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Old 30th December 2010, 13:24   #10881  |  Link
Kisa_AG
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Quote:
Originally Posted by TDiTP_ View Post
It's not bug. eac3to never was programmed for such downmix (3.0->2.0). Only 5.1->2.0 and only DPLII (even 5.0->2.0 doesn't work).
Ok, thanks, now it's clear. But IMHO in such case it should be mentioned in eac3to help.
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Old 31st December 2010, 14:59   #10882  |  Link
a1adin
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Hi guys,

first of all, thanks for this very nice program.
sorry if this is a noob question:
I like to convert 5.1 aac to AC3.
Please let me know the correct command line to do it.

since I do not have nero, first I convert the aac to AC3 using vlc media player,
then I get the AC3 but with the wrong channel mapping.
Then I use eac3to to correct the mapping:
0 1 2 3 4 5
5.1 AAC FC FL FR SL SR LFE
5.1 AC3 FL FC FR SL SR LFE

so the command line is:
eac3to input.ac3 output.ac3 -1,0,2,3,4,5

However, when I listen to it, I still get the wrong channel mapping.
Please let me know what is wrong and how to fix it.
Thanks.

Last edited by a1adin; 31st December 2010 at 15:02.
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Old 31st December 2010, 17:41   #10883  |  Link
tebasuna51
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Quote:
Originally Posted by a1adin View Post
...
I like to convert 5.1 aac to AC3.
Please let me know the correct command line to do it.
since I do not have nero,...
Without Nero plugin for eac3to you can use Faad and Aften:
Faad -w -b 4 input.aac | Aften -readtoeof 1 -b 640 - output.ac3

Quote:
... first I convert the aac to AC3 using vlc media player, then I get the AC3 but with the wrong channel mapping.
With my test, always I convert AAC with VLC 1.1.4 I get a wrong channel map like this:
FL, FR, SL, SR, FC, LFE

no mater if the output is WAV, FLAC or AC3.
With AC3 the SR channel is empty.

Then I recommend you convert the AAC to FLAC (or WAV), to avoid the SR problem and two lossy encodes, with VLC and after use:
eac3to intermediate.flac output.ac3 -0,1,4,5,2,3
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Old 1st January 2011, 08:46   #10884  |  Link
a1adin
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Hi tebasuna51,

Thanks for a very informative reply!

Quote:
Originally Posted by tebasuna51 View Post
With my test, always I convert AAC with VLC 1.1.4 I get a wrong channel map like this:
FL, FR, SL, SR, FC, LFE
so that's the culprit I could not make it right.
thanks for letting me know, I almost die with frustration figuring that out.

Quote:
Originally Posted by tebasuna51 View Post
Without Nero plugin for eac3to you can use Faad and Aften:
Faad -w -b 4 input.aac | Aften -readtoeof 1 -b 640 - output.ac3
could you give the link for FAAD?

Also, I am going to install nero7.
What part of nero should I install to be able to use the aac decoder for eac3to?
Please advise.
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Old 1st January 2011, 08:56   #10885  |  Link
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Quote:
Originally Posted by a1adin View Post
Also, I am going to install nero7.
What part of nero should I install to be able to use the aac decoder for eac3to?
Please advise.
Nero Burning ROM is all I needed.
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Old 1st January 2011, 11:11   #10886  |  Link
tebasuna51
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Quote:
Originally Posted by a1adin View Post
...
could you give the link for FAAD?

Also, I am going to install nero7.
What part of nero should I install to be able to use the aac decoder for eac3to?
For Faad decoder (FAAD2v20100614 CVS), and other free decoders/encoders, you can go to http://www.rarewares.org/

For Nero 7 you need install also the Blu-Ray/HD DVD plugin.
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Old 1st January 2011, 17:06   #10887  |  Link
a1adin
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@ tebasuna51 & robpdotcom
thanks for the reply!

what is the best nero for eac3to: 7.8.5.0 or 7.10.1.0 ?

Quote:
Originally Posted by tebasuna51 View Post
Faad -w -b 4 input.aac | Aften -readtoeof 1 -b 640 - output.ac3
sorry if this is a noob question:
what is the "straight line" before Aften means?
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Old 2nd January 2011, 06:14   #10888  |  Link
tebasuna51
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Quote:
Originally Posted by a1adin View Post
...
what is the "straight line" before Aften means?
Is the "pipe" command, the standard output (STDOUT, -w parameter) of Faad is redirected to standard input (STDIN, input replaced with '-') of Aften.

