eac3to - audio conversion tool

[piratburner]

DTS-HD 7.1 works fine for me, HDDVD --> BD with DTSHD-MA 7.1, using tsMuxeR. All demuxed using eac3to. Maybe try demuxing the individual track instead of the whole disc.

Leon, as in the Professional on BD? Not in North America yet :(, one of my favs!
Yes !
Léon 1994 Ultimate Edition 1080p Blu-ray GER AVC DTS-HD MA 7.1

With English sound :)

Here is a trackname for one DTS-HD when using -demux ....
00005 - 3 - DTS Hi-Res, German, 7.1 channels, 24 bits, 2864kbps, 48khz.dtshr


When playing a original m2ts file from Léon i have sound from my TviX but after TsMuxer = No Sound :-(

[piratburner]

DTS-HD 7.1 works fine for me, HDDVD --> BD with DTSHD-MA 7.1, using tsMuxeR. All demuxed using eac3to. Maybe try demuxing the individual track instead of the whole disc.



Are you always using eac3to to demux a BD and then TsMuxer ??


I have always used TsMuxer from original BD and make a "movie only" m2ts, So I will see how it works when using eac3to first :-)

[rica]

If the -demux feature is used does this demux the entire BD/HDDVD? Or is -demux used for specific streams in a track. Normally I would specify which stream I want along with it's destiation without using -demux.

I don't use -demux option. I specify the files as well.

[piratburner]

Are you always using eac3to to demux a BD and then TsMuxer ??


I have always used TsMuxer from original BD and make a "movie only" m2ts, So I will see how it works when using eac3to first :-)

Ok I got this 2 files from eac3to ... using -demux
00005 - 2 - h264, 1080p24.h264
00005 - 4 - DTS Hi-Res, English, 7.1 channels, 24 bits, 2864kbps, 48khz.dtshr

Add this 2 into TsMuxeR -> M2TS muxing -> play the file from TviX And I got NO sound :devil:

Used version 1.8.8

[nautilus7]

Obviously the problem is NOT eac3to. Why do you keep posting here? It would be better to open a new thread and discuss your problem there.

[shroomM]

Hey,

first, a big thank you for the program, it is a godsend :)
But I do have a couple of quick questions...

I have a RAW/PCM, 5.1 channels, 16 bits, 48khz track and would like to encode it to Dolby Digital 5.1. I know I can encode using eac3to by just specifying the output file as .ac3, but here's what I'm more interested in...

When I demux the PCM track, here's the eac3to's output...

[a04] Reading RAW/PCM...
[a04] Swapping endian...
[a04] Remapping channels...
[a04] Swapping endian...
[a04] Remapping channels...
[a04] Creating file "D:\bluray\eng.pcm"...

If I later run eac3to with the syntax eac3to D:\bluray\eng.pcm D:\bluray\eng_flac.flac, the output is....

This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
The RAW/PCM file seems to be big endian.
The RAW/PCM file seems to have a bitdepth of 16 bits.
The RAW/PCM file seems to have 6 channels.
RAW/PCM, 5.1 channels, 0:08:28, 16 bits, 4608kbps, 48khz
This is probably a Blu-ray PCM track. Will remap channels accordingly.
Reading RAW/PCM...
Swapping endian...
Remapping channels...
Encoding FLAC with libFlac...

I'm now wondering ... is the ordering of channels in the FLAC file OK or is it swapped?

The second question is about downmixing 5.1 to stereo (I know it's pretty much obsolete, but I need it in this case)...

Assuming that the FLAC file above is OK (channels and all), can I just load that FLAC in i.e. meGUI (provided that I have madFlac installed). Would that downmix be OK?

Is the alternative, adding a -down2 to eac3to when generating the FLAC also OK ?

[himan2001]

2. Is there a way with eac3to to demux in 1 step TrueHD track directly from BD/HD into ac3 core and lossless part (for Scenarist)?
I tried it with HD DVD yesterday and i made 2 steps: 1) thd+ac3 track creation 2) demuxing to ac3 and mlp. It's not a problem for me, just curious.


Still the same "issue" with DTS and the -core option.

