eac3to - audio conversion tool

[tebasuna51]

...
But, shouldnt I be setting the source to "29.970" for NTSC? Thats what it shows in mediainfo as frame reate for the video.

One more time, the video frame rate isn't related with audio.
Try yourself.

[sreemv]

One more time, the video frame rate isn't related with audio.
Try yourself.

Sorry for being a newbie but I was under the impression that I am having to compensate/stretch the audio to match the video frame rate difference[29.970 vs 23.976/24]? No?

[TinTime]

Sorry for being a newbie but I was under the impression that I am having to compensate/stretch the audio to match the video frame rate difference[29.970 vs 23.976/24]? No?

No. If the video framerate is changed by adding/dropping frames/fields then the duration is the same and the audio requires no adjustment.

[Forteen88]

How come, when I convert the audio from VOB-files (AC3) to AAC, I sometimes get this type of message: "a03 [NeroAacEnc] Processed 7260 second. Aborted at file position 4153661440.", but sometimes the ~third try it works OK?

[mrr19121970]

How come, when I convert the audio from VOB-files (AC3) to AAC, I sometimes get this type of message: "a03 [NeroAacEnc] Processed 7260 second. Aborted at file position 4153661440.", but sometimes the ~third try it works OK?

the 1st 2 times you get a read error, the 3rd you didn't.

[Forteen88]

the 1st 2 times you get a read error, the 3rd you didn't.That never happens when I use MKVMerge, so I doubt that the harddrive is bad. Does eac3to got better errorcheck than MKVMerge, or what?
Also, when I convert to FLAC, I never got that problem.

EDIT: @MikeEby I run it on a copy (ripped) that is already on the harddrive. But as said, it seems to be an issue that has not been properly fixed yet.

[MikeEby]

That never happens when I use MKVMerge, so I doubt that the harddrive is bad. Does eac3to got better errorcheck than MKVMerge, or what?
Also, when I convert to FLAC, I never got that problem.

You get this error when your ripping directly off the Blu-ray or when the copy of the disk is already on the hard drive? I've never seen the error when AnyDVD has made a copy to hard drive. I have seen it when ripping directly off the Blu-Ray disk.

Mike

[Snowknight26]

It's an issue with the way NeroAacEnc and eac3to work together. Ever since I reported the issue, madshi tried to fix, but by doing so caused the same issue to appear under different circumstances. The only sure fire to not run into the issue is to encode individual demuxed tracks to AAC.

[IanD]

A little help here? :helpful:

To sum up: Blade Runner HD-DVD, seamless branching with 3 different versions of the movie. eac3to gives me the list of playlist, listing all of them with 3 audio. When I demux the first playlist, i get this "video and audio overlaps" problem, and I get 6 audio streams instead of 3. More importantly, none of the demuxed audio track is correctly in sync with the video, starting from 1:39:58.
An option with HD-DVD titles and AnyDVD HD is to edit the playlist .XPL file to remove any unwanted material. Once you have it pared down to the absolute minimum, eac3to and ClownBD are usually able to work with it.

I have seen some titles that embed a different video structure at the start of an .evo, which throws off eac3to: by looking into the .XPL file and editing it to remove the offending reference, all is well. The only problem is that chapters times are then incorrect and must be adjusted accordingly. Interestingly, I think Bluray is starting to prefix different video at the beginning of some files too.

I strongly suggest, if working with HD-DVD, to become familiar with the basics of editing .XPL playlists as an additional tool: that way you have exact control over what is happening, instead of relying on software you have little control over.

[shogo_kawada]

I strongly suggest, if working with HD-DVD, to become familiar with the basics of editing .XPL playlists as an additional tool: that way you have exact control over what is happening, instead of relying on software you have little control over.
Thanks for your reply, but I already solved my Blade Runner situation using the process described in this (http://forum.doom9.org/showthread.php?p=1459110#post1459110)post. I also managed to add the italian soundtrack (taken from a cheap dvd boxset), slowed down with eac3to

[jd213]

Hmm, all of the HD DVD to BD-R conversions I've done with thd+ac3 tracks encoded by eac3to give me an "Unknown Disc" error with my Oppo BDP-80, although they work fine in my PS3 and Japanese Sony BD recorder. Discs without thd+ac3 work fine in the Oppo.