The wav output of Faad is send to Aften through internal SO buffers (fast), without write the file to disk (slow).
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Old 3rd January 2011, 19:37   #10889  |  Link
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got a problem with a 2.0 DTS-HD MA track, always get this error message when trying to convert it to flac:

Code:
eac3to v3.24
command line: eac3to track.dtsma track.flac
------------------------------------------------------------------------------
DTS Master Audio, 2.0 channels, 24 bits, 48kHz
(core: DTS, 2.0 channels, 24 bits, 960kbps, 48kHz)
Decoding with ArcSoft DTS Decoder...
Encoding FLAC with libFlac...
Creating file "track.flac"...
The ArcSoft DTS Decoder reported an error while decoding.  <ERROR>
Aborted at file position 1871970304.  <ERROR>
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Old 3rd January 2011, 20:08   #10890  |  Link
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960kb/sec core? I've never seen an official disk using that. Is it some homebrewn source?

Cheers,

-jj-
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Old 3rd January 2011, 20:31   #10891  |  Link
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Quote:
Originally Posted by M-Blaster View Post
Hey, when demuxing a movie with DTS HD master audio
"eac3to l:\mainevo.evo e:\ren.mkv -core"
I am getting only big flac audio files. that can not integrated in the mkv movie.

How can I get dts files (1508kbit core only ) from an movie with DTS Master?
And how do I get an unaltered full DTS-HD Master Audio track? What extension do I want to specify for output?

The resulting audio track will then be fed to mkvmerge gui, to be multiplexed in an existing matroska container.
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Old 3rd January 2011, 21:23   #10892  |  Link
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Quote:
Originally Posted by jj666 View Post
960kb/sec core? I've never seen an official disk using that. Is it some homebrewn source?

Cheers,

-jj-
its the commentary track of the Master's of Cinema version of Metropolis (and the error happens somewhere halfway during the process, not directly at the beginning; demuxing this track was fine, as well as remuxing the video and all other tracks)

Last edited by Thunderbolt8; 3rd January 2011 at 21:26.
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Old 4th January 2011, 09:53   #10893  |  Link
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Quote:
Originally Posted by palpyhat View Post
And how do I get an unaltered full DTS-HD Master Audio track? What extension do I want to specify for output?

The resulting audio track will then be fed to mkvmerge gui, to be multiplexed in an existing matroska container.
eac3to "path\to\input.extension" 3: "save\path\to\output.dtshd"

Where input.extension can be some m2ts, evo, mpls or whatever source has the dts-hd audio. The "3:" in this instance is the specific track that the dts-hd audio is on. You can find the track number that holds the dts-hd file in your particular file by running:

eac3to "path\to\input.extension"
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Old 4th January 2011, 13:15   #10894  |  Link
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The extension .dtsma also works for DTS Master Audio, if you need to distinguish them from DTS Hi-Res (.dtshr).
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Old 4th January 2011, 13:39   #10895  |  Link
dansrfe
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.dtsma is probably a better choice, correct.
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Old 4th January 2011, 14:50   #10896  |  Link
palpyhat
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Thank you.

Also a couple of technical questions.

1. In a "DTS-HD Master Audio" audio track, which data is stored first in each packet/frame? The DTS core, or the required deltas to reconstruct the DTS-HD-MA lossless stream? I assume it is the former, but I am not sure.

2. If we assume that C=core, D=Deltas and L=Lossless then we have L=C+D, and in each "DTS-HD Master Audio" audio track we have the following information -> C, D. Is it possible, from any existing piece of software to rebuild the L stream? If so, can it be muxed into a Matroska container, and will any decoder understand it?
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Old 4th January 2011, 15:05   #10897  |  Link
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Quote:
Originally Posted by palpyhat View Post
2. If we assume that C=core, D=Deltas and L=Lossless then we have L=C+D, and in each "DTS-HD Master Audio" audio track we have the following information -> C, D. Is it possible, from any existing piece of software to rebuild the L stream? If so, can it be muxed into a Matroska container, and will any decoder understand it?
What you have described here looks like regular decode process -> DTS-HD MA aware software decoder (ArcSoft for instance) decodes compressed data into lossless stream. This stream you can save as LPCM with eac3to and mux into MKV later of course.
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Old 4th January 2011, 17:24   #10898  |  Link
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Nero 7 and Win7x86

I have Nero 7 (premium) installed, along with the BluRay/HDDVD Plugin, they show as being properly licensed in the Nero ProductSetup tool, on Win7x86; however eac3to -test still reports that "Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed".

This same version was working fine with no tweaks on WinXP, but not after a clean install of Win7 and Nero.

Can someone please enlighten me as to how to get eac3to to recognize the decoder on Win7x86?
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Last edited by therealjoeblow; 4th January 2011 at 17:38.
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Old 8th January 2011, 13:04   #10899  |  Link
TDiTP_
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After this post of tebasuna51 i decided to check all possible DTS-HD 7.1-layouts. I suppose that for all configurations Arcsoft makes correction for standard location of speakers in a 7.1 audio system and i suppose that we should use Wave_Format_Ex with MaskChannel = 0x063F for all layouts. I calculated the coefficients for every mix (some test samples were used and i have a DTS-HD M.A.S.).

