Hope you had a pleasant day and have a little bit of energy left to help a
newbie like me :o)
I just got my first DSP-board (TMS320C6713 DSK) and I want to implement an
algorithm that improves the SNR of a noisy speech signal.
I have hooked up my PC's "line out" to the "line in" on the DSK. And I have
connected my headset to "line out" on the DSK. I have found some
applications (came along with Code Composer Studio) that can play back audio
in realtime, but they are all big programs.
So I was wondering if somebody out there could send me the minimum source
code that is required for acquiring and playing audio samples in real time?
I need something like this:
Algorithm Pseudo Code:
1) record a frame of audio to an input buffer
2) process samples in buffer to obtain better SNR
3) place result in output buffer
4) play samples in output buffer
5) go to 1
Thank you very much :o)