[FFmpeg-user] How to convert music into DTS encoded WAV file?
Carl Eugen Hoyos
ceffmpeg at gmail.com
Thu Aug 11 02:31:58 EEST 2016
Sorry for the late answer, I can't be online all day long;-)
2016-08-10 11:40 GMT+02:00 Nomis101 🐝 <Nomis101 at web.de>:
> Hi, ffmpeg user list. I have a problem that I can not solve and hope you
> can help me.
> I found out, that I can listen to 5.1 surround music over iTunes /
> Airplay if the music is an dts-encoded WAV file (44.1 kHz, 16 Bit). This
> means I have to convert all my surround music, mostly *.dts, *.flac or
> *.ac3, into this format. Everything I've tried did not produce a WAV
> file that imports into iTunes.
First thing: There is a reason for "-strict -2"! FFmpeg does provide a dts
encoder but it is very experimental and I suspect it is not transparent no
matter how high the bitrate. (It allows to test below commands though.)
I believe there is a binary encoder that you can use (I don't know for
sure though) to encode your input files to dts first and then remux as
You can mux dts in wav, the correct twocc is 0x2001, I have no idea why
vlc doesn't support it (it should with --demux=ffmpeg) but this is not what
you are searching for: You want dts in spdif in wav looking like pcm
stereo. Your software player will send the pcm stereo that it sees to your
hardware, the hardware will detect spdif and start its spdif demuxer which
detects dts and sends the dts data to the dts hardware decoder.
FFmpeg supports an spdif muxer (non-experimental) that you can use
to mux a raw spdif file. You can then tell FFmpeg that your spdif file is
actually pcm stereo and mux that into wav.
$ ffmpeg -i input -strict -2 -acodec dts -f spdif out1.spdif
or (much better because it involves no experimental encoding!):
$ ffmpeg -i dts_input -acodec copy -f spdif out2.spdif
$ ffmpeg -f s16le -ac 2 -ar 44100 -i out.spdif -acodec copy out.wav
Note the "-ac 2" no matter if your input is stereo or not!
(Untested atm, but this should allow you to reach your goal, or ask again.)
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