Hi folks,

After experimenting with codec2 / fdmdv I realised that typical PC
sound hardware is not FUN. I had a hard time finding a suitable
soundcard / microphone combination that would yield a reasonable
audio input for experimentation. I found that sampling the mic at 48ks
and then downsampling with sox to 8ks yielded a much improved result.
With this procedure I managed to produce something that after passing
through the c2enc-mod-demod-c2dec chain was usable.

(those of you with better audio h/w might disagree :-) 

Noting that the initial sample rate should preferably be higher, I
started thinking about how input sample resolution would affect the
audio, and with an idea of using a MCU for transceiver interface, I
decided to try codec2 out with 12 and 10 bit resolution. Not to
interfere with the codec2 hacking, I wrote a small program that read
in the raw input file for codec2, and masked it, removing the 4 or 6
least significant bits.

Listening to the result, I could hear very little difference with a
well adjusted audio file, (thanx peter). 

Next step was to pass the result through the codec2 chain, and the
result is to my judgment very interesting, in that there is very little
difference in the final recovered audio file.

So, what do you think, this experiment is reproducible with only
a small filter program, and maybe says something about possible future
implementations.

Regards,

Gullik /SM6FBD


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