I personly dislike 7.1 downmixing from DTS-HD 7.1 files, when
a "clean" 5.1 core is present.

But is not possible to reencode directly in 1 step to ac3 from the -core. eac3to always sees the 7.1 MA and try to downmix from the extention. At the moment, first the -core must be exported, than you can ac3-encode from the 5.1 core.

[rica]

Still the same "issue" with DTS and the -core option.

I personly dislike 7.1 downmixing from DTS-HD 7.1 files, when
a "clean" 5.1 core is present.

But is not possible to reencode directly in 1 step to ac3 from the -core. eac3to always sees the 7.1 MA and try to downmix from the extention. At the moment, first the -core must be exported, than you can ac3-encode from the 5.1 core.

Core is not an independent file as in THD+ac3 (BD);
it can not be read individually since core is an obligation for DTS-HD format: system works in this manner: core+extensions.
So if you want to re-encode it to ac3, eac3to must decode and downmix before encoding in case of extension is 7.1.

EDIT:
Some points:
If you want to extract core dts (5.1) as is, use -core option.
When you try to re-encode (or to make a simple revision like convert 24 to 16), not only eac3to but any program has to decode the original file first.
If you don't ask eac3to extracting core, it is gonna directly try to decode DTS-HD which is 7.1 in your case.
But default encoder (aften) is not able to encode to 7.1; so the original has to be downconverted to 5.1 by eac3to.
Unless you use a pro encoder by Dolby, you will never able to get an 7.1 dolby re-encode; even in this case, as you guess, eac3to must firstly decode your DTS-HD.
So difference between you and the guy from whom you taken the quote is;
he is getting/or extracting a seperate ac3 (as is) from thd+ac3 combination -it is not a core,
but he has to make a second command line since it needs a lossless thd as well.
you, on the other hand, are re-encoding DTS-HD, not core.
So the way you follow;
extracting core dts
and later,
converting it to 5.1 is normal process.
BTW; shift+arrow up repeats your previous command.

[tebasuna51]

...
I'm now wondering ... is the ordering of channels in the FLAC file OK or is it swapped?
Must be Ok. If you detect any problem please report it.

The second question is about downmixing 5.1 to stereo (I know it's pretty much obsolete, but I need it in this case)...

Assuming that the FLAC file above is OK (channels and all), can I just load that FLAC in i.e. meGUI (provided that I have madFlac installed). Would that downmix be OK?
Why not?

Is the alternative, adding a -down2 to eac3to when generating the FLAC also OK ?
Of course. Test yourself.

If the raw/pcm track from your BD is the 'x' you can use directly:
eac3to D:\bluray x: D:\bluray\eng_flac2.flac -down2

[ACrowley]

Not all Ok here.
If we name source channels like Rear: RL, RR, RC and

RL' = +90º phase shift RL
RR' = -90º phase shift RR

seems the phase shift is make at encoder phase and is the same for 'matrix' and 'discrete' DTS-ES in all the tests.

The output channels after encode to 3/3.1 DTS-ES discrete and decoded with ArcSoft:

1) 3/3.1 -> 5.1 (like previous test)

SL = RL' + 0.71 x RC
SR = RR' + 0.71 x RC
Ok

2) 3/3.1 -> 6.1

BC = RC
SL = RL'
SR = RR'
Ok

3) 3/3.1 -> 7.1

SL = 0.40 x RL' + 0.60 x RC
SR = 0.40 x RR' + 0.60 x RC
BL = 0.75 x RL'
BR = 0.75 x RR'
Seems wrong: RC only in Side and most Rear part in Back, I think must be the other way, Side <-> Back

----------------------------
The output channels after encode to 3/3.1 DTS-ES matrix and decoded with ArcSoft:

1) 3/3.1 -> 5.1

SL = RL' + 0.71 x RC
SR = RR' + 0.71 x RC
Ok

2) 3/3.1 -> 7.1

SL = 0.71 x RL' + 0.50 x RC
SR = 0.71 x RR' + 0.50 x RC
BL = 0.71 x RL' + 0.50 x RC
BR = 0.71 x RR' + 0.50 x RC
For me seems ArcSoft can't recover the RC mixed in RL-RR because RC must be present only in Back channels.

