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Old 27th July 2009, 20:59   #9141  |  Link
TinTime
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Quote:
Originally Posted by Thunderbolt8 View Post
is there a way to use -slowdown for 24fps sources, to slow them down to 23.976fps? it seems that eac3to always automatically assumes the source has 25 fps and it therefore uses a 1,024fps slowdown instead of only 0,024fps. the resulting length in my case is ~5 mins longer, while it should only be ~7 seconds longer.
You can use -changeTo23.976

If you've already demuxed the audio you need to specify the source speed too, like...

Code:
eac3to.exe input output -24.000 -changeTo23.976
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Old 27th July 2009, 21:22   #9142  |  Link
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-24.000 after input output? or is it input -24.000 output -changeto23.976, as usual?
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Old 27th July 2009, 21:26   #9143  |  Link
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Yes, after input output as above. Maybe it works the other way too - I've never tried it.
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Old 27th July 2009, 21:33   #9144  |  Link
Thunderbolt8
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nope, works only the way you described. thanks
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Old 28th July 2009, 03:06   #9145  |  Link
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I rencently converted a 25fps DTS-HD MA to 24fps LPCM. The source was 16 bit, the output LPCM was 24 bit. I did not use any extra parameters, except the "-changeto" switch.

I'd just like to understand why it went from 16 bit DTS-HD MA to 24 bit LPCM. If anybody could explain that would be great.

Here's an image of the process... sorry couldn't wait for the log file, on my way to work...

EDIT: Oh yeah, I first tried converting to w64. It produced a 16 bit LPCM, but couldn't mux with tsMuxeR, got several sync errors, I used the same convert switch. So I decided to try converting to PCM, everything muxes fine... and here I am.

Thanks.


Last edited by odin24; 28th July 2009 at 03:09.
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Old 28th July 2009, 09:00   #9146  |  Link
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Quote:
Originally Posted by odin24 View Post
I'd just like to understand why it went from 16 bit DTS-HD MA to 24 bit LPCM. If anybody could explain that would be great.
Easy, the timestretch you make is a lossy conversion then the output can't be exact and is better a 24 bit aproach.
I recommend you convert to ac3 640 Kb/s because you can't preserve the lossless quality.

Quote:
EDIT: Oh yeah, I first tried converting to w64. It produced a 16 bit LPCM, but couldn't mux with tsMuxeR, got several sync errors, I used the same convert switch. So I decided to try converting to PCM, everything muxes fine... and here I am.
If you have sync problems with w64 you got the same problems with LPCM, the audio data is identical.
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Old 28th July 2009, 10:16   #9147  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
Easy, the timestretch you make is a lossy conversion then the output can't be exact and is better a 24 bit aproach.
I recommend you convert to ac3 640 Kb/s because you can't preserve the lossless quality.


If you have sync problems with w64 you got the same problems with LPCM, the audio data is identical.
I believe you when you say they are identical, however, the sync error only occured when muxing with w64, no such error occured while muxing pcm. I don't actually mean "lip sync", I mean the muxing process could not complete because of an error... something about "byte sync error, or something like that.

So, when stretching, or shortening the time it is considered lossy and the LPCM track will not equal the source DTS-HD MA... will it even be close? Wouldn't the LPCM still be better than AC3 @ 640k?

Thanks for your response.
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Old 28th July 2009, 13:18   #9148  |  Link
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Quote:
Originally Posted by odin24 View Post
I believe you when you say they are identical, however, the sync error only occured when muxing with w64, no such error occured while muxing pcm. I don't actually mean "lip sync", I mean the muxing process could not complete because of an error... something about "byte sync error, or something like that.
Then seems TsMuxer still don't support w64.

Quote:
So, when stretching, or shortening the time it is considered lossy and the LPCM track will not equal the source DTS-HD MA... will it even be close?
Yes, but you can't recover the original audio then must be considered lossy.

Quote:
Wouldn't the LPCM still be better than AC3 @ 640k?
Of course because the uncompressed data in LPCM is the source to encode to ac3, then you add more loss when encode. But the LPCM bitrate is 6912 Kb/s (seems the video )
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Old 28th July 2009, 18:17   #9149  |  Link
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Quote:
Originally Posted by odin24 View Post
So, when stretching, or shortening the time it is considered lossy and the LPCM track will not equal the source DTS-HD MA... will it even be close? Wouldn't the LPCM still be better than AC3 @ 640k?
Stretching and shortening requires resampling, so yes, it is a lossy operation. But if done right (and I'm sure eac3to does it right) the loss should be minimal and is not comparable to the losses caused by perceptual codecs (whether or not you can actually hear either of them is a different question ).