I do not know the BD on which layouts from 1 till 5 were used but it can appear in the future.

=============================================
Results:

0).

Not compliant with Wave_Format_Ex; eac3to/arcsoft decodes it to Standart Mask = 0x063F (FL FR FC LFE BL BR SL SR). Mix:
Code:
FL = 0.68039*L			FR  = 0.68039*R
FC = 0.68039*C			LFE = 0.68039*LFE
SL = 0.60065*Ls			SR  = 0.60065*Rs 
BL = 0.31953*Ls + 0.68039*Lsr	BR  = 0.31953*Rs + 0.68039*Rsr
In other words (in dBs terms):
Code:
FL = L(-3.3 dB)			FR  = R(-3.3 dB)
FC = C(-3.3 dB)			LFE = LFE(-3.3 dB)
SL = Ls(-4.4 dB)		SR  = Rs(-4.4 dB) 
BL = Ls(-9.9 dB) + Lsr(-3.3 dB)	BR  = Rs(-9.9 dB) + Rsr(-3.3 dB)
The two conditions were satisfied:
  • Decoder avoids overflow: sum of the coefficients for every speaker less or equal 1.
  • Decoder preserves acoustic power contribution (~0.4629) for every speaker:
    Code:
    L:	0.68039^2 = 0.4629			LFE:	0.68039^2 = 0.4629
    R:	0.68039^2 = 0.4629			C:	0.68039^2 = 0.4629
    Ls:	0.31953^2 + 0.60065^2 = 0.4629		Rs:	0.31953^2 + 0.60065^2 = 0.4629
    Lsr:	0.68039^2 = 0.4629			Rsr:	0.68039^2 = 0.4629
So.. this is correct mix.

1).

Not compliant with Wave_Format_Ex; eac3to/arcsoft decodes it to Standart Mask = 0x063F (FL FR FC LFE BL BR SL SR). Mix:
Code:
FL = 0.58579*L + 0.41421*Lw	FR  = 0.58579*R + 0.41421*Rw
FC = 0.58579*C			LFE = 0.58579*LFE
SL = 0.51713*Ls + 0.41421*Lw	SR  = 0.51713*Rs + 0.41421*Rw
BL = 0.27514*Ls			BR  = 0.27514*Rs
In other words (in dBs terms):
Code:
FL = L(-4.6 dB) + Lw(-7.6 dB)	FR  = R(-4.6 dB) + Rw(-7.6 dB)
FC = C(-4.6 dB)			LFE = LFE(-4.6 dB)
SL = Ls(-5.7 dB) + Lw(-7.6 dB)	SR  = Rs(-5.7 dB) + Rw(-7.6 dB)
BL = Ls(-11.2 dB)		BR  = Rs(-11.2 dB)
The two conditions were satisfied:
  • Decoder avoids overflow: sum of the coefficients for every speaker less or equal 1.
  • Decoder preserves acoustic power contribution (~0.3431) for every speaker:
    Code:
    L:	0.58579^2 = 0.3431			LFE:	0.58579^2 = 0.3431
    R:	0.58579^2 = 0.3431			C:	0.58579^2 = 0.3431
    Lw:	0.41421^2 + 0.41421^2 = 0.3431		Ls:	0.51713^2 + 0.27514^2 = 0.3431
    Rw:	0.41421^2 + 0.41421^2 = 0.3431		Rs:	0.51713^2 + 0.27514^2 = 0.3431
    I do not give a similar calculations for the other layouts below, you can double-check the figures yourselves
This is correct mix too.

2).

This layout is compatible with Wave_Format_Ex with Mask = 0x560F (FL FR FC LF SL SR TFL TFR), but eac3to/arcsoft decodes it to Mask = 0x063F. Mix:
Code:
FL = 0.58579*L + 0.41421*Lh	FR  = 0.58579*R + 0.41421*Rh
FC = 0.58579*C			LFE = 0.58579*LFE
SL = 0.51713*Ls			SR  = 0.51713*Rs
BL = 0.27514*Ls			BR  = 0.27514*Rs
In other words (in dBs terms):
Code:
FL = L(-4.6 dB) + Lh(-7.6 dB)	FR  = R(-4.6 dB) + Rh(-7.6 dB)
FC = C(-4.6 dB)			LFE = LFE(-4.6 dB)
SL = Ls(-5.7 dB)		SR  = Rs(-5.7 dB)
BL = Ls(-11.2 dB)		BR  = Rs(-11.2 dB)
Decoder avoids overflow and preserves acoustic power contribution (~0.3431) for all speakers except Lh, Rh (~0.1716).

3).