3) 3/3.1 -> 6.1, eac3to crash with this bugreport.txt (http://pastebin.com/m276c45a0)

The report was made with 2.70 but is the same with 2.72, I send the 2.70 bug because was made after a reset with the minimum of soft running.

mhhh...youre right.
I compared only the BC Channels.
And the BC looks fully identical as you can see in my Post

But i think this can be fixed ?
I hope so ,because i have a lot of dts es 6.1 "Matrix" Tracks on my HDD ,waitnig for a proper Method to decode all 7 CH for a reencode ( all PAL Audio Tracks/Target is PAL to NTSC-reencode to dts es 6.1 "discrete")

On the other Side, i dont think the 6.1 Output from matrix 6.1from Arcsoft is sooooo wrong..better then nothing ?!

[tebasuna51]

But i think this can be fixed ?
I hope so ,because i have a lot of dts es 6.1 "Matrix" Tracks on my HDD ,waitnig for a proper Method to decode all 7 CH for a reencode ( all PAL Audio Tracks/Target is PAL to NTSC-reencode to dts es 6.1 "discrete")
Do you want reencode yours 6.1 "Matrix" to 6.1 "Discrete"?
Do you have a 6.1 audio equipment without support for 6.1 "Matrix" but yes for 6.1 "Discrete"?

Seems ArcSoft can't do the job by the moment.

On the other Side, i dont think the 6.1 Output from matrix 6.1from Arcsoft is sooooo wrong..better then nothing ?!

I can't obtain (crash) a 6.1 output from a 6.1 "Matrix"

The 7.1 output is a waste of space, with SL = BR and SR = BR, I'm sure any 7.1 audio equipment can do the job at play time.

[ACrowley]

Do you want reencode yours 6.1 "Matrix" to 6.1 "Discrete"?
Do you have a 6.1 audio equipment without support for 6.1 "Matrix" but yes for 6.1 "Discrete"?

Seems ArcSoft can't do the job by the moment.



I can't obtain (crash) a 6.1 output from a 6.1 "Matrix"

The 7.1 output is a waste of space, with SL = BR and SR = BR, I'm sure any 7.1 audio equipment can do the job at play time.


yes, i have this AV Receiver with a 7.1 Speaker Setup
http://www.kenwood.de/products/home/kompo/avcc/KRF-V7300D/kenwood (HDMI 1.3.a/7.1, dts-hd ,trueHD,DD+,dts es Discrete/Matrix, 7.1 PCM etc) .

I have a lot of dts es 6.1 matrix Tracks and i want reencode them to dts es 6.1 discrete.
Because : They are in PAL and i need them in NTSC. So i make a Timestretch and a reencode with dts pro series. Ofcourse i want keep the BC Channel. I get on 5.1 from dts es matrix Tracks with current Methods.

Mh.. no, i get clean output from dts es 6.1 matrix tracks with arcsoft -7. As you can see in my Post.

[tebasuna51]

I have a lot of dts es 6.1 matrix Tracks and i want reencode them to dts es 6.1 discrete.
Because : They are in PAL and i need them in NTSC. So i make a Timestretch and a reencode with dts pro series. Ofcourse i want keep the BC Channel. I get on 5.1 from dts es matrix Tracks with current Methods.
I have a method to extract the BC channel but is not easy, work fine with test samples (channels separated in time) but I can test real samples. Maybe you can make the test:
- Decode the 6.1 matrix to 6 monowavs (5.1)
- Merge (WaveWizard or Sox) the Back (or Side) channels to stereo.
- Decode this stereo file like DolbyPrologic with Foobar2000 and the Free Surround plugin decoder (Center=1, Dimension=-0.5, A=0, B=0).
- The FL,FR,FC from the resultant 5.1 output are your BL,BR,BC channels.

With the test file:
FL = 0.9448 x BL + 0.0004 x BC
FR = 0.9448 x BR + 0.0004 x BC
FC = 0.0235 x BL + 0.0248 x BR + BC
LF < 0.0000
SL < 0.0000
SR < 0.0000

Mh.. no, i get clean output from dts es 6.1 matrix tracks with arcsoft -7. As you can see in my Post.

