc question has previously been answered.

With lord of the rings 6.1 dts-hd-ma audio


C:\> eac3to lotr.mkv
MKV, 1 video track, 5 audio tracks, 5 subtitle tracks, 2:07:41, 24p /1.001
1: h264/AVC, English, 1080p24 /1.001 (16:9)
2: DTS Master Audio, English, 6.1 channels, 24 bits, 48kHz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48kHz)
"Surround 6.1"
...


I decode it to flac (arcsoft 1.1.0.0)

C:\> eac3to lotr.mkv 1:lord_of_the_rings.flac


What order will the tracks be in the flac file for example if I open it in audacity?

I have verified that the flac is bit-for-bit identical (including channel order) to the makemkv decode results. This post (http://www.makemkv.com/forum2/viewtopic.php?f=4&t=5710#p23881) on the makemkv forum says that the channel order should therefore be

L R C LFE Ls Rs Cs

However, if I produce my own 7 channel flac file (with libsoundfile/audiolab), I find that downmixing it with eac3to


eac3to my_7_channel.flac my_7_channel.stereo.flac -downstereo


emperically uses the following matrix

[[ 1. 0. 0.7071 0. 0.7071 1. 0. ]
[ 0. 1. 0.7071 0. 0.7071 0. 1. ]]


which suggests that it is interpeting my file to have the order

L R C LFE Cs Ls Rs

Is this discrepancy due to differing channel mask between my file and that created by eac3to? Or is my first conclusion incorrect that eac3to places the centre surround channel last in the flac when decoding from dts-hd-ma?

[nevcairiel]

The "official" FLAC 6.1 order is L R C LFE Cb Ls Rs .. unfortunately 6.1 is annoying and both orders are commonly used.
See here: http://xiph.org/flac/format.html#frame_header (scroll down to <4> channel assignment)

7 channels: front left, front right, front center, LFE, back center, side left, side right

The MakeMKV post you referenced is old, since FLAC did add official 7 and 8 channel orders now.
However, it may be nice and write metadata into the FLAC headers to indicate the channel order, which eac3to should hopefully read.

[tebasuna51]

This post (http://www.makemkv.com/forum2/viewtopic.php?f=4&t=5710#p23881) on the makemkv forum says that the channel order should therefore be

L R C LFE Ls Rs Cs

And the info in the post is wrong.

I tested MakeMKV decoding to wav a DTS-MA 6.1 and output the standard 6.1 channel order (the same than eac3to output):

FL FR FC LF BC SL SR

where
BC = Back Center
SL = Side Left
SR = Side Rigth

Talking about channel order please use always the channel names and order defined here (http://www-mmsp.ece.mcgill.ca/documents/AudioFormats/WAVE/Docs/multichaudP.pdf) to avoid confusions.

The channel names and order inside a encoded format can be differents but, to transcode, we needs always decode to standard WAV order and channels, to allow the new encoder convert the audio to their internal order and channels.

Standard order for 7.1 is (like I put in my precedent post):

FL FR FC LF BL BR SL SR

[Thunderbolt8]

regarding 2.1 DTS-HD MA tracks: Does the arcsoft bug (LFE channel cannot be decoded) still exist in each arcsoft version? Or is there a version which is not affected by the bug?

when I play such a track with MPC-HC & LAV Audio with arcsoft capabilities, I wont hear the LFE channel then? what happens when I transform the 2.1 track with eac3to, using the arcsoft decoder, e.g. into wavs. will the LFE channel then be processed incorrectly? but when I just demux the track nothing happens?

[stax76]

M2TS, 1 video track, 2 audio tracks, 5 subtitle tracks, 24p /1.001
1: Chapters, 24 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, 24 bits, 1509kbps, 48kHz)
4: AC3 Surround, English, 2.0 channels, 320kbps, 48kHz
5: Subtitle (PGS), English
6: Subtitle (PGS), Swedish
7: Subtitle (PGS), Norwegian
8: Subtitle (PGS), Danish
9: Subtitle (PGS), Finnish
a03 The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
a03 The libav DTS decoder doesn't decode the full DTS-HD information.
Creating file "V:\Blurays and DVDs\Saving Mr Banks\Saving Mr Banks_Chapters.txt"...
v02 Extracting video track number 2...
a03 Extracting audio track number 3...
a03 Extracting DTS core...
a03 Decoding with libav/ffmpeg...
a03 Reducing depth from 64 to 32 bits...
a03 Encoding AAC <0.50> with NeroAacEnc...
v02 Creating file "V:\Blurays and DVDs\Saving Mr Banks\Saving Mr Banks.h264"...
a03 0:21:12 The source file seems to be damaged (sync byte missing).
v02 0:21:13 The source file seems to be damaged (sync byte missing).
v02 0:21:13 The source file seems to be damaged (discontinuity).
a03 0:21:12 The source file seems to be damaged (discontinuity).
a03


Can somebody tell me what could be the reason for this error?