Not compliant with Wave_Format_Ex; eac3to/arcsoft decodes it to Standart Mask = 0x063F (FL FR FC LFE BL BR SL SR). Mix:
Code:
FL = 0.62897*L			FR  = 0.62897*R
FC = 0.62897*C			LFE = 0.62897*LFE
SL = 0.55525*Ls + 0.44474*Lhs	SR  = 0.55525*Rs + 0.44474*Rhs
BL = 0.29543*Ls			BR  = 0.29543*Rs
In other words (in dBs terms):
Code:
FL = L(-4.0 dB)				FR  = R(-4.0 dB)
FC = C(-4.0 dB)				LFE = LFE(-4.0 dB)
SL = Ls(-5.1 dB) + Lhs(-7.0 dB)		SR  = Rs(-5.1 dB) + Rhs(-7.0 dB)
BL = Ls(-10.6 dB)			BR  = Rs(-10.6 dB)
Decoder avoids overflow and preserves acoustic power contribution (~0.3956) for all speakers except Rhs, Lhs (~0.1978).

4).

Compliant with Wave_Format_Ex, is decoded by eac3to/arcsoft to Mask = 0x0F0F (FL FR FC LFE BC SL SR TC).
Code:
FL = 0.58579*L			FR  = 0.58579*R
FC = 0.62897*C + 0.41421*Ch	LFE = 0.58579*LFE
SL = 0.27514*Ls + 0.41421*Cs	SR  = 0.27514*Rs + 0.41421*Cs
BC = 0.51713*Ls			TC  = 0.51713*Rs
In other words (in dBs terms):
Code:
FL = L(-4.6 dB)			FR  = R(-4.6 dB)
FC = C(-4.0 dB) + Ch(-7.6 dB)	LFE = LFE(-4.6 dB)
SL = Ls(-11.2 dB) + Cs(-7.6 dB)	SR  = Rs(-11.2 dB) + Cs(-7.6 dB)
BC = Ls(-5.7 dB)		TC  = Rs(-5.7 dB)
Decoder avoids overflow and preserves acoustic power contribution (~0.3433) for all speakers except C, Ch (~0.3956 and ~0.1716).

Looks like ArcSoft decodes as for standard MaskChannel = 0x063F and eac3to uses incorrect Mask.
After such decoding we need rename channels: SL->BL, SR->BR, BC->SL, TC->SR and it will be for "standard location of speakers in a 7.1 audio system".

5).

Compliant with Wave_Format_Ex, is decoded by eac3to/arcsoft to Mask = 0x270F (FL FR FC LF BC SL SR TFC).
Code:
FL = 0.72317*L			FR  = 0.72317*R
FC = 0.72317*C			LFE = 0.72317*LFE
SL = 0.33967*Ls + 0.51135*Cs	SR  = 0.33967*Rs + 0.51135*Cs
BC = 0.63841*Ls + 0.36158*Oh	TFC = 0.63841*Rs + 0.36158*Oh
In other words (in dBs terms):
Code:
FL = L(-2.8 dB)			FR  = R(-2.8 dB)
FC = C(-2.8 dB)			LFE = LFE(-2.8 dB)
SL = Ls(-9.4 dB) + Cs(-5.8 dB)	SR  = Rs(-9.4 dB) + Cs(-5.8 dB)
BC = Ls(-3.9 dB) + Oh(-8.8 dB)	TFC = Rs(-3.9 dB) + Oh(-8.8 dB)
Decoder avoids overflow and preserves acoustic power contribution (~0.5230) for all speakers except Oh (~0.2615).

Again, looks like ArcSoft decodes as for standard MaskChannel = 0x063F and eac3to uses incorrect Mask.
After such decoding we need rename channels: SL->BL, SR->BR, BC->SL, TFC->SR and it will be for "standard location of speakers in a 7.1 audio system".

=============================================
Conclusion:

For all layouts Arcsoft makes correction for standard location of speakers in a 7.1 audio system and i suppose that we should use Wave_Format_Ex with MaskChannel = 0x063F for all layouts and remap (rename) channels for some layouts.

BTW, ArcSoft 1.1.0.0 can decode "normal setup" (-logdts: $84b), but it can't decode 16-bit non-standart 7.1-layouts correctly (noise in FL and FR channels). ArcSoft 1.1.0.7 and 1.1.0.8 decodes all 7.1-layouts correctly.

Last edited by TDiTP_; 28th December 2011 at 21:09. Reason: add info
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Old 8th January 2011, 13:35   #10900  |  Link
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I have a movie, Edward Scissorhands, which has a DTS-HD MA 4.0ch track. Trying to convert the track to Dolby Digital but I receive the error:

Code:
a03 The AC3 encoder received a non-supported data format (pcm, 4, 24, -).
Can libaften not encode a 4ch track with no LFE?
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