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28th October 2008, 11:39 | #6761 | Link | ||
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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:
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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28th October 2008, 12:39 | #6762 | Link | |
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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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28th October 2008, 14:45 | #6763 | Link | ||
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- 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:
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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. Last edited by Thunderbolt8; 29th October 2008 at 01:07. |
28th October 2008, 19:01 | #6765 | Link |
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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"... |
28th October 2008, 19:19 | #6767 | Link | |
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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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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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28th October 2008, 21:05 | #6770 | Link | |
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28th October 2008, 21:46 | #6771 | Link | |
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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. |
28th October 2008, 22:19 | #6776 | Link | |
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Then delete the ones you do not need, keeping the center channel. |
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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? |
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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