I'll be sure to ask in a forum dedicated to Oppo players since it seems to be more of a player problem, but any ideas what might be causing this?

edit: nevermind, it looks like all of the TrueHD titles I re-authored coincidentally were all burned to LTH discs, which won't play on the BDP-80.

[TDiTP_]

May be bug?

1). eac3to 3.24 can't decode any MLP encoded in SurCode Dolby MLP Encoder 1.0.29. (sample (http://www.mediafire.com/download.php?m26rmm80dk9feyk))


eac3to v3.24
command line: eac3to ID20.mlp ID20.wav -libav
------------------------------------------------------------------------------
MLP, 5.1 channels, 16 bits, 48kHz
Decoding with libav/ffmpeg...
The libav decoder output an unexpected bitdepth (-1). <ERROR>
Aborted at file position 262144. <ERROR>


eac3to v3.24
command line: eac3to ID20.mlp ID20.nero.wav -nero
------------------------------------------------------------------------------
MLP, 5.1 channels, 16 bits, 48kHz
Disabling DRC for Nero (E-)AC3 decoding...
Decoding with DirectShow (Nero Audio Decoder 2)...
The DirectShow audio decoder didn't accept the input stream. <ERROR>
Aborted at file position 262144. <ERROR>

At the same time older versions of eac3to (for example 3.17) work perfectly (bit-in-bit). As well as http://sourceforge.net/projects/dvdadecoder/ .

2). ArcSoft DTS Decoder 1.1.0.0 and 1.1.0.7 with eac3to can't normal decode DTS(-HD) 2.1. LFE-channel is clear.
In the same time Sonic (4.3.0.169) works well.
Samples: DTS 2.1 (http://www.mediafire.com/?g2n295cd847w9x1), DTS-96/24 2.1 (http://www.mediafire.com/?87a1nso5jho362v), DTS-HD MA 2.1 (http://www.mediafire.com/?0hz1kpe8ibyqffp).

[Inspector.Gadget]

ArcSoft DTS Decoder 1.1.0.0 and 1.1.0.7 with eac3to can't normal decode DTS(-HD) 2.1. LFE-channel is clear.

Arrrgh, can't Arcsoft just make something that ALWAYS WORKS?

[Thunderbolt8]

whats worse is they dont even seem to be able or willing to fix it in updates. those problems must have been known to them since quite a while.

[xkodi]

whats worse is they dont even seem to be able or willing to fix it in updates. those problems must have been known to them since quite a while.

since Sonic works well for those DTS(-HD) 2.1 tracks then i don't see why we should care about Arcsoft, but it would be very handy if we have a chart giving information which DTS(-HD) with which decoder (Arcsoft, Sonic, etc) works good.

[dansrfe]

Yep, I can confirm the bug. Well this just sucks.

[TDiTP_]

AFAIK ArcSoft bugs are:

1). Any version can't ignore DN in DTS-HD MA and DTS-HD Hi-Res.
Solition: Sonic, but we can't use it with > 6.1
BTW reference decoder of DTS-HD StreamPlayer can't ignore dialog normalization too (that is why file-header cutting won't help us)

2). DTS-ES Discrete 6.1/6.0 (in eac3to terms: DTS-ES 6.1/6.0) and DTS-HD with this core can be decoded with only ArcSoft 1.1.0.0. Other versions do not work.

3). DTS(-HD) 1.0 can be decoded with only ArcSoft 1.1.0.0 and 1.1.0.8.

4). Any version can't decode DTS-HD MA 7.1 ("strange setup").
Solution: if we have DTS-HD StreamPlayer we can cut file-header (first 140 byte) of similar stream obtained in DTS-HD Master Audio Suite and then merge it with our DTS-HD MA 7.1. This new track can be decoded in StreamPlayer.

5). Any version can't decode DTS(-HD) 2.1.
Solution: Sonic, Nero (libav gives too low level of LFE-channel)

[Frogger13]

Well beside being very time consuming this does not solve the problem at all as I had one time to restart single file encoding about 10 times until it finally succeded. In EAC3to v3.22 This error never happened to me and still doesn't (I reverted to that version since 3.24 is horrible)!