I don't know what is my problem, I'm wait to madshi can see anything at bug report.

[Thunderbolt8]

Got a problem with a H.264 TV cap:
(actually wanted to slow it down, both video and audio, to 23.976 fps, but this doesnt work as well ofc)

eac3to v2.72
command line: eac3to G:\bla.ts G:\bla.mkv
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 0:00:05
1: h264/AVC, 1080i50 (16:9)
2: AC3, English, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB, 57ms
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[v01] Muxing video to Matroska...
[a02] Removing AC3 dialog normalization...
[a02] Applying (E-)AC3 delay...
[a02] This doesn't seem to be a valid PES packet.
Aborted at file position 1277952.

heres a 10mb sample: http://www.sendspace.com/file/yveygw

[piratburner]

I trying to Convert .dtshr to PCM . I got info about remapping channel, is that a problem or have I forgot something ? Se my command below.

L:\leon>eac3to "DTS Hi-Res, English, 7.1 channels, 24 bits, 2864kbps, 48khz.dtshr" LEON.PCM
DTS Hi-Res, 7.1 channels, 2:12:50, 24 bits, 2864kbps, 48khz
Decoding with ArcSoft DTS Decoder...
Swapping endian...
Remapping channels...
Creating file "LEON.PCM"...

[tebasuna51]

I got info about remapping channel, is that a problem or have I forgot something ?

Don't worry. Eac3to only inform you (DTS internal channel order is different than LPCM order). All is ok.

[piratburner]

Don't worry. Eac3to only inform you (DTS internal channel order is different than LPCM order). All is ok.

Thanks :-)

I have a problem with DTS-HD 7.1 with tsMuxer so I will give this a try instead, Love eac3to :)

Ahhh PCM are not allowed in TsMuxer so I can't do a new m2ts file, Or do I need something more with my PCM track in eac3to before I add that to TsMuxer?

EDIT

Found this little tool :-) Pcm2Tsmu :-) so now it's possible to use PCM in TsMuxer :-)

[rica]

tebasuna,
why not edit your signature and put pcm2tsmu link?

[Octo-puss]

I got a question. Is there any room for performance improvement? It looks to me like the data extracting is pretty slow (I don't understand how it works though, so it might be normal of course). The other day I took one of my movies and dug the AC3 audio out of it - it wasn't even converting from other format - and it was pretty slow...

[piratburner]

I trying to Convert .dtshr to PCM . I got info about remapping channel, is that a problem or have I forgot something ? Se my command below.

L:\leon>eac3to "DTS Hi-Res, English, 7.1 channels, 24 bits, 2864kbps, 48khz.dtshr" LEON.PCM
DTS Hi-Res, 7.1 channels, 2:12:50, 24 bits, 2864kbps, 48khz
Decoding with ArcSoft DTS Decoder...
Swapping endian...
Remapping channels...
Creating file "LEON.PCM"...

OK I use Pcm2Tsmu for my pcm track, but adding the file to TsMexer, then is telling me thats is a 5.1 track. And playing the m2ts file is giving me funny nois. Is it a bug in eac3to when converting to PCM

[nautilus7]

I got a question. Is there any room for performance improvement? It looks to me like the data extracting is pretty slow (I don't understand how it works though, so it might be normal of course). The other day I took one of my movies and dug the AC3 audio out of it - it wasn't even converting from other format - and it was pretty slow...I am not right person to answer that, but using different HDDs for source and destination files improves speed quite enough.

[Octo-puss]

There's logic behind that, but if the extraction goes at around under 5MB/s, I believe there's a fair bit of reserve :)

[nautilus7]

Well... What is your CPU? I have a Core 2 Duo 6600.

[Thunderbolt8]

does anyone know how to create a single channel wav file (only the center channel) of a multi channel flac/ac3/lpcm/dtshd/truehd source track?
when I do "eac3to source.track outcome.wav" then I only get one huge file, which includes all the source channels and not only the center one I want to have.

[Octo-puss]

Well... What is your CPU? I have a Core 2 Duo 6600.

E6750 (overclocked, but that doesn't matter much).

