Been reading through codec2.c in the newest release, and focusing on
codec2_encode_1200.
The comment above the method:
" Encodes 320 speech samples (40ms of speech) into 48 bits. "
I would like to verify the following from you guys:
1) The 40ms of speech is only true at 8kHz. If we pass in recorded audio
at a higher rate then the 320 samples would equate to some smaller amount
of time
2) The buffer (of 320 samples) does not change in quantity as the rate of
the source file changes. The buffer size is constant (320 samples)
3) These 320 samples will always output 48 bits, even if those 48 bits
corresponds to varying lengths of time due to different sample lengths for
different files.
Hope this makes sense. I'm pretty sure this is true, but I don't yet fully
understand the codec's internals.
Thank you,
Keith
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