If you find the resulting PCM file too large, I would recommend to reduce the word length back to 16 bits (-down16 in eac3to) and then use a lossless compression such as FLAC. Lossy encoding e.g. to AC3 is a much more severe manipulation of the signal.
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Old 28th July 2009, 23:48   #9150  |  Link
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I wonder, does eac3to recognize AC3 4.0 channels correctly as such? what does 4.0 mean, 2 front and 2 rear channels? because eac3to recognized it as 3.1 so there might be a difference? its from an old movie and unfortunately I don't know whether its really supposed to be 2 front 2 rear.
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Old 29th July 2009, 01:19   #9151  |  Link
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Quote:
Originally Posted by Thunderbolt8 View Post
I wonder, does eac3to recognize AC3 4.0 channels correctly as such? what does 4.0 mean, 2 front and 2 rear channels? because eac3to recognized it as 3.1 so there might be a difference?
Yep, eac3to can difference 4 channel ac3 with:
AC3, 2/2 channels (FL,FR,BL,BR)
AC3, 3/1 channels (FL,FR,FC,BC)
AC3, 2/1.1 channels (FL,FR,LF,BC)
AC3, 3.1 channels (FL,FR,FC,LF)

for more info see this post:
http://forum.doom9.org/showthread.ph...10#post1134710
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Old 29th July 2009, 03:25   #9152  |  Link
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k, I guess that 3/1 thing should be correct then

thanks!
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Old 30th July 2009, 15:15   #9153  |  Link
pascalwil
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Hi

Anyone to help me use SurCode DVD DTS Encoder alone (ie outside of eac3to) in Command Line mode?
I would like to batch convert the 6 wav files I create with eac3to into one DTSWAV file using a script instead of the Surcode GUI.
An example of the script to use would be most welcome to automate the process of creating DTS audio CDs from my 5.1 DVDs.
Thanks for your help.

Cheers
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Old 30th July 2009, 21:13   #9154  |  Link
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am I going nuts, or do DTSHD(MA and HR)>FLAC conversions always end up w/ the dialogs center channel being too low?

TrueHD or LPCM>FLAC is perfectly fine(same dialogs volume as core DTS), and I've tried both Sonic 4.3 and Arcsoft(they give identical CRC btw).

I've encountered this problem many times, DTSHD>FLAC is unusable for me...I use ffdshow to decode if that means anything(DRC is disabled).

I've also tried to enable -keepDialnorm in eac3to, but it doesn't chance anything(same CRC for the FLAC).

Last edited by leeperry; 30th July 2009 at 21:15.
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Old 30th July 2009, 21:19   #9155  |  Link
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does it change when you use madflac to decode?
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Old 30th July 2009, 21:36   #9156  |  Link
leeperry
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yep, center is still too low compared to the core DTS...even w/ madflac.

this time it was a DTS HR 24bit track that I'm downsampling to 16 :

Quote:
eac3to track.dts N:\eng_16.flac -sonic -down16
DTS Hi-Res, 5.1 channels, 1:51:33, 24 bits, 3840kbps, 48khz
(core: DTS, 5.1 channels, 1:51:33, 24 bits, 1509kbps, 48khz)
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Reducing depth from 24 to 16 bits...
Encoding FLAC with libFlac...
Creating file "N:\eng_16.flac"...
eac3to processing took 15 minutes, 19 seconds.
Done.
I'll try to come up w/ some proper sample, was just wondering if anyone else encountered this problem
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Old 31st July 2009, 00:41   #9157  |  Link
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Quote:
Originally Posted by leeperry View Post
am I going nuts, or do DTSHD(MA and HR)>FLAC conversions always end up w/ the dialogs center channel being too low?

TrueHD or LPCM>FLAC is perfectly fine(same dialogs volume as core DTS), and I've tried both Sonic 4.3 and Arcsoft(they give identical CRC btw)...
I can't reproduce your problem:
Code:
                       Core                 dtshd
---------------  ------------------   --------------------
RMS power ch0:   3.21%   (-29.88 dB)   3.40%   (-29.37 dB)
RMS power ch1:   2.82%   (-30.99 dB)   2.99%   (-30.49 dB)
RMS power ch2:   4.93%   (-26.15 dB)   5.22%   (-25.64 dB)
RMS power ch3:   1.59%   (-35.96 dB)   1.86%   (-34.61 dB)
RMS power ch4:   0.88%   (-41.11 dB)   0.93%   (-40.60 dB)
RMS power ch5:   0.69%   (-43.22 dB)   0.73%   (-42.72 dB)
Please upload a sample.
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Old 31st July 2009, 01:16   #9158  |  Link
leeperry
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ok, thanks for the test...I was hoping for some measurements...how did you do that? you're comparing the FLAC'ed DTS-HD against the core DTS?

I've already had the problems many times on DTS HD tracks

for instance on this stream "DTS Master Audio, 5.1 channels, 24 bits, 48khz"(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz), the dialogs were hardly audible, where they were perfectly fine on the core...and -keepDialnorm doesn't work(maybe by design?)

Last edited by leeperry; 31st July 2009 at 03:20.
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Old 31st July 2009, 01:54   #9159  |  Link
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^If you upload a sample, they may be able to help.
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Old 31st July 2009, 03:10   #9160  |  Link
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Quote:
Originally Posted by leeperry View Post
I was hoping for some measurements...how did you do that? you're comparing the FLAC'ed DTS-HD against the core DTS?
I campare a dtshd decoded to wav with ArcSoft (DTS-MA) and decoded with libav (DTS core), the values are statistics from Wavosaur (free audio editor)
Quote:
...and -keepDialnorm doesn't work(maybe on design?)
Don't use -keepDialnorm never with ac3 or dts (I don't know with TrueHD), let eac3to work.
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