Post-processing techniques used before saving the call recordings?

Hello there!
I have observed that my FreePBX server is receiving the audio data in G.711-PCMU(bit-depth of 8) codec through SIP Trunk connection. But it is saving the call recordings with bith-depth of 16 and it is quite clear and smooth.
So, I want to know what are the post-processing techniques it is using to upscale the audio and where the script for this post-processing is actually implemented?
I will be thankful for any help and guidance.

which is a simple table lookup, which is the obvious way of doing it.

More generally, Asterisk will use the same code it uses to transcode between channels.

Note that G.711 is not linear. It covers the full 16 dynamic range.

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