I'm sampling the audio signal through mic jack on C6713 board.
The code is based on the example project "dsk_app". The audio can be
back from the speaker jack. However, when I tranform the signal in the
buffer to double precision values, it seemed that (later) half of the data
in the buffer has been truncked (all zeored). Is there any one can give me
some advice to solve this issue? Thanks a lot.
Do you know a company who employs DSP engineers?
Is it already listed at http://dsprelated.com/employers.php ?