So if things where buggy then they where still way better than now.

Put simply: EAC3to v3.24 is useless for AC3 to AAC encoding and you must revert to v3.22!

It's an issue with the way NeroAacEnc and eac3to work together. Ever since I reported the issue, madshi tried to fix, but by doing so caused the same issue to appear under different circumstances. The only sure fire to not run into the issue is to encode individual demuxed tracks to AAC.

PS: I did never Encode from a muxed file. I always encode from extracted DTS or AC3 tracks. Both of them cause the Error, the only difference being that the error happens more often when more than 1 file is encoded at a time, but it still happens when encoding 1 file at a time.

[Thunderbolt8]

4). Any version can't decode DTS-HD MA 7.1 ("strange setup").
Solution: if we have DTS-HD StreamPlayer we can cut file-header (first 140 byte) of similar stream obtained in DTS-HD Master Audio Suite and then merge it with our DTS-HD MA 7.1. This new track can be decoded in StreamPlayer.
but whats the channel matrix then afterwards for that 7.1 track, is it that one of a strange setup track or of a normal 7.1 one? (they are slightly different)

[tebasuna51]

AFAIK ArcSoft bugs are:
...

4). Any version can't decode DTS-HD MA 7.1 ("strange setup").
Solution: if we have DTS-HD StreamPlayer we can cut file-header (first 140 byte) of similar stream obtained in DTS-HD Master Audio Suite and then merge it with our DTS-HD MA 7.1. This new track can be decoded in StreamPlayer.
I don't think so.
I think ArcSoft decode correctly DTS-HD MA 7.1 ("strange setup"), and search other "solution" is useless.

Let me explain the problem:

When decode standard 7.1 we can obtain a exact correlation between DTS speakers and WAV speakers:
Lss -> Side Left
Lsr -> Back Left
(the same for Right speakers)

But with "strange setup" we have:
Ls -> ?
Lsr -> Back Left
Don't exist, in WAV specs, a channel equivalent to Ls, we have only
SPEAKER_BACK_LEFT
SPEAKER_BACK_RIGHT
SPEAKER_BACK_CENTER
SPEAKER_SIDE_LEFT
SPEAKER_SIDE_RIGHT
(and others not related)

Then ArcSoft decode:
Side_Left = 0.6007 x Ls
Back_Left = 0.6804 x Lsr + 0.3196 x Ls
the Ls channel is separated in two components with this conditions:

0.6804 + 0.3196 = 1
to avoid overflow, and

(0.6007)^2 + (0.3196)^2 = (0.6804)^2
to preserve the audio power contribution of Ls.

All other channels must be attenuated also like:
Front_Left = 0.6804 x L
...
to preserve the volume balance.

And don't exist a better solution to the problem.
For what you want extract the exact Ls channel?
You can't convert it to other audio format, and you can't play it ,unless you change the speakers position physically in your audio room each time you want play a "strange setup".

When you play a DTS "strange setup" in your receiver it stop the play and send a message like "Please, change the Side speakers to Ls-Rs possitions and press Play to continue"?

I think not, or make a mix like ArcSoft do or you play incorrectly the audio.

[TDiTP_]

tebasuna51
Great explanation. Thanks.

For what you want extract the exact Ls channel?
Typical (but unusual) situation is decode "strange setup", then mix C-channel with translating voice (AFAIK it's popular in my country only :)) and then encode in the same scheme "strange setup".
Now I think that can use ArcSoft.

[Joniii]

I demuxed 2 BD's of The Aviator to use audio from import BD to Region A BD. German import has english DTS-HD Master Audio and I would like to use that with Region A BD with VC-1 because main track in it is AC3. Only problem is that the audio is not completely in synch.

I was thinking of converting DTS core and AC3 to wav with something like Wavelab and check the graph to see where the audio starts and add the delay when remuxing but that propably isn't the easiest way plus wavelab isn't freeware. Anyone have any ideas?

[dansrfe]

I don't quite understand why you you are bringing the Region A BD into the mix when your imported German disc has English DTS-HD Master Audio and a 1080p HD picture. Imported BD's, especially in AVC, are BETTER than VC-1 Region A BD's. Authoring/encoding houses in Germany actually know how to author/encode.

