[openal] Audio quality and upsampling

alberto box alberto.box66 at gmail.com
Fri Jul 1 03:37:23 EDT 2016

I'd need an advice to solve an audio quality issue on my Windows openal
application ...

My implementation receive telephony quality audio streams from the network
and uses OpenAL to merge and play them on the speakers. OpenAL audio input
is 8KHz 16 bits.

The final result on the speakers seems a bit distorted, a sort of metallic
sound, thus I investigated further the problem using audacity to record the
PC audio output (using Windows WASAPI loopback interface).

It becames clear that the upsampling from 8KHz to the 44.1KHz (the rate
requested by the final audio device) introduces distortion, because the
added samples are computed as a linear interpolation between the adiacent
original samples. Trying to upsamples the original audio using audacity
produces a better audio and the samples are really smooth (it uses a
different kind of interpolation).

I guess the upsampling is internally performed by OpenAL (am I wrong?); I'm
using a old version of the library (oalinst.exe properties shows version / 2007).

My question is: what can I do to improve the upsampling? Do I need to
update OpeanAL libraries to a newer version (I found oalinst.exe)?
Do I need to migrate to OpenAL soft?

Thank you!
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