Currently it's not possible that GUIs can read colored eac3to output, because of this I don't have the full error message.

[Snowknight26]

The reason is probably that "the source file seems to be damaged."

[tebasuna51]

regarding 2.1 DTS-HD MA tracks: Does the arcsoft bug (LFE channel cannot be decoded) still exist in each arcsoft version? Or is there a version which is not affected by the bug?

Using dtsdecoderdll v1.1.0.0 the LFE is decoded without problems.

[GCRaistlin]

Does the DRC bug (http://forum.doom9.org/showthread.php?p=1404212#post1404212) still exist in 3.27?

[tebasuna51]

Does the DRC bug (http://forum.doom9.org/showthread.php?p=1404212#post1404212) still exist in 3.27?

Seems you talk about the NERO DRC bug (http://forum.doom9.org/showthread.php?p=1399113#post1399113)

Now the default decoder for AC3 is libav instead Nero for that problem.

[GCRaistlin]

tebasuna51, using libav leads to another issue (http://forum.doom9.org/showthread.php?p=1508936#post1508936), doesn't it?

[nevcairiel]

tebasuna51, using libav leads to another issue (http://forum.doom9.org/showthread.php?p=1508936#post1508936), doesn't it?

Those issues have long been fixed.

[r0lZ]

Another question about the Arcsoft DTS decoder. Is it better to use it when decoding "regular" 5.1 DTS tracks? Or is it useful only for the recent 7.1 evolutions? AFAIK, a decoder that is able to decode a specific format (without bugs) should give exactly the same result than another decoder, and therefore there is no advantage to use the slow Arcsoft decoder when decoding a DTS tracks that the libav decoder can also decode properly. Am I right?

[tebasuna51]

tebasuna51, using libav leads to another issue (http://forum.doom9.org/showthread.php?p=1508936#post1508936), doesn't it?
What is the problem?

Of course there are very light differences between decoders but jruggle (ffmpeg developer in AC3 area) minimize the problem.
eac3to uses libav to decode AC3 to 64 bits float (the best resolution), and libav was actualized in eac3to v3.27 (the test was made with 3.24).
I'm trust in the AC3 expert (jruggle) about that.

BTW, if you prefer use a class "A" free decoder liba52 you can use BeHappy with NicAudio to decode AC3.

[tebasuna51]

Another question about the Arcsoft DTS decoder....is it useful only for the recent 7.1 evolutions?

libav can't decode DTS-HD, only the 'core' is decoded, then can't be used to obtain 7.1 output and lose info from DTS-HR or DTS-MA (lossless).

... Is it better to use it when decoding "regular" 5.1 DTS tracks?... AFAIK, a decoder that is able to decode a specific format (without bugs) should give exactly the same result than another decoder, and therefore there is no advantage to use the slow Arcsoft decoder when decoding a DTS tracks that the libav decoder can also decode properly. Am I right?

Lossy decoders not always output exactly the same result.
There are light differences (like with AC3 in precedents posts).
ArcSoft is a "certified" decoder, libav a free decoder.
It's your choice.

[r0lZ]

OK, thanks for the information.

[Thunderbolt8]

got an error message:

X:\file.m2ts>eac3to 1) 2: X:\file.mkv
M2TS, 1 video track, 11 audio tracks, 10 subtitle tracks, 1:48:42, 24p /1.001
1: Chapters, 37 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
...
v02 Extracting video track number 2...
v02 The h264 muxer doesn't support this stream type yet.
v02 Please send a 20MB sample to dear@madshi.net
v02 Muxing video to Matroska...
Aborted at file position 1048576.

sample can be found here https://www.sendspace.com/file/f7n75i (its a file from 2013)

[bmcelvan]

I am having a problem trying trying to decode a TrueHD to anything, wavs, w64, wav or flac. I get the following log:

bitstream parsing for track 2 failed
demuxing this track may still produce correct results - or not.
This audio conversion is not supported.

The eac3to info says:
2: TrueHD, English, 7.1 channels, 48KHz

The mediainfo says:
Audio #1
ID : 2
Format : TrueHD
Codec ID : A_TRUEHD
Duration : 2h 2mn
Bit rate mode : Variable
Maximum bit rate : 7 719 Kbps
Channel(s) : 8 channels
Channel positions : Front: L C R, Side: L R, Back: L R, LFE
Sampling rate : 48.0 KHz
Compression mode : Lossless
Language : English
Default : Yes
Forced : No

I have demuxed using mkvtoolnix and changed the extension to .ac3, .thd+ac3 and .truehd and when I try to look at it with eac3to it just says The format of the source file could not be detected.

tsmuxer won't recognize it and vidcoder (handbrake) doesn't even acknowledge it is there.

Could this be some new form of protection from Dolby. Do I need to update a codec or something.

Could it by Dolby atmos - is there a decoder for that?

[bmcelvan]

*Edit: My apologies, due to bad searching I didn't find the posts about using ffmpeg.