[sshd]

Small bug: When converting 24p mkv to 23.976

eac3to.exe file.mkv 1: vid.mkv 2: aud.flac -slowdown (works)
eac3to.exe file.mkv 1: vid.mkv 2: aud.flac 3: subs.srt -slowdown (-slowdown ignored)

[tebasuna51]

eac3to.exe file.mkv 1: vid.mkv 2: aud.flac -slowdown (works)
Only with flac, maybe you need:
eac3to.exe file.mkv 1: vid.mkv -slowdown 2: aud.flac -slowdown
eac3to.exe file.mkv 1: vid.mkv 2: aud.flac 3: subs.srt -slowdown (-slowdown ignored)
Yep, -slowdown don't work with subs.

[Joniii]

I don't quite understand why you you are bringing the Region A BD into the mix when your imported German disc has English DTS-HD Master Audio and a 1080p HD picture. Imported BD's, especially in AVC, are BETTER than VC-1 Region A BD's. Authoring/encoding houses in Germany actually know how to author/encode.

I thought of that too since it's AVC and has a high bitrate compared to VC-1 but I don't really understand the source of these german releases, I mean do they have the same quality masters? Is there a program that opens both AVC and VC-1 similar to VirtualDub so that I can go to specific frame and take a screenshots to compare?

[dansrfe]

I do believe The Aviator has the same master used for the Region A & B Discs. easiest way to compare is just open the file in vlc and take screenshots. another way is to index both .m2ts files with ffms2 and then do a stackhorizontal for side-by-side analysis of the same frame. But personally I would go with the AVC version with the higher bitrate.

[Joniii]

I do believe The Aviator has the same master used for the Region A & B Discs. easiest way to compare is just open the file in vlc and take screenshots. another way is to index both .m2ts files with ffms2 and then do a stackhorizontal for side-by-side analysis of the same frame. But personally I would go with the AVC version with the higher bitrate.
Thx for the info :).

[Joniii]

Another question.

I'm trying to add AC3 track from Cars PAL DVD to Cars Region A BD. Everything goes fine except I think eac3to is remapping the channels incorrectly, I mean DVD ac3 should have the same channel order than in BD or am I wrong?

D:\eac3to>eac3to e:\c.ac3 d:\cf.ac3 -slowdown
AC3 EX, 5.1 channels, 1:51:44, 384kbps, 48kHz
The Nero decoder doesn't seem to work, will use libav instead.
Decoding with libav/ffmpeg...
Remapping channels...
Changing FPS from 25.000 to 23.976...
Encoding AC3 <640kbps> with libAften...
Creating file "d:\cf.ac3"...
Clipping detected, a 2nd pass will be necessary.
Starting 2nd pass...
Decoding with libav/ffmpeg...
Remapping channels...
Changing FPS from 25.000 to 23.976...
Encoding AC3 <640kbps> with libAften...
Applying -0,28dB gain...
Creating file "d:\cf.ac3"...
eac3to processing took 25 minutes, 40 seconds.
Done.

[tebasuna51]

...
Everything goes fine except I think eac3to is remapping the channels incorrectly, I mean DVD ac3 should have the same channel order than in BD or am I wrong?

Don't worry, the message:

Remapping channels...

is from ac3 internal order to uncompressed order, but the final ac3 have the same channel map than original.

[Thunderbolt8]

there has been a lot of checking done with different versions of the arcsoft decoder files. but are there maybe other versions of the sonic decoder which might give us more benefits?

[xkodi]

there has been a lot of checking done with different versions of the arcsoft decoder files. but are there maybe other versions of the sonic decoder which might give us more benefits?

i don't believe so, because i ran some tests before:

http://forum.doom9.org/showthread.php?p=1212881#post1212881

and also here (at the end of the post) is summary and conclusions from the tests:

http://forum.doom9.org/showthread.php?p=1158014#post1158014

in fact i tried recently even the latest Sonic version "4.3.0.238" and it can't work with eac3to at all.

