First off, there seems to be a sparse number of low level audio programming tutorials out there so if anyone knows of some good resources I would be very appreciative. Now on to the topic at hand!
I am using WASAPI for the audio engine that I am trying to develop and I am having difficulties reformatting read audio file data into a format that the audio device is prepared to receive. Specifically I am wondering how to on the fly convert the sampling rate and bit depth of a read audio file into the requested format of the device. I saw an example where linear interpolation was used but read elsewhere that such a filter introduces too many artifacts such as aliasing, so again I'm wondering what practical algorithms would be for reformatting audio data in a real time application.