[openal] Audio quality and upsampling

alberto box alberto.box66 at gmail.com
Sat Jul 2 04:03:53 EDT 2016

Hi Chris, thank you for your answer.

In your opinion the api exposed by my old creative version are the same api
exposed by the newer openal soft? I mean is there a chance for me to update
the libraries only, without working on the source code?

Il 02/lug/2016 06:52, "Chris Robinson" <chris.kcat at gmail.com> ha scritto:

> On 07/01/2016 12:37 AM, alberto box wrote:
>> 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).
> Creative's software driver only does linear resampling, unfortunately.
> OpenAL Soft defaults to linear, but has options for 4- or 8-point sinc
> resampling, and also has a variable 12- to 24-point sinc with antialiasing.
> Another option is to resample it manually, using a high quality resampler
> to increase it to 44.1khz before giving it to OpenAL, then the linear
> resampling noise will be less noticeable.
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