[xkodi]

I think ArcSoft decode correctly DTS-HD MA 7.1 ("strange setup"), and search other "solution" is useless.

if that is the case then why we want to decode DTS-HD MA 5.1 in bit-perfect way (and both Sonic and Arcsoft do that), because if you look the channel layout diagrams here:

http://forum.doom9.org/showthread.php?p=1443085#post1443085

then it's obvious there is no match of the speaker positions between Mircosoft 5.1 WAV file and DTS-HD MA 5.1 - Microsoft 5.1 WAV files use only 0x3F as channel mask for 5.1 WAV files, which is FL, FR, FC, LFE, SL, SR.

so, maybe i'm wrong somehow, but please explain.

also, the same post gives information that DTS-HD MA 7.1 ("strange setup") tracks can be decoded with Sonic bit-perfectly to 5.1 channels that match DTS-HD MA 5.1 speaker positions, which is another thing that seems correct at least to me to be done.

P.S. and the same question(s) for DTS-HD MA 7.1 not-"strange setup" tracks based on DTS and Microsoft channel layout diagrams and respectively speaker positions.

[EDIT] BTW, in the mentioned old post, DTS channel layout diagrams are taken directly from DTS Labs software encoder software and Microsoft ones are from here:

http://download.microsoft.com/download/9/c/5/9c5b2167-8017-4bae-9fde-d599bac8184a/SpkrConfig5.docx

page 6 and 7, where is also explained how Microsoft 5.1 WAV is played back on 7.1 channel system in Windows.

[TDiTP_]

Then ArcSoft decode:
Side_Left = 0.6007 x Ls
Back_Left = 0.6804 x Lsr + 0.3196 x Ls

These coefficients must be function of the angle between C-speaker and Rs-speaker. May be anybody know angle? I can't find any information about it.
The angle between C-speaker and Rsr-speaker must be 150º, is not it?

[tebasuna51]

These coefficients must be function of the angle between C-speaker and Rs-speaker. May be anybody know angle? I can't find any information about it.
The angle between C-speaker and Rsr-speaker must be 150º, is not it?

Yes, must be function of the angle, but doesn't exist a common opinion about the exact angles between speakers:
http://en.wikipedia.org/wiki/File:Standard_7.1_surround_sound_speaker_placement.png

The same for 5.1 config, the Surround channels can be located between 110º and 135º.

[tebasuna51]

...
then it's obvious there is no match of the speaker positions between Mircosoft 5.1 WAV file and DTS-HD MA 5.1 - Microsoft 5.1 WAV files use only 0x3F as channel mask for 5.1 WAV files, which is FL, FR, FC, LFE, SL, SR.

so, maybe i'm wrong somehow, but please explain.
Please don't add more confussion.
Of course MS WAV channels don't have a exact correlation with Surround channels, for that admit two ChannelMask for 5.1 config:
0x003F channel mask for FL, FR, FC, LF, BL, BR
0x060F channel mask for FL, FR, FC, LF, SL, SR

Both must be played in the exact way:
- A 5.1 system must consider the last channels as Surround channels (120º more o less).
- A 7.1 system must split the Surround channels between Side and Back channels.

also, the same post gives information that DTS-HD MA 7.1 ("strange setup") tracks can be decoded with Sonic bit-perfectly to 5.1 channels that match DTS-HD MA 5.1 speaker positions, which is another thing that seems correct at least to me to be done.

P.S. and the same question(s) for DTS-HD MA 7.1 not-"strange setup" tracks based on DTS and Microsoft channel layout diagrams and respectively speaker positions.
I never tested the Sonic decoder and don't know what is exactly your questions, sorry.

For me the standard 7.1 have a exact correlation with MS channels (and standard location of speakers in a 7.1 audio system), but we need make some correction for "strange setup"

Also a receiver than accept 7.1 but have only 5.1 speakers must do some correction not only ignore the channels.

[xkodi]

Please don't add more confussion. sorry, but i'm confused and i believe i'm not the only one having the same questions regarding this subject.

