[FFmpeg-user] How to convert music into DTS encoded WAV file?

[Peter White]

Hi!

10.08.2016 11:40 Nomis101:
 > 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.
 >
 > The format that I need is the following:
 >
 > /$ ffmpeg -i /Users/Music/test.wav
 > Input #0, wav, from '/Users/Music/test.wav':
 >   Duration: 00:04:47.37, bitrate: 1411 kb/s
 >     Stream #0:0: Audio: dts (DTS) ([1][0][0][0] / 0x0001), 44100 Hz,
 > 5.1(side), fltp
 >
 > You can find a short sample of a proper formatted file here. With
 > Quicktime you will only get some noise. To hear the music you need VLC,
 > which can decode the dts.
 > http://polysom.verilite.de/tmp/test.zip
 >
 > The most promising command so far was:/
 > //$ ffmpeg -i /Volumes/Macintosh/EREDV946.flac -ar 44100 -b:a 1411k
 > -acodec pcm_s16le /Volumes/Macintosh/EREDV946.wav//
 > [...]
 > //$ ffmpeg -i /Volumes/Macintosh/EREDV946.wav -strict -2 -acodec dts

Is there a reason you did this in two steps? ffmpeg should be able to
just:

$ ffmpeg -i test.wav -strict -2 -c:a dts -ar 44100 -b:a 1411k output.wav

 > //Output #0, wav, to '/Volumes/Macintosh/EREDV946_2.wav'://
 > //  Metadata://
 > //    ISFT            : Lavf57.41.100//
 > //    Stream #0:0: Audio: dts (dca) ([1] [0][0] / 0x2001), 44100 Hz,
 > 5.1(side), s32, 1411 kb/s//

This is not quite the same as your sample. Here the sample format is
s32 as opposed to fltp in the sample. Sadly though, I am unable to set
the sample format to fltp. It seem this is not supported. The dca
encoder is still experimental, might be why.

$ ffmpeg -i test.wav -strict -2 -c:a dts -ar 44100 -b:a 1411k 
-sample_fmt fltp output.wav
Input #0, wav, from 'test.wav':
   Duration: 00:00:05.03, bitrate: 1411 kb/s
     Stream #0:0: Audio: dts (DTS) ([1][0][0][0] / 0x0001), 44100 Hz, 
5.1(side), fltp
[dca @ 0x252ce00] Specified sample format fltp is invalid or not supported
Output #0, wav, to 'output.wav':
     Stream #0:0: Unknown: none
     Metadata:
       encoder         : Lavc57.51.100 dca
Stream mapping:
   Stream #0:0 -> #0:0 (dts (dca) -> dts (dca))
Error while opening encoder for output stream #0:0 - maybe incorrect 
parameters such as bit_rate, rate, width or height

 > But this does not play with VLC or import into iTunes.

For me it doesn't as well. Interistingly enough, it does play in MPV.

 > I think maybe
 > because the thing in the bracket is not correct? It should be
 > "([1][0][0][0] / 0x0001)", but it is "([1] [0][0] / 0x2001)".

Out of my depth here, but that is certainly possible.

 > Can somebody help me on how to generate a properly dts-encoded WAV file
 > using ffmpeg that imports into iTunes?

It might help, if you can get some info which formats iTunes accepts.
I cannot imagine that it is that narrow in playing such files.


Best,
Peter