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Old 28th October 2008, 11:39   #6761  |  Link
tebasuna51
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Quote:
Originally Posted by ACrowley View 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")
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.

Quote:
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.
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Old 28th October 2008, 12:39   #6762  |  Link
ACrowley
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Quote:
Originally Posted by tebasuna51 View Post
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/...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.

Last edited by ACrowley; 28th October 2008 at 12:50.
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Old 28th October 2008, 14:45   #6763  |  Link
tebasuna51
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Quote:
Originally Posted by ACrowley View Post
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

Quote:
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.
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Old 28th October 2008, 18:46   #6764  |  Link
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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)

Code:
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

Last edited by Thunderbolt8; 29th October 2008 at 01:07.
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Old 28th October 2008, 19:01   #6765  |  Link
piratburner
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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"...
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Old 28th October 2008, 19:10   #6766  |  Link
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Quote:
Originally Posted by piratburner View Post
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.
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Old 28th October 2008, 19:19   #6767  |  Link
piratburner
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Quote:
Originally Posted by tebasuna51 View Post
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 :-)

Last edited by piratburner; 28th October 2008 at 19:47.
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Old 28th October 2008, 20:07   #6768  |  Link
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tebasuna,
why not edit your signature and put pcm2tsmu link?
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Old 28th October 2008, 20:17   #6769  |  Link
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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...
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Old 28th October 2008, 21:05   #6770  |  Link
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Quote:
Originally Posted by piratburner View Post
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
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Old 28th October 2008, 21:46   #6771  |  Link
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Quote:
Originally Posted by Octo-puss View Post
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.
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Old 28th October 2008, 22:01   #6772  |  Link
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There's logic behind that, but if the extraction goes at around under 5MB/s, I believe there's a fair bit of reserve
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Old 28th October 2008, 22:06   #6773  |  Link
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Well... What is your CPU? I have a Core 2 Duo 6600.
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Old 28th October 2008, 22:12   #6774  |  Link
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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.
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Old 28th October 2008, 22:19   #6775  |  Link
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Well... What is your CPU? I have a Core 2 Duo 6600.
E6750 (overclocked, but that doesn't matter much).
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Old 28th October 2008, 22:19   #6776  |  Link
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Quote:
Originally Posted by Thunderbolt8 View Post
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.
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Old 28th October 2008, 22:23   #6777  |  Link
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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?
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Old 28th October 2008, 22:25   #6778  |  Link
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Quote:
Originally Posted by odin24 View Post
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!
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Old 28th October 2008, 22:27   #6779  |  Link
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If you only need center, you can use undocumented switch -mono to save some space and hdd activity.
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Old 28th October 2008, 23:24   #6780  |  Link
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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
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