Of course MS WAV channels don't have a exact correlation with Surround channels, for that admit two ChannelMask for 5.1 config:
0x003F channel mask for FL, FR, FC, LF, BL, BR
0x060F channel mask for FL, FR, FC, LF, SL, SR

the channel mask of 0x060F exists only in theory (and not even in theory if you ask me) and in the document i referred to:

http://download.microsoft.com/download/9/c/5/9c5b2167-8017-4bae-9fde-d599bac8184a/SpkrConfig5.docx

on page 9 Microsoft make it very clear that 5.1 WAV files no matter of the actual channel mask in newer versions of Windows are treatеd as with channel mask of 0x3F and effectively there is no channel mask of 0x060F, because even that channel mask is actually used it's the same for Windows as channel mask of 0x3F and thus in practice there is only one channel mask for 5.1 WAV files:

In Windows XP with SP2 and Windows Server 2003 with SP1, if the system mixer produces a 5.1-channel output stream, the mixer always sets the stream's channel mask to 0x3F. The system mixer behaves this way even if it receives a 5.1-channel input stream with a channel mask of 0x60F. With this behavior, an audio driver never receives a 5.1-channel stream with a channel mask of 0x60F from the mixer.

Both must be played in the exact way:

i agree with this statement, because in practice Windows doesn't make difference between 0x3F and 0x60F channel mask as "SpkrConfig5.docx" document states

- A 5.1 system must consider the last channels as Surround channels (120º more o less).

i'm not sure about the exact angle and what "Surround channels" means in this case, but that is clearly defined in "SpkrConfig5.docx" document that in 5.1 setup Microsoft expect last two channel to be "Side Left" (SL) and "Side Right" (SR).

so, do you say that SL and SR in Microsoft world is what in DTS world is "Ls" ans "Rs"? because that is in contradiction with DTS 5.1 speaker positions:

http://www.imagebam.com/image/addcf997797266

and Microsoft 5.1 speaker positions as defined in page 6 "Playing a 5.1-Channel Stream on a 7.1 Speaker Configuration" of "SpkrConfig5.docx" document. that's the most confusing part at least for me.

- A 7.1 system must split the Surround channels between Side and Back channels.

that's not correct according page 6 "Playing a 5.1-Channel Stream on a 7.1 Speaker Configuration" of "SpkrConfig5.docx" document, where it's stated by Microsoft that the Back channels are just silence in such case. so, i can't agree here unless there is some document backing up your statement.

I never tested the Sonic decoder and don't know what is exactly your questions, sorry.

For me the standard 7.1 have a exact correlation with MS channels (and standard location of speakers in a 7.1 audio system)

how's that? at least from my point of view it's only in your opinion, because i don't think there is 1:1 match in speaker position between:

DTS-HD MA 7.1 (normal setup):

http://www.imagebam.com/image/43bb3497794528

and Microsoft 7.1 setup:

http://www.imagebam.com/image/160e7497795034

considering how DTS-HD MA 5.1 speaker position is:

http://www.imagebam.com/image/addcf997797266

and that there is no match in speaker position of Microsoft and DTS for 5.1 channels setup at least from what's defined in "SpkrConfig5.docx" document.

anyway, i don't want to argue about it, but there is no doubt it's very confusing.

[tebasuna51]

...
the channel mask of 0x060F exists only in theory (and not even in theory if you ask me)
Say you that to madshi and Flac developers, eac3to and flac decoder always output 0x060F.
and in the document i referred to:

http://download.microsoft.com/download/9/c/5/9c5b2167-8017-4bae-9fde-d599bac8184a/SpkrConfig5.docx
The document isn't clear, see in this extract of KsMedia.h:
// Speaker Positions:
#define SPEAKER_FRONT_LEFT 0x1
#define SPEAKER_FRONT_RIGHT 0x2
#define SPEAKER_FRONT_CENTER 0x4
#define SPEAKER_LOW_FREQUENCY 0x8
#define SPEAKER_BACK_LEFT 0x10
#define SPEAKER_BACK_RIGHT 0x20
#define SPEAKER_FRONT_LEFT_OF_CENTER 0x40
#define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80
#define SPEAKER_BACK_CENTER 0x100
#define SPEAKER_SIDE_LEFT 0x200
#define SPEAKER_SIDE_RIGHT 0x400
...
#define KSAUDIO_SPEAKER_5POINT1 (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT)
...
#define KSAUDIO_SPEAKER_5POINT1_SURROUND (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT)
Then both exist :
KSAUDIO_SPEAKER_5POINT1 = 0x3F with BL, BR
KSAUDIO_SPEAKER_5POINT1_SURROUND = 0x60F with SL, SR
but both must be played the same.

