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Old 27th February 2015, 14:32   #1  |  Link
eddman
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FFmpeg commands for converting 24-bit multichannel DTS to 16-bit stereo W64?

I'm looking for ffmpeg commands for the aforementioned conversion.

Also, is there a way to use the dolby pro logic method, like eac3to? I assume that gives a better result than simple, right?

Last edited by eddman; 27th February 2015 at 15:51.
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Old 1st March 2015, 12:53   #2  |  Link
Reino
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Something like this?
Code:
ffmpeg.exe -hide_banner -i input.dts -af "aresample=matrix_encoding=dplii" -ac 2 output.w64
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Old 1st March 2015, 17:35   #3  |  Link
eddman
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Quote:
Originally Posted by CoRoNe View Post
Something like this?
Code:
ffmpeg.exe -hide_banner -i input.dts -af "aresample=matrix_encoding=dplii" -ac 2 output.w64
Thanks. I don't see a bit depth command. Does ffmpeg convert to 16-bit by default? If so, what'd be the command to convert to/keep the same 24-bit depth?
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Old 1st March 2015, 17:58   #4  |  Link
poisondeathray
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Yes, 16bit by default. Your title asked for 16bit

-c:a pcm_s24le for 24bit little endian
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Old 1st March 2015, 18:05   #5  |  Link
eddman
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Quote:
Originally Posted by poisondeathray View Post
Yes, 16bit by default. Your title asked for 16bit

-c:a pcm_s24le for 24bit little endian
I know. I was just surprised that it didn't keep the original bit depth by default. It's kind of an odd thing.

Actually, I wanted to know the proper bit converting command; 16-bit was an example.

Thanks again.

Quote:
little endian
I don't really know what that is and why it should be like that.

Last edited by eddman; 1st March 2015 at 18:09.
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Old 2nd March 2015, 00:59   #6  |  Link
foxyshadis
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Quote:
Originally Posted by eddman View Post
I don't really know what that is and why it should be like that.
Just the byte order to correctly read the samples, and since it's always tagged, it doesn't matter. Technically: x86/x64 are little endian, Motorola 6800 and its descendants were big endian, and all that means is that if the file doesn't match the CPU then the bytes have to be swapped internally to work with them. (ARM can be both endians.)

tl;dr: Always use le.
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Old 2nd March 2015, 10:20   #7  |  Link
pandy
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Sometimes surround matrices may give suboptimal results (some TV's equipped with voice processing fail) then this perhaps can be useful - works fine for me.

Code:
@ffmpeg -threads %NUMBER_OF_PROCESSORS%*1.5 -i %1 -vn -c:a pcm_s16le -af "aformat=sample_fmts=fltp,pan=stereo|FL < FL + 1.414FC + .5BL + .5SL + 0.25LFE|FR < FR + 1.414FC + .5BR + .5SR + 0.25LFE,aresample=resampler=soxr:osr=48000:dither_method=shibata,aformat=sample_fmts=s16:channel_layouts=stereo[out]" -y %1.w64
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