Comments (4)
The libav/ffmpeg decoder/parser can handle all but the http://samples.libav.org/A-codecs/DTS/dts/Master%20Audio%205.1%2024bit.dts sample file without error.
from dcadec.
Note that a bunch of those files are just cut rather badly, so the last audio frame erroring out is pretty "normal".
from dcadec.
This is expected. As @Nevcairiel mentioned, these files are cut badly at the end, leaving just one core frame without EXSS part. The number of decoded channels (or bitdepth, or sample rate) changes in the decoder, the WAV file writer notices that and returns -1 (invalid argument) error because it can't handle change of audio parameters once the WAV header is written. Of course a proper output sink would reconfigure itself and try to continue in this case, something the current primitive WAV writer doesn't do.
I don't know how exactly FFmpeg handles this, maybe it doesn't decode enough of extensions so that its decoder output configuration never changes, or maybe it handles change of parameters gracefully somehow.
from dcadec.
http://samples.libav.org/A-codecs/DTS/dts/Master%20Audio%205.1%2024bit.dts
Actually, dcadec decodes this one without any error in my case.
from dcadec.
Related Issues (20)
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