Other question is how the system mixer work:
- If the input is a 5.1 0x60F the output is 5.1 0x3F, no problem.
- If the input is 7.1 and need 5.1 output, discards channels 4 and 5 (BL,BR) and send a 5.1 0x3F with original channels SL, SR. This undesired behaviour can be modified using filters like ffdshow.

But all of this is irrelevant to your question:
so, do you say that SL and SR in Microsoft world is what in DTS world is "Ls" ans "Rs"?
No, I say than no matter how you call the last 2 channels (Back or Side) in a 5.1 audio they are always the "Ls, Rs" channels in DTS world (~120º), or Surround channels in AC3 world (~110º)

Forget also all the figures about exact speakers position in the MS docx, the audio is created like DTS (or DD) say and real audio systems must have the speakers located like DTS-DD say:
- For a 5.1 system: Surround channels at 110º-120º (not Side 90º nor Back 150º)
- For a 7.1 system: Side channels at ~90º, Back channels ~150º

[TDiTP_]

Yes, must be function of the angle, but doesn't exist a common opinion about the exact angles between speakers:
http://en.wikipedia.org/wiki/File:Standard_7.1_surround_sound_speaker_placement.png

I thought DTS Inc. published their angle recommendation.

Then another question please. How did you obtain that coefficients?

[tebasuna51]

Then another question please. How did you obtain that coefficients?
- I make 8 monowav test channel, each one with a max peak at 0 dB (normalized).

- I encode the 8 channels to 7.1 "strange setup" and after decoded with ArcSoft.

- The max peak of each channel in decoded wav say me the coefficients

[Skinleech]

I took delivery of a Blackgold 3620 DVB-T2 decoder yesterday so I can cap Freeview HD streams in the UK. I have got it set up with DVBViewer to record the incoming .ts streams - the broadcasts are 1440x1088/50i with AAC audio streams (AAC-LC afaik). Eac3to can correctly identify the streams in the source .ts but it can't proces the AAC tracks in anyway. For example, eac3to says the following when checking a recorded stream:

eac3to v3.24
command line: eac3to "L:\Recorded TV\12-10_19-18-52_BBC HD_Wild China.ts"
------------------------------------------------------------------------------
TS, 1 video track, 2 audio tracks, 1 subtitle track, 0:01:41, 50i
1: h264/AVC, 1440x1080 50i (16:9)
2: AAC, English, unknown parameters, -1718ms
3: AAC, English, unknown parameters, -1252ms
4: Subtitle (DVB), English
Bitstream parsing for tracks 2 and 3 failed. <WARNING>
Demuxing these tracks may still produce correct results - or not. <WARNING>

If I demux the tracks I get AAC files, but these still cannot be processed by eac3to. If I try playback in MPC, the audio is detected as a mpeg stream, and fails to play - no AAC filter is loaded. DVBViewer plays the recorded streams without issue. Is there any way to get eac3to to parse the AAC tracks, so I can convert them to ac3 - I can then edit the streams in VideoRedo.

Here's a sample:

http://www.freefilehosting.net/12-1019-18-52bbchdwildchina

[tebasuna51]

I took delivery of a Blackgold 3620 DVB-T2 decoder yesterday so I can cap Freeview HD streams in the UK. I have got it set up with DVBViewer to record the incoming .ts streams - the broadcasts are 1440x1088/50i with AAC audio streams (AAC-LC afaik).
...
Your Track 2 is a AAC LOAS-LATM.

