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

Nomis101 🐝 Nomis101 at web.de
Wed Aug 17 21:52:38 EEST 2016

Am 17.08.16 um 06:56 schrieb Peter White:
> I don't see where resampling could have happened, so my guess is that
> you are basically playing 48kHz audio at 44.1kHz. The difference in
> length sounds about right for that (48/44.1~=1.088).
> Do you hear a change in pitch as well?
You are right. It works if I use source material that has 44.1 kHz or if
I do at the first place
$ ffmpeg -i input.foo -strict -2 -acodec dts -ar 44100 -f spdif foo.spdif
I'm not sure if there is a change in pitch, it sounds a bit more dull,
so maybe there is.

Am 17.08.16 um 07:05 schrieb Carl Eugen Hoyos:
> I tried (hard) to tell you in advance.
I know, but sadly I do not know any other DTS encoder for Mac. Do you
know if ffmpeg DTS encoder quality will be improved?

Am 17.08.16 um 07:05 schrieb Carl Eugen Hoyos:
> How can I reproduce this?

Do the following:
1. Find any source material with 48 kHz sampling rate (mine was
alac/flac, but don't think this matters).
2. Open this file in VLC, check that it plays nicely and note the duration
3. Do
$ ffmpeg -i /test.flac -strict -2 -acodec dts -f spdif /test.spdif
4. Than do
$ ffmpeg -f s16le -ac 2 -ar 44100 -i /test.spdif -acodec copy /test.wav
5. Open WAV file in VLC (I had to use --demux=ffmpeg to open it in VLC)
6. Play the file and note the longer duration and that file sounds slower

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