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

[Peter White]

10.08.2016 18:28, Nomis101:
> I've tried a combination of both current suggestions, but this gives me
> an error:
>
> /$ ffmpeg -i /Volumes/Macintosh/EREDV946.flac -strict -2 -c:a dts -ar
> 44100 -b:a 1411k -sample_fmt fltp -tag:a 0x0001
> /Volumes/Macintosh/EREDV946.wav
 > [...]
> [dca @ 0x7fc21680d800] Specified sample format fltp is invalid or not supported

This is the problem. fltp seems not to be supported (yet) by this
codec. I get the same with the git master from today (N-81311-g75e7f20).

> Am 10.08.16 um 16:09 schrieb Peter White:
>> Is there a reason you did this in two steps? ffmpeg should be able to
>> just:
> This is, because the flac is 24 bit and I wanted to convert it first
> into 16 bit wav.

That does not make much sense, since the final file is supposed to be
fltp, which is 32 bit, anyway. At least that is what your sample
suggests. See what the dca encoder does, when you don't specify
-sample_fmt:

$ ffmpeg -i test.wav -c:a dts -strict -2 -ar 44100 output.wav
Output #0, wav, to 'output.wav':
   Metadata:
     ISFT            : Lavf57.46.100
     Stream #0:0: Audio: dts (dca) ([1] [0][0] / 0x2001), 44100 Hz, 
5.1(side), s32, 1411 kb/s

It defaults to s32. I don't know if this DTS-in-WAV is supposed to take
any other sample formats than fltp. But the sample you supplied actually
is fltp, hence 32 bit. So, there really is no point in downgrading the
sample format to 16 bits if DTS in a WAV needs to be 32 bits anyway.

But another thing popped into my head. Since it is a 6-channel audio,
the bitrate should be more than 1411k. This looks an awful lot like a
stereo bitrate, which would actually be 1411.2k (16*44.1k*2). Is it
possible that this is a matrix encoding, like Dolby Prologic? I don't
know if it matters at all but I think it's worth pointing out, just in
case.


Best,
Peter