After extract with eac3to you need convert this aac to a standard AAC with vlc:
"D:\your path\vlc.exe" -I dummy "China_2 DELAY -1718ms.aac" --sout #std{access=file,mux=mp4,dst="China_2_nolatm.mp4"} vlc://quit

Now you have a playable mp4:
Audio format : AAC-LC
Channels ....: 6
SampleRate ..: 48000 Hz.
Duration trak: 100.096 sec., (0h. 1m. 40.096s.)
BitRate .....: 315 Kb/s

You can use the mp4 to mux with video in a mkv/mp4 container or decode with NeroAacDec

Edit: your Track 3 (also LOAS-LATM) seems a:
Audio format : AAC-HE v2 (LC+SBR+PS)
Channels ....: 2
SampleRate ..: 48000 Hz.
Duration trak: 99.584 sec., (0h. 1m. 39.584s.)
BitRate .....: 63 Kb/s
but I get only silence.

[Skinleech]

Your Track 2 is a AAC LOAS-LATM.

After extract with eac3to you need convert this aac to a standard AAC with vlc:

Thank you very much - just tried these steps and I now have a playable audio file! I assume the conversion is not lossless, i.e. some kind of header removal? Is it possible to go direct to ac3 via VLC? I don't use it at all.

With regards to the 2nd audio track, it is likely supposed to be silence - the 2nd track is designed to carry an audio description service, but only 2 - 3 programmes a day actually use the track.

Thank you for your assistance.

[tebasuna51]

I assume the conversion is not lossless, i.e. some kind of header removal?...
Yes, the conversion is lossless, the audio data in LOAS-LATM stream is the same than audio data in .mp4.

vlc remove LOAS and LATM headers and put the header parameters in the mp4 container.

[SeeMoreDigital]

eac3to v3.24
command line: eac3to "L:\Recorded TV\12-10_19-18-52_BBC HD_Wild China.ts"So BBC have decided to broadcast using AAC audio. The same as ITV. This move could set a precedent for all futue FTA HDTV channels in the UK :eek:

I wonder.... Is there any chance some kind person on this forum can "whip-up" an dedicated application to strip these LOAS/LATM headers from AAC streams?


Cheers

[Snowknight26]

Any chance of implementing working outputting to NUL? Currently if you output a track to NUL:\file.ext, eac3to just stalls a few seconds in, and NUL doesn't work at all.

[xkodi]

I wonder.... Is there any chance some kind person on this forum can "whip-up" an dedicated application to strip these LOAS/LATM headers from AAC streams?


there is already such application(s) available:

LOAS_LATM_to_ADTS.exe: LOAS/LATM AAC to ADTS AAC

(ADTS_to_RAW.exe: ADTS AAC to RAW AAC
LOAS_LATM_to_RAW.exe: LOAS/LATM AAC to RAW AAC)

File name: AAC_Tools_NA_0.1.zip File size: 148.64 KB (http://www.fileserve.com/file/5wzwNpg)

as well as the zip file includes the source code in Delphi - the same programming language in which eac3to is written and thus such conversion(s) could be included easily in eac3to too

[tebasuna51]

there is already such application(s) available:

LOAS_LATM_to_ADTS.exe: LOAS/LATM AAC to ADTS AAC

(ADTS_to_RAW.exe: ADTS AAC to RAW AAC
LOAS_LATM_to_RAW.exe: LOAS/LATM AAC to RAW AAC)

File name: AAC_Tools_NA_0.1.zip File size: 148.64 KB (http://www.fileserve.com/file/5wzwNpg)

as well as the zip file includes the source code in Delphi - the same programming language in which eac3to is written and thus such conversion(s) could be included easily in eac3to too

Yes, but don't work for me.

See this post http://forum.doom9.org/showthread.php?t=157467

Is the same soft than proposed in http://www.dvbviewer.tv/forum/topic/35738-loaslatm-%26gt%3B-adts-conversion/

[Skinleech]

Yes, the conversion is lossless, the audio data in LOAS-LATM stream is the same than audio data in .mp4.

vlc remove LOAS and LATM headers and put the header parameters in the mp4 container.

Great, thanks. I'm currently messing about with a decent workflow so I can pull the files into VRD4 to edit ad breaks etc.

Is it possible to apply the associated audio delay to the .mp4 output by VRD during the header removal, rather than setting it in the container?

[tebasuna51]

...
Is it possible to apply the associated audio delay to the .mp4 output by VRD during the header removal, rather than setting it in the container?
Sorry, I don't know a tool to convert +delay to initial AAC silence like eac3to/DelayCut can do with AC3/DTS/MP3.