There's an issue sometimes where when processing the audio from a
desktop sharing file, the STARTPTS value is invalid somehow, and this
results in bad timestamps in the output stream. Depending on the
selected codec/container combination, this might result in errors such
as spam of the message:
Application provided invalid, non monotonically increasing dts to muxer in stream
or simply a hang during processing.
Since we're already using aresample in async mode to correct timestamp
gaps, we can use "asetpts=N" here to simply set the audio frame pts
values to the sample number directly, giving proper monotonic timestamps
with no gaps.
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