[odin24]

does anyone know how to create a single channel wav file (only the center channel) of a multi channel flac/ac3/lpcm/dtshd/truehd source track?
when I do "eac3to source.track outcome.wav" then I only get one huge file, which includes all the source channels and not only the center one I want to have.

eac3to source.audio output.wavs

Then delete the ones you do not need, keeping the center channel.

[MichaelAnders]

Using v2.72, it seems as if the channel order is corrupt?

M2TS, 1 video track, 2 audio tracks, 1:21:00
1: VC-1, 1080p24 /1.001 (16:9)
2: DTS Master Audio, German, 5.1 channels, 24 bits, 48khz
3: DTS Master Audio, English, 7.1 channels, 16 bits, 48khz
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "e.flac"...
[a03] The original audio track has a constant bit depth of 16 bits.

If I convert the German audio to FLAC (same as before, this time only the 2nd stream), everything is fine and the file sounds perfect in MPCHC and also in foobar2000:
[a02] Original audio track: max 24 bits, average 16 bits, most common 16 bits.

The English FLAC however, is totally wrong. It seems as if FL and FR are now SL and SR, the subwoofer is also on some other channel...

If I play the English stream in the M2TS file, the "properties" in MPCHC shows "Audio: DTS 48000Hz 6ch [Audio]".

I don't have any older version of eac3to to check this, but something is wrong... Maybe eac3to finds 8 channels, but it's actually 6?

[Thunderbolt8]

eac3to source.audio output.wavs

Then delete the ones you do not need, keeping the center channel.
d'oh, missed the plural. too easy, thanks! :D

[nautilus7]

If you only need center, you can use undocumented switch -mono to save some space and hdd activity.

[Thunderbolt8]

hm the result seems to be slightly different in that case, when I cancel the process after some mb (I only need the beginning of the file) and try to open it with wavepad it tells me that I need the ACM plugin, while the cancelled center channel from '.wavs' works fine :S

[nautilus7]

It should be the wav header in the begging of each file.

[tebasuna51]

OK I use Pcm2Tsmu for my pcm track, but adding the file to TsMexer, then is telling me thats is a 5.1 track. And playing the m2ts file is giving me funny nois. Is it a bug in eac3to when converting to PCM

Eac3to don't have a bug please go to Pcm2Tsmu thread (http://forum.doom9.org/showthread.php?p=1207732#post1207732).

[Snowknight26]

8: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
"Commentary By Director Jonathan Demme And Screenplay Co-Writer Daniel Pyne"
[...]
[a08] Reducing depth from 64 to 24 bits...
When converting to wav.

[Thunderbolt8]

a technical question: when comparing 2 source.track -> .wav files of the same channel from different source types (e.g. center channel of 5.1 ac3 and flac or left and right channel of 2.0 ac3 with left and right channel of 5.1 flac) and looking for their different delays is it technically correct then to look where the first tiny bit of sound occurs in the channel graph? what I mean is that a flac track of course has a higher quality as an ac3 track and this difference must be located somewhere. so could it be possible that the flac track has numerical more digits in the graphs at the beginning of the track compared to the ac3 track so that this kind of delay meassurement is wrong, or is this difference of sound quality only to be found in the amplitude of the graph, that it only spread in the vertical, but not the horizontal direction of the graph?

[svgame]

thank you Madshi
just there has a problem.i make 3 step
1) use eac3to demux TrueHD track directly from BD/HD
2) type command "eac3to 00001.thd audio.thd" "eac3to 00001.thd audio.ac3",demuxing to ac3 core and lossless part
3) rename audio.thd to audio.mlp,and import to Scenarist.at 100%,its hung!!!about 2 minute，get a error message "External component has thrown an exception"
where is problem?Madshi

[ACrowley]

I have a method to extract the BC channel but is not easy, work fine with test samples (channels separated in time) but I can test real samples. Maybe you can make the test:
- Decode the 6.1 matrix to 6 monowavs (5.1)
- Merge (WaveWizard or Sox) the Back (or Side) channels to stereo.
- Decode this stereo file like DolbyPrologic with Foobar2000 and the Free Surround plugin decoder (Center=1, Dimension=-0.5, A=0, B=0).
- The FL,FR,FC from the resultant 5.1 output are your BL,BR,BC channels.