Is there a way to determine if the above track I was talking about is Dolby Atmos. Is that what the A in A_TrueHD stands for? If Yes, then:

I just want to be certain. If I have an .mkv file or a Bluray folder with a 5.1 or 7.1 TrueHD (atmos) track and use the following command, it will output the lossless TrueHD extracted audio in a flac file with proper channel mapping?

ffmpeg -i "my file or folder" -map 0:1 "extracted_audio.flac"

[ndjamena]

(The "A" is for audio)

[Sparktank]

Is there a way to determine if the above track I was talking about is Dolby Atmos.

The case of the movie should tell you if it has Dolby Atmos.

[tebasuna51]

...
ffmpeg -i "my file or folder" -map 0:1 "extracted_audio.flac"

That must work for the second track of a input file (tested .m2ts or .mkv).
I don't know what happens with a folder input.

For questions about ffmpeg syntax please use this thread:
ffmpeg syntax (http://forum.doom9.org/showthread.php?t=171251)

[eddman]

I tried replacing the avcodec-54.dll and avutil-52.dll files with newer versions from the latest version of Zeranoe FFmpeg, but eac3to gives me an error that the dll is missing.

What am I doing wrong?

As a side question; is there any other tool/method for converting DTS or dolby audio besides eac3to?

[LigH]

You can certainly convert DVD Video compatible dts or Dolby Digital audio with other tools, like BeHappy or MeGUI, maybe TEncoder. HD audio from Blu-rays is probably not widely supported though.

[tebasuna51]

I tried replacing the avcodec-54.dll and avutil-52.dll files with newer versions from the latest version of Zeranoe FFmpeg, but eac3to gives me an error that the dll is missing.

What am I doing wrong?
The dll's for eac3to aren't the standard dll's, must be compiled for eac3to by madshi.

As a side question; is there any other tool/method for converting DTS or dolby audio besides eac3to?
The answer is in your first question: ffmpeg

Try TAudioConverter (http://forum.doom9.org/showthread.php?t=165577) (ffmpeg)
Or AviSynth based BeHappy/MeGUI like LigH say.

[shh]

Hello!
Config:
C:\TMP>C:\eac3to\eac3to.exe -test
eac3to (v3.27) is up to date
Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed
...
ArcSoft DTS Decoder (1.1.0.8) works fine
Sonic Audio Decoder (3.27.0.0) doesn't seem to be installed

I got some problems with a DTS-HD MA Mono file:
10MB snippet: http://www.file-upload.net/download-10353765/test.dtsma.html
C:\TMP>C:\eac3to\eac3to.exe test.dtsma
DTS Master Audio, 1.0 channels, 24 bits, 48kHz
(core: DTS, 1.0 channels, 24 bits, 768kbps, 48kHz)
Transcoding to WAV / FLAC / ... results in a sped up audio file.
Only way to handle this is extracting the DTS-core and convert the DTS afterwords.
However, MPC-HC is able to play the original .dtsma file. So the ArcSoft DTS Decoder does work for this file?
Any hints on what to do?

[Music Fan]

You should maybe try ArcSoft DTS Decoder v1.1.0.0 instead of 1.1.0.8 for this file.

[shh]

v1.1.0.0 workend. :eek:
Thanks a lot!

[73ChargerFan]

This is much simpler. Extract the TrueHD Atmos stream to a multi-channel wav via ffmpeg, and then run that wav through eac3to to dts or whatever.
ffmpeg -i 00000.m2ts -map 0:1 extract.wav
eac3to extract.wav extract.dts

Madshi, any ETA when you'll be able update to discard the Atmos extensions? I'm finding it in more and more BDs.

Thanks.

[Batman007]

Guys
I'm trying to slow down an audio from 25.000 FPS to 23.976 FPS through ea3to
I put this code into cmd
C:\eac3to\ea3to.exe "D:\MeGUI 2418\MeGUI output\AVSEQ02 Tc0 L2 2ch 48 224 DELAY 0ms.mp3" "101 D.mp3" -changeTo23.976

But I'm getting this error :
http://i.imgur.com/kbhusvs.jpg

Please tell me what mistake I'm making or what I'm missing

Thank you ......

[LigH]

ea>c<3to.exe

[r0lZ]

Try:
eac3to.exe "D:\MeGUI 2418\MeGUI output\AVSEQ02 Tc0 L2 2ch 48 224 DELAY 0ms.mp3" "101 D.mp3" -slowdown

-slowdown convert 25.000 and 24.000 content to 23.976 fps

[xxx666yyy777]

Guys
I'm trying to slow down an audio from 25.000 FPS to 23.976 FPS through ea3to
I put this code into cmd


But I'm getting this error :
http://i.imgur.com/kbhusvs.jpg

Please tell me what mistake I'm making or what I'm missing

Thank you ......

BTW - you can usually use the "Tab" button to auto-complete directories and filenames in the MS-DOS window...