With the test file:
FL = 0.9448 x BL + 0.0004 x BC
FR = 0.9448 x BR + 0.0004 x BC
FC = 0.0235 x BL + 0.0248 x BR + BC
LF < 0.0000
SL < 0.0000
SR < 0.0000



I don't know what is my problem, I'm wait to madshi can see anything at bug report.

so the outout is identical to the source BC Channel ?
Will i twork with DD EX tracks?

By the Way, i compared all Surround/Back Channles from dts es discrete with the test matrix reencode and arcsoft -7.
Theyre all looking fine and identical compared to the original dts es discrete....

However, Madshi will make it ,as always

[tebasuna51]

so the outout is identical to the source BC Channel ?
The analog extraction isn't perfect, like you see there are BL and BR parts (2%) added to the BC channel.

Will i twork with DD EX tracks?
Seems works with dolby-waterfal_51EX sample but I don't have channel test EX to quantify the separation.

By the Way, i compared all Surround/Back Channles from dts es discrete with the test matrix reencode and arcsoft -7.
Theyre all looking fine and identical compared to the original dts es discrete....

Then if work for you don't need nothing more.

[ACrowley]

Then if work for you don't need nothing more.

yeah..it should be tested more..im not sure
Strange, i cant reproduce my Test. eac3to/arcsoft -7 crashes on the dts es matrix reencode. But it works with other matrix dts tracks!

Madshi should take a closer look, he is the master:)

[rica]

Hi guys.

How can i shorten this:

eac3to E:\VIDEO_TS\VTS_01.vob+E:\VIDEO_TS\VTS_01_2.vob 2: C:\KB\video.m2v 6: C:\KB\dtsaudio.dts 5:

This:

eac3to E:\VIDEO_TS\[VTS_01_1+VTS_01_2].vob

or this:

eac3to E:\VIDEO_TS\[VTS_01_1.vob+VTS_01_2.vob]

doesn't work?


_ _ _ _

[madshi]

I need to remux no. 4: DTS-ES 6.1 track 1536k and 3: DD 5.1EX 640k track.
Here is the log:

but... PDVD reports that I've got DTS-ES 5.1 1536k track and not 6.1...
Previously during playing BD movie using the same PDVD - reported: DTS-ES 6.1 1536k.
Please explain what I did wrong.
Don't trust in what PDVD says. Run "eac3to demuxedtrack.dts". What does eac3to say?

btw, madshi, is there any chance eac3to to report if AC3 track is DD-EX?
Will add that to my to do list.

Not all Ok here.
If we name source channels like Rear: RL, RR, RC and

RL' = +90º phase shift RL
RR' = -90º phase shift RR

seems the phase shift is make at encoder phase and is the same for 'matrix' and 'discrete' DTS-ES in all the tests.
Ouch. Don't you consider that "bad"? I mean just imagine we want to speed correct (e.g. apply PAL speedup) to a DTS track. Doesn't that mean that the phase changes with every new reencode? So if I do "eac3to source.dts dest.dts -speedup" the phase of the surround channels will be different?

1) 3/3.1 -> 5.1 (like previous test)

SL = RL' + 0.71 x RC
SR = RR' + 0.71 x RC
Ok
Ok, that's fine, that's how eac3to is also doing 6.1 -> 5.1 downmixing.

3) 3/3.1 -> 7.1

SL = 0.40 x RL' + 0.60 x RC
SR = 0.40 x RR' + 0.60 x RC
BL = 0.75 x RL'
BR = 0.75 x RR'
Seems wrong: RC only in Side and most Rear part in Back, I think must be the other way, Side <-> Back
Agreed.

2) 3/3.1 -> 7.1

SL = 0.71 x RL' + 0.50 x RC
SR = 0.71 x RR' + 0.50 x RC
BL = 0.71 x RL' + 0.50 x RC
BR = 0.71 x RR' + 0.50 x RC
For me seems ArcSoft can't recover the RC mixed in RL-RR because RC must be present only in Back channels.
Agreed.

3) 3/3.1 -> 6.1, eac3to crash with this bugreport.txt (http://pastebin.com/m276c45a0)

The report was made with 2.70 but is the same with 2.72, I send the 2.70 bug because was made after a reset with the minimum of soft running.
It seems that the ArcSoft decoder crashes internally somewhere. That is probably a bug in the ArcSoft decoder. You have version 1.1.0.0.

@ACrowley, are you using a different ArcSoft decoder version? FWIW, I'm getting the same crash as tebasuna51...

Still the same "issue" with DTS and the -core option.

I personly dislike 7.1 downmixing from DTS-HD 7.1 files, when
a "clean" 5.1 core is present.

But is not possible to reencode directly in 1 step to ac3 from the -core. eac3to always sees the 7.1 MA and try to downmix from the extention. At the moment, first the -core must be exported, than you can ac3-encode from the 5.1 core.
There's a bug in eac3to. If you use the "-core" option, eac3to nevertheless thinks that it will get 7.1 from the decoder. Will have to fix that.

However, I'd strongly recommend that you reconsider your transcoding method. The "clean" 5.1 core will result in worse audio quality than the eac3to produced 7.1 downmix because the 7.1 source has a higher quality in every single channel compared to the core 5.1 track. Even if you're afraid that eac3to's 7.1 -> 5.1 downconversion is worse compared to what the studio did, still the front channels and LFE will have a higher quality if you use the full DTS-HD track. So IMHO using only the core is a bad idea.

Got a problem with a H.264 TV cap:
(actually wanted to slow it down, both video and audio, to 23.976 fps, but this doesnt work as well ofc)

eac3to v2.72
command line: eac3to G:\bla.ts G:\bla.mkv
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 0:00:05
1: h264/AVC, 1080i50 (16:9)
2: AC3, English, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB, 57ms
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[v01] Muxing video to Matroska...
[a02] Removing AC3 dialog normalization...
[a02] Applying (E-)AC3 delay...
[a02] This doesn't seem to be a valid PES packet.
Aborted at file position 1277952.

heres a 10mb sample: http://www.sendspace.com/file/yveygw
Looks like a broken/corrupted file.

does anyone know how to create a single channel wav file (only the center channel) of a multi channel flac/ac3/lpcm/dtshd/truehd source track?
when I do "eac3to source.track outcome.wav" then I only get one huge file, which includes all the source channels and not only the center one I want to have.
This will decode the first 50MB of the source file and store only the center channel into a WAV file:

"eac3to sourcefile test.wav -mono -50mb"

a technical question: when comparing 2 source.track -> .wav files of the same channel from different source types (e.g. center channel of 5.1 ac3 and flac or left and right channel of 2.0 ac3 with left and right channel of 5.1 flac) and looking for their different delays is it technically correct then to look where the first tiny bit of sound occurs in the channel graph?
I'd suggest using Audacity or a similar wave editor and compare the form of the graphs. It's very easy to see how much delay an audio track needs this way.

Using v2.72, it seems as if the channel order is corrupt?

M2TS, 1 video track, 2 audio tracks, 1:21:00
1: VC-1, 1080p24 /1.001 (16:9)
2: DTS Master Audio, German, 5.1 channels, 24 bits, 48khz
3: DTS Master Audio, English, 7.1 channels, 16 bits, 48khz
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "e.flac"...
[a03] The original audio track has a constant bit depth of 16 bits.

If I convert the German audio to FLAC (same as before, this time only the 2nd stream), everything is fine and the file sounds perfect in MPCHC and also in foobar2000:
[a02] Original audio track: max 24 bits, average 16 bits, most common 16 bits.

The English FLAC however, is totally wrong. It seems as if FL and FR are now SL and SR, the subwoofer is also on some other channel...

If I play the English stream in the M2TS file, the "properties" in MPCHC shows "Audio: DTS 48000Hz 6ch [Audio]".

I don't have any older version of eac3to to check this, but something is wrong... Maybe eac3to finds 8 channels, but it's actually 6?
Please try "eac3to English.flac Englisch.wav -50mb". That will decode the first 50MB of the FLAC file to a WAV file. Then load this WAV file in e.g. Audacity and check whether the channel order is correct or not.

Or if you want me to check it, please upload a small sample of the original 7.1 English DTS track, please?

8: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
"Commentary By Director Jonathan Demme And Screenplay Co-Writer Daniel Pyne"
[...]
[a08] Reducing depth from 64 to 24 bits...
When converting to wav.
Are you trying to test whether I can read your mind?

thank you Madshi
just there has a problem.i make 3 step
Please stop asking the same question over and over again. I've already answered it a while ago.

[madshi]

How can i shorten this
By calling eac3to from inside "E:\VIDEO_TS". Then you don't need to use any paths for the source files.

I got a question. Is there any room for performance improvement? It looks to me like the data extracting is pretty slow (I don't understand how it works though, so it might be normal of course). The other day I took one of my movies and dug the AC3 audio out of it - it wasn't even converting from other format - and it was pretty slow...
When extracing one AC3 track from a (M2)TS file I'm getting about 13-15 MB. That's ok, I think?

[rica]

By calling eac3to from inside "E:\VIDEO_TS". Then you don't need to use any paths for the source files

Thanks madshi but this time i get this:

C:\>eac3to E:\VIDEO_TS VTS_01_1.vob+VTS_01_2.vob
HD DVD / Blu-Ray disc structure not found.

[Snowknight26]

Are you trying to test whether I can read your mind?
No, but I didn't think a sample was necessary. I was able to convert it to wav using a previous version (can't remember which), but it doesn't work ever since you added the deeper bitdepth analysis, leading me to believe its a small bug introduced in your code only recently.

When extracing one AC3 track from a (M2)TS file I'm getting about 13-15 MB. That's ok, I think?
I have a server capable of doing 750MB/s average read with a C2Q, but it always takes eac3to about 10 minutes to demux a audio stream from your average length movie. Lets hope there is room for improvement. ;)

[madshi]

Thanks madshi but this time i get this:

C:\>eac3to E:\VIDEO_TS VTS_01_1.vob+VTS_01_2.vob
HD DVD / Blu-Ray disc structure not found.

I think you need to learn how to use command lines... ;)

You should have "E:\VIDEO_TS\>eac3to VTS_01_1.vob+VTS_01_2.vob".

No, but I didn't think a sample was necessary.
I'm not asking for a sample. You posted a log (which doesn't contain any error messages) without any further comment or question. What am I supposed to do with the log? I don't even know what you want from me.

[Snowknight26]

I'm not asking for a sample. You posted a log (which doesn't contain any error messages) without any further comment or question. What am I supposed to do with the log? I don't even know what you want from me.

[a08] Reducing depth from 64 to 24 bits...
Since when are AC3 tracks 64 bit?

[rica]

I think you need to learn how to use command lines... ;)

You should have "E:\VIDEO_TS\>eac3to VTS_01_1.vob+VTS_01_2.vob".



Agree :stupid: :p

[madshi]

Since when are AC3 tracks 64 bit?
AC3 tracks decode to floating point. The latest eac3to version internally uses 64bit floating point now.

[Snowknight26]

Ah, my mistake then. Good to hear though.

[tebasuna51]

RL' = +90º phase shift RL
RR' = -90º phase shift RR

seems the phase shift is make at encoder phase and is the same for 'matrix' and 'discrete' DTS-ES in all the tests.
Ouch. Don't you consider that "bad"? I mean just imagine we want to speed correct (e.g. apply PAL speedup) to a DTS track. Doesn't that mean that the phase changes with every new reencode? So if I do "eac3to source.dts dest.dts -speedup" the phase of the surround channels will be different?

Yes I think is "bad".
I don't know if DTS PRO ENCODER have a switch to apply or not the phase shift to surround channels (like Dolby Encoders have)

[MichaelAnders]

Or if you want me to check it, please upload a small sample of the original 7.1 English DTS track, please?
I don't have the tool you talked about, so here is the DTS file. I demuxed it via tsmuxergui, just the first 90 seconds. Interestingly, tsmuxergui also says it is 6 channels...

http://www.sendspace